Re: [OSL | CCIE_Voice] Wireshark pcap questions for SIP and H323 call flows
For #2, just filter “sip” in your captured session to see just the sip message flow. To see a ladder diagram, click Telephony on the top menu and select VoIP Calls. Wireshark will then show a pop up window with all captured voice calls. Double click the call to see a ladder diagram. This is a blog post I wrote quite some time ago on how to configure a Counterpath SIP soft phone for use with CUCM and includes the steps I mentioned in Wireshark. http://www.markholloway.com/blog/?p= On Mar 23, 2014, at 2:50 AM, Shrinivas Varanasy voip...@me.com wrote: Open the file with wireshark and filter SIP, select follow UDP stream. Use translatorX to get the call flow ladder diagram Sent from my iPad On Mar 22, 2014, at 7:44 AM, virajith vir...@rediffmail.com wrote: Hello Guys, Could anyone help me with the following :- 1) I am looking at converting a sip and H323 pcap ( wireshark file ) into a plain text file ( .txt format) . Are there any easy options to do this ? I saw some tcpdump options however not sure on what to use. 2) Also are there easy options for call flow diagrams . I notice there a few filtering options however what is the best to view the call flow ? - Vir Get your own FREE website, FREE domain FREE mobile app with Company email. Know More ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] RIP CCIE-Voice :-)
From a Cisco partner business perspective I don’t see how it would make sense to have your current CCIE’s retake the lab when they could just as easily migrate to Collaboration after taking their written renewal. Time is money, and consuming your top resources with another lab certification with no foreseeable gain seems like a waste. I imagine most folks working for partners are already deep into the same platforms on the Collaboration blueprint. At minimum, they should at least have the UC NFR software setup in their own lab already. On Feb 15, 2014, at 12:10 AM, m george m.george00...@gmail.com wrote: Will anyone here who already passed voice lab preparing to undertake collaboration lab for 2nd CCIE title ? I have talked to many folks me my colleagues we plan to convert our titles to Collab IE with written rather than going for another hectic lab. What's your guys take on this ? What will you do ? On Sat, Feb 15, 2014 at 3:54 AM, Abel ... midga...@gmail.com wrote: Upgrading my home lab already, kind of expensive with new 29xx. But just for the knowledge sake. On Sat, Feb 15, 2014 at 7:15 AM, wilson.sam...@bt.com wrote: Aha Nicolas, you have a point sir. Anyway, I just wanted to make the passage of the track / version somewhat memorable that's all. No need to get serious on this now (note to myself as well) Lets get the Colloboration done.. Btw, who is attempting it on tihs forum and how you have prepared for it? Lab Gear?? Regards From: Mergenthal, Chase [chase.mergent...@bestbuy.com] Sent: Friday, February 14, 2014 3:01 PM To: Nicolas MICHEL; Samuel,W,Wilson,JKH3 R Cc: Online Study Subject: RE: [OSL | CCIE_Voice] RIP CCIE-Voice :-) It’s funny you mention that, on my second or so attempt; at the end of the exam UCCX wasn’t working at all… I got 100% on UCCX… -- Chase Mergenthal From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Nicolas MICHEL Sent: Friday, February 14, 2014 1:41 PM To: wilson.sam...@bt.com Cc: Online Study Subject: Re: [OSL | CCIE_Voice] RIP CCIE-Voice :-) Wilson, I am already a CCIE in RS so I know what to expect when I am taking a CCIE exam. When you skip the UCCX task because you ran out of time and when you score report says : UCCX = 100%, to me it means complete nonsense :) Lots of people have different opinion and point of view :) Cheers Nicolas ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
Something doesn’t seem to add up in my head. Supp Services shouldn’t effect DTMF. Did you change anything related to the SIP Trunk on CUCM? Or anything DTMF related on a dial-peer? On Jan 30, 2014, at 6:22 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Hello Somphol/Justin, I have resolved the issue by adding the command no supplementary-service sip moved-temporarily. Thanks a lot Somphol for pointing the document to me. Thank you Justin for providing me the inputs. Regards, Viki On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney justin.s.car...@gmail.com wrote: I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered else you wouldn't be getting to cue using g729 that is coming over the wan (maybe the tag command just got lost on the copy/paste of the config to the email?). (Also for the sccp config you're missing the same tag command for the cfb and the conference hardware command. You have the sccp ccm pointing to the cucm ip after cme, are you trying to register sccp resources to cucm?) You can run debug ccsip messages on cme to ensure you see the dtmf comes across the sip trunk from cucm. Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check this is set the same inside cue. For an alternate test, when you place the same call can you leave a message ( 2 sec) and hang up without pressing pound? Does the mwi come on and can the cme phone retrieve the voicemail after entering the pin? If so use the same debug ccsip messages cmd to see the expected/normal debug output for the dtmf on this working scenario. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html. Regards, --Somphol. ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
I understand how DTMF works on SIP Trunks, what I’m not clear on is how “no supp services” would have an impact on his DTMF issue. I’m trying to understand the logic of something changing with RFC2833 or SIP NOTIFY to the point where # is now recognized, yet without changing anything related to DTMF. Wouldn’t supp services only impact the signlaing behavior of the SIP 302 message itself? But not DTMF? On Jan 30, 2014, at 8:00 AM, Bill Lake whl...@gmail.com wrote: Inbound SIP trunk from ITSP and CUE http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml He would see the issue in the debugs On Thu, Jan 30, 2014 at 6:51 AM, Mark Holloway m...@markholloway.com wrote: Something doesn’t seem to add up in my head. Supp Services shouldn’t effect DTMF. Did you change anything related to the SIP Trunk on CUCM? Or anything DTMF related on a dial-peer? On Jan 30, 2014, at 6:22 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Hello Somphol/Justin, I have resolved the issue by adding the command no supplementary-service sip moved-temporarily. Thanks a lot Somphol for pointing the document to me. Thank you Justin for providing me the inputs. Regards, Viki On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney justin.s.car...@gmail.com wrote: I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered else you wouldn't be getting to cue using g729 that is coming over the wan (maybe the tag command just got lost on the copy/paste of the config to the email?). (Also for the sccp config you're missing the same tag command for the cfb and the conference hardware command. You have the sccp ccm pointing to the cucm ip after cme, are you trying to register sccp resources to cucm?) You can run debug ccsip messages on cme to ensure you see the dtmf comes across the sip trunk from cucm. Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check this is set the same inside cue. For an alternate test, when you place the same call can you leave a message ( 2 sec) and hang up without pressing pound? Does the mwi come on and can the cme phone retrieve the voicemail after entering the pin? If so use the same debug ccsip messages cmd to see the expected/normal debug output for the dtmf on this working scenario. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html. Regards, --Somphol. ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
In the larger debug attachment the SDP includes a=fmtp:18 in the 200 OK coming from the CME site (IP 3.3.3.3). In the other capture I didn’t see any SDP. If no DTMF offer is present during call setup, this would assume plain old in-band DTMF, which won’t work on a compressed codec like G.729. You press digits and nothing happens. G729 requires RFC 2833, SIP NOTIFY, or KPML to function properly. On Jan 30, 2014, at 1:05 PM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Hello All, I have attached the debug ccsip messages output before and after using the command. I do not have the answer why it resolved the dtmf-issue. If you guys find something, please share it. Thanks, Viki On Thu, Jan 30, 2014 at 4:16 PM, Moataz moataz_m...@yahoo.com wrote: no supplementary service affect only call forwarding and call transfer , i do not know how it solve DTMF Regards, Moataz Tolba On Thursday, 30 January 2014, 15:17, Mark Holloway m...@markholloway.com wrote: I understand how DTMF works on SIP Trunks, what I’m not clear on is how “no supp services” would have an impact on his DTMF issue. I’m trying to understand the logic of something changing with RFC2833 or SIP NOTIFY to the point where # is now recognized, yet without changing anything related to DTMF. Wouldn’t supp services only impact the signlaing behavior of the SIP 302 message itself? But not DTMF? On Jan 30, 2014, at 8:00 AM, Bill Lake whl...@gmail.com wrote: Inbound SIP trunk from ITSP and CUE http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml He would see the issue in the debugs On Thu, Jan 30, 2014 at 6:51 AM, Mark Holloway m...@markholloway.com wrote: Something doesn’t seem to add up in my head. Supp Services shouldn’t effect DTMF. Did you change anything related to the SIP Trunk on CUCM? Or anything DTMF related on a dial-peer? On Jan 30, 2014, at 6:22 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Hello Somphol/Justin, I have resolved the issue by adding the command no supplementary-service sip moved-temporarily. Thanks a lot Somphol for pointing the document to me. Thank you Justin for providing me the inputs. Regards, Viki On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney justin.s.car...@gmail.com wrote: I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered else you wouldn't be getting to cue using g729 that is coming over the wan (maybe the tag command just got lost on the copy/paste of the config to the email?). (Also for the sccp config you're missing the same tag command for the cfb and the conference hardware command. You have the sccp ccm pointing to the cucm ip after cme, are you trying to register sccp resources to cucm?) You can run debug ccsip messages on cme to ensure you see the dtmf comes across the sip trunk from cucm. Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check this is set the same inside cue. For an alternate test, when you place the same call can you leave a message ( 2 sec) and hang up without pressing pound? Does the mwi come on and can the cme phone retrieve the voicemail after entering the pin? If so use the same debug ccsip messages cmd to see the expected/normal debug output for the dtmf on this working scenario. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html. Regards, --Somphol. ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com
Re: [OSL | CCIE_Voice] Lab Flavors
What is it with RTP and fish? The day I took my lab there they served chicken la cordon bleu, which is just as bad when your stomach is in knots. In San Jose we went to the cafeteria and I had salad. Much better, and I passed. On Dec 16, 2013, at 12:37 PM, Bill Lake whl...@gmail.com wrote: Bring lunch just in case you get salmon patties From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Brad Williams Sent: Monday, December 16, 2013 11:35 AM To: Wayne Lawson Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab Flavors No worries. Was just looking for any helpful RTP advice. Thanks. On Monday, December 16, 2013, Wayne Lawson wrote: Yeah - Brad, what exactly are you asking? If you're looking for NDA specific info - you're not going to get any answers here. Regards, Wayne A. Lawson II Founder CEO - iPexpert CCIE #5244 / Emeritus :: World-Class Cisco Certification Training Mobile: +1.810.334.1564 :: Free Videos :: Free Training / Product Offerings :: CCIE Blog :: Twitter On Dec 16, 2013, at 12:30 PM, Daniel Pagan dpa...@fidelus.com wrote: Brad: I can be wrong here, but this seems like a question that falls under the NDA umbrella. - Daniel From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Brad Williams Sent: Monday, December 16, 2013 12:12 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab Flavors Hello, I am about to sit for my lab at RTP. Does anyone have any insight as to which labs are being offered and in what frequency at RTP? Thank you. ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] CCIE Voice to CCIE Collaboration Details
3 months? You have until 2016 to convert from CCIE Voice to CCIE Collaboration. --snip-- Pass the CCIE Collaboration Written Exam and then permanently convert your CCIE Voice certification to a CCIE Collaboration certification between November 21, 2013 and February 13, 2016. On Sep 18, 2013, at 6:03 PM, shawn roger shawn.roge...@gmail.com wrote: Cisco is really smart :) They need money from anywhere ha ha :) But only 1 problem they should do option 2 till collaboration end :) not only for 3 months that means if a person do not pass in collaboration his voice also useless + collaboration also useless for him They are dam smart On Wed, Sep 18, 2013 at 11:03 PM, Brian Schear brian.sch...@vitalsite.com wrote: They have the details out for CCIE Voice to CCIE Collaboration migration now. https://learningnetwork.cisco.com/docs/DOC-21915 Brian Schear CCIE #36045 (Voice) This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] IE voice retired
This is a valid question. I hope the answer is yes. On Jun 2, 2013, at 10:28 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: My question is : If we only pass ccie voice lab then get ccie number and we keep renewing it . When cisco retired it in Feb 2014. Does my ccie number count for silver or gold partnership discount? From: Kamran Ahsanullah kamran.ahsanul...@gmail.com; To: ccie_voice@onlinestudylist.com; Subject: Re: [OSL | CCIE_Voice] IE voice retired Sent: Sun, Jun 2, 2013 7:42:32 PM If you have your voice ccie or pass the exam before the Collaboration CCIE goes live then yes, you get to keep your CCIE number. If you then pass Collaboration CCIE you will become a dual Voice/Collaboration CCIE. On 2 June 2013 19:00, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. IE voice retired (Karen Johnson) -- Message: 1 Date: Sun, 2 Jun 2013 06:40:03 -0700 (PDT) From: Karen Johnson karen.johnson...@yahoo.ca To: William Bell b...@ucguerrilla.com, Martin Sloan martinsloa...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, vma...@ipexpert.com vma...@ipexpert.com Subject: [OSL | CCIE_Voice] IE voice retired Message-ID: 1370180403.14208.yahoomail...@web163901.mail.gq1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 hi all, ? Let say we keep re-certify our IE Voice and still hold the number. Would it be still count to Cisco partnership when?IE voice retired?? ? Tks K -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130602/8aff702f/attachment-0001.html -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 88, Issue 8 * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
. Rant over. On May 28, 2013, at 20:14, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Yes its really frustrating what Cisco is doing to us. Ok let me tell you this. People now have invested a lot of money in pursuing their CCIE Voice that includes (Verious Workbook fees , Rack Rentals , Home Lab building , travel expenses and Lab fees attempts for whatever times) So when people achieve CCIE Voice nowadays a year or two later it would be considered old and grandfathered. Also , Cisco has released a new lab for 2 months while they are planning to abolish the whole syllabus. Why they do that to us They already make money out of everything especially lab multiple times of lab attempts per each person. CCIE Voice achievers has to send cisco request for Migration without Lab test. CCVP it was automatically migrated to CCNP Voice without any additional tests. CCNA is migrated to CCNA R/S without any additional tests. In case of Video part then I suggest whether they force CCIE Voice people to make CCNA VIDEO or CCNP Video if they will release or they make just a migration lab track that includes VIDEO stuff only for a cheaper fee something like $500. Thats same for MICROSOFT they abolished MCSE to change it to MCITP people usually just add 2 tracks to become full MCITP same when they migrate to new MCSE (Microsoft Certified Solutions Experts) there is only an upgrade track rather than taking the whole 5 tracks again. Cisco obviously has to do something like that.It's really unfair retiring the whole cisco voice totally. Guys to make the new Collaboration lab that would cost anyone over 50K to buy telepresence , X9XX routers stuff , 9971 Video Phones , TV's and etc.. Even the rack rentals would be 5 times the old voice track as the equipment would be way more expensive. Seriously , We have to agree all of us from multiple different voice study group to have a migration track to Collaboration please share your thoughts guys On 28 May 2013 18:56, Mark Holloway m...@markholloway.com wrote: Bummer, I was really hoping CCIE Voice candidates would transition to Collaboration without any additional lab exams. On May 28, 2013, at 7:08 PM, Vik Malhi vma...@ipexpert.com wrote: For my initial reaction read here: http://bit.ly/12MNK5t Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
Bummer, I was really hoping CCIE Voice candidates would transition to Collaboration without any additional lab exams. On May 28, 2013, at 7:08 PM, Vik Malhi vma...@ipexpert.com wrote: For my initial reaction read here: http://bit.ly/12MNK5t Vik Malhi – CCIE #13890 Managing Partner - IPexpert, Inc. Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Blueprint Change
You should not concern yourself with the blue print change at this point. Focus on passing the current blue print and you will make it happen. If the blue print changes, you have 6 months to continue studying, but even when Cisco starts testing the new blue print it's going to take some time for IPExpert (and others) to get their new study material completed to match the new blue print. You're only going to make it harder on yourself if you hold your breath for the next blue print. On Dec 13, 2012, at 1:09 PM, Marko Milivojevic mar...@ipexpert.com wrote: Remember that even when they announce the new blueprint, it will take 6 months before it goes active. -- Marko Milivojevic - CCIE #18427 (SP RS) Senior CCIE Instructor - IPexpert On Thu, Dec 13, 2012 at 9:20 AM, Michael Davis michaeldavis1...@yahoo.com wrote: The Blueprint for voice will change, and I was also told soon. When I took my first attempt at my CCIE voice this week, I was told it weill not be as quickly as you would think by the proctor. Vik, on the other hand, is convinced it will sooner that latter. You just have to keep checking the Cisco web site for updates. Michael ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cisco ripped me off
If memory serves me correctly, if they reach a point in the lab grading where you've already reached the failing mark, they don't proceed to grade the rest of the exam and you just get Fail for remaining sections. I cannot verify that though and it has been two years since I passed. I took my first CCIE Voice attempt in RTP and additional attempts in San Jose. I felt the network in San Jose was much faster. It feels like the RTP network must be connecting back to San Jose, because it was slow in comparison. There were some odd behaviors in RTP that made my palms sweaty because I thought things were tanking on the back end of the network, but everything eventually proceeded normally. I never felt that way in San Jose. On Oct 31, 2012, at 10:40 AM, Leslie Meade wrote: Plus, I still think that they will stop marking as soon as they know you won't pass. Sp parts you know you nailed, you will see a zero Sent from my iPad On Oct 31, 2012, at 7:35 AM, Krishna vinayak_...@yahoo.commailto:vinayak_...@yahoo.com wrote: Cory, Technically speaking, the grading has to be evaluated by taking the seating position where we took the exam rather doing it remotely for their convenience. i used switchport mode trunk, switchport trunk native vlan data on sb and sc. Can anyone expect fail in the exam after evaluating the tasks thrice and check everything line by line, and the end showing the score report as fail... This is completely insane. I was wondering if i can legally proceed so that justification will be done for the right candidates. Thank you krishna. From: Cory Gray corygray22...@hotmail.commailto:corygray22...@hotmail.com To: 'Krishna' vinayak_...@yahoo.commailto:vinayak_...@yahoo.com; 'Online Study' ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com Sent: Wednesday, October 31, 2012 7:41 AM Subject: RE: [OSL | CCIE_Voice] Cisco ripped me off Krishna, I am sorry to hear that. I suffered something similar during my last attempt but after much thinking I think I know what happened and maybe the same happened to you. Even though IPexpert recommends using switchport mode trunk on ESW interfaces I still had been using switch mode access because it never failed. I also did this because using switchport mode trunk would show nothing in the show vlan-switch command so I was scared this was how it was being graded and would miss the points. IPexpert recommends this because they say the other way has been known to stop working for no reason. When I got my score report the next day, I could see several sections wrong that I knew I configured right. Doing the math I believe when they went to grade my exam the next day that my CUCME phones were no longer registered. I will use switchport mode trunk for now on. What did you do? That is my only theory. Maybe you have one different that can help others if you choose not to take it again. I will be back 11/30 and am hoping to do as well as I did last time but pass :) Thanks, Cory From: ccie_voice-boun...@onlinestudylist.commailto:ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Krishna Sent: Wednesday, October 31, 2012 8:08 AM To: Online Study Subject: Re: [OSL | CCIE_Voice] Cisco ripped me off all, yesterday i took my second attempt in rtp, and i am 200 % sure that i pass the exam. I got 1 hour left even after testing it thrice, but looking at the score report i was shocked, and i completely disagree with my score report. F... CCIE lab script evaluation.. i am completely pissed off the way it showed the results... no more CCIE in my life... i appreciate all my friends who helped me in this journey. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.comhttp://www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Passed!!!
Congratulations. It's hard to believe there are 10,000 more CCIE's in just two years. I earned mine on October 28, 2010. I just took my written again a couple of days ago. Mark CCIE #27384 (Voice) On Oct 24, 2012, at 11:51 AM, Bruno Nonogaki wrote: Hello guys, I have just received my results, and I was approved on my CCIE Voice today, second attempt. Many thanks to everybody on this list!!! :) Now it is time to take a rest of this one-year long jorney... Bruno Nonogaki, CCIE #37170 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CFUR for and by fields
CUCM 7.01 has a bug. The VM Profile is a work around. Every time I would reset my CUCM VM's sometimes I would get the expected display as shown below, other times I wouldn't. A few other folks confirmed this as well. It's a display issue with 7.01 and VMware On Mar 31, 2011, at 12:19 PM, adam compton wrote: Roger, From hq ph 1 (5001) For: +16178631001 ( 1... ) by : +16178631001 ( 1... ) That's exactly what I want to see! Any tips how you achieved this? Bo, This is where I lose you: 4. Create a VM Profile called SBPH1 and mask +16178631001. This will require you to configure Alternative Numbers in CUC later for the VM to work I know that CUC configuration is required to get the voicemail to work, but I am only worried about the phone display at this point. I've got CFUR working, just not with the correct display. Michael, I have enabled redirect IE under the HQ gateway. I've noticed if I don't have it i don't get a for and by field at all. To my knowledge, the redirect ie is turned on under the serial interface by default. A heads up, right now I have the following display forwarded 5001 for: 1001 by: 1001 Thanks for all your help! On Thu, Mar 31, 2011 at 1:51 PM, Rogers Ochieng rogersochi...@gmail.com wrote: I've only been able to get below: From hq ph 1 (5001) For: +16178631001 ( 1... ) by : +16178631001 ( 1... ) if i restrict calling name i get From (5001) For: +16178631... by : +16178631001 ( 1... ) Using the VM profile solution i've managed below without (1...) or + From (5001) For: 16178631001 by : 16178631001 I don't know other solution, anyone out there with an idea? On 31 March 2011 19:04, adam compton com...@gmail.com wrote: All, I am desperately seeking help on how to alter the For and By fields on a CFUR to an SRST phone and a branch site. This is what I need to display on the phone: Forwarded 5001 For: +16178631001 ( 1... ) by : +16178631001 ( 1... ) I've been trying for a month to figure out how to display this with no luck. I've watched every video I know of in the IPexpert catalog, and nothing references it that I've seen. Any help would be appreciated, because i am at my wits end on this one. Adam Compton ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cbarge in SRST nt working !
Usually this is because you are using Single Button Cbarge on CUCM which breaks Cbarge in SRST. It's a known issue. Use standard Cbarge in CUCM and it should work in SRST. On Mar 29, 2011, at 9:08 AM, adam compton wrote: Do you see the CBarge softkey but the conference gives you a busy signal? or is it that CBarge doesn't show up at all? One thing I can think is to make sure the dspfarm is registered to the correct SCCP ccm group. in a SRST scenario, you would probably need a sccp ccm group for call manager and one for CME. On Tue, Mar 29, 2011 at 11:35 AM, Rahul Kapor rahul.kapo...@gmail.com wrote: Hi Roger , I added soft key template no help. Hi Bo type cannot be added because mac address is not specified ephone config is ephone 1 privacy off Can body send me working config for CBarge in SRST ? thx, Rahul On Tue, Mar 29, 2011 at 12:36 PM, Rogers Ochieng rogersochi...@gmail.com wrote: Add ephone-template 1 softkeys remote-in-use Cbarge NewCall And also no huntstop on the conference ephone-dn On 28 March 2011 19:01, Rahul Kapor rahul.kapo...@gmail.com wrote: Hi all , Cbarge in SRST not working here is my config ephone-dn-template 1 call-forward busy 914082026002 call-forward noan 914082026002 timeout 3 ephone-template 1 softkeys idle Redial Newcall Cfwdall ephone-dn 10 octo-line number conference ad-hoc ephone 1 privacy off device-security-mode none ephone 2 privacy off device-security-mode none telephony-service sdspfarm units 1 sdspfarm tag 1 HQ-CONF no privacy conference hardware srst mode auto-provision none srst ephone template 1 srst dn template 1 srst dn line-mode octo max-ephones 15 max-dn 15 ip source-address 14.160.116.40 port 2000 system message you are in fallback voicemail 914082026002 max-conferences 12 gain -6 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 1.1.1.1 14.160.116.40 transfer-system full-consult create cnf-files version-stamp 7960 Mar 27 2011 01:04:02 Phones gets registered to SRST and shared line is seen on phone display. i created octo dn for conf and conf bridge is registered to SRST. Please let me know if i am missing any thing. thx, Rahul ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cancelled Classes?
I thought I was the only one who picked up on that. :-) Although, it's 81 degrees in Phoenix with sunny skies. So technically San Jose is in a crisis situation compared to Phoenix. On Feb 16, 2011, at 7:46 AM, Rrcrumm wrote: Lol the weather has the city on lockdown Sent from my iPhone On Feb 15, 2011, at 11:02 PM, Vik Malhi vma...@ipexpert.com wrote: No - San Jose classes are not affected. We have had some terrible weather in San Jose this week which could affect classes too. For example it was only 60 degrees and had some light rain earlier today. So assuming there are no adverse weather conditions we should be good to go:-) Vik Malhi - CCIE#13890 Managing Partner / Instructor - IPexpert Inc Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join IPexpert's Free CCIE Peer Groups Study Communities at www.IPexpert.com/communities On Feb 15, 2011, at 18:58, Bill Lake whl...@gmail.com wrote: Hey this is not going to effect any of the classes in San Jose? I have one scheduled for April and I have to buy air tickets and take vacation at work. On Tue, Feb 15, 2011 at 7:28 PM, Wayne Lawson groupst...@ipexpert.com wrote: Sam, I'd love the opportunity to chat with you. Call me. 810.334.1564. Regards, Wayne A. Lawson II - CCIE #5244 (RS) Founder, President CEO - IPexpert, Inc., Proctor Labs, Inc. Platinum Solutions Group, LLC. Mailto: wlaw...@ipexpert.com Telephone: +1.810.334.1564 eFax: +1.810.454.0244 ::Message sent from iPhone IPexpert Proctor Labs are premier providers of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities and our public website at www.ipexpert.com or www.proctorlabs.com. CCIE-focused job community located at www.platinumplacementservices.com. Connect @ www.WayneLawson.com. On Feb 15, 2011, at 7:11 PM, Sam smgm...@gmail.com wrote: Anyone heard about anything that might be going on with instructors? A colleague I work with has had two of the voice classes he was scheduled to attend cancelled. They were both in Columbus. I am wondering if it might have anything to do with Amy Ryan leaving to work for Cisco. I wanted to go to one, but San Jose is just too far for me to drive (paying my own way and I'm in Chicago, so I need to drive), and if they aren't going to hold any more in Columbus then I'm not sure what I am going to do. Anyone heard anything? Sam ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Need assistance in setting up a Home Lab
I have a few pieces of gear that I'm willing to part with if anyone is interested. Primarily 2811's and 2821 fully equipped, which works great with the IPX labs. It's best to take this discussion off the list, but I'd much rather see my equipment find a home with someone who is working towards the CCIE rather than list it on ebay. On Feb 4, 2011, at 7:07 PM, Ashwani Ranpise wrote: ccied1ot, How much do you want for 2811 with AIM-CUE and NM-HD2ve ? I am in North Carolina. Thanks, Ashwani From: ccie_voice-boun...@onlinestudylist.com on behalf of ccieid1ot Sent: Fri 2/4/2011 3:54 PM To: khaled Saholy Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Need assistance in setting up a Home Lab I still have some pieces left. I'm willing to let it go for cheap. Let me know. Oh and thanks. Was a long journey and I probably wouldn't have done it again. duy ccie #27737 voice tmobile g2 On Feb 4, 2011 1:48 PM, khaled Saholy khaled_sah...@hotmail.com wrote: Thanks cciedot for the information and congratulation for your ccie. I'll check the price of your lab setup and see my budget. Note: The email before the last one was not sent by me , it's duplicated. Regards. Khaled Date: Fri, 4 Feb 2011 13:22:31 -0600 Subject: Re: RE: [OSL | CCIE_Voice] Need assistance in setting up a Home Lab From: ccieid...@gmail.com To: khaled_sah...@hotmail.com CC: ccie_voice@onlinestudylist.com The nm-hd-2ve has 30 channels onboard so no pvdm2 is needed. I used the 2811 because it supports cme 7.x. that is needed as the other routers doesn't support it. duy ccie #27737 voice tmobile g2 On Feb 4, 2011 1:07 PM, khaled Saholy khaled_sah...@hotmail.com wrote: It seems both modules are the same but NM-HD-2VE with predefined no of DSP and with NM-HDV2 , expansion is supported. For Site C , why you chose 2811 and not like 2621xm or 3745 as they are cheaper? Anyhow, are all features that should be tested in lab V3 are there in your lab, I mean didn't you face any kind of limitation during your practic for the lab? Khaled Date: Fri, 4 Feb 2011 12:29:23 -0600 Subject: Re: [OSL | CCIE_Voice] Need assistance in setting up a Home Lab From: ccieid...@gmail.com To: khaled_sah...@hotmail.com CC: ccie_voice@onlinestudylist.com So does the NM-HD-2VE. It supports the new way of configuring transcoders and conf-b. Here was my setup. HQ 2610XM NM-HDV-24 VWic-1MFT-T1 WIC-1T SB 1760 PVDM-4 VWIC-1MFT-T1 WIC-1T SC 2811 AIM-CUE VWIC-1MFT-E1 WIC-1T NM-HD-2VE PSTN/WAN 3640 NM-HD-2VE VWIC-2MFT-T1 VWIC-1MFT-T1 NM-4T 2011/2/4 khaled Saholy khaled_sah...@hotmail.com Hi I chose NM-HDV2 because it supports PVDM2 , see this note from Cisco site: PVDM and PVDM2 modules are not interchangeable. Use PVDM modules with the NM-HDV network module only, and use PVDM2 modules with the NM-HDV2 network module only. http://www.cisco.com/en/US/docs/routers/access/interfaces/nm/hardware/installation/guide/Conntvoi.html It seems you already got IE number in voice , what's the hardware you used for lab? Regards. Khaled Date: Fri, 4 Feb 2011 10:38:46 -0600 Subject: Re: [OSL | CCIE_Voice] Need assistance in setting up a Home Lab From: ccieid...@gmail.com To: khaled_sah...@hotmail.com CC: ccie_voice@onlinestudylist.com I have some vwics wic-1t aim-cue and 3550 switch, also a nm-hd-2ve. does the same as nm-hdv2. But has 2 slots. duy ccie #27737 voice tmobile g2 On Feb 4, 2011 8:54 AM, khaled Saholy khaled_sah...@hotmail.com wrote: Hi friends, I'm planning to buy home lab for CCIE Voice and I spent some days in searching for the right stuff. So, I'd like to share it here to know if I go straight and order it or I missed any part. About the lab which contain Cisco 2801 or 2811 , I could use them but you know because of budget limitation I thought of other models. I already have a PC with QuadCore CPU and 8 GB of RAM and here is the lab components: PSTN/FrameRelay router: 3745 2 x NM-HDV2 1 x VWIC-2MFT-T1 1 x VWIC-1MFT-E1 2 x PVDM2-16 3 x WIC-1T HQ Router: 2621XM 1 x NM-HDV2 1 x VWIC-1MFT-T1 1 x WIC-1T 2 x PVDM2-16 Branch1 Router: 2621XM 1 x NM-HDV2 1 x VWIC-1MFT-T1 1 x WIC-1T 1 x PVDM2-16 Branch2 Router: 3745 1 x AIM-CUE 1 x NM-HDV2 1 x VWIC-1MFT-E1 1 x WIC-1T 1 x PVDM2-16 1 x 7960 IP Phone 1 x 7940 IP Phone 2 x 7961 IP Phone 2 x 7941 IP Phone 1 x WS-C3550-24PWR-SMI 4 x CAB-6060-3FT BACK-TO-BACK CABLE FOR WIC-1T TO 1T Waiting for your feedback. Thanks and regards. Khaled Al-Saholy -- duy CCIE #27737 Voice ___ For more
Re: [OSL | CCIE_Voice] VUE ID for the LAB.
I only had to show my driver's license. I took my first attempt in RTP and the second in San Jose. On Jan 26, 2011, at 10:22 AM, Matteo B. wrote: Hello People... next Friday i'm going to sit for the lab...( cross your finger!! ) On the confirmation email there is a note that say i've to bring with me my VUE ID. On my profile on vue web site there is no mention of a VUE ID. On my written score report there is a Candidate ID, that is the one i use to log in into the ccie site, is this the id i have to bring with me? otherwise where i can find this VUE ID? cheers Matteo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] why trunk mode on ESW?
I recommend that you know both ways. The lab might tell you what you cannot do which means there is only one other option remaining in order to get the question right. On Jan 18, 2011, at 11:18 PM, bruno wrote: Dear all in vol1 network infrastrure, why we need to configure trunk mode on esw,why not access mode .i have test the access mode is ok. SITEB(config)#int range f0/1/0 -3 SITEB(config-if-range)# switchport trunk native vlan 602 SITEB(config-if-range)# switchport mode trunk SITEB(config-if-range)# switchport voice vlan 502 SITEB(config-if-range)#description ***CONNECT TO IP PHONE*** SITEC#show interfaces f0/1/0 switchport Name: Fa0/1/0 Switchport: Enabled Administrative Mode: trunk Operational Mode: trunk Administrative Trunking Encapsulation: dot1q Operational Trunking Encapsulation: dot1q Negotiation of Trunking: Disabled Access Mode VLAN: 0 ((Inactive)) Trunking Native Mode VLAN: 602 (DATA-VLAN) Trunking VLANs Enabled: ALL Trunking VLANs Active: 1,502,602 Protected: false Priority for untagged frames: 0 Override vlan tag priority: FALSE Voice VLAN: 502 Appliance trust: none Best Regards, bruno ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Barge is not working in SRST
If you configure Single Button cBarge in CUCM then it won't work in SRST. On Jan 18, 2011, at 11:29 AM, Amit Batra wrote: Hello guys May be I am wrong. But I kind of remember this issue. People talked a lot about it. And is a known bug. If this is the same issue which we are talking about . I guess restarting the router was the fix. Regards Amit Sent from my iPhone On 19/01/2011, at 4:50, Vik Malhi vma...@ipexpert.com wrote: I would first test a ad hoc conference call. Does this work? sh sccp connections to confirm the hw cfb is being used. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Mritunjay Kumar mjs...@gmail.com Date: Tue, 18 Jan 2011 15:49:00 +0530 To: Miron Kobelski findko...@gmail.com, Shrini linuxbos...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Barge is not working in SRST Hi Shrini, http://onlinestudylist.com/archives/ccie_voice/2010-November/071295.html privacy off under ephone-template does not work as this is a bug. it is tested and discussed at http://onlinestudylist.com/archives/ccie_voice/2010-November/071295.html but i tried with ephone template but did not work either :( attaching sh runn . Regards, Mritunjay On Tue, Jan 18, 2011 at 2:16 PM, Miron Kobelski findko...@gmail.com wrote: Hi, I suggest to follow Randall's suggestion, as this has already been discussed many times... regards kobel On Mon, Jan 17, 2011 at 11:35, Mritunjay Kumar mjs...@gmail.com wrote: Hi All, thanks for reply and sorry for delayed response. set the max-ephone to 2 and phone type is set but still not working :( is octo number shared by two phone which is used for Cbarge is adhoc conference number. Is there any bug or workaround. same config was working earlier! Regards, Mritunjay On Sun, Jan 16, 2011 at 8:55 PM, Randall Saborio ill2...@gmail.com wrote: This problem has come up quite a few times on this mailing list. You can go to the archives and do a search and you should find the solution: http://onlinestudylist.com/archives/ccie_voice/ On Sun, Jan 16, 2011 at 6:26 AM, Mritunjay Kumar mjs...@gmail.com wrote: Hi All , Cbarge in SRST is not working here is the config telephony-service sdspfarm units 2 sdspfarm tag 1 BR1-CNF no privacy conference hardware srst mode auto-provision none srst dn line-mode dual max-ephones 20 max-dn 20 ip source-address 14.160.116.40 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp Jan 01 2002 00:00:00 BR1# ephone-template 1 softkeys remote-in-use CBarge ephone-dn 1 octo-line number no-reg primary ephone-dn 2 octo-line number conference ad-hoc ephone-dn 3 dual-line number 3001 no-reg primary preference 5 ephone-dn 4 dual-line number 3002 no-reg primary preference 5 ephone 1 privacy off device-security-mode none mac-address 0026.CBBE.E8C9 ephone-template 1 button 1:3 2:1 ephone 2 privacy off device-security-mode none mac-address 0026.CBBE.EC4F ephone-template 1 button 1:4 2:1 hardware conf is registered . privacy is disabled under telephony service and in ephone. CME version 7.1 any missing config here ? Please suggest. Regards, Mritunjay ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Randall da ill Saborio CCIE Voice Wannabe #10054675811 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding
Re: [OSL | CCIE_Voice] Inbound SIP call to CUCM from CUBE goes unanswered
Like Matt said, perform a sip debug. You will most likely see that CUCM is responding with something. It may be something like 404 not found. If that is the case look at the Called number in the SIP Invite and make sure it matches your dial plan in CUCM. On Dec 31, 2010, at 9:00 AM, Matthew Berry wrote: You could do a debug ccsip messages on the CUBE and see what's taking place in the SIP messages between the gateway and CUCM. Send that on over for us to take a look at. Matthew Berry Sr. Voice Engineer - CCIE 26721 F906EF75-D025-431C-B55C-27FF496CF05D[6].png CDW Advanced Technology Services 7145 Boone Avenue North | Brooklyn Park, MN 55428 Single Number Reach: +1.763.592.5987 matthew.ber...@cdw.com From: ccielab...@gmail.com ccielab...@gmail.com Date: Fri, 31 Dec 2010 08:59:20 -0600 To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Inbound SIP call to CUCM from CUBE goes unanswered I'm testing a cube configuration in my lab setup. I have H.323 coming from CME to CUBE running on R1 and then SIP to the CUCM via a SIP trunk. I see the proper dialpeers being triggered in CUBE, but the CUCM doesn't seem to respond to the SIP call setup inbound. Calls from CUCM to CME via CUBE work , so I'm pretty confident the SIP trunk is functional. Short of trying to look through CUCM traces, is there a good debug on R1/Cube that would provide some insight into whats going on? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Thank you
I want to say thank you to everyone on the OSL who has participated in any of my discussions or helped resolve issues that I encountered. I went to San Jose for my second attempt on Friday and received the news yesterday that I passed. CCIE #27384. Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Policy Map - set dscp vs. set ip dscp
I want to create a qos policy to police sip traffic on my HQ 3750 and remark excess to dscp 0. Can someone explain the difference in a policy-map between 'set dscp 24' and 'set ip dscp 24'? Also, is it accurate to set the burst to 8000 or should it be a minimum of 16000 burst, or is it personal preference? mls qos map policed-dscp 24 26 to 0 class-map match-any cmap-sip match access-group 100 access-list 100 remark SIP access-list 100 permit udp any eq 5060 any access-list 100 permit tcp any eq 5060 any policy-map pmap-sip class cmap-sip set dscp 24 police 32000 8000 exceed-action policed-dscp-transmit policy-map pmap-mgcp class cmap-mgcp set ip dscp 24 police 32000 8000 exceed-action policed-dscp-transmit ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call Forward Unregistered
I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and VMWare. If you go to the Device Phone and click on the Site B phones Line and specifically assign the Voicemail Profile to the Line it might work. I had success a couple of times doing this, but then after resetting my rack the last time and assigning the VM profile to the Line I still had this issue. On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote: Scenario: In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway cme HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits dialing in SRST.(Wan failure) I use call forward unregistered feature. When I call from HQ Phone-1 call routed through HQ Gateway. When I call from Site-C Phone-1 call routed through the GK first and then HQ Gateway. Below is the display I am getting on my Site-B phone display. Forward HQ Phone 1 (2001) For 3001 By3001 Forward Site-C Phone 1 (4001) For 3001 By3001 My question how can I achieve below display in FOR and BY field it should be E.164 number format and than 4 digits internal ID Forward (2001) For +19723033001 (3...) By+19723033001 (3...) Forward (4001) For +19723033001 (3...) By+19723033001 (3...) Thanking you in anticipation folks. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call Forward Unregistered
I think the main thing to understand is that it should work using E164 in For/By under normal circumstances and everything else we are suggesting is a work around to a known bug with CUCM 7.0 and VMWare. On Oct 17, 2010, at 3:56 PM, Daniel Berlinski wrote: Hello guys If you want to manipulate this with CUCM the place to change the redirected number is the VM profile as indicated by Mark. Alternatively you could attach an additional rule to the translation-profile plugged inbound to the POTS call leg in the branch router in SRST mode and configure it to change the redirect-called number from to the e164 that you are after. Cheers On Mon, Oct 18, 2010 at 11:36 AM, Mark Holloway m...@markholloway.com wrote: I ran into this same issue. Supposedly it's a problem with CUCM 7.0 and VMWare. If you go to the Device Phone and click on the Site B phones Line and specifically assign the Voicemail Profile to the Line it might work. I had success a couple of times doing this, but then after resetting my rack the last time and assigning the VM profile to the Line I still had this issue. On Oct 17, 2010, at 3:28 PM, Afzal Bhutta wrote: Scenario: In SRST mode: HQ MGCP gateway and Site-B H322 gateway Site-C H322 gateway cme HQ and Site C phones are being able to call SiteB Phone 1 using 4 digits dialing in SRST.(Wan failure) I use call forward unregistered feature. When I call from HQ Phone-1 call routed through HQ Gateway. When I call from Site-C Phone-1 call routed through the GK first and then HQ Gateway. Below is the display I am getting on my Site-B phone display. Forward HQ Phone 1 (2001) For 3001 By3001 Forward Site-C Phone 1 (4001) For 3001 By3001 My question how can I achieve below display in FOR and BY field it should be E.164 number format and than 4 digits internal ID Forward (2001) For +19723033001 (3...) By+19723033001 (3...) Forward (4001) For +19723033001 (3...) By+19723033001 (3...) Thanking you in anticipation folks. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE Upgrading and Licensing
I had to upgrade my CUE from 2.1.2 to 7.01. http://www.markholloway.com/blog/?p=595 On Oct 12, 2010, at 3:33 AM, Amr Sherif wrote: Hello Experts, I have CUE version 3.2.3 and i want to upgrade to 7.0 which is the exam version . The license for my CUE is CME mode and it's embedded inside the system when i bought it so i dont have the license as external file , So here is my concern ; If i upgrade to version 7.0 and download CME CCM licenses from my cisco account ,is this will work okay or the license have to be purchased not be downloaded. Another Concern, If somehow after upgrading to ver. 7.0 the license is not work ,or any problem just show up ,can i reverse back to my version 3.2.3 and the built-in license by executing the command offline then restore factory default. Please any help would be appreciated. Best regards, Amr Sherif Senior Network Voice Engineer CCNA,CCNP,CCVP and CCIE Voice Written (Certified) CCIE Voice Lab (In Progress) ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Do Not Disturb odd behavior
Has anyone ever experienced an issue where you assign DnD to a softkey template in CUCM, assign that template to a phone (which also has voicemail in Unity Connection), but when a call comes into that phone with DnD, it still rings instead of going to voicemail even if DnD is Active? Even if I press DnD while the call is coming in it still does not go to voicemail. I've got Forward Busy, Internal, External, No Answer, and Unregistered set to go to voicemail. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Do Not Disturb odd behavior
Hmm, I actually had to go to the Device Phone HQPH1 and scroll down to Do Not Disturb and hard-set it to Call Reject (instead of Default Profile Behavior). I thought the default behavior was in fact Reject. On Oct 12, 2010, at 11:46 AM, Mark Holloway wrote: Has anyone ever experienced an issue where you assign DnD to a softkey template in CUCM, assign that template to a phone (which also has voicemail in Unity Connection), but when a call comes into that phone with DnD, it still rings instead of going to voicemail even if DnD is Active? Even if I press DnD while the call is coming in it still does not go to voicemail. I've got Forward Busy, Internal, External, No Answer, and Unregistered set to go to voicemail. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SRST in the ACTUAL LAB
I always shut down the Serial interface of the Frame Relay WAN link. On Oct 10, 2010, at 9:44 PM, Pithog Oil wrote: Hi experts, What is the best and quickest way to invoke SRST in the labs, for me the only way i have tested as at now is the ip expert proctorlabs way, of creating sub only group and stoping ths SUB, please is there a better way to do this in the labs. Pithog oil ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call Forward Unregistered
Ok, the secret to getting it to work every time is going to Device Phone Line and setting the voicemail profile to Default (or some voicemail profile). Even though None should use the system default voicemail profile, if you don't hard-set a voicemail profile the CFUR won't always show the external mask when the call is forwarded, but if you force a voicemail profile on the Line it will work. Thanks to both of you for your help. :) On Oct 9, 2010, at 8:58 AM, Vik Malhi wrote: Mark- can you try adding a new VM Profile for 3XXX with a MASK of the full number (the # that you want to display on the Unregistered phone). The only way to manipulate the Redirecting # in UCM is using the VM Profile. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Mark Holloway m...@markholloway.com Date: Fri, 8 Oct 2010 16:14:37 -0700 To: Mark Holloway m...@markholloway.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Call Forward Unregistered I have had it working before, but it's odd because sometimes when I reset the lab rack I can get it work and other times it does not work the way I want. I'm trying to figure out if I keep overlooking something. On Oct 8, 2010, at 4:08 PM, Mark Holloway wrote: I do not want to modify 5XXX. I want to modify 3XXX (the DN that is invoking CFUR) which is the Redirecting number. On Oct 8, 2010, at 4:02 PM, Prashant Patel wrote: Hi Mark, The easiest way is to use calling party Transformation on the outbound gateway. For example - 5002 calling 3002 out of local gateway. create a pt and assign it to a css. Assign css to the gateway calling party transformation css and uncheck use dp box. Now create a calling party transformation for 5XXX in the pt and modify the ANI to use extenal mask. This will modify the ANI from 5xxx to external mask everytime the 5xxx makes a call out of that gateway. HTH Prashant On Fri, Oct 8, 2010 at 6:39 PM, Mark Holloway m...@markholloway.com wrote: I'm trying to get my CFUR to work so it shows the External Mask in the For and By part of the call presentation but instead I am only getting it to show the 4 digit extension. For example, lets say HQ 5001 calls BR1 3001 (3001 is unregistered and has CFUR set in CUCM to dial out the PSTN because that site is in SRST mode). The presentation on the BR1 phones is Forwarded HqPh1 5001, For 3001 By 3001. Instead of 3001 I want to display the External Mask. Does anyone know the proper way to do this? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call Forward Unregistered
By the way, I don't get why it works this way, but it does work. It's just another one of those odd things you just have to know. Vik, I know you said voicemail is the only place a Redirecting number is modified, and Marcelo mentioned there is an issue CFUR and redirecting behavior in VMWare (I experience different behavior each time I reset my rack too), so as odd as it is I think it's important to know the Voicemail profile assignment is a valid fix. On Oct 9, 2010, at 9:39 AM, Mark Holloway wrote: Ok, the secret to getting it to work every time is going to Device Phone Line and setting the voicemail profile to Default (or some voicemail profile). Even though None should use the system default voicemail profile, if you don't hard-set a voicemail profile the CFUR won't always show the external mask when the call is forwarded, but if you force a voicemail profile on the Line it will work. Thanks to both of you for your help. :) On Oct 9, 2010, at 8:58 AM, Vik Malhi wrote: Mark- can you try adding a new VM Profile for 3XXX with a MASK of the full number (the # that you want to display on the Unregistered phone). The only way to manipulate the Redirecting # in UCM is using the VM Profile. -- Vik Malhi – CCIE #13890 Managing Partner / Instructor - IPexpert, Inc. Mailto: vma...@ipexpert.com Telephone: +1.810.326.1444 ext 420 Fax: +1.810.454.0130 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Wireless, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Mark Holloway m...@markholloway.com Date: Fri, 8 Oct 2010 16:14:37 -0700 To: Mark Holloway m...@markholloway.com Cc: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Call Forward Unregistered I have had it working before, but it's odd because sometimes when I reset the lab rack I can get it work and other times it does not work the way I want. I'm trying to figure out if I keep overlooking something. On Oct 8, 2010, at 4:08 PM, Mark Holloway wrote: I do not want to modify 5XXX. I want to modify 3XXX (the DN that is invoking CFUR) which is the Redirecting number. On Oct 8, 2010, at 4:02 PM, Prashant Patel wrote: Hi Mark, The easiest way is to use calling party Transformation on the outbound gateway. For example - 5002 calling 3002 out of local gateway. create a pt and assign it to a css. Assign css to the gateway calling party transformation css and uncheck use dp box. Now create a calling party transformation for 5XXX in the pt and modify the ANI to use extenal mask. This will modify the ANI from 5xxx to external mask everytime the 5xxx makes a call out of that gateway. HTH Prashant On Fri, Oct 8, 2010 at 6:39 PM, Mark Holloway m...@markholloway.com wrote: I'm trying to get my CFUR to work so it shows the External Mask in the For and By part of the call presentation but instead I am only getting it to show the 4 digit extension. For example, lets say HQ 5001 calls BR1 3001 (3001 is unregistered and has CFUR set in CUCM to dial out the PSTN because that site is in SRST mode). The presentation on the BR1 phones is Forwarded HqPh1 5001, For 3001 By 3001. Instead of 3001 I want to display the External Mask. Does anyone know the proper way to do this? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CME phones and max number of supported calls
If I want limit BR2Ph1 to 3 incoming calls and BR2Ph2 to 6 incoming calls, how can I control the total number of incoming calls to each phone if there is more than one ephone-dn assign to the phone? For example, if 6001 is an octo line assigned to Ph1, 6002 is an octo line assigned to Ph2, and 6003 is an octo line shared on both phones. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site
Hmm, PSTN to BR1 and IP to IP (inter and intra site) play multicast MoH piano music from route flash just fine, but for some reason when calling from BR1 to the PSTN and pressing HOLD on the BR1 phone it plays beep beep beep. Usually the issue is PSTN to IP because you need a voice class codec on the SUB/PUB dial peers that support G711, which I have, and PSTN to BR1 piano music streams multicast ok. Not sure what would cause IP to PSTN calls to fail streaming MoH and play beep beep beep. Any ideas? On Oct 7, 2010, at 1:36 PM, ayman labib wrote: Thanks for the reply. As it turns out. Loopback interface is a required step. Now everything is working. Thanks Next challenge is to get Site HQ and SRST to use MoH with CME using the Gatekeeper. Thanks From: ayman labib ayman_la...@yahoo.com To: amr thabt amrth...@gmail.com Cc: ccie_voice@onlinestudylist.com Sent: Thu, October 7, 2010 3:49:41 PM Subject: Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site Thanks for the reply. I do have the max ephone etc.. I removed my config to keep it short. I tried it with bind command and without. Same Issue. I don't have Lo0 configured. Everything is configured using the fa0/1 interface. Please have a look at the screen shots of my config. I really appreciate everyone's help. 2 days and it's driving me crazy. call-manager-fallback secondary-dialtone 9 max-conferences 8 gain -6 transfer-system full-consult ip source-address 192.168.31.10 port 2000 strict-match max-ephones 10 max-dn 10 transfer-pattern .T voicemail 912123945020 call-forward pattern .T call-forward busy 12123945020 call-forward noan 12123945020 timeout 20 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 192.168.31.10 time-zone 8 ! From: amr thabt amrth...@gmail.com To: ayman labib ayman_la...@yahoo.com Cc: ccie_voice@onlinestudylist.com Sent: Thu, October 7, 2010 3:07:59 PM Subject: Re: [OSL | CCIE_Voice] MoH to PSTN from SRST site Hi Ayman, I have three comments that may help 1 Do you add max-dn and max-ephone under call-manager-fallback 2-in ccm-manager music-on-hold bind fa0/1 remove the bind use only ccm-manager music-on-hold 3- in multicast command add both loopback and VLan SVI ip address. HTH AMR On Thu, Oct 7, 2010 at 9:56 PM, ayman labib ayman_la...@yahoo.com wrote: Just wondering if anyone encountered this problem. I still can't get MOH when calling the PSTN phone and the site is not in SRST mode. According to the sh command below. The call manager has done its job but the GWY is not responding. Any ideas? MOH local and between HQ works fine. Just need a sanity check. Thanks for all your help SRST-Site#sh ccm-manager music-on-hold Current active multicast sessions : 1 Multicast RTP port Packets Call CodecIncoming Address number in/outidInterface === 239.1.1.1 16384 0/0 12 g711ulaw ccm-manager music-on-hold bind fa0/1 call-manager-fallback ip source-address 192.168.31.10 port 2000 strict-match moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 192.168.31.10 http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html#wp1046789 From: ayman labib ayman_la...@yahoo.com To: ccie_voice@onlinestudylist.com Cc: ccie_voice@onlinestudylist.com Sent: Wed, October 6, 2010 9:45:12 AM Subject: MoH to PSTN from SRST site Hello Experts, Follow up to Mark's email about Moh to PSTN. I don't hear the Piano music as well. Inter-site and Intra-site with HQ works. I see the Muticast on the gateway is invoked and on the server, but don't hear anything. Any idea? Thanks in advance admin:show perf query class Cisco MOH Device ==query class : - Perf class (Cisco MOH Device) has instances and values: MOH_2 - MOHHighestActiveResources = 1 MOH_2 - MOHMulticastResourceActive = 0 MOH_2 - MOHMulticastResourceAvailable = 25 MOH_2 - MOHOutOfResources = 0 MOH_2 - MOHTotalMulticastResources = 25 MOH_2 - MOHTotalUnicastResources = 250 MOH_2 - MOHUnicastResourceActive = 0 MOH_2 - MOHUnicastResourceAvailable= 250 MOH_3 - MOHHighestActiveResources = 1 MOH_3 - MOHMulticastResourceActive = 1 MOH_3 - MOHMulticastResourceAvailable = 24 MOH_3 - MOHOutOfResources = 0 MOH_3 - MOHTotalMulticastResources = 25 MOH_3 - MOHTotalUnicastResources = 250 MOH_3 - MOHUnicastResourceActive = 0 MOH_3 - MOHUnicastResourceAvailable= 250
[OSL | CCIE_Voice] UCCX Prompt
Does anyone know if/what UCCX wav file says Please try again later Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Unity Connection - Error when trying to record a customer greeting
Has anyone ever seen this before? I login to Unity Connection then click on my BR1PH1 user so I can record a custom greeting. inline: PastedGraphic-2.png When I press the Record button I get the following error. inline: PastedGraphic-3.png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Call Forward Unregistered
I'm trying to get my CFUR to work so it shows the External Mask in the For and By part of the call presentation but instead I am only getting it to show the 4 digit extension. For example, lets say HQ 5001 calls BR1 3001 (3001 is unregistered and has CFUR set in CUCM to dial out the PSTN because that site is in SRST mode). The presentation on the BR1 phones is Forwarded HqPh1 5001, For 3001 By 3001. Instead of 3001 I want to display the External Mask. Does anyone know the proper way to do this? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call Forward Unregistered
I do not want to modify 5XXX. I want to modify 3XXX (the DN that is invoking CFUR) which is the Redirecting number. On Oct 8, 2010, at 4:02 PM, Prashant Patel wrote: Hi Mark, The easiest way is to use calling party Transformation on the outbound gateway. For example - 5002 calling 3002 out of local gateway. create a pt and assign it to a css. Assign css to the gateway calling party transformation css and uncheck use dp box. Now create a calling party transformation for 5XXX in the pt and modify the ANI to use extenal mask. This will modify the ANI from 5xxx to external mask everytime the 5xxx makes a call out of that gateway. HTH Prashant On Fri, Oct 8, 2010 at 6:39 PM, Mark Holloway m...@markholloway.com wrote: I'm trying to get my CFUR to work so it shows the External Mask in the For and By part of the call presentation but instead I am only getting it to show the 4 digit extension. For example, lets say HQ 5001 calls BR1 3001 (3001 is unregistered and has CFUR set in CUCM to dial out the PSTN because that site is in SRST mode). The presentation on the BR1 phones is Forwarded HqPh1 5001, For 3001 By 3001. Instead of 3001 I want to display the External Mask. Does anyone know the proper way to do this? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Call Forward Unregistered
I have had it working before, but it's odd because sometimes when I reset the lab rack I can get it work and other times it does not work the way I want. I'm trying to figure out if I keep overlooking something. On Oct 8, 2010, at 4:08 PM, Mark Holloway wrote: I do not want to modify 5XXX. I want to modify 3XXX (the DN that is invoking CFUR) which is the Redirecting number. On Oct 8, 2010, at 4:02 PM, Prashant Patel wrote: Hi Mark, The easiest way is to use calling party Transformation on the outbound gateway. For example - 5002 calling 3002 out of local gateway. create a pt and assign it to a css. Assign css to the gateway calling party transformation css and uncheck use dp box. Now create a calling party transformation for 5XXX in the pt and modify the ANI to use extenal mask. This will modify the ANI from 5xxx to external mask everytime the 5xxx makes a call out of that gateway. HTH Prashant On Fri, Oct 8, 2010 at 6:39 PM, Mark Holloway m...@markholloway.com wrote: I'm trying to get my CFUR to work so it shows the External Mask in the For and By part of the call presentation but instead I am only getting it to show the 4 digit extension. For example, lets say HQ 5001 calls BR1 3001 (3001 is unregistered and has CFUR set in CUCM to dial out the PSTN because that site is in SRST mode). The presentation on the BR1 phones is Forwarded HqPh1 5001, For 3001 By 3001. Instead of 3001 I want to display the External Mask. Does anyone know the proper way to do this? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] QoS Policy Map
I'm trying to create a policy map that matches the skinny signaling protocol that will police it and rewrite the exceeded packets from dscp 24 to 0. I am pretty sure I have the policy map created correctly but when I do 'show policy-map interface interface' I am not seeing the counters increment. Am I missing something? ## Cat 3750 ## mls qos map policed-dscp 24 to 0 access-list 100 remark SKINNY access-list 100 permit tcp any eq 2000 any access-list 100 permit tcp any eq 2001 any access-list 100 permit tcp any eq 2002 any class-map match-any class-map-skinny match access-group 100 policy-map policy-map-voip-signal class class-map-skinny set dscp cs3 police 32000 8000 exceed-action policed-dscp-transmit interface FastEthernet1/0/1 description Trunk Port to Router switchport trunk encapsulation dot1q switchport mode trunk service-policy input policy-map-voip-signal sw1#show policy-map int fast 1/0/1 FastEthernet1/0/1 Service-policy input: policy-map-voip-signal Class-map: class-map-skinny (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: access-group 100 0 packets, 0 bytes 5 minute rate 0 bps Class-map: class-default (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: any 0 packets, 0 bytes 5 minute rate 0 bps sw1#show mls qos QoS is enabled QoS ip packet dscp rewrite is enabled ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
My gateway is H323. I can do the same exact configuration on an MGCP gateway and it works. I always need to do no mgcp/mgcp as well. Have you run into this situation on an H323 gateway? On Oct 5, 2010, at 6:53 PM, Kalyan iyer wrote: Hey Mark, I ran into the same problem with MOH. You have the correct configuration. However, If your BR1 RTR is a MGCP GW, like I had you will need to do a no mgcp / mgcp to make the MOH work. Thanks Kalyan On Sun, Oct 3, 2010 at 9:39 PM, David Lee d16...@gmail.com wrote: Hey Mark, Check the MRGL of the voice gateway. The phone where you press hold -- from this phone is the source determined. But the MOH is taken from the MRGL configured on the holdee, in this case the VG. Thanks, -Dave On Sun, Oct 3, 2010 at 9:08 PM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. MoH SRST (Stream from Flash)` (Mark Holloway) 2. Re: MoH SRST (Stream from Flash)` (Prashant Patel) 3. Re: MoH SRST (Stream from Flash)` (James Key) 4. Re: MoH SRST (Stream from Flash)` (Mark Holloway) -- Message: 1 Date: Sun, 3 Oct 2010 17:17:44 -0700 From: Mark Holloway m...@markholloway.com To: CCIE Voice Maillist ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)` Message-ID: 85912468-288c-4ecc-9e9e-f9d9d22a3...@markholloway.com Content-Type: text/plain; charset=us-ascii I thought I had this figured out but I'm slipping up somewhere. Could use some help. :) I'm configuring multicast moh at BR1 using G.711 and streaming from BR1 router flash. BR1 is an H323 gateway. call-manager-fallback max-dn 24 max-ephones 2 ip source address 10.20.30.254 this is the voice vlan default gateway moh music-on-hold.au piano music file in flash multicast moh 239.1.1.1 port 16384 route voice vlan ip = 192.168.65.254 loop0 ip = 192.1.65.254 ip multicast-routing is enabled ip pim dense mode is configured on voice vlan interface and loop0 interface cucm moh audio source and PUB are configured for multicast routing (1 hop) and assigned to mrg br1_mcast_moh which is assigned to mrgl br1 which is assigned to br1 device pool I have a device pool 'MoH' that has a region 'MoH' and is assigned G711 to all other regions. This region is assign to device pool MoH, and device pool MoH is assign to the MoH servers. When HQ calls BR1 and HQ presses hold, the BR1 phone hears piano music. When PSTN calls BR1 and BR1 presses hold, PSTN hears beep beep beep r2# debug ephone moh EPHONE music-on-hold debugging is enabled Oct 4 00:09:50.794: MoH route If Vlan302 ETHERNET 192.168.65.254 via ARP Oct 4 00:09:50.794: MoH route If Loopback0 46 192.1.65.254 via 192.1.65.254 r2#debug ccm-m music-on-hold all Call Manager music-on-hold all debugging is on r2# Oct 4 00:16:08.156: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.156: moh_process_ccb: dstadr 192.168.65.30, callid -1, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.164: moh_process_ccb: dstadr 192.168.65.30, callid 11, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.164: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.032: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:17.264: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 disconnected from 911 , call lasted 9 seconds Oct 4 00:16:17.268: moh_update_rtp: callID 12 dstCallID 11 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20101003/e5357589/attachment-0001.html -- Message: 2 Date: Sun, 3 Oct 2010 20:20:43 -0400 From: Prashant Patel prashantpatel...@gmail.com To: Mark Holloway m...@markholloway.com Cc: CCIE Voice Maillist ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)` Message-ID: aanlktin8bnyo
Re: [OSL | CCIE_Voice] Single Number Reach
Fantastic. Thank you for taking the time to do this and for sharing it with everyone. On Oct 4, 2010, at 10:08 AM, Graham Hopkins wrote: Mark, having done some further tests, I now have this working - the key here is that the calling number transformation pattern matches the calling number at the time the route pattern was matched. So this is likely to be 2001 as I presume that the external phone number masked is applied as a transform on the route pattern. Therefore alter your calling party transform pattern to 2XXX ( or whatever the best pattern fro HQ is) and prefix the 555. Other sites will still show the full E.164 number. Graham On 1 Oct 2010, at 18:00, Mark Holloway wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
I created two calling party transformation patterns. 2XXX becomes HQ 7 digit external number 3XXX becomes BR1 10 digit external number No need to touch BR2 since everything else uses E164 anyway. On Oct 4, 2010, at 11:00 AM, Graham Hopkins wrote: Yeah just glad I finally got my head around it. For anyone having issues with how the various patterns are matched and the transformations are applied I suggest that is worth setting up the following: Translation pattern - RP RL Transformation Pattern at the Gateway and then applying changes at different points and monitoring the results to get a good understanding of where patterns are matched and which transformations overwrite other ones. Dial Number Analyzer is also pretty handy here if something doesn't match and you think it should Graham On 4 Oct 2010, at 18:41, Mark Holloway wrote: Fantastic. Thank you for taking the time to do this and for sharing it with everyone. On Oct 4, 2010, at 10:08 AM, Graham Hopkins wrote: Mark, having done some further tests, I now have this working - the key here is that the calling number transformation pattern matches the calling number at the time the route pattern was matched. So this is likely to be 2001 as I presume that the external phone number masked is applied as a transform on the route pattern. Therefore alter your calling party transform pattern to 2XXX ( or whatever the best pattern fro HQ is) and prefix the 555. Other sites will still show the full E.164 number. Graham On 1 Oct 2010, at 18:00, Mark Holloway wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
It's the strangest thing. I couldn't get Multicast MoH to work on my BR1 H323 router. I wiped out my call-manager-fallback configuration, re-entered everything, put my router in SRST mode (to practice other things) and just for the hell of it I tried testing Multicast MoH over the PSTN and it worked. I then brought up the Serial interface so the router came out of SRST mode and Multicast MoH is still working as expected. I didn't test Multicast MoH after rebuilding call-manager-fallback but before putting it into SRST. So I'm not sure exactly which one fixed it. However, I did try rebuilding call-manager-fallback a couple of times yesterday and it didn't fix it. My working configuration: call-manager-fallback max-dn 14 max-ephone 2 ip source-address Voice Vlan IP moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route Voice Vlan IP Loopback0 r2(config-subif)#do sh run | sec ccm-m ccm-manager music-on-hold bind Voice Vlan Number On Oct 4, 2010, at 7:30 PM, Prashant Patel wrote: Hi Mark, When you do a show perf query class Cisco MOH device on the server that has the MOH servers registered if you see an increment on the MOHOutOfResources then there is probably a codec mismatch and this increments the counter. The Device Pool assigned to the MOH server needs to have a region that does g711 with all HQ or BR1 or BR2 regions. HTH Prashant On Sun, Oct 3, 2010 at 9:08 PM, Mark Holloway m...@markholloway.com wrote: Sorry James..my mistake. It shouldn't be 10.20.30.254 but rather it should be 192.168.65.254. He it is again (proper) call-manager-fallback max-dn 24 max-ephones 2 ip source address 192.168.65.254 this is the voice vlan default gateway on Vlan302 moh music-on-hold.au piano music file in flash multicast moh 239.1.1.1 port 16384 route voice vlan ip = 192.168.65.254 loop0 ip = 192.1.65.254 ccm-manager music-on-hold bind Vlan302 ip multicast-routing is enabled ip pim dense mode is configured on voice vlan interface and loop0 interface cucm moh audio source and PUB are configured for multicast routing (1 hop) and assigned to mrg br1_mcast_moh which is assigned to mrgl br1 which is assigned to br1 device pool I have a device pool 'MoH' that has a region 'MoH' and is assigned G711 to all other regions. This region is assign to device pool MoH, and device pool MoH is assign to the MoH servers. When HQ calls BR1 and HQ presses hold, the BR1 phone hears piano music. When PSTN calls BR1 and BR1 presses hold, PSTN hears beep beep beep r2# debug ephone moh EPHONE music-on-hold debugging is enabled Oct 4 00:09:50.794: MoH route If Vlan302 ETHERNET 192.168.65.254 via ARP Oct 4 00:09:50.794: MoH route If Loopback0 46 192.1.65.254 via 192.1.65.254 r2#debug ccm-m music-on-hold all Call Manager music-on-hold all debugging is on r2# Oct 4 00:16:08.156: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.156: moh_process_ccb: dstadr 192.168.65.30, callid -1, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.164: moh_process_ccb: dstadr 192.168.65.30, callid 11, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.164: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.032: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:17.264: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 disconnected from 911 , call lasted 9 seconds Oct 4 00:16:17.268: moh_update_rtp: callID 12 dstCallID 11 On Oct 3, 2010, at 5:44 PM, James Key wrote: Mark, Looking at your config, a little confused on your ip source address under call-manager fallback and what you have for your route under multicast. One is listed as voice vlan gateway and the other is voice vlan ip, but two different networks. What you have listed for your CUCM config looks correct. Also, do you also have ccm-manager music-on-hold defined on the br1 router? believe this is needed for multicast even though an H323 gateway. James From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway [...@markholloway.com] Sent: Sunday, October 03, 2010 7:17 PM To: CCIE Voice Maillist Subject: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)` I thought I had this figured out but I'm slipping up somewhere. Could use some help. :) I'm configuring multicast moh at BR1 using G.711 and streaming from BR1 router flash. BR1 is an H323 gateway. call
Re: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
Ok, I should be more careful before emailing. When BR1 ph1 came out of SRST I made a call from the PSTN to ph1 and put the call on hold and I heard piano music. ph2 was still in SRST mode. When phone 2 came out of SRST mode and I repeat the same call, I hear beep beep beep. This is so odd. On Oct 4, 2010, at 10:16 PM, Mark Holloway wrote: It's the strangest thing. I couldn't get Multicast MoH to work on my BR1 H323 router. I wiped out my call-manager-fallback configuration, re-entered everything, put my router in SRST mode (to practice other things) and just for the hell of it I tried testing Multicast MoH over the PSTN and it worked. I then brought up the Serial interface so the router came out of SRST mode and Multicast MoH is still working as expected. I didn't test Multicast MoH after rebuilding call-manager-fallback but before putting it into SRST. So I'm not sure exactly which one fixed it. However, I did try rebuilding call-manager-fallback a couple of times yesterday and it didn't fix it. My working configuration: call-manager-fallback max-dn 14 max-ephone 2 ip source-address Voice Vlan IP moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route Voice Vlan IP Loopback0 r2(config-subif)#do sh run | sec ccm-m ccm-manager music-on-hold bind Voice Vlan Number On Oct 4, 2010, at 7:30 PM, Prashant Patel wrote: Hi Mark, When you do a show perf query class Cisco MOH device on the server that has the MOH servers registered if you see an increment on the MOHOutOfResources then there is probably a codec mismatch and this increments the counter. The Device Pool assigned to the MOH server needs to have a region that does g711 with all HQ or BR1 or BR2 regions. HTH Prashant On Sun, Oct 3, 2010 at 9:08 PM, Mark Holloway m...@markholloway.com wrote: Sorry James..my mistake. It shouldn't be 10.20.30.254 but rather it should be 192.168.65.254. He it is again (proper) call-manager-fallback max-dn 24 max-ephones 2 ip source address 192.168.65.254 this is the voice vlan default gateway on Vlan302 moh music-on-hold.au piano music file in flash multicast moh 239.1.1.1 port 16384 route voice vlan ip = 192.168.65.254 loop0 ip = 192.1.65.254 ccm-manager music-on-hold bind Vlan302 ip multicast-routing is enabled ip pim dense mode is configured on voice vlan interface and loop0 interface cucm moh audio source and PUB are configured for multicast routing (1 hop) and assigned to mrg br1_mcast_moh which is assigned to mrgl br1 which is assigned to br1 device pool I have a device pool 'MoH' that has a region 'MoH' and is assigned G711 to all other regions. This region is assign to device pool MoH, and device pool MoH is assign to the MoH servers. When HQ calls BR1 and HQ presses hold, the BR1 phone hears piano music. When PSTN calls BR1 and BR1 presses hold, PSTN hears beep beep beep r2# debug ephone moh EPHONE music-on-hold debugging is enabled Oct 4 00:09:50.794: MoH route If Vlan302 ETHERNET 192.168.65.254 via ARP Oct 4 00:09:50.794: MoH route If Loopback0 46 192.1.65.254 via 192.1.65.254 r2#debug ccm-m music-on-hold all Call Manager music-on-hold all debugging is on r2# Oct 4 00:16:08.156: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.156: moh_process_ccb: dstadr 192.168.65.30, callid -1, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.164: moh_process_ccb: dstadr 192.168.65.30, callid 11, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.164: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.032: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:17.264: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 disconnected from 911 , call lasted 9 seconds Oct 4 00:16:17.268: moh_update_rtp: callID 12 dstCallID 11 On Oct 3, 2010, at 5:44 PM, James Key wrote: Mark, Looking at your config, a little confused on your ip source address under call-manager fallback and what you have for your route under multicast. One is listed as voice vlan gateway and the other is voice vlan ip, but two different networks. What you have listed for your CUCM config looks correct. Also, do you also have ccm-manager music-on-hold defined on the br1 router? believe this is needed for multicast even though an H323 gateway. James From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com
Re: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
Yeah, I have a voice-class codec assigned to the dial-peers with g711u first and g729r8 second. On Oct 4, 2010, at 10:21 PM, Daniel Berlinski wrote: Check that your voip dial-peers facing the CUCM leg have voice-class codec configured with g729 and g711 support. On Tue, Oct 5, 2010 at 6:16 PM, Mark Holloway m...@markholloway.com wrote: It's the strangest thing. I couldn't get Multicast MoH to work on my BR1 H323 router. I wiped out my call-manager-fallback configuration, re-entered everything, put my router in SRST mode (to practice other things) and just for the hell of it I tried testing Multicast MoH over the PSTN and it worked. I then brought up the Serial interface so the router came out of SRST mode and Multicast MoH is still working as expected. I didn't test Multicast MoH after rebuilding call-manager-fallback but before putting it into SRST. So I'm not sure exactly which one fixed it. However, I did try rebuilding call-manager-fallback a couple of times yesterday and it didn't fix it. My working configuration: call-manager-fallback max-dn 14 max-ephone 2 ip source-address Voice Vlan IP moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route Voice Vlan IP Loopback0 r2(config-subif)#do sh run | sec ccm-m ccm-manager music-on-hold bind Voice Vlan Number On Oct 4, 2010, at 7:30 PM, Prashant Patel wrote: Hi Mark, When you do a show perf query class Cisco MOH device on the server that has the MOH servers registered if you see an increment on the MOHOutOfResources then there is probably a codec mismatch and this increments the counter. The Device Pool assigned to the MOH server needs to have a region that does g711 with all HQ or BR1 or BR2 regions. HTH Prashant On Sun, Oct 3, 2010 at 9:08 PM, Mark Holloway m...@markholloway.com wrote: Sorry James..my mistake. It shouldn't be 10.20.30.254 but rather it should be 192.168.65.254. He it is again (proper) call-manager-fallback max-dn 24 max-ephones 2 ip source address 192.168.65.254 this is the voice vlan default gateway on Vlan302 moh music-on-hold.au piano music file in flash multicast moh 239.1.1.1 port 16384 route voice vlan ip = 192.168.65.254 loop0 ip = 192.1.65.254 ccm-manager music-on-hold bind Vlan302 ip multicast-routing is enabled ip pim dense mode is configured on voice vlan interface and loop0 interface cucm moh audio source and PUB are configured for multicast routing (1 hop) and assigned to mrg br1_mcast_moh which is assigned to mrgl br1 which is assigned to br1 device pool I have a device pool 'MoH' that has a region 'MoH' and is assigned G711 to all other regions. This region is assign to device pool MoH, and device pool MoH is assign to the MoH servers. When HQ calls BR1 and HQ presses hold, the BR1 phone hears piano music. When PSTN calls BR1 and BR1 presses hold, PSTN hears beep beep beep r2# debug ephone moh EPHONE music-on-hold debugging is enabled Oct 4 00:09:50.794: MoH route If Vlan302 ETHERNET 192.168.65.254 via ARP Oct 4 00:09:50.794: MoH route If Loopback0 46 192.1.65.254 via 192.1.65.254 r2#debug ccm-m music-on-hold all Call Manager music-on-hold all debugging is on r2# Oct 4 00:16:08.156: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.156: moh_process_ccb: dstadr 192.168.65.30, callid -1, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.164: moh_process_ccb: dstadr 192.168.65.30, callid 11, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.164: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.032: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:17.264: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 disconnected from 911 , call lasted 9 seconds Oct 4 00:16:17.268: moh_update_rtp: callID 12 dstCallID 11 On Oct 3, 2010, at 5:44 PM, James Key wrote: Mark, Looking at your config, a little confused on your ip source address under call-manager fallback and what you have for your route under multicast. One is listed as voice vlan gateway and the other is voice vlan ip, but two different networks. What you have listed for your CUCM config looks correct. Also, do you also have ccm-manager music-on-hold defined on the br1 router? believe this is needed for multicast even though an H323 gateway. James From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com
Re: [OSL | CCIE_Voice] Single Number Reach
Hmm, for Single Number Reach? If a call comes in to HQ3 and simultaneously rings the PSTN/SNR number, and the PSTN answers the calls, I believe there is a way for the HQ3 phone to display on its screen that it is in use. Turning off Privacy for HQ3 didn't change that behavior. :/ On Oct 3, 2010, at 4:30 AM, Roger Källberg wrote: Turn off privacy Roger Källberg CCIE # 26199 (Voice) Unified Communication Consultant Cygate AB From: Mark Holloway [mailto:m...@markholloway.com] Sent: den 2 oktober 2010 06:24 To: Graham Hopkins Cc: CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] Single Number Reach Is there a specific setting to force the ip phone to display an in use message in the event the pstn phone answers the incoming call? On Oct 1, 2010, at 11:42 AM, Graham Hopkins wrote: Same here , I was beginning to think that no patterns are matched in calling number transformations - but I tested with a pattern of ! and a mask of 12345 and that works. So it would appear that there is a mismatch between \+1480.! and the calling number, which does seem odd as if you leave it alone it gets sent to the PSTN as +1480XXX. It would appear that it should match as the pattern ! with XXX works, but as Mark says this doesn't do what he requires Graham On 1 Oct 2010, at 19:23, Mark Holloway wrote: The only issue with this is you don't know if the calling party is Subscriber, National, or International, so you can't use XXX because if BR2 or BR1 calls HQ3 the From number would only show the first 7 digits. On Oct 1, 2010, at 11:21 AM, sisiaji wrote: yeah, you are right, I was referring to RP/RL transformations... i tested it and i got the same in my lab so i guess, as you already mentioned before, the way to do it is to actually put Calling Party Transform Mask to be XXX on the RL (for RG member). On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway m...@markholloway.com wrote: When doing it under Call Routing Transformation Pattern Calling Party Transformation you have to use \+ When doing it on the Calling Party transform mask on a Route Pattern or Route List you don't use \ On Oct 1, 2010, at 10:44 AM, sisiaji wrote: calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting
Re: [OSL | CCIE_Voice] Single Number Reach
Never mind, I believe I have done it correctly. When the PSTN phone answers an SNR call, then only way I can get HQPH3 to show In Use Remote is to actually press the Line 1 button on HQPH3 and then it displays In Use Remote on the phone. On Oct 3, 2010, at 10:18 AM, Mark Holloway wrote: Hmm, for Single Number Reach? If a call comes in to HQ3 and simultaneously rings the PSTN/SNR number, and the PSTN answers the calls, I believe there is a way for the HQ3 phone to display on its screen that it is in use. Turning off Privacy for HQ3 didn't change that behavior. :/ On Oct 3, 2010, at 4:30 AM, Roger Källberg wrote: Turn off privacy Roger Källberg CCIE # 26199 (Voice) Unified Communication Consultant Cygate AB From: Mark Holloway [mailto:m...@markholloway.com] Sent: den 2 oktober 2010 06:24 To: Graham Hopkins Cc: CCIE Voice Maillist Subject: Re: [OSL | CCIE_Voice] Single Number Reach Is there a specific setting to force the ip phone to display an in use message in the event the pstn phone answers the incoming call? On Oct 1, 2010, at 11:42 AM, Graham Hopkins wrote: Same here , I was beginning to think that no patterns are matched in calling number transformations - but I tested with a pattern of ! and a mask of 12345 and that works. So it would appear that there is a mismatch between \+1480.! and the calling number, which does seem odd as if you leave it alone it gets sent to the PSTN as +1480XXX. It would appear that it should match as the pattern ! with XXX works, but as Mark says this doesn't do what he requires Graham On 1 Oct 2010, at 19:23, Mark Holloway wrote: The only issue with this is you don't know if the calling party is Subscriber, National, or International, so you can't use XXX because if BR2 or BR1 calls HQ3 the From number would only show the first 7 digits. On Oct 1, 2010, at 11:21 AM, sisiaji wrote: yeah, you are right, I was referring to RP/RL transformations... i tested it and i got the same in my lab so i guess, as you already mentioned before, the way to do it is to actually put Calling Party Transform Mask to be XXX on the RL (for RG member). On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway m...@markholloway.com wrote: When doing it under Call Routing Transformation Pattern Calling Party Transformation you have to use \+ When doing it on the Calling Party transform mask on a Route Pattern or Route List you don't use \ On Oct 1, 2010, at 10:44 AM, sisiaji wrote: calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR
[OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
I thought I had this figured out but I'm slipping up somewhere. Could use some help. :) I'm configuring multicast moh at BR1 using G.711 and streaming from BR1 router flash. BR1 is an H323 gateway. call-manager-fallback max-dn 24 max-ephones 2 ip source address 10.20.30.254 this is the voice vlan default gateway moh music-on-hold.au piano music file in flash multicast moh 239.1.1.1 port 16384 route voice vlan ip = 192.168.65.254 loop0 ip = 192.1.65.254 ip multicast-routing is enabled ip pim dense mode is configured on voice vlan interface and loop0 interface cucm moh audio source and PUB are configured for multicast routing (1 hop) and assigned to mrg br1_mcast_moh which is assigned to mrgl br1 which is assigned to br1 device pool I have a device pool 'MoH' that has a region 'MoH' and is assigned G711 to all other regions. This region is assign to device pool MoH, and device pool MoH is assign to the MoH servers. When HQ calls BR1 and HQ presses hold, the BR1 phone hears piano music. When PSTN calls BR1 and BR1 presses hold, PSTN hears beep beep beep r2# debug ephone moh EPHONE music-on-hold debugging is enabled Oct 4 00:09:50.794: MoH route If Vlan302 ETHERNET 192.168.65.254 via ARP Oct 4 00:09:50.794: MoH route If Loopback0 46 192.1.65.254 via 192.1.65.254 r2#debug ccm-m music-on-hold all Call Manager music-on-hold all debugging is on r2# Oct 4 00:16:08.156: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.156: moh_process_ccb: dstadr 192.168.65.30, callid -1, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.164: moh_process_ccb: dstadr 192.168.65.30, callid 11, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.164: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.032: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:17.264: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 disconnected from 911 , call lasted 9 seconds Oct 4 00:16:17.268: moh_update_rtp: callID 12 dstCallID 11 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)`
Sorry James..my mistake. It shouldn't be 10.20.30.254 but rather it should be 192.168.65.254. He it is again (proper) call-manager-fallback max-dn 24 max-ephones 2 ip source address 192.168.65.254 this is the voice vlan default gateway on Vlan302 moh music-on-hold.au piano music file in flash multicast moh 239.1.1.1 port 16384 route voice vlan ip = 192.168.65.254 loop0 ip = 192.1.65.254 ccm-manager music-on-hold bind Vlan302 ip multicast-routing is enabled ip pim dense mode is configured on voice vlan interface and loop0 interface cucm moh audio source and PUB are configured for multicast routing (1 hop) and assigned to mrg br1_mcast_moh which is assigned to mrgl br1 which is assigned to br1 device pool I have a device pool 'MoH' that has a region 'MoH' and is assigned G711 to all other regions. This region is assign to device pool MoH, and device pool MoH is assign to the MoH servers. When HQ calls BR1 and HQ presses hold, the BR1 phone hears piano music. When PSTN calls BR1 and BR1 presses hold, PSTN hears beep beep beep r2# debug ephone moh EPHONE music-on-hold debugging is enabled Oct 4 00:09:50.794: MoH route If Vlan302 ETHERNET 192.168.65.254 via ARP Oct 4 00:09:50.794: MoH route If Loopback0 46 192.1.65.254 via 192.1.65.254 r2#debug ccm-m music-on-hold all Call Manager music-on-hold all debugging is on r2# Oct 4 00:16:08.156: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.156: moh_process_ccb: dstadr 192.168.65.30, callid -1, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.164: moh_process_ccb: dstadr 192.168.65.30, callid 11, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.164: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:08.180: %ISDN-6-CONNECT: Interface Serial0/0/0:0 is now connected to 911 N/A Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.028: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:09.032: moh_update_rtp: callID 12 dstCallID 11 Oct 4 00:16:17.264: %ISDN-6-DISCONNECT: Interface Serial0/0/0:0 disconnected from 911 , call lasted 9 seconds Oct 4 00:16:17.268: moh_update_rtp: callID 12 dstCallID 11 On Oct 3, 2010, at 5:44 PM, James Key wrote: Mark, Looking at your config, a little confused on your ip source address under call-manager fallback and what you have for your route under multicast. One is listed as voice vlan gateway and the other is voice vlan ip, but two different networks. What you have listed for your CUCM config looks correct. Also, do you also have ccm-manager music-on-hold defined on the br1 router? believe this is needed for multicast even though an H323 gateway. James From: ccie_voice-boun...@onlinestudylist.com [ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway [...@markholloway.com] Sent: Sunday, October 03, 2010 7:17 PM To: CCIE Voice Maillist Subject: [OSL | CCIE_Voice] MoH SRST (Stream from Flash)` I thought I had this figured out but I'm slipping up somewhere. Could use some help. :) I'm configuring multicast moh at BR1 using G.711 and streaming from BR1 router flash. BR1 is an H323 gateway. call-manager-fallback max-dn 24 max-ephones 2 ip source address 10.20.30.254 this is the voice vlan default gateway moh music-on-hold.au piano music file in flash multicast moh 239.1.1.1 port 16384 route voice vlan ip = 192.168.65.254 loop0 ip = 192.1.65.254 ip multicast-routing is enabled ip pim dense mode is configured on voice vlan interface and loop0 interface cucm moh audio source and PUB are configured for multicast routing (1 hop) and assigned to mrg br1_mcast_moh which is assigned to mrgl br1 which is assigned to br1 device pool I have a device pool 'MoH' that has a region 'MoH' and is assigned G711 to all other regions. This region is assign to device pool MoH, and device pool MoH is assign to the MoH servers. When HQ calls BR1 and HQ presses hold, the BR1 phone hears piano music. When PSTN calls BR1 and BR1 presses hold, PSTN hears beep beep beep r2# debug ephone moh EPHONE music-on-hold debugging is enabled Oct 4 00:09:50.794: MoH route If Vlan302 ETHERNET 192.168.65.254 via ARP Oct 4 00:09:50.794: MoH route If Loopback0 46 192.1.65.254 via 192.1.65.254 r2#debug ccm-m music-on-hold all Call Manager music-on-hold all debugging is on r2# Oct 4 00:16:08.156: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.156: moh_process_ccb: dstadr 192.168.65.30, callid -1, port 21836, codec 16, moh_en 0, moh_addr 0.0.0.0 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.160: moh_update_rtp: callID 12 dstCallID -1 Oct 4 00:16:08.164
[OSL | CCIE_Voice] Single Number Reach
I'm having a hard time when an internal extension calls another internal extension that uses SNR, the From phone number on the PSTN phone is 4 digits instead of 7. For example, extension 2001 calls 2003, and 2003 simultaneously rings a PSTN phone number. The display on the PSTN phone says HqPh1 (2001) instead of the 7 digit or 10 digit number. I have created PT_SNR which is assigned to CSS_SNR. I have CSS_SNR assigned to the Remote Destination Profile for both CSS and Redirecting CSS. My SNR number is +14086347694 and I have a route pattern that contains \+1408.6347694 which egresses the RL_HQ_ONLY (this is not Standard Local Route Group). I also created a Translation Pattern with PT_SNR and I have checked Use External Phone Number Mask. I was expecting this to take the 4 digit Calling number and insert the External mask instead. I tried following the steps in the Mock Lab guide (I believe it is Lab 6) but I still cannot get it working. Any assistance would be appreciated. Perhaps someone has a blog post? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.!(replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
When doing it under Call Routing Transformation Pattern Calling Party Transformation you have to use \+ When doing it on the Calling Party transform mask on a Route Pattern or Route List you don't use \ On Oct 1, 2010, at 10:44 AM, sisiaji wrote: calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
The only issue with this is you don't know if the calling party is Subscriber, National, or International, so you can't use XXX because if BR2 or BR1 calls HQ3 the From number would only show the first 7 digits. On Oct 1, 2010, at 11:21 AM, sisiaji wrote: yeah, you are right, I was referring to RP/RL transformations... i tested it and i got the same in my lab so i guess, as you already mentioned before, the way to do it is to actually put Calling Party Transform Mask to be XXX on the RL (for RG member). On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway m...@markholloway.com wrote: When doing it under Call Routing Transformation Pattern Calling Party Transformation you have to use \+ When doing it on the Calling Party transform mask on a Route Pattern or Route List you don't use \ On Oct 1, 2010, at 10:44 AM, sisiaji wrote: calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Single Number Reach
Is there a specific setting to force the ip phone to display an in use message in the event the pstn phone answers the incoming call? On Oct 1, 2010, at 11:42 AM, Graham Hopkins wrote: Same here , I was beginning to think that no patterns are matched in calling number transformations - but I tested with a pattern of ! and a mask of 12345 and that works. So it would appear that there is a mismatch between \+1480.! and the calling number, which does seem odd as if you leave it alone it gets sent to the PSTN as +1480XXX. It would appear that it should match as the pattern ! with XXX works, but as Mark says this doesn't do what he requires Graham On 1 Oct 2010, at 19:23, Mark Holloway wrote: The only issue with this is you don't know if the calling party is Subscriber, National, or International, so you can't use XXX because if BR2 or BR1 calls HQ3 the From number would only show the first 7 digits. On Oct 1, 2010, at 11:21 AM, sisiaji wrote: yeah, you are right, I was referring to RP/RL transformations... i tested it and i got the same in my lab so i guess, as you already mentioned before, the way to do it is to actually put Calling Party Transform Mask to be XXX on the RL (for RG member). On Fri, Oct 1, 2010 at 7:52 PM, Mark Holloway m...@markholloway.com wrote: When doing it under Call Routing Transformation Pattern Calling Party Transformation you have to use \+ When doing it on the Calling Party transform mask on a Route Pattern or Route List you don't use \ On Oct 1, 2010, at 10:44 AM, sisiaji wrote: calling party transformation is done without prefix \ On Fri, Oct 1, 2010 at 7:00 PM, Mark Holloway m...@markholloway.com wrote: The crazy thing is I tried this but I couldn't get it to work. PT_HQ_CALLING_OUT assigned to CSS_HQ_CALLING_OUT assigned to CALLING Number Transform on the Outbound portion of the HQ gateway. Calling Party Transformation: PT_HQ_CALLING_OUT pattern is \+1480.! (replace 480 with what your HQ area code is) Strip Predot That should make the outbound From number +14805552001 appear as 5552001 on the PSTN phone. and I should see 5552001 in the isdn q931 debug output. I'm still seeing the full E164 number. On Oct 1, 2010, at 9:22 AM, Graham Hopkins wrote: Well I'm just showing the full E.164 as that's what the lab I'm looking at looks for. However I guess you could strip the HQ area code at the gateway with the calling party transformation. In the real world (plan to visit that soon) then the remote destination is likely to be a mobile phone which isn't really local to any gateway - at least not here in the UK so would be a national call from anywhere. Graham On 1 Oct 2010, at 17:10, Mark Holloway wrote: Sorry, I meant Translation Patterns, not Profiles. Still working on the From number presentation. I'm assuming that if HQ1 calls HQ3 the PSTN phone should show a 7 digit From number, but if BR1 calls 2003 the PSTN should show a 10 digit From number. Would you guys agree? On Oct 1, 2010, at 8:56 AM, Mark Holloway wrote: Graham, same thing here. This is a summary of what I've done to get it working correctly. I eliminated using Translation Profiles as I didn't find them necessary for this. Create PT_SNR which is assigned to CSS_SNR Create a Remote Destination Profile and assign CSS_SNR to both Calling Search Space and Rerouting Calling Search Space. Build/associate your end user with this Remote Destination Profile. Build a Route List (RL_SNR) that includes just the HQ gateway and set the Calling Party External Phone Mask to On. Doing this in the Route Pattern won't work. Set Called Party to Subscriber (assuming the Remote Destination number is a local number). Lastly, build a Route Pattern that matches your Remote Destination Profile external number and assign it to PT_SNR and RL_SNR. The only thing about this method is that when calls from 2001 ring 2003 which rings the PSTN, this method is using the external mask which means HQ1's external mask is E164. Typically when a Subscriber call egresses the HQ gateway you would want the From number to be 7 digits. Are you guys putting a Calling Party Transformation on your HQ gateway to strip off the HQ area code for Subscriber calls? For all other purposes of presenting 7, 10, or E164, I have always used the Calling Party Transform in either the Route Pattern or Route List's Route Group. Thanks, Mark On Oct 1, 2010, at 7:25 AM, Graham Hopkins wrote: Just hit the same problem in Vol2 Lab4 and I can confirm that this doesn't work at the RP level but does work at the RL level. Is this a known bug ? Graham On 1 Oct 2010, at 13:35, Tam Nhu wrote: Hi Mark, The EPNM does not work at RP for SNR. Have you try to set EPNM at RL level? TN
[OSL | CCIE_Voice] So close, yet so far away..
FAIL. Based on my score report it was not by much. I will go back in 30 days to try again. It was my first Voice attempt. Next time I will go to San Jose instead of RTP just because it's closer. Although, the proctor in RTP was very nice and helpful. I wish she would go to San Jose. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Unity Connection Greeting
Is there an option so instead of the VM greeting being Sorry, Jon Doe is not available it says Sorry, extension 5001 is not available. Basically, I want it to say the extension instead of the name. Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Unity Connection Greeting
Thanks. :) On Sep 24, 2010, at 3:01 PM, bkvalent...@gmail.com wrote: Remove the display name. - Reply message - From: Mark Holloway m...@markholloway.com Date: Fri, Sep 24, 2010 5:53 pm Subject: [OSL | CCIE_Voice] Unity Connection Greeting To: osl osl ccie_voice@onlinestudylist.com Is there an option so instead of the VM greeting being Sorry, Jon Doe is not available it says Sorry, extension 5001 is not available. Basically, I want it to say the extension instead of the name. Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Anyone get MWI to work with CUE in SRST?
When BR2 is part of CUCM and CUE is integrated with CUCM through JTAPI I could never get MWI to work if BR2 is in SRST. Has anyone been able to get this working? Cisco docs say with older version of CUE it doesn't work. I think they are referring to 2.x or 3.x, not 7.x. Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Anyone get MWI to work with CUE in SRST?
Yes that did the trick. Thank you! On Sep 24, 2010, at 4:20 PM, CCIE Voice GMAIL wrote: If I understand your question correctly, I believe the command you need to add is this: sip-ua mwi-server ipv4:CUE address if you are using unsolicited you just need to add the unsolicited keyword sip-ua mwi-server ipv4:CUE address unsolicited http://www.ciscosystems.cg/en/US/docs/ios/voice/command/reference/vr_m3.html #wp1373612 Hope this helps. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Friday, September 24, 2010 3:53 PM To: osl osl Subject: [OSL | CCIE_Voice] Anyone get MWI to work with CUE in SRST? When BR2 is part of CUCM and CUE is integrated with CUCM through JTAPI I could never get MWI to work if BR2 is in SRST. Has anyone been able to get this working? Cisco docs say with older version of CUE it doesn't work. I think they are referring to 2.x or 3.x, not 7.x. Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RES: CCIE Voice Passed
Parabens! On Sep 24, 2010, at 8:11 PM, Marcelo Alexandria wrote: Hello Guys , Since last year i m trying to pass in the lab , so in last Friday I was make my 2nd attempt in version 3. On Monday I got my results and I cant believe ..i passed!!! So , I never post nothing here in the list because my English is not very good….., but I follow the daily posts….. I need say “thank you” to all and to Viki for your free trainings in the site.All of you help me a lot in my Studies…. I need say a special thank you to Angel Peres for your posts…help all ever So guys …thank you again….i only can say the test is very hard and I got very hard test in my lab…but this time I was very calm and prepared. So the ipexpert help me a lot yet. Thank you all again Marcelo Alexandria from Brazil CCIE Voice #27021 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Display of globalized number on 7965
If I send a call from the HQ through the PSTN to BR2 and the calling number from HQ is in E.164 format, the BR2 phone's main display (ie. large fonts) doesn't show the + in front of 1XX but on the bottom part of the 7965's screen it does show +1XX in smaller fonts. Is this considered acceptable if we are required to send E.164 as the calling number? I would think so. The ISDN debug on BR2 does show the calling number as +1XX. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCM Corporate Directory (BLF Presence Status)
Have you enabled the BLF For Call Lists in Enterprice Parameters? Yes, I have BLF For Call List set to Enabled. I Reset (under Enterprise) and then performed a Restart on the phones and I am still not able to get the corporate directory to show status. do you have dn partition included in your subscribe css? btw, afair it's not used for blf, only for directory list presence information. Yes, Subscribe CSS is simply CSS_HQ_DEVICE which contains PT_INTERNAL. PT_INTERNAL is assigned to all HQ and BR1 Lines. HQ phones are SUBSCRIBED to CSS_HQ_DEVICE I have End Users for all HQ phones and they are associated with the Device and Line When I press Directories Corporate Directory Search I can see all the users ok, but I see the grey square next to their name with no presence change when an HQ line goes off hook. I do see the BLF Speed Dial Presence working on HqPh3 when HqPh1's primary Line goes off hook. I'm stumped. On Sep 11, 2010, at 2:28 AM, Roger Källberg wrote: Hi Mark, Have you enabled the BLF For Call Lists in Enterprice Parameters? Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Från: Mark Holloway [...@markholloway.com] Skickat: den 11 september 2010 07:47 Till: osl osl Ämne: [OSL | CCIE_Voice] CUCM Corporate Directory (BLF Presence Status) I have HqPh1 and HqPh3 both assigned to the same SUBSCRIBE CSS and HqPh3 has BLF Speed Dial assigned to watch HqPh1's primary extension and everything works great on HqPh3's line key that watches HqPh1. However, I am trying to access the corporate directory on HqPh3 and expect to see presence status for HqPh1's main number. If HqPh1 goes off hook I am not seeing anything change in the corporate directory listing on HqPh3. I have End Users created for my Hq phones and assigned their primary extension. The Hq phones are both in Standard Presence Group. If I go to System Presence Group and set Allow then restart the phone it doesn't make a difference. Any suggestions? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCM Corporate Directory (BLF Presence Status)
Yes, each End User's Extension Mobility section is populated with the same Presence Group and SUBSCRIBE Calling Search Space. On Sep 11, 2010, at 11:57 AM, Brian Mulgrew wrote: Hi Mark - are your end users set for a Presence Group? Thk Brian On 11/09/2010, Mark Holloway m...@markholloway.com wrote: Have you enabled the BLF For Call Lists in Enterprice Parameters? Yes, I have BLF For Call List set to Enabled. I Reset (under Enterprise) and then performed a Restart on the phones and I am still not able to get the corporate directory to show status. do you have dn partition included in your subscribe css? btw, afair it's not used for blf, only for directory list presence information. Yes, Subscribe CSS is simply CSS_HQ_DEVICE which contains PT_INTERNAL. PT_INTERNAL is assigned to all HQ and BR1 Lines. HQ phones are SUBSCRIBED to CSS_HQ_DEVICE I have End Users for all HQ phones and they are associated with the Device and Line When I press Directories Corporate Directory Search I can see all the users ok, but I see the grey square next to their name with no presence change when an HQ line goes off hook. I do see the BLF Speed Dial Presence working on HqPh3 when HqPh1's primary Line goes off hook. I'm stumped. On Sep 11, 2010, at 2:28 AM, Roger Källberg wrote: Hi Mark, Have you enabled the BLF For Call Lists in Enterprice Parameters? Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Från: Mark Holloway [...@markholloway.com] Skickat: den 11 september 2010 07:47 Till: osl osl Ämne: [OSL | CCIE_Voice] CUCM Corporate Directory (BLF Presence Status) I have HqPh1 and HqPh3 both assigned to the same SUBSCRIBE CSS and HqPh3 has BLF Speed Dial assigned to watch HqPh1's primary extension and everything works great on HqPh3's line key that watches HqPh1. However, I am trying to access the corporate directory on HqPh3 and expect to see presence status for HqPh1's main number. If HqPh1 goes off hook I am not seeing anything change in the corporate directory listing on HqPh3. I have End Users created for my Hq phones and assigned their primary extension. The Hq phones are both in Standard Presence Group. If I go to System Presence Group and set Allow then restart the phone it doesn't make a difference. Any suggestions? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Sent from my mobile device ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCM Corporate Directory (BLF Presence Status)
Thanks Tam. I thought the Primary Extension was the only required association on the End User page. Of course the Device Phone also had the End Use associated. Since the Phone Number entry on the End User page is not a required field I didn't even think to populate it. :) On Sep 11, 2010, at 1:14 PM, Tam Nhu wrote: Hi Mark, In the End User page, you need to have the Telephone Number field configured with a DN of that user; the Corporate Directory only looks at this field for Presence status. TN. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUCM Corporate Directory (BLF Presence Status)
I have HqPh1 and HqPh3 both assigned to the same SUBSCRIBE CSS and HqPh3 has BLF Speed Dial assigned to watch HqPh1's primary extension and everything works great on HqPh3's line key that watches HqPh1. However, I am trying to access the corporate directory on HqPh3 and expect to see presence status for HqPh1's main number. If HqPh1 goes off hook I am not seeing anything change in the corporate directory listing on HqPh3. I have End Users created for my Hq phones and assigned their primary extension. The Hq phones are both in Standard Presence Group. If I go to System Presence Group and set Allow then restart the phone it doesn't make a difference. Any suggestions? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] srr-queue Shape vs Share on 3750
If I want a priority queue to have 25% of the port bandwidth, I have configured shape 4. I want queues 2, 3, and 4 to share 40%, 40%, and 20% of the remaining bandwidth. All the examples I have seen for shape/share show a value of 1 for priority queue in share regardless of the fact shape is set to 4 (25% of available bandwidth). srr-queue bandwidth shape 4 0 0 0 -- 4 = one 1/4th of total bandwidth; 0 = use Share instead srr-queue bandwidth share 1 40 40 20 -- queue 2, 3, 4 will share 40%, 40%, 20% of bandwidth Why does share's priority queue need a value of 1 if shape is already 4? Is it an indicator saying there is a value for shape so use that instead? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] srr-queue Shape vs Share on 3750
Daniel, Thanks for explaining. Makes sense now. :) You are correct in that it's queue 1, not the priority queue that I am sizing. On Sep 8, 2010, at 11:44 PM, Daniel Berlinski wrote: Hi Mark I think you need to place a value in that first queue position for the share command line other than zero otherwise IOS gives you an error. That value is ignored because as you have already pointed out, your shape to 25% of the bandwidth is already in place and take precedence on that queue. When you say priority-queue, do you mean simply put Queue 1 or you mean that you believe that you are sizing the priority-queue to 25% of the available interface bandwidth? In your intended config would you add priority-queue out as well? I ask the question because there is no way to size the depth of your priority queue on egress, only on ingress. Cheers On Thu, Sep 9, 2010 at 6:16 PM, Mark Holloway m...@markholloway.com wrote: If I want a priority queue to have 25% of the port bandwidth, I have configured shape 4. I want queues 2, 3, and 4 to share 40%, 40%, and 20% of the remaining bandwidth. All the examples I have seen for shape/share show a value of 1 for priority queue in share regardless of the fact shape is set to 4 (25% of available bandwidth). srr-queue bandwidth shape 4 0 0 0 -- 4 = one 1/4th of total bandwidth; 0 = use Share instead srr-queue bandwidth share 1 40 40 20 -- queue 2, 3, 4 will share 40%, 40%, 20% of bandwidth Why does share's priority queue need a value of 1 if shape is already 4? Is it an indicator saying there is a value for shape so use that instead? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Secure CRT Key Mappings
In the real Cisco lab can you assign your own Key Mappings in Secure CRT (for copy/paste functionality)? For example, I am running Secure CRT 4 at home and I can assign Page-Up to copy and Page-Down to paste. I hope CTRL-C and CTRL-V work in Notepad. :/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again) what am I missing???
In UCM how do you determine whether you are assigning single button cBarge or normal cBarge? On Aug 7, 2010, at 9:35 AM, cisco voip wrote: That bug is for srst mode auto provision none.. for provision all, it should work The problem you are facing of having cbarge for split second is because you had single button cbarge when phones were registered to CUCM, disable that setting and make it normal cbarge, they will start to work in srst mode as well On Fri, Aug 6, 2010 at 5:05 PM, Ashar Siddiqui siddas...@gmail.com wrote: I am glad that the solution proposed by Cisco is exactly what I did months back after trying different solutions. Ash. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of CCIE Voice GMAIL Sent: 06 August 2010 03:13 To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again) what am I missing??? I thought I’d share this with everyone as this have been extremely frustrating for me. Apparently this is a known bug (well…recently known). CSCti11843 From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of MARSHALL, JODY C (ATTBCS) Sent: Wednesday, August 04, 2010 4:55 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again) what am I missing??? I have read (several) post on this and have tested several different ways. None of which have I been able to make work. Can you please take a look and see if I am missing something. The first configuration is from auto-provision all. I had the phones registers then unregister bounced the router and register again. Cbarge does not work. I see the remote-in-use state for a second then disappears. I then registered the phones to CUCM and removed telephony-service reloaded the router and reconfigured telephony-service with auto-provision none with the second configuration posted. Cbarge does not work. 124-20.T5.bin telephony-service sdspfarm units 5 sdspfarm tag 2 sitebcfb conference hardware srst mode auto-provision all srst ephone template 1 srst ephone description SRST : Aug 03 2010 13:28:28 : Aug 03 2010 21:20:20 srst dn template 1 srst dn line-mode octo max-ephones 4 max-dn 30 preference 3 ip source-address 10.12.202.1 port 2000 system message CCIEVOICE time-zone 8 date-format dd-mm-yy voicemail 2220 max-conferences 8 gain -6 web admin system name administrator password ccievoice transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Aug 03 2010 21:20:26 ! R2#sho sccp SCCP Admin State: UP Gateway IP Address: 10.12.202.1, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 10.12.202.1, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 3 Trustpoint: N/A Call Manager: 10.12.200.21, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 2 Trustpoint: N/A Call Manager: 10.12.200.22, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 1 Trustpoint: N/A Conferencing Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.12.202.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: g729br8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 R2#sho ephone ephone-1[0] Mac:001A.6D57.021D TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 privacy:0 IP:10.10.255.156 35781 7961 keepalive 82 max_line 6 available_line 5 button 1: dn 1 number 3001 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 2 number 3012 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared privacy button is enabled Preferred Codec: g711ulaw ephone-2[1] Mac:0019.56A3.A0D8 TCP socket:[2] activeLine:0 REGISTERED in SCCP
Re: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again) what am I missing???
I had the same problem. When the phones go into SRST mode and then I call from the PSTN to Ph1's shared line and put the call on hold, I go to Ph2 and press Ph2's shared line and see the cBarge softkey for a split second then it changes to a ghosted Redial softkey. I could never get it to work. On Aug 4, 2010, at 4:55 AM, MARSHALL, JODY C (ATTBCS) wrote: I have read (several) post on this and have tested several different ways. None of which have I been able to make work. Can you please take a look and see if I am missing something. The first configuration is from auto-provision all. I had the phones registers then unregister bounced the router and register again. Cbarge does not work. I see the remote-in-use state for a second then disappears. I then registered the phones to CUCM and removed telephony-service reloaded the router and reconfigured telephony-service with auto-provision none with the second configuration posted. Cbarge does not work. 124-20.T5.bin telephony-service sdspfarm units 5 sdspfarm tag 2 sitebcfb conference hardware srst mode auto-provision all srst ephone template 1 srst ephone description SRST : Aug 03 2010 13:28:28 : Aug 03 2010 21:20:20 srst dn template 1 srst dn line-mode octo max-ephones 4 max-dn 30 preference 3 ip source-address 10.12.202.1 port 2000 system message CCIEVOICE time-zone 8 date-format dd-mm-yy voicemail 2220 max-conferences 8 gain -6 web admin system name administrator password ccievoice transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Aug 03 2010 21:20:26 ! R2#sho sccp SCCP Admin State: UP Gateway IP Address: 10.12.202.1, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 10.12.202.1, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 3 Trustpoint: N/A Call Manager: 10.12.200.21, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 2 Trustpoint: N/A Call Manager: 10.12.200.22, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 1 Trustpoint: N/A Conferencing Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.12.202.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: g729br8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 R2#sho ephone ephone-1[0] Mac:001A.6D57.021D TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 privacy:0 IP:10.10.255.156 35781 7961 keepalive 82 max_line 6 available_line 5 button 1: dn 1 number 3001 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 2 number 3012 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared privacy button is enabled Preferred Codec: g711ulaw ephone-2[1] Mac:0019.56A3.A0D8 TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 privacy:0 IP:10.10.255.153 14441 7961 keepalive 82 max_line 6 available_line 5 button 1: dn 3 number 3002 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 2 number 3012 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared privacy button is enabled Preferred Codec: g711ulaw ephone-dn-template 1 call-forward busy 2220 call-forward noan 2220 timeout 20 mwi sip ! ! ephone-template 1 privacy-button softkeys remote-in-use CBarge Newcall! ! ephone-dn 1 octo-line number 3001 label 3001 description +9723033001 name SiteBPh1 preference 3 ephone-dn-template 1 ! ! ephone-dn 2 octo-line number 3012 label 3012 description 3012 name 3012 preference 3
Re: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again) what am I missing???
Sorry, I didn't mean to say put the call on hold, what I meant was I let the call sit idle while going to Ph2 and pressing the shared line. I would see the cBarge softkey for a split second then it changes to Redial. On Aug 4, 2010, at 8:39 AM, Mark Holloway wrote: I had the same problem. When the phones go into SRST mode and then I call from the PSTN to Ph1's shared line and put the call on hold, I go to Ph2 and press Ph2's shared line and see the cBarge softkey for a split second then it changes to a ghosted Redial softkey. I could never get it to work. On Aug 4, 2010, at 4:55 AM, MARSHALL, JODY C (ATTBCS) wrote: I have read (several) post on this and have tested several different ways. None of which have I been able to make work. Can you please take a look and see if I am missing something. The first configuration is from auto-provision all. I had the phones registers then unregister bounced the router and register again. Cbarge does not work. I see the remote-in-use state for a second then disappears. I then registered the phones to CUCM and removed telephony-service reloaded the router and reconfigured telephony-service with auto-provision none with the second configuration posted. Cbarge does not work. 124-20.T5.bin telephony-service sdspfarm units 5 sdspfarm tag 2 sitebcfb conference hardware srst mode auto-provision all srst ephone template 1 srst ephone description SRST : Aug 03 2010 13:28:28 : Aug 03 2010 21:20:20 srst dn template 1 srst dn line-mode octo max-ephones 4 max-dn 30 preference 3 ip source-address 10.12.202.1 port 2000 system message CCIEVOICE time-zone 8 date-format dd-mm-yy voicemail 2220 max-conferences 8 gain -6 web admin system name administrator password ccievoice transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Aug 03 2010 21:20:26 ! R2#sho sccp SCCP Admin State: UP Gateway IP Address: 10.12.202.1, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 10.12.202.1, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 3 Trustpoint: N/A Call Manager: 10.12.200.21, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 2 Trustpoint: N/A Call Manager: 10.12.200.22, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 1 Trustpoint: N/A Conferencing Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.12.202.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: g729br8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 R2#sho ephone ephone-1[0] Mac:001A.6D57.021D TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 privacy:0 IP:10.10.255.156 35781 7961 keepalive 82 max_line 6 available_line 5 button 1: dn 1 number 3001 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 2 number 3012 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared privacy button is enabled Preferred Codec: g711ulaw ephone-2[1] Mac:0019.56A3.A0D8 TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 privacy:0 IP:10.10.255.153 14441 7961 keepalive 82 max_line 6 available_line 5 button 1: dn 3 number 3002 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 2 number 3012 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared privacy button is enabled Preferred Codec: g711ulaw ephone-dn-template 1 call-forward busy 2220 call-forward noan 2220 timeout 20 mwi sip ! ! ephone-template 1 privacy-button softkeys remote-in-use CBarge Newcall
Re: [OSL | CCIE_Voice] CBarge in SRST mode
Graham, Are you configuring this in your own lab or using Proctor Labs? I am using my own lab (2800's, 12.4.24T3, 7965 phones) and I couldn't get cBarge to work in SRST with auto provision none. Others using Proctor Labs said they could get it to work. Perhaps it's a difference between IOS versions and/or phone types. I literally tried everything. On Jul 26, 2010, at 6:59 AM, Graham Hopkins wrote: Been following the thread on this and have concerns about the ephone-template not appearing to work. The only but I can find that relates to this is CSCsx15347 which refers to a G.729 codec in the ephone -template not being used until after a reboot. The only way I can get this to work without specifying privacy off under the ephone is to run with srst mode auto-provision all and then save the config and reboot - the ephone-template then works privacy button as well . Config below. Anyone have any further thoughts on how to do this without using auto-provision all. Anyone found a way to do it with auto provision none and the ephone template - no manual configuration of the ephone? telephony-service sdspfarm units 4 sdspfarm tag 1 br1-conf no privacy conference hardware srst mode auto-provision all srst ephone template 1 srst dn line-mode octo max-ephones 4 max-dn 8 ip source-address 10.10.201.1 port 2000 system message CCIE SRST Fallback voicemail 912123945600 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Jul 21 2010 11:48:33 ephone-template 1 privacy off privacy-button softkeys remote-in-use Newcall CBarge ephone 1 mac-address 0026.CB3D.2888 ephone-template 1 button 1:1 2:2 3:3 ! ! ! ephone 2 mac-address 0021.D8B8.EDDF ephone-template 1 button 1:4 2:3 ! Regards Graham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
What ephone-dn information are you manually entering into the router's config versus what you let the router 'learn'? Thanks..I appreciate the help. On Jul 21, 2010, at 6:47 AM, Randall Saborio wrote: Funny you two bring up this as I was just finishing my tests to replicate the problem. I agree the problem is all because of privacy remaining on on the phones. What finally resolves the problem is just to preconfigure the ephones with the privacy off set, like on that url you mentioned: ephone 1 privacy off ! ephone 2 privacy off To my disappointment, it seems the ephone-template with privacy off is all useless, and also couldn't get the privacy button to ever show up. Just did the individal ephones with the privacy off, and all worked fine. On Wed, Jul 21, 2010 at 7:30 AM, Berry, Matthew J. mjbe...@krollontrack.com wrote: My configuration has worked. You need to make sure that the ephone configuration has “privacy off” in order for the cBarge to work with auto provision none. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com From: Bryan [mailto:ccieiwi...@gmail.com] Sent: Wednesday, July 21, 2010 8:28 AM To: Berry, Matthew J. Cc: Mark Holloway; osl osl Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) Sorry to jump in on the topic. Matt, just curious were you successful with this configuration? It does not work for me with auto-provision none and an ephone-template under the srst ephone template configuration. Another strange thing I have noticed in SRST is when I issue a show telephony-service all, and scroll down to the ephone-template section. It says privacy default, and I have not figured out how to get rid of it or if it is even possible. On Wed, Jul 21, 2010 at 9:20 AM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Mark – Try removing all your learned ephone configuration, change the auto provision mode to none, and then add the ephone template under srst ephone template. See if that will work for you. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Wednesday, July 21, 2010 12:33 AM To: Mark Holloway Cc: osl osl Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) I found this blog post showing an example config that I have loaded on my router that I think should be the correct way to configure cbarge in srst. However, I still can't get it to work. When my phones 'fallback' and I call from the pstn into the shared line, the other phone can't barge the call because when I go off-hook on the shared line the 'cBarge' softkey will display for a fraction of a second the it turns into a ghosted 'Redial' softkey. http://ccieash.wordpress.com/2010/06/21/hardware-conferencing-ios/ On Jul 20, 2010, at 2:30 PM, Mark Holloway wrote: The odd part is this.. Once the phones fall back to srst and I place a call from the PSTN to 2005, I go to the second phone and press the second line key for 2005. I expect the phone to display Remote in Use and offer the cBarge and NewCall softkeys. However, I can see the cBarge and NewCall softkeys appear for a split second, then they disappear and the normal softkeys (CallFwd, etc) appear. On Jul 20, 2010, at 2:18 PM, Mark Holloway wrote: Angel - What kind of phones did you test srst cBarge with? I can't get this to work with my 7965 phones. I needed to add more details under the ephone configuration in order for ephone-template 1 to be applied to the phone which should make the cBarge softkey available during srst. Otherwise, if I reference telephony-service 'srst ephone template 1' it doesn't seem to load properly on the 7965's when they fall back. Only by explicitly assigning the ephone-template 1 under the ephone works (which requires the mac address to be assigned as well). Even so, when a call comes in from the PSTN to my shared line 2005 during srst, I cannot get the other phone to display the cBarge softkey. When I go off-hook on the second phone on line 2005, I get dial-tone but the phone is treating this like any normal DN wanting to make an outbound call. I have made sure Privacy = off but still no luck. telephony-service sdspfarm units 1 sdspfarm tag 1 BR1-CONF conference hardware srst mode auto-provision none srst dn line-mode dual-octo max-ephones 2 max-dn 20 no-reg primary ip source-address 192.168.1.254 port 2000 system message SRST MODE time-zone 8 voicemail 917752011015 max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T r2-br1(config)#do sh sccp SCCP Admin State: UP Gateway Local Interface: Vlan10 IPv4 Address: 192.168.1.254
Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
Angel - What kind of phones did you test srst cBarge with? I can't get this to work with my 7965 phones. I needed to add more details under the ephone configuration in order for ephone-template 1 to be applied to the phone which should make the cBarge softkey available during srst. Otherwise, if I reference telephony-service 'srst ephone template 1' it doesn't seem to load properly on the 7965's when they fall back. Only by explicitly assigning the ephone-template 1 under the ephone works (which requires the mac address to be assigned as well). Even so, when a call comes in from the PSTN to my shared line 2005 during srst, I cannot get the other phone to display the cBarge softkey. When I go off-hook on the second phone on line 2005, I get dial-tone but the phone is treating this like any normal DN wanting to make an outbound call. I have made sure Privacy = off but still no luck. telephony-service sdspfarm units 1 sdspfarm tag 1 BR1-CONF conference hardware srst mode auto-provision none srst dn line-mode dual-octo max-ephones 2 max-dn 20 no-reg primary ip source-address 192.168.1.254 port 2000 system message SRST MODE time-zone 8 voicemail 917752011015 max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T r2-br1(config)#do sh sccp SCCP Admin State: UP Gateway Local Interface: Vlan10 IPv4 Address: 192.168.1.254 Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 192.168.1.254, Port Number: 2000 Priority: 1, Version: 7.0, Identifier: 1 Conferencing Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.254, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 ephone-template 1 privacy off privacy-button conference drop-mode local softkeys remote-in-use CBarge Newcall softkeys idle Redial Newcall Cfwdall softkeys seized Cfwdall Endcall Meetme Pickup Redial softkeys connected Hold Endcall Trnsfer Park Confrn ConfList Select Join ephone-dn 1 octo-line number 2001 no-reg primary label 2001 description 7753012001 name Br1Ph1 call-forward busy 917752011015 call-forward noan 917752011015 timeout 20 ephone-dn 2 octo-line number 2002 no-reg primary label 2002 description 7753012002 name Br1Ph2 call-forward busy 917752011015 call-forward noan 917752011015 timeout 20 ephone-dn 3 octo-line number 2005 no-reg primary label 2005 ephone-dn 10 octo-line number 2010 no-reg primary conference ad-hoc no huntstop ephone 1 privacy off device-security-mode none mac-address 0022.90BA.2ECC ephone-template 1 type 7965 button 1:1 2:3 ephone 2 privacy off device-security-mode none mac-address 0022.90BA.2CB6 ephone-template 1 type 7965 button 1:2 2:3 On Jun 15, 2010, at 4:32 AM, Angel Perez wrote: Hi: These are my observations for srst and cbarge First of all you will need a cnf bridge configured, the best way is adding srst ip add as a third option in sccp ccm group, once your cnf bridge is registered to srst router (it take some more times than phones) you will need a dn octo line (recomended) configured as conference ad-hoc 1: srst auto all: Then once ephones are registered to srst these combinations worked for me: == telephony-service privacy ! (default) ephone 1 no privacy privacy-button ! from the button disable or enable it == telephony-service privacy / no privacy ! you can also manage from here ephone 1 no privacy privacy-button ! === telephony-service privacy ! (default) ephone 1 privacy-button privacy on / privacy off ! enable and disable from ephone === If you enable/disable localy you can't enable/disable globaly ephone 1 privacy on/off ! enable/disable privacy telephony-service privacy/no privacy! this won't enable/disable privacy becouse you have enable/disable it localy at ephone 2: srst auto none: follow vc approach described above hth Date: Tue, 15 Jun 2010 06:09:52 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) Angel - I think you are right. The only way I can see of configuring privacy on/off would be through the ephone section itself. Privacy isn't an option with an ephone-template, otherwise you could have set it there. You could possibly set no privacy under telephony-service, but that would be a global setting. I am not at my lab right now so I cannot verify if that would actually propagate down to SRST-provisioned phones. Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/15/2010
Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
The odd part is this.. Once the phones fall back to srst and I place a call from the PSTN to 2005, I go to the second phone and press the second line key for 2005. I expect the phone to display Remote in Use and offer the cBarge and NewCall softkeys. However, I can see the cBarge and NewCall softkeys appear for a split second, then they disappear and the normal softkeys (CallFwd, etc) appear. On Jul 20, 2010, at 2:18 PM, Mark Holloway wrote: Angel - What kind of phones did you test srst cBarge with? I can't get this to work with my 7965 phones. I needed to add more details under the ephone configuration in order for ephone-template 1 to be applied to the phone which should make the cBarge softkey available during srst. Otherwise, if I reference telephony-service 'srst ephone template 1' it doesn't seem to load properly on the 7965's when they fall back. Only by explicitly assigning the ephone-template 1 under the ephone works (which requires the mac address to be assigned as well). Even so, when a call comes in from the PSTN to my shared line 2005 during srst, I cannot get the other phone to display the cBarge softkey. When I go off-hook on the second phone on line 2005, I get dial-tone but the phone is treating this like any normal DN wanting to make an outbound call. I have made sure Privacy = off but still no luck. telephony-service sdspfarm units 1 sdspfarm tag 1 BR1-CONF conference hardware srst mode auto-provision none srst dn line-mode dual-octo max-ephones 2 max-dn 20 no-reg primary ip source-address 192.168.1.254 port 2000 system message SRST MODE time-zone 8 voicemail 917752011015 max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T r2-br1(config)#do sh sccp SCCP Admin State: UP Gateway Local Interface: Vlan10 IPv4 Address: 192.168.1.254 Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 192.168.1.254, Port Number: 2000 Priority: 1, Version: 7.0, Identifier: 1 Conferencing Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.254, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 1 ephone-template 1 privacy off privacy-button conference drop-mode local softkeys remote-in-use CBarge Newcall softkeys idle Redial Newcall Cfwdall softkeys seized Cfwdall Endcall Meetme Pickup Redial softkeys connected Hold Endcall Trnsfer Park Confrn ConfList Select Join ephone-dn 1 octo-line number 2001 no-reg primary label 2001 description 7753012001 name Br1Ph1 call-forward busy 917752011015 call-forward noan 917752011015 timeout 20 ephone-dn 2 octo-line number 2002 no-reg primary label 2002 description 7753012002 name Br1Ph2 call-forward busy 917752011015 call-forward noan 917752011015 timeout 20 ephone-dn 3 octo-line number 2005 no-reg primary label 2005 ephone-dn 10 octo-line number 2010 no-reg primary conference ad-hoc no huntstop ephone 1 privacy off device-security-mode none mac-address 0022.90BA.2ECC ephone-template 1 type 7965 button 1:1 2:3 ephone 2 privacy off device-security-mode none mac-address 0022.90BA.2CB6 ephone-template 1 type 7965 button 1:2 2:3 On Jun 15, 2010, at 4:32 AM, Angel Perez wrote: Hi: These are my observations for srst and cbarge First of all you will need a cnf bridge configured, the best way is adding srst ip add as a third option in sccp ccm group, once your cnf bridge is registered to srst router (it take some more times than phones) you will need a dn octo line (recomended) configured as conference ad-hoc 1: srst auto all: Then once ephones are registered to srst these combinations worked for me: == telephony-service privacy ! (default) ephone 1 no privacy privacy-button ! from the button disable or enable it == telephony-service privacy / no privacy ! you can also manage from here ephone 1 no privacy privacy-button ! === telephony-service privacy ! (default) ephone 1 privacy-button privacy on / privacy off ! enable and disable from ephone === If you enable/disable localy you can't enable/disable globaly ephone 1 privacy on/off ! enable/disable privacy telephony-service privacy/no privacy! this won't enable/disable privacy becouse you have enable/disable it localy at ephone 2: srst auto none: follow vc approach described above hth Date: Tue, 15 Jun 2010 06:09:52 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) Angel - I think you are right. The only way I can see of configuring privacy on/off would be through the ephone section itself. Privacy isn't an option with an ephone
Re: [OSL | CCIE_Voice] CBarge on UCM not working
Ok, I was missing Privacy = Off on both phones. Now, when trying to cBarge, I immediately get a fast busy and the phone I am trying to cbarge from displays Cannot Complete Conference. I know the hardware conference on BR1 is registered to UCM, it is assigned to an MRG which is assigned to an MRGL that is assigned to DP_BR1. When I configured my dspfarm profile on BR1 the max sessions I could configure is 2. I believe that should still be adequate. On Jul 14, 2010, at 11:12 PM, ccieid1ot wrote: Set privacy to off. On Thu, Jul 15, 2010 at 12:30 AM, Mark Holloway m...@markholloway.com wrote: So, I have two phones Br1Ph1 and Br1Ph2 sharing the same DN. Both have the same softkey template I created called Standard User-CBarge which includes Remote in Use - CBarge. Both phones are in Device Pool BR1 which includes the BR1_MRGL which has MRG_BR1_HW_CONF assigned. This MRG contains the BR1-CONF which is my hardware conferencing on the BR1 router and shows as registered with UCM. On the Device Phone page for both phones, I have Single Button Barge set to CBarge. If I understand CBarge correctly, when Br1Ph1 answers a call I should have the option on Br1Ph2 to press the CBarge softkey? Br1Ph2's lamp lights red when Br1Ph1 is on the shared line, but I never see a CBarge softkey. Perhaps I am missing something significant. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CBarge on UCM not working
Man, I forgot I was working with Locations and AAR last night and I forgot to remove the low bandwidth I set for BR1 to test AAR. cBarge is working now based on setting Privacy to Off. Thank you to everyone who helped. It is greatly appreciated. :) On Jul 14, 2010, at 11:28 PM, Mark Holloway wrote: Ok, I was missing Privacy = Off on both phones. Now, when trying to cBarge, I immediately get a fast busy and the phone I am trying to cbarge from displays Cannot Complete Conference. I know the hardware conference on BR1 is registered to UCM, it is assigned to an MRG which is assigned to an MRGL that is assigned to DP_BR1. When I configured my dspfarm profile on BR1 the max sessions I could configure is 2. I believe that should still be adequate. On Jul 14, 2010, at 11:12 PM, ccieid1ot wrote: Set privacy to off. On Thu, Jul 15, 2010 at 12:30 AM, Mark Holloway m...@markholloway.com wrote: So, I have two phones Br1Ph1 and Br1Ph2 sharing the same DN. Both have the same softkey template I created called Standard User-CBarge which includes Remote in Use - CBarge. Both phones are in Device Pool BR1 which includes the BR1_MRGL which has MRG_BR1_HW_CONF assigned. This MRG contains the BR1-CONF which is my hardware conferencing on the BR1 router and shows as registered with UCM. On the Device Phone page for both phones, I have Single Button Barge set to CBarge. If I understand CBarge correctly, when Br1Ph1 answers a call I should have the option on Br1Ph2 to press the CBarge softkey? Br1Ph2's lamp lights red when Br1Ph1 is on the shared line, but I never see a CBarge softkey. Perhaps I am missing something significant. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CBarge on UCM not working
So, I have two phones Br1Ph1 and Br1Ph2 sharing the same DN. Both have the same softkey template I created called Standard User-CBarge which includes Remote in Use - CBarge. Both phones are in Device Pool BR1 which includes the BR1_MRGL which has MRG_BR1_HW_CONF assigned. This MRG contains the BR1-CONF which is my hardware conferencing on the BR1 router and shows as registered with UCM. On the Device Phone page for both phones, I have Single Button Barge set to CBarge. If I understand CBarge correctly, when Br1Ph1 answers a call I should have the option on Br1Ph2 to press the CBarge softkey? Br1Ph2's lamp lights red when Br1Ph1 is on the shared line, but I never see a CBarge softkey. Perhaps I am missing something significant. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue
Thanks, Graham. I'm glad I could contribute something beneficial! :) On Jul 13, 2010, at 10:40 AM, Graham Hopkins wrote: Well the answer on the SIP gateway is to rewrite the SIP message using voice service voip sip sip-profiles 1 ! voice class sip-profiles 1 response 183 sip-header Remote-Party-ID modify (.*):9(.*) \1:\2 Thanks to Mark Holloway's blog for pointing me in the right direction. Jul 13 17:31:59.297: //-1//SIP/Info/sip_profiles_application_modify_remove_header: Header before modification : Remote-Party-ID: sip:95621...@10.10.110.3;party=called;screen=no;privacy=off Jul 13 17:31:59.297: //-1//SIP/Info/sip_profiles_application_modify_remove_header: Header after modification : Remote-Party-ID: sip:5621...@10.10.110.3;party=called;screen=no;privacy=off Sent: SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 10.10.210.11:5060;branch=z9hG4bK80124aba121 From: Br2 Ph1 sip:5623...@10.10.210.11;tag=bdc70633-cf9d-4ffb-8d2d-b6a883aec812-49066039 To: sip:5621...@10.10.110.3;tag=26568D0-1811 Date: Tue, 13 Jul 2010 17:31:59 GMT Call-ID: 890d9700-c3c1a30f-ad7-bd20...@10.10.210.11 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: sip:5621...@10.10.110.3;party=called;screen=no;privacy=off Contact: sip:5621...@10.10.110.3:5060;transport=tcp Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 235 v=0 o=CiscoSystemsSIP-GW-UserAgent 6425 6036 IN IP4 10.10.110.3 s=SIP Call c=IN IP4 10.10.110.3 t=0 0 m=audio 19552 RTP/AVP 18 19 c=IN IP4 10.10.110.3 On 13 Jul 2010, at 08:13, Graham Hopkins wrote: Yes similar to what I was doing. Also tried doing the same with a SIP gateway, which is a real pain as the SIP trunk from CUCM doesn't pass the type and plan. Also does anyone know if there is a SIP equivalent of no supplementary-service h225-notify cid-update - or any other way of preventing the 9 appearing on the phone display. Regards Graham On 13 Jul 2010, at 03:53, Mark Holloway wrote: Ok, so this is how set my H.323 gateway to operate. For example, a single POTS dial peer to handle Local calls (7 digit called, 7 digit calling number) for normal operation with UCM and when the router is in SRST mode. dial-peer voice 4 voip description Calls from UCM add 9 translation-profile incoming ADD9 incoming called-number . voice translation-profile ADD9 translate called 50 voice translation-rule 50 rule 1 /\(.*\)/ /9\1/ dial-peer voice 920 pots description LOCAL translation-profile outgoing LOCAL destination-pattern 9[2-9]..$ port 0/0/0:23 voice translation-profile LOCAL translate calling 11 translate called 10 voice translation-rule 10 rule 1 // // type unknown subscriber plan unknown isdn ! voice translation-rule 11 rule 1 /\(^2...$\)/ /222\1/ On Jul 9, 2010, at 12:22 PM, Graham Hopkins wrote: With the two sets of dial-peers you do need to take care that overlapping patterns don't cause problems in SRST for example I hit issues with [2-9].. and 91[2-9]..[2-9].. I decided to go with the translation pattern to put the 9 back on to the digits sent by CUCM, but this 9 will still show up on the phone unless you use voice service voip no supplementary-service h225-notify cid-update Regards Graham Hopkins On 9 Jul 2010, at 19:21, Mark Holloway wrote: Sounds like you have the PSTN to CUCM part working ok. This is what I have been doing. On the H323 router create the following dial-peer dial-peer voice 10 pots destination-pattern [2-9]..$ port 0/0/0:23 On CUCM have a Route Pattern that handles \+1414.[2-9]XX for calls originated by BR1 phones and strip the predot. This way you can assign the call type as Subscriber within the Route Pattern and if local calls are supposed to send a 7 digit calling number you can set the calling party transformation mask within the Route Pattern to XXX. You could have a second dial-peer on your H323 router for SRST dial-peer voice 910 pots destination-pattern 9[2-9]..$ port 0/0/0:23 translation-profile outgoing LOCAL There are really two different ways to handle H323 gateway dial-peers. You can strip the 9 in CUCM then add it back on the H323 gateway through a translation-profile and only have one set of dial-peers. Or, build your dial-peers for local, LD, international, and 911 without the 9, copy/paste in notepad and put a 9 in front of the dial-peer number and the destination-pattern then paste it into your router. You will have two sets of dial-peers for SRST and normal operation. On Jul 9, 2010, at 10:28 AM
Re: [OSL | CCIE_Voice] CUE not stating PSTN Calling Party Number
Thanks. In the GUI it's under Voicemail VM Configuration Play Caller ID for External Callers = YES On Jul 11, 2010, at 11:52 PM, Graham Hopkins wrote: Add the line voicemail callerid not sure where it is in the GUI - must check Graham On 12 Jul 2010, at 06:42, Mark Holloway m...@markholloway.com wrote: I'm not quite sure what's causing this issue, but when any PSTN number calls Br2Ph1 or Br2Ph2 I can see the Calling party information fine in the ISDN setup and on the display of the phones, but if I let it go to voicemail and then check messages from the phones after MWI lights up, CUE always says An unknown caller left you a message. I'm not sure why CUE isn't stating the Calling Party number? Any ideas? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue
I proceeded to use the method where all my H323 dial-peers start with 9 in the destination-pattern. I imagine it's more work to have UCM keep the 9 on the dialed number because of TEHO to multiple gateways, it gets very busy to know when to prepend and not prepend in UCM route lists. Assuming 9 is stripped on UCM and the H323 gateway is adding 9 before sending the call to a POTS dial peer, is a VoIP dial-peer being created to match any incoming call and then it is sent through a translation-profile so it can match a POTS dial peer? On Jul 9, 2010, at 12:22 PM, Graham Hopkins wrote: With the two sets of dial-peers you do need to take care that overlapping patterns don't cause problems in SRST for example I hit issues with [2-9].. and 91[2-9]..[2-9].. I decided to go with the translation pattern to put the 9 back on to the digits sent by CUCM, but this 9 will still show up on the phone unless you use voice service voip no supplementary-service h225-notify cid-update Regards Graham Hopkins On 9 Jul 2010, at 19:21, Mark Holloway wrote: Sounds like you have the PSTN to CUCM part working ok. This is what I have been doing. On the H323 router create the following dial-peer dial-peer voice 10 pots destination-pattern [2-9]..$ port 0/0/0:23 On CUCM have a Route Pattern that handles \+1414.[2-9]XX for calls originated by BR1 phones and strip the predot. This way you can assign the call type as Subscriber within the Route Pattern and if local calls are supposed to send a 7 digit calling number you can set the calling party transformation mask within the Route Pattern to XXX. You could have a second dial-peer on your H323 router for SRST dial-peer voice 910 pots destination-pattern 9[2-9]..$ port 0/0/0:23 translation-profile outgoing LOCAL There are really two different ways to handle H323 gateway dial-peers. You can strip the 9 in CUCM then add it back on the H323 gateway through a translation-profile and only have one set of dial-peers. Or, build your dial-peers for local, LD, international, and 911 without the 9, copy/paste in notepad and put a 9 in front of the dial-peer number and the destination-pattern then paste it into your router. You will have two sets of dial-peers for SRST and normal operation. On Jul 9, 2010, at 10:28 AM, Joaquim Fernandes wrote: HI Team, I have an issue with this question. Question === when pstn number 414363 call phones at site b they should display 7 digits on the phone display. For example when pstn calling ph 1 or ph 2 at branch B it should display 363 on the screen. My solution = I have added +1 in Device pool of Branch B to make it globalised when the call comes in the H323 Branch B router. I have created \+1414.363 calling party transformation mask. I have created \+1414.363 route pattern with Branch B as the gateway. (branch b is the H323 gateway). So on the Route pattern i have just done predot and in the branch b route list i have done NANP-Predot and prefix 9. I have done vice versa as well but things doesnt work. IN the branch B router i have a dial-peer for the local calls. dial-peer voice 1 pots destination-pattern 9[2-9].. port 0/0/0:23 translation-profile outgoing local translation-rule 1 rule 1 /^8.../ /363\0/ translation-rule 2 rule 1 // // type any sub plan any isdn translation-profile lcoal translate called 2 translate calling 1 Note: If i make a dial-peer without 9 i.e (...) Then the display is perfect. but i dont feel this would be the solution. because in srst this would be an issue. Issue = The issue is when PSTN phone 414363 calls Brach B ph1 or ph2 the caller id is 363 and in the missed call its globalized number +1414363 as per the question. But when i do redial using missed calls from Branch B ph1 or ph2 the calling number on the ip phones is displayed as 9363 (9 is the secondary dial tone) and the call goes through. Evrything works fine except for the display on ph1 or ph2, there is 9. How do i get rid of it 9. I hope i have made my point very clear of what issue i am facing. The question state the display on the phone should be only 363 and not 9363. Regards, JF ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please
Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue
Ok, so this is how set my H.323 gateway to operate. For example, a single POTS dial peer to handle Local calls (7 digit called, 7 digit calling number) for normal operation with UCM and when the router is in SRST mode. dial-peer voice 4 voip description Calls from UCM add 9 translation-profile incoming ADD9 incoming called-number . voice translation-profile ADD9 translate called 50 voice translation-rule 50 rule 1 /\(.*\)/ /9\1/ dial-peer voice 920 pots description LOCAL translation-profile outgoing LOCAL destination-pattern 9[2-9]..$ port 0/0/0:23 voice translation-profile LOCAL translate calling 11 translate called 10 voice translation-rule 10 rule 1 // // type unknown subscriber plan unknown isdn ! voice translation-rule 11 rule 1 /\(^2...$\)/ /222\1/ On Jul 9, 2010, at 12:22 PM, Graham Hopkins wrote: With the two sets of dial-peers you do need to take care that overlapping patterns don't cause problems in SRST for example I hit issues with [2-9].. and 91[2-9]..[2-9].. I decided to go with the translation pattern to put the 9 back on to the digits sent by CUCM, but this 9 will still show up on the phone unless you use voice service voip no supplementary-service h225-notify cid-update Regards Graham Hopkins On 9 Jul 2010, at 19:21, Mark Holloway wrote: Sounds like you have the PSTN to CUCM part working ok. This is what I have been doing. On the H323 router create the following dial-peer dial-peer voice 10 pots destination-pattern [2-9]..$ port 0/0/0:23 On CUCM have a Route Pattern that handles \+1414.[2-9]XX for calls originated by BR1 phones and strip the predot. This way you can assign the call type as Subscriber within the Route Pattern and if local calls are supposed to send a 7 digit calling number you can set the calling party transformation mask within the Route Pattern to XXX. You could have a second dial-peer on your H323 router for SRST dial-peer voice 910 pots destination-pattern 9[2-9]..$ port 0/0/0:23 translation-profile outgoing LOCAL There are really two different ways to handle H323 gateway dial-peers. You can strip the 9 in CUCM then add it back on the H323 gateway through a translation-profile and only have one set of dial-peers. Or, build your dial-peers for local, LD, international, and 911 without the 9, copy/paste in notepad and put a 9 in front of the dial-peer number and the destination-pattern then paste it into your router. You will have two sets of dial-peers for SRST and normal operation. On Jul 9, 2010, at 10:28 AM, Joaquim Fernandes wrote: HI Team, I have an issue with this question. Question === when pstn number 414363 call phones at site b they should display 7 digits on the phone display. For example when pstn calling ph 1 or ph 2 at branch B it should display 363 on the screen. My solution = I have added +1 in Device pool of Branch B to make it globalised when the call comes in the H323 Branch B router. I have created \+1414.363 calling party transformation mask. I have created \+1414.363 route pattern with Branch B as the gateway. (branch b is the H323 gateway). So on the Route pattern i have just done predot and in the branch b route list i have done NANP-Predot and prefix 9. I have done vice versa as well but things doesnt work. IN the branch B router i have a dial-peer for the local calls. dial-peer voice 1 pots destination-pattern 9[2-9].. port 0/0/0:23 translation-profile outgoing local translation-rule 1 rule 1 /^8.../ /363\0/ translation-rule 2 rule 1 // // type any sub plan any isdn translation-profile lcoal translate called 2 translate calling 1 Note: If i make a dial-peer without 9 i.e (...) Then the display is perfect. but i dont feel this would be the solution. because in srst this would be an issue. Issue = The issue is when PSTN phone 414363 calls Brach B ph1 or ph2 the caller id is 363 and in the missed call its globalized number +1414363 as per the question. But when i do redial using missed calls from Branch B ph1 or ph2 the calling number on the ip phones is displayed as 9363 (9 is the secondary dial tone) and the call goes through. Evrything works fine except for the display on ph1 or ph2, there is 9. How do i get rid of it 9. I hope i have made my point very clear of what issue i am facing. The question state the display on the phone should be only 363 and not 9363. Regards, JF ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUE not stating PSTN Calling Party Number
I'm not quite sure what's causing this issue, but when any PSTN number calls Br2Ph1 or Br2Ph2 I can see the Calling party information fine in the ISDN setup and on the display of the phones, but if I let it go to voicemail and then check messages from the phones after MWI lights up, CUE always says An unknown caller left you a message. I'm not sure why CUE isn't stating the Calling Party number? Any ideas? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] VWIC2-2MFT-T1 shared
You can have a data T1 and a PRI T1, or a data E1 and a PRI E1, but you can't split T1/E1 across the same card. On Jul 9, 2010, at 7:05 AM, Kevin Damisch wrote: I’ve seen this question before but can’t find it. On a VWIC2-2MFT-T1/E1 card, can you configure or are there issues with having a data T1 on one port and an E1 PRI on the other port on the same card? I thought the VWIC2 cards supported that, but not the older ones. This communication (including any attachments) is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If you are not the intended recipient, any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify Vital Support Systems at 515 334 5700 and delete or destroy all copies and the original document. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Globalisation/Localisation Issue
Sounds like you have the PSTN to CUCM part working ok. This is what I have been doing. On the H323 router create the following dial-peer dial-peer voice 10 pots destination-pattern [2-9]..$ port 0/0/0:23 On CUCM have a Route Pattern that handles \+1414.[2-9]XX for calls originated by BR1 phones and strip the predot. This way you can assign the call type as Subscriber within the Route Pattern and if local calls are supposed to send a 7 digit calling number you can set the calling party transformation mask within the Route Pattern to XXX. You could have a second dial-peer on your H323 router for SRST dial-peer voice 910 pots destination-pattern 9[2-9]..$ port 0/0/0:23 translation-profile outgoing LOCAL There are really two different ways to handle H323 gateway dial-peers. You can strip the 9 in CUCM then add it back on the H323 gateway through a translation-profile and only have one set of dial-peers. Or, build your dial-peers for local, LD, international, and 911 without the 9, copy/paste in notepad and put a 9 in front of the dial-peer number and the destination-pattern then paste it into your router. You will have two sets of dial-peers for SRST and normal operation. On Jul 9, 2010, at 10:28 AM, Joaquim Fernandes wrote: HI Team, I have an issue with this question. Question === when pstn number 414363 call phones at site b they should display 7 digits on the phone display. For example when pstn calling ph 1 or ph 2 at branch B it should display 363 on the screen. My solution = I have added +1 in Device pool of Branch B to make it globalised when the call comes in the H323 Branch B router. I have created \+1414.363 calling party transformation mask. I have created \+1414.363 route pattern with Branch B as the gateway. (branch b is the H323 gateway). So on the Route pattern i have just done predot and in the branch b route list i have done NANP-Predot and prefix 9. I have done vice versa as well but things doesnt work. IN the branch B router i have a dial-peer for the local calls. dial-peer voice 1 pots destination-pattern 9[2-9].. port 0/0/0:23 translation-profile outgoing local translation-rule 1 rule 1 /^8.../ /363\0/ translation-rule 2 rule 1 // // type any sub plan any isdn translation-profile lcoal translate called 2 translate calling 1 Note: If i make a dial-peer without 9 i.e (...) Then the display is perfect. but i dont feel this would be the solution. because in srst this would be an issue. Issue = The issue is when PSTN phone 414363 calls Brach B ph1 or ph2 the caller id is 363 and in the missed call its globalized number +1414363 as per the question. But when i do redial using missed calls from Branch B ph1 or ph2 the calling number on the ip phones is displayed as 9363 (9 is the secondary dial tone) and the call goes through. Evrything works fine except for the display on ph1 or ph2, there is 9. How do i get rid of it 9. I hope i have made my point very clear of what issue i am facing. The question state the display on the phone should be only 363 and not 9363. Regards, JF ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] 3750 QoS Police
I'm attempting to police VoIP signaling on Fast1/0/1 of a 3750 switch that is configured as a trunk port connecting to the HQ router. I can't apply the service-policy in the output direction. Am I thinking about this the wrong way because I can apply it in the inbound direction. # show run interface FastEthernet1/0/1 description ** To R1-HQ Gigabit Ethernet 0/0 ** switchport trunk encapsulation dot1q switchport mode trunk HQ-3750(config-if)#service-policy output VOIP-SIGNAL police command is not supported for this interface The interface does not support the specified policy configuration and/or parameter values. Warning: Assigning a policy map to the output side of an interface not supported HQ-3750(config-if)#service-policy input VOIP-SIGNAL HQ-3750(config-if)#do sh run interface FastEthernet1/0/1 description ** R1-HQ Gigabit Ethernet 0/0 ** switchport trunk encapsulation dot1q switchport mode trunk service-policy input VOIP-SIGNAL mls qos map policed-dscp 24 to 8 mls qos map cos-dscp 0 8 16 24 32 46 48 56 mls qos class-map match-any SIGNAL match ip dscp cs3 policy-map VOIP-SIGNAL class SIGNAL police 32000 8000 exceed-action policed-dscp-transmit ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] NTP
If a router (for example, HQ) is configured with the ntp server x.x.x.x command to sync time from another source, but I want another device (such as PUB) to get its time from the HQ router, do I also need to configure the HQ router with ntp server stratum X or can UCM simply get the time sync from HQ without the stratum command? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] NTP
Yikes, I meant ntp master stratum X not ntp server stratum X On Jul 8, 2010, at 3:57 PM, Mark Holloway wrote: If a router (for example, HQ) is configured with the ntp server x.x.x.x command to sync time from another source, but I want another device (such as PUB) to get its time from the HQ router, do I also need to configure the HQ router with ntp server stratum X or can UCM simply get the time sync from HQ without the stratum command? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] NTP
Cool, thanks Graham and Randall. On Jul 8, 2010, at 4:11 PM, Graham Hopkins wrote: Default stratum is 8 so a simple ntp master will work Graham On 8 Jul 2010, at 23:59, Mark Holloway m...@markholloway.com wrote: Yikes, I meant ntp master stratum X not ntp server stratum X On Jul 8, 2010, at 3:57 PM, Mark Holloway wrote: If a router (for example, HQ) is configured with the ntp server x.x.x.x command to sync time from another source, but I want another device (such as PUB) to get its time from the HQ router, do I also need to configure the HQ router with ntp server stratum X or can UCM simply get the time sync from HQ without the stratum command? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] isdn plan
Are you setting plan/type for both the called and calling numbers or just one of them? For example, if a task says the pstn provider wants the called party number type set and you set the plan/type for the called number, are you just leaving the calling portion set to CallManager or are you setting the plan/type for that as well? On Jul 7, 2010, at 11:43 AM, Berry, Matthew J. wrote: I make a habit of always setting the plan to ISDN. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Wednesday, July 07, 2010 1:40 PM To: OSL osl Subject: [OSL | CCIE_Voice] isdn plan When tasked with setting the call type to unknown, subscriber, national, or international, are you guys also setting the plan to isdn or are you just specifying the type and leaving the plan as unknown even though all the pstn access is isdn? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] isdn plan
When tasked with setting the call type to unknown, subscriber, national, or international, are you guys also setting the plan to isdn or are you just specifying the type and leaving the plan as unknown even though all the pstn access is isdn? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] isdn plan
I do too but I was wondering if the task doesn't specify setting the plan would we get knocked for doing it? I wouldn't think so but wasn't sure if I was the only one doing it. On Jul 7, 2010, at 11:43 AM, Berry, Matthew J. wrote: I make a habit of always setting the plan to ISDN. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Wednesday, July 07, 2010 1:40 PM To: OSL osl Subject: [OSL | CCIE_Voice] isdn plan When tasked with setting the call type to unknown, subscriber, national, or international, are you guys also setting the plan to isdn or are you just specifying the type and leaving the plan as unknown even though all the pstn access is isdn? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] HQ BR2 - CUE Transcoding
If calls should complete using G.729 from HQ/BR1 to CUE on BR2 which is G.711u, can the transcoding be configured on the BR2 router locally or does it need to happen via the originating party's transcoding resources in UCM? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] HQ BR2 - CUE Transcoding
Thanks, everyone. I configured the Transcoder locally on BR2. Now my issue is when I call from HQ to BR2 CUE, the call is answered by CUE but I do not hear the CUE attendant. The HQ phone shows RTP Sender packets incrementing but my Rcvr packets is not incrementing. Local BR2 phones work fine, so I know CUE is up and running. Has anyone experienced one-way audio with CUE before while Transcoding? r3-br2#show sccp Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.254, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 r3-br2#show sdspfarm units mtp-3 Device:CME-XCODE TCP socket:[7] REGISTERED in SCCP ver 17/10 actual_stream:8 max_stream 8 IP:192.168.1.254 31790 MTP Dixieland keepalive 19 Supported codec: G711Ulaw G711Alaw G729a G729ab r3-br2# show run | sec teleph telephony-service sdspfarm units 5 sdspfarm transcode sessions 6 sdspfarm tag 2 CME-XCODE r3-br2#show dspfarm profile 2 Dspfarm Profile Configuration Profile ID = 2, Service = TRANSCODING, Resource ID = 2 Profile Description : Profile Service Mode : Non Secure Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 4 Number of Resource Available : 4 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 On Jul 6, 2010, at 10:23 AM, Graham Hopkins wrote: You'll need to do it at BR2 - if you do it at HQ/BR1 it will be G.711 across the WAN. Graham On 6 Jul 2010, at 17:41, Mark Holloway wrote: If calls should complete using G.729 from HQ/BR1 to CUE on BR2 which is G.711u, can the transcoding be configured on the BR2 router locally or does it need to happen via the originating party's transcoding resources in UCM? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] HQ BR2 - CUE Transcoding
Matthew, Thanks for the suggestions. I do have the dial-peer set to G711u. However I didn't add G729r8 in the dspfarm profile but that didn't seem to change the result. I performed a shutdown, added the codec, no shut, but I'm still not receiving RTP to HqPh1. Argh, this one is a bugger! I thought I had this nailed too, but that's what happens when too much time goes by in between the repetition of practice. dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP I'm doing a reload now. I doubt that will fix it but since this is my own lab you never know.. On Jul 6, 2010, at 10:46 AM, Berry, Matthew J. wrote: Mark - Make sure that g729r8 is added under the dspfarm profile. Also, make sure you CUE dial-peer is hardcoded to be G711ulaw. Otherwise, it will try to use the default which is g729. Matthew Berry, CCVP, Sr. Unified Communications Engineer mjbe...@kroll.com -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Mark Holloway Sent: Tuesday, July 06, 2010 12:44 PM To: Graham Hopkins Cc: OSL osl Subject: Re: [OSL | CCIE_Voice] HQ BR2 - CUE Transcoding Thanks, everyone. I configured the Transcoder locally on BR2. Now my issue is when I call from HQ to BR2 CUE, the call is answered by CUE but I do not hear the CUE attendant. The HQ phone shows RTP Sender packets incrementing but my Rcvr packets is not incrementing. Local BR2 phones work fine, so I know CUE is up and running. Has anyone experienced one-way audio with CUE before while Transcoding? r3-br2#show sccp Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.254, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 r3-br2#show sdspfarm units mtp-3 Device:CME-XCODE TCP socket:[7] REGISTERED in SCCP ver 17/10 actual_stream:8 max_stream 8 IP:192.168.1.254 31790 MTP Dixieland keepalive 19 Supported codec: G711Ulaw G711Alaw G729a G729ab r3-br2# show run | sec teleph telephony-service sdspfarm units 5 sdspfarm transcode sessions 6 sdspfarm tag 2 CME-XCODE r3-br2#show dspfarm profile 2 Dspfarm Profile Configuration Profile ID = 2, Service = TRANSCODING, Resource ID = 2 Profile Description : Profile Service Mode : Non Secure Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 4 Number of Resource Available : 4 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 On Jul 6, 2010, at 10:23 AM, Graham Hopkins wrote: You'll need to do it at BR2 - if you do it at HQ/BR1 it will be G.711 across the WAN. Graham On 6 Jul 2010, at 17:41, Mark Holloway wrote: If calls should complete using G.729 from HQ/BR1 to CUE on BR2 which is G.711u, can the transcoding be configured on the BR2 router locally or does it need to happen via the originating party's transcoding resources in UCM? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] HQ BR2 - CUE Transcoding
No luck. Here is the BR2 configuration. voice-card 0 dspfarm dsp services dspfarm sccp local Vlan500 sccp ccm 192.168.1.254 identifier 1 priority 1 version 7.0 sccp ! sccp ccm group 2 bind interface Vlan500 associate ccm 1 priority 1 associate profile 2 register CME-XCODE ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729r8 codec g729br8 codec g729abr8 codec g729ar8 maximum sessions 4 associate application SCCP dial-peer voice 4000 voip description CUE destination-pattern 4000 session protocol sipv2 session target ipv4:192.168.1.253 dtmf-relay sip-notify codec g711ulaw no vad telephony-service sdspfarm units 5 sdspfarm transcode sessions 4 sdspfarm tag 2 CME-XCODE conference hardware no auto-reg-ephone authentication credential administrator cisco max-ephones 2 max-dn 10 ip source-address 192.168.1.254 port 2000 url services http://192.168.1.253/voiceview/common/login.do url authentication http://192.168.1.254/CCMCIP/authenticate.asp time-format 24 voicemail 4000 max-conferences 2 gain -6 call-forward pattern .T web admin system name administrator password cisco dn-webedit time-webedit transfer-system full-consult transfer-pattern .T When HQ calls CUE I'll get the following output on the BR2 router even though the HQ phone (192.168.50.29) doesn't increment Rcvr Packets. Call from HQ to Br2Ph1 or Br2Ph2 work fine (of course, no transcoding required). r3-br2#show voip rtp connections VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 17 18 1653227762 192.168.1.254 192.168.50.29 2 18 17 1844016904 192.168.1.254 192.168.1.253 Found 2 active RTP connections On Jul 6, 2010, at 11:26 AM, Graham Hopkins wrote: Mark is that the full telephony-service below or an extract? You'll need max-dn, max-ephone and a source address to fire up sccp fully I can see the dspfarm profile has registered so just taking a guess really. However did this myself this afternoon - had some dtmf-relay issues but transcoder was ok - post the whole config if you like. Graham On 6 Jul 2010, at 18:43, Mark Holloway m...@markholloway.com wrote: Thanks, everyone. I configured the Transcoder locally on BR2. Now my issue is when I call from HQ to BR2 CUE, the call is answered by CUE but I do not hear the CUE attendant. The HQ phone shows RTP Sender packets incrementing but my Rcvr packets is not incrementing. Local BR2 phones work fine, so I know CUE is up and running. Has anyone experienced one-way audio with CUE before while Transcoding? r3-br2#show sccp Transcoding Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.1.254, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 r3-br2#show sdspfarm units mtp-3 Device:CME-XCODE TCP socket:[7] REGISTERED in SCCP ver 17/10 actual_stream:8 max_stream 8 IP:192.168.1.254 31790 MTP Dixieland keepalive 19 Supported codec: G711Ulaw G711Alaw G729a G729ab r3-br2# show run | sec teleph telephony-service sdspfarm units 5 sdspfarm transcode sessions 6 sdspfarm tag 2 CME-XCODE r3-br2#show dspfarm profile 2 Dspfarm Profile Configuration Profile ID = 2, Service = TRANSCODING, Resource ID = 2 Profile Description : Profile Service Mode : Non Secure Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 4 Number of Resource Available : 4 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 On Jul 6, 2010, at 10:23 AM, Graham Hopkins wrote: You'll need to do it at BR2 - if you do it at HQ/BR1 it will be G.711 across the WAN. Graham On 6 Jul 2010, at 17:41, Mark Holloway wrote: If calls should complete using G.729 from HQ/BR1 to CUE on BR2 which is G.711u, can the transcoding be configured on the BR2 router locally or does it need to happen via the originating
Re: [OSL | CCIE_Voice] Music on hold from router flash (Piano music)
Is your Site B router MGCP or H323? With an H323 gateway I could get the router to stream the local piano music while the MoH server is set to one hop in UCM. With an MGCP gateway I couldn't get this to work and it always streams from UCM unless the router is in SRST mode then it plays piano music. I am also using a home lab. I tried to isolate why this wasn't working but could never come up with a root cause. On Jul 2, 2010, at 7:52 AM, Afzal Bhutta wrote: Sorry Folks not providing details in first attempt. Thanks for all and special thanks to Matthew Berry and Randall Saborio for their interest and figured out this issue. Let’s make thing more understandable. I am working in my home lab. I am trying to spoof call manager. My target is to get music from router flash for HQ and for siteB not from call manager. Call manager is configured as I explain below. I have adjusted my ServerMax Hops to 15 for the M.cast it is working fine for HQ and SiteB but I am hearing music from Call manger not from router flash (Piano music) What I performed on the routers. I have enabled Muticast-routing on HQ and site B I have use IP pim dense mode commands on Voice-vlan interfaces, Loopback0, and serial interfaces which are connected to frame relay (WAN links) both for HQ and Site B. CCM-manager music on hold command is also on both sites. Site B is providing SRST. SRST is configured using telephony command. Troubleshooting: When I adjusted ServerMax Hops to 2 still I can hear music from call manager. I tested it in this way. Call from HQ to SiteB, HQ-ph is put on hold and I can hear music from the router flash (Piano music) If site B is put on hold I can hear call manger music (Actually it should be from router flash- Am I right?) When I adjusted my ServerMax Hops to 1 for the M.cast it is not working I can not hear any music just silence even no beeps. Even within HQ phone, when they call each other I put one of them on hold I can not hear any music not from call manger nor from router flash. Yes I can hear music from router flash when I call from HQ to Site B with adjusted my ServerMax Hops to 2 and put HQ phone on hold but when I put hold for SIteB Phone nothing I can hear completely silent even no beeps Here is Call manager config details, MOH is multicast on 239.1.1.1 port 16384.Allow multicasting is enable on CUCM-PUB. CallManager MoH Server Increment Multicast on = IP Address CallManager MoH ServerMax Hops = 1 MOH Audio Source:? SampleAudioSource (1) = Allow Multicasting In Media Resource Group =? Use Multicast for MOH Audio (This is enable) CME is completely separate side,It is not participating in this Scenario. IP Voice Media Streaming App is enabled for G729 and G722 in service parameter.(Cisco IP Voice Media Streaming App = 711 uulaw and 729 Annex A selected) I have MOH region with G711ulaw enable with all other region with codec G711ulaw. HQ device pool using MRGL SiteB device pool using MRGL MRGL contains MOH-PUB-MULTI-RG All phones within site (Intra-site) using G711ulaw where as between site (Inter-site) they are using G729ulaw. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUE/CCM JTapi and MWI
Anyone have any luck getting MWI to work when CUE is integrated with CCM? According to Cisco's documentation you do not need MWI numbers when JTAPI is used. Voicemail is working as expected but I cannot get MWI to light up on the phones using that are using the UCM Voicemail Profile I have created for CUE (assigned to Br2Ph1 and Br2Ph2). CUE successfully shows that it is registered with the CTI Route Point which explains why Voicemail works. My Application User (JTAPI-CUE) has the CTI Route Point and additional 3 ports associated with it under Controlled Devices and I have assigned Standard CTI User under Permissions Information. I even ticked Accept Unsolicited Notification but that didn't make a difference. I appreciate any assistance.. ccn subsystem jtapi ctiport 3111 3112 3113 ccm-manager address 10.10.210.11 10.10.210.10 ccm-manager credentials hidden kqp8kECeSyBmpARJPQkSHY8Uxj6U33PNSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmPSd8ZZNgd+Y9J3xlk2B35j0nfGWTYHfmP end subsystem ccn subsystem sip gateway address 10.57.126.1 mwi sip unsolicited end subsystem ccn trigger http urlname msgnotifytrg application msgnotification enabled maxsessions 2 end trigger ccn trigger http urlname mwiapp application ciscomwiapplication enabled maxsessions 1 end trigger ccn trigger jtapi phonenumber 3110 application voicemail enabled maxsessions 3 end trigger ccn trigger sip phonenumber 3110 application voicemail enabled maxsessions 3 end trigger ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
