[OSL | CCIE_Voice] silly dtmf,fax,modem relay question
i was reading about the rfc2833 dtmf support. one doubt really got me thinking... i hope its not too silly for this group.. incase its too silly, then i have hey its in the blueprint excuse ;) so when we use dtmf relay.. or any relay..say modem or fax.. when mean out band, we mean atleast 2 channels right?? one for the usual rtp and the other for the fax,modem,dtmf channel right?? So in vic-2fxs we shouldnt be able to do relay right??. but cisco says we can. Before Cisco IOS Release 12.4(6)XE, SCCP gateways supported fax passthrough only. In Cisco IOS Release 12.4(6)XE and later releases, SCCP Gateway Controlled Fax Relay adds support http://www.cisco.com/en/US/docs/ios/voice/fxs/configuration/guide/fxsrelay.html#wp1049432 well... i am just a little confused as to how do we do out of band if there are no more than one channel?? i am sure by now that my understanding of outofband is also flawed... do correct me. joel. -- it's not the years in your life that count. It's the life in your years. Abraham Lincoln
Re: [OSL | CCIE_Voice] CTI RP
I integrated IPCC with CCM. IPCC and CCM are both collocated on the same box. Created new application with trigger point 1700 ON the CCM PUB and SUB the CTI RP shows me it is registered. But when I call 1700 from the IP Phone or from PSTN it keeps coming busy all time. Rest the CTI RP, reset the IIS server ( Don't think that will play any part), I even rebooted both PUB and SUB, don't know why it is busy all the time. Any help is much appreciated. Thanks Narinder CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system.
[OSL | CCIE_Voice] lab v2 / v3 questions
Hello, I prepared for voice v2 but I couldn't find a place for lab before Agust. v3 will be started middle of July. I need a guide for the voice lab exam which way I can continue; - Continue with v2 and wait till find a place or continue with v3 and take the lab in Agust? - Do you thing there are too many difference between v2 and v3? - I have still vouchers for lab and will I have option to login v2 or v3 for preparation? Thanks, M.Deniz KIZILCABOLUK Alcatel-Lucent Bell NV IPTC I'ntl System - Integr. Test Copernicuslaan 50, 2018 Antwerp, Belgium E-mail : deniz.kizilcabo...@alcatel-lucent.com Mobile : +32 477961935 Fax: +90 5339793549
Re: [OSL | CCIE_Voice] silly dtmf,fax,modem relay question
I believe that statement means it supports dtmf relay to or from that device, not actually exiting out to the analog device FROM the card. - Original Message - From: Joel Jose joeljose...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 04, 2009 5:47 AM Subject: [OSL | CCIE_Voice] silly dtmf,fax,modem relay question i was reading about the rfc2833 dtmf support. one doubt really got me thinking... i hope its not too silly for this group.. incase its too silly, then i have hey its in the blueprint excuse ;) so when we use dtmf relay.. or any relay..say modem or fax.. when mean out band, we mean atleast 2 channels right?? one for the usual rtp and the other for the fax,modem,dtmf channel right?? So in vic-2fxs we shouldnt be able to do relay right??. but cisco says we can. Before Cisco IOS Release 12.4(6)XE, SCCP gateways supported fax passthrough only. In Cisco IOS Release 12.4(6)XE and later releases, SCCP Gateway Controlled Fax Relay adds support http://www.cisco.com/en/US/docs/ios/voice/fxs/configuration/guide/fxsrelay.html#wp1049432 well... i am just a little confused as to how do we do out of band if there are no more than one channel?? i am sure by now that my understanding of outofband is also flawed... do correct me. joel. -- it's not the years in your life that count. It's the life in your years. Abraham Lincoln
Re: [OSL | CCIE_Voice] CTI RP
In my experience, when this happens the problem is usually that there is something wrong with the script. If you open up the script you are using with the script editor, and use the validate function, it will tell you if there is something it doesn't like. If the script has validation problems, it will NOT answer the phone. You can also plug in one of your canned scripts to see if they work (the aa.aef or icd.aef). If they don't answer, you're likely missing some configuration. Remember that in addition to the CTI Route point, you must also configure the CTI Ports themselves (the call control groupthese are the actual ports that are answering the phone) and the Cisco Media Termination Dialog group (Under Cisco Media on the subsystems menu). The Media Termination group is what provides the ability to interact with the caller (i.e. speak to them get digits from them, etc). If these are not set up, then you can't configure them onto the JTAPI trigger. And if they aren't configured on the JTAPI trigger (under the settings for Call Control Group and Primary Dialog Group), then a script that needs to answer the phone and interact with callers.can't. So, you'd likely get a busy signal. On a more basic setting, you could bounce the CRS Node engine, make sure the services are upand if you're also running extension mobility/IPMA Console/etc. you need to change the port number in Tomcat on IPCC to fix the conflict there. HTH Cliff - Original Message - From: Kumar, Narinder narinder.ku...@uxcg.com.au To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 04, 2009 8:51 AM Subject: Re: [OSL | CCIE_Voice] CTI RP I integrated IPCC with CCM. IPCC and CCM are both collocated on the same box. Created new application with trigger point 1700 ON the CCM PUB and SUB the CTI RP shows me it is registered. But when I call 1700 from the IP Phone or from PSTN it keeps coming busy all time. Rest the CTI RP, reset the IIS server ( Don't think that will play any part), I even rebooted both PUB and SUB, don't know why it is busy all the time. Any help is much appreciated. Thanks Narinder CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system.
Re: [OSL | CCIE_Voice] CTI RP
In addition to what Cliff has said- also ensure the JTAPI and RM-CM Subsystem's are showing as active in the CRS control center. Call from the HQ phone where there is no need for a transcoder to be invoked and you do not need any Location bandwidth available. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Cliff McGlamry cl...@mcglamry.net Date: Wed, 4 Mar 2009 10:52:23 -0500 To: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CTI RP In my experience, when this happens the problem is usually that there is something wrong with the script. If you open up the script you are using with the script editor, and use the validate function, it will tell you if there is something it doesn't like. If the script has validation problems, it will NOT answer the phone. You can also plug in one of your canned scripts to see if they work (the aa.aef or icd.aef). If they don't answer, you're likely missing some configuration. Remember that in addition to the CTI Route point, you must also configure the CTI Ports themselves (the call control groupthese are the actual ports that are answering the phone) and the Cisco Media Termination Dialog group (Under Cisco Media on the subsystems menu). The Media Termination group is what provides the ability to interact with the caller (i.e. speak to them get digits from them, etc). If these are not set up, then you can't configure them onto the JTAPI trigger. And if they aren't configured on the JTAPI trigger (under the settings for Call Control Group and Primary Dialog Group), then a script that needs to answer the phone and interact with callers.can't. So, you'd likely get a busy signal. On a more basic setting, you could bounce the CRS Node engine, make sure the services are upand if you're also running extension mobility/IPMA Console/etc. you need to change the port number in Tomcat on IPCC to fix the conflict there. HTH Cliff - Original Message - From: Kumar, Narinder narinder.ku...@uxcg.com.au To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 04, 2009 8:51 AM Subject: Re: [OSL | CCIE_Voice] CTI RP I integrated IPCC with CCM. IPCC and CCM are both collocated on the same box. Created new application with trigger point 1700 ON the CCM PUB and SUB the CTI RP shows me it is registered. But when I call 1700 from the IP Phone or from PSTN it keeps coming busy all time. Rest the CTI RP, reset the IIS server ( Don't think that will play any part), I even rebooted both PUB and SUB, don't know why it is busy all the time. Any help is much appreciated. Thanks Narinder CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system.
Re: [OSL | CCIE_Voice] silly dtmf,fax,modem relay question
Joel, First of all no question is too silly on this post so thank you for your question. It's a good question- we normally set dtmf-relay to use a certain method because the document we read says to do it that way and then we move on! But I think to answer the question properly we are best of answering two other questions: (1) Why do we need dtmf-relay (2) what does in-band The answer to (1) is dtmf being sent to an IVR application across the WAN is unreliable because usually g729 (or some other compressed codec) is used across the WAN and whilst this is great for voice- DTMF, Fax and Modem signals are more susceptible to becoming distorted using the compression techniques used for Low Bit Rate codecs and also packet loss/jitter/etc. This could mean you press 1 for sales but it ends up arriving at the IVR as 2 for support:-) The answer to (2) is in-band means we receive the DTMF tone and place it into an RTP packet with payload identifier of G729 or whatever the codec being used is. In other words, the DTMF tone is carried in the voice stream. We want to achieve two things- we STILL NEED voice to use g729 but we need a more reliable method for transporting (or relaying) DTMF/Fax/Modem signals. SO we can use H245 signaling (two types of signling here) or RFC 2833 (RTP-NTE). The H245 signaling channel was opened up before the RTP media stream- H245 is used to inform the other party of codecs to use and which UDP port to communicate on for the RTP session. H323 supports these two methods but not SIP. RTP-NTE is mistakenly known as in-band when actually it is not (no matter what the docs state!). There is a subtle difference between the two- with RFC 2833 we are placing the DTMF into an RTP packet but the payload identifier is NTE as opposed to G729 and this offers a more reliable way to transport the DTMF tone. SIP and H323 support this method. SIP also has another out-of-band method which places the DTMF tone into a SIP NOTIFY message which some SIP applications support. H323 does not support this. SCCP phones use a method very similar to SIP phones and that is DTMF tones are placed into SCCP messages - again this is out-of-band. So a VIC-2FXS has no need to support DTMF-RELAY. The phone supports DTMF and the gateway receives the DMTF tone and relays this across the WAN using the technique specified in the VOIP dial-peer. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Cliff McGlamry cl...@mcglamry.net Date: Wed, 4 Mar 2009 10:36:24 -0500 To: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] silly dtmf,fax,modem relay question I believe that statement means it supports dtmf relay to or from that device, not actually exiting out to the analog device FROM the card. - Original Message - From: Joel Jose joeljose...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 04, 2009 5:47 AM Subject: [OSL | CCIE_Voice] silly dtmf,fax,modem relay question i was reading about the rfc2833 dtmf support. one doubt really got me thinking... i hope its not too silly for this group.. incase its too silly, then i have hey its in the blueprint excuse ;) so when we use dtmf relay.. or any relay..say modem or fax.. when mean out band, we mean atleast 2 channels right?? one for the usual rtp and the other for the fax,modem,dtmf channel right?? So in vic-2fxs we shouldnt be able to do relay right??. but cisco says we can. Before Cisco IOS Release 12.4(6)XE, SCCP gateways supported fax passthrough only. In Cisco IOS Release 12.4(6)XE and later releases, SCCP Gateway Controlled Fax Relay adds support http://www.cisco.com/en/US/docs/ios/voice/fxs/configuration/guide/fxsrelay.htm l#wp1049432 well... i am just a little confused as to how do we do out of band if there are no more than one channel?? i am sure by now that my understanding of outofband is also flawed... do correct me. joel. -- it's not the years in your life that count. It's the life in your years. Abraham Lincoln
Re: [OSL | CCIE_Voice] lab v2 / v3 questions
- Continue with v2 and wait till find a place or continue with v3 and take the lab in Agust? Ultimately this is your decision- I would suggest you look every day and over the next few weeks if there are no lab dates available then you may want to move onto the new blueprint. I know if a few people who recently got lab dates in Tokyo. * Do you thing there are too many difference between v2 and v3? This is a loaded question. There are massive differences between the two blueprints but the technologies used in v2 are still applicable in v3 and so your v2 prep will not be in vain. There are new things in v3 that do not exist in v2 such as device mobility, single number reach, sip endpoints and presence to name but a few. * I have still vouchers for lab and will I have option to login v2 or v3 for preparation? Vouchers for Proctorlabs? They are valid from 1 year of purchase. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: KIZILCABOLUK DENIZ deniz.kizilcabo...@alcatel-lucent.com.tr Date: Wed, 4 Mar 2009 15:28:13 +0100 To: OSL Group ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] lab v2 / v3 questions Hello, I prepared for voice v2 but I couldn¹t find a place for lab before Agust. v3 will be started middle of July. I need a guide for the voice lab exam which way I can continue; - Continue with v2 and wait till find a place or continue with v3 and take the lab in Agust? - Do you thing there are too many difference between v2 and v3? - I have still vouchers for lab and will I have option to login v2 or v3 for preparation? Thanks, M.Deniz KIZILCABOLUK Alcatel-Lucent Bell NV IPTC I'ntl System - Integr. Test Copernicuslaan 50, 2018 Antwerp, Belgium E-mail : deniz.kizilcabo...@alcatel-lucent.com Mobile : +32 477961935 Fax: +90 5339793549
Re: [OSL | CCIE_Voice] CTI RP
No Need of transcoder all G711. All the points cliff has suggested is already checked and are in place. I removed my script and used aa.aef no change. Port conflict checked. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vik Malhi Sent: Thursday, 5 March 2009 4:35 AM To: Cliff McGlamry; OSL Group Subject: Re: [OSL | CCIE_Voice] CTI RP In addition to what Cliff has said- also ensure the JTAPI and RM-CM Subsystem's are showing as active in the CRS control center. Call from the HQ phone where there is no need for a transcoder to be invoked and you do not need any Location bandwidth available. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Cliff McGlamry cl...@mcglamry.net Date: Wed, 4 Mar 2009 10:52:23 -0500 To: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CTI RP In my experience, when this happens the problem is usually that there is something wrong with the script. If you open up the script you are using with the script editor, and use the validate function, it will tell you if there is something it doesn't like. If the script has validation problems, it will NOT answer the phone. You can also plug in one of your canned scripts to see if they work (the aa.aef or icd.aef). If they don't answer, you're likely missing some configuration. Remember that in addition to the CTI Route point, you must also configure the CTI Ports themselves (the call control groupthese are the actual ports that are answering the phone) and the Cisco Media Termination Dialog group (Under Cisco Media on the subsystems menu). The Media Termination group is what provides the ability to interact with the caller (i.e. speak to them get digits from them, etc). If these are not set up, then you can't configure them onto the JTAPI trigger. And if they aren't configured on the JTAPI trigger (under the settings for Call Control Group and Primary Dialog Group), then a script that needs to answer the phone and interact with callers.can't. So, you'd likely get a busy signal. On a more basic setting, you could bounce the CRS Node engine, make sure the services are upand if you're also running extension mobility/IPMA Console/etc. you need to change the port number in Tomcat on IPCC to fix the conflict there. HTH Cliff - Original Message - From: Kumar, Narinder narinder.ku...@uxcg.com.au To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 04, 2009 8:51 AM Subject: Re: [OSL | CCIE_Voice] CTI RP I integrated IPCC with CCM. IPCC and CCM are both collocated on the same box. Created new application with trigger point 1700 ON the CCM PUB and SUB the CTI RP shows me it is registered. But when I call 1700 from the IP Phone or from PSTN it keeps coming busy all time. Rest the CTI RP, reset the IIS server ( Don't think that will play any part), I even rebooted both PUB and SUB, don't know why it is busy all the time. Any help is much appreciated. Thanks Narinder CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer viruses, Getronics Australia does not warrant that this email or any attachments are free of viruses or any other defects or errors. It is the duty of the recipient to virus scan and otherwise test any information contained in this email before loading onto any computer system. CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not
Re: [OSL | CCIE_Voice] CTI RP
Narinder, I usually get a message stating a system problem when I hit IPCC and there is something wrong with my script / configurations. I have received busy signals when I have had issues with calling search spaces on my CTI ports or CTI route points. If the ports are registered, I would take a look at the CSS on your CTI/RP ports. Scott Hardesty | Solutions Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:sharde...@presidio.com D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kumar, Narinder Sent: Wednesday, March 04, 2009 5:15 PM To: Vik Malhi; Cliff McGlamry; OSLGroup Subject: Re: [OSL | CCIE_Voice] CTI RP No Need of transcoder all G711. All the points cliff has suggested is already checked and are in place. I removed my script and used aa.aef no change. Port conflict checked. -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vik Malhi Sent: Thursday, 5 March 2009 4:35 AM To: Cliff McGlamry; OSL Group Subject: Re: [OSL | CCIE_Voice] CTI RP In addition to what Cliff has said- also ensure the JTAPI and RM-CM Subsystem's are showing as active in the CRS control center. Call from the HQ phone where there is no need for a transcoder to be invoked and you do not need any Location bandwidth available. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: vma...@ipexpert.com Join our free online support and peer group communities: http://www.IPexpert.com/communities IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. From: Cliff McGlamry cl...@mcglamry.net Date: Wed, 4 Mar 2009 10:52:23 -0500 To: OSL Group ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CTI RP In my experience, when this happens the problem is usually that there is something wrong with the script. If you open up the script you are using with the script editor, and use the validate function, it will tell you if there is something it doesn't like. If the script has validation problems, it will NOT answer the phone. You can also plug in one of your canned scripts to see if they work (the aa.aef or icd.aef). If they don't answer, you're likely missing some configuration. Remember that in addition to the CTI Route point, you must also configure the CTI Ports themselves (the call control groupthese are the actual ports that are answering the phone) and the Cisco Media Termination Dialog group (Under Cisco Media on the subsystems menu). The Media Termination group is what provides the ability to interact with the caller (i.e. speak to them get digits from them, etc). If these are not set up, then you can't configure them onto the JTAPI trigger. And if they aren't configured on the JTAPI trigger (under the settings for Call Control Group and Primary Dialog Group), then a script that needs to answer the phone and interact with callers.can't. So, you'd likely get a busy signal. On a more basic setting, you could bounce the CRS Node engine, make sure the services are upand if you're also running extension mobility/IPMA Console/etc. you need to change the port number in Tomcat on IPCC to fix the conflict there. HTH Cliff - Original Message - From: Kumar, Narinder narinder.ku...@uxcg.com.au To: ccie_voice@onlinestudylist.com Sent: Wednesday, March 04, 2009 8:51 AM Subject: Re: [OSL | CCIE_Voice] CTI RP I integrated IPCC with CCM. IPCC and CCM are both collocated on the same box. Created new application with trigger point 1700 ON the CCM PUB and SUB the CTI RP shows me it is registered. But when I call 1700 from the IP Phone or from PSTN it keeps coming busy all time. Rest the CTI RP, reset the IIS server ( Don't think that will play any part), I even rebooted both PUB and SUB, don't know why it is busy all the time. Any help is much appreciated. Thanks Narinder CONFIDENTIALITY - The information contained in this electronic mail message is confidential and is intended solely for the addressee(s). If you are not an authorised recipient of this message please contact Getronics Australia immediately by reply email and destroy/delete this message from your computer. Any unauthorised form of reproduction of this message, or part thereof, is strictly prohibited. DISCLAIMER - Unless specifically indicated otherwise, the views and opinions expressed in this email are those of the sender and not Getronics Australia. While we endeavour to protect our network from computer
Re: [OSL | CCIE_Voice] IP PIM
Dense mode is a push method of doing multicast and sparse dense is a pull method of doing multicast. On Tue, Mar 3, 2009 at 12:56 PM, hasan khalife hasan_khal...@hotmail.comwrote: WHAT IS THE DIFFERENEC BTW IP PIM DENSE-MODE IP PIM SPARSE-DENSE-MODE -- See all the ways you can stay connected to friends and familyhttp://www.microsoft.com/windows/windowslive/default.aspx
Re: [OSL | CCIE_Voice] IP PIM
PIM Dence mode is more for local area network or samll size network, more of broadcast, push. Not recommended for WAN PIM sparce-mode is recommended for traffic over WAN. OR LAN. ip pim sparce-dense mode activates both space and dence mode - sparce first and dense, This command is mainly used for Auto-RP, where you can have multiple RPs and selection process to find RP use dense mode, once RP is selected, it uses sparce mode for actual multicast. /Jin Jung... Ryan Trauernicht wrote: Dense mode is a push method of doing multicast and sparse dense is a pull method of doing multicast. On Tue, Mar 3, 2009 at 12:56 PM, hasan khalife hasan_khal...@hotmail.com mailto:hasan_khal...@hotmail.com wrote: WHAT IS THE DIFFERENEC BTW IP PIM DENSE-MODE IP PIM SPARSE-DENSE-MODE See all the ways you can stay connected to friends and family http://www.microsoft.com/windows/windowslive/default.aspx
Re: [OSL | CCIE_Voice] FXS restricted without using COR
CCIE OSL wrote: Quick question, I am trying to configure my FXS port to only allow call local and 911. I know how to do it with COR, But I do not want to create COR, it seems bit too much for simple restriction for PSTN phone. Is there any other way to restrict this phone with out using COR? Thanks.. /Jin Jung...
[OSL | CCIE_Voice] IPBlue and IPCC Express
Has anyone been able to get the IP Phone service to work with IP Blue? I'm getting some really strange behaviors from IP Blue when attempting to utilize services configured on the phone. I've got services showing up on the phones that aren't configured on them, etc. Very strange. And IPPA will NOT work. Just curious if anyone else has seen this. Cliff
