[OSL | CCIE_Voice] h323 Fast start configuration
Hi All, In lab 8 we have to do FS with SiteB GW. If we use G711u codec for FS then it fails the call for HQ phones when they use SiteB GW for routing as backup route because HQ and SiteB uses G729 as inter region codec. I tried to change the g711u to g729r8 in mtp configuration for FS and under outbound FS enable drop down option. Then in that case it is working for all the calls. Any other solution anybody can think of for this FS issue..?? Regards, Piyush Jain From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Tuesday, May 14, 2013 8:20:08 PM Subject: CCIE_Voice Digest, Vol 87, Issue 32 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. Re: CCIE_Voice Digest, Vol 87, Issue 30 (jainpiyush2...@ymail.com) 2. Calling Name for MVA (CCIEing) 3. Re: Calling Name for MVA (heshamcentr...@gmail.com) -- Message: 1 Date: Tue, 14 May 2013 11:17:46 +0530 From: jainpiyush2...@ymail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 30 Message-ID: b691de99-251f-441c-9818-54f948ea0d62.maildroid@localhost Content-Type: text/plain; charset=utf-8 For early media, Mtp codec depends on your requirement. If your inter region codec requirement is g729 then define that in the mtp codec configuration and you don't need to use pass through.. I have tested this in my lab.. Thanks and regards, Piyush Jain Sent from my android device. -Original Message- From: ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Mon, 13 May 2013 9:31 PM Subject: CCIE_Voice Digest, Vol 87, Issue 30 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. Re: MRG for MTP, SIP Early Offer (FAISAL AL-EMAD) 2. Cisco ip phone over internet (Dharambir kumar varma) 3. Registering phone (Dharambir kumar varma) 4. Re: h323 fast start (Kirill Groshev) -- Message: 1 Date: Mon, 13 May 2013 09:14:13 +0300 From: FAISAL AL-EMAD eng_ale...@hotmail.com To: Bill Lake whl...@gmail.com, Ben John benjoh...@hotmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MRG for MTP, SIP Early Offer Message-ID: dub123-w1493846c19fce1225b3a33fc...@phx.gbl Content-Type: text/plain; charset=windows-1256 Dear Bill, You are right, with Early offer we need MTP but in the configuration of MTP should enable codec g711ulaw and pass-through only? Best Regards Eng. Faisal alemadNetworks UC Engineer Three things in life that make you a great person 1. Hard Work 2. Sincerity 3. success Date: Sun, 12 May 2013 19:40:32 -0500 From: whl...@gmail.com To: benjoh...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MRG for MTP, SIP Early Offer Early offer requires media resources Delayed offer does not require it SIP Early Offer Support over Unified CM SIP Trunks SIP negotiates media exchange by means of the Session Description Protocol (SDP), where one side offers a set of capabilities to which the other side answers, thus converging on a set of media characteristics. SIP allows the initial offer to be sent either by the caller in the initial INVITE message (Early Offer) or, if the caller chooses not to, the called party can send the initial offer in the first reliable response (Delayed Offer). By default, Unified CM SIP trunks send the INVITE without an initial offer (Delayed Offer). In general SIP Delayed Offer is preferred for Unified CM SIP trunks because MTPs are not needed to establish a Delayed Offer call for voice, video, or encrypted media. If SIP Early Offer is desired, Unified CM has two configurable options to enable a SIP trunk to send the offer in the INVITE: ?Media Termination Point Required ?Early Offer Support for Voice and Video Calls (Insert
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 55
Hi Piyush, i have tried with with G711ulaw on SB gateway its working fine for me with redundant to HQ call routing here what i have done I have created MTP and Xcode on Site B router sccp ccm group 1 ass ccm 1 prio 1 ass ccm 2 prio 2 ass pro 1 reg SB-XCODE ass pro 2 reg SB-MTP dspfarm pro 1 trans max sess 4 ass app sccp no shut dspfram pro 2 mtp codec g711ulaw max sess soft 8 ass app sccp no shut and on CUCM u have to create MRG n MRGL and assign ths MRGL to SB Gateway and check MTP required in CUCM Gateway page you can try ths configuration and let me know ur feedback ... On Tue, May 21, 2013 at 10:08 AM, ccie_voice-requ...@onlinestudylist.comwrote: 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. (no subject) (ie ravindra) 2. (MGCP Teardown) (ie ravindra) 3. Re: (no subject) (Shabeeb Mohammed) 4. h323 Fast start configuration (Piyush Jain) -- Message: 1 Date: Tue, 21 May 2013 04:26:42 +0530 From: ie ravindra ieravin...@gmail.com To: CCIE Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] (no subject) Message-ID: cabadenywv08supms0meguj29ihtrbt+enrvpouse56vcyxu...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Dear All, Whats is the real meaning of MGCP tear down. Is it means dropping a call or , What ? thanks for your valuable input. Ravi, -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130521/91fde8a8/attachment-0001.html -- Message: 2 Date: Tue, 21 May 2013 04:41:48 +0530 From: ie ravindra ieravin...@gmail.com To: CCIE Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] (MGCP Teardown) Message-ID: CABADEnz40hKrB=bWpHQA8eNomj3v6nU= 57ufmvq+hnvmmm-...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Dear All, Whats is the real meaning of MGCP tear down. Is it means dropping a call or , What ? thanks for your valuable input. Ravi, -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130521/c6e48219/attachment-0001.html -- Message: 3 Date: Tue, 21 May 2013 10:36:49 +0530 From: Shabeeb Mohammed shabeebc...@gmail.com To: ie ravindra ieravin...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] (no subject) Message-ID: caojzbky0gwx7s_2udiduvw6euuqsfxfuwzm6638mtg1a+pn...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hey ravi, I believe it means that mgcp packets was disrupted in between transmission resulting in packet loss etc. This error implies that there is a bug in the ios. Try upgrading the ios and check Regards Shabeeb On 21 May 2013 04:28, ie ravindra ieravin...@gmail.com wrote: Dear All, Whats is the real meaning of MGCP tear down. Is it means dropping a call or , What ? thanks for your valuable input. Ravi, ___ 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 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130521/92320b15/attachment-0001.html -- Message: 4 Date: Tue, 21 May 2013 00:08:52 -0700 (PDT) From: Piyush Jain jainpiyush2...@ymail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] h323 Fast start configuration Message-ID: 1369120132.4245.yahoomail...@web122203.mail.ne1.yahoo.com Content-Type: text/plain; charset=iso-8859-1 Hi All, In lab 8 we have to do FS with SiteB GW. If we use G711u codec for FS then it fails the call for HQ phones when they use SiteB GW for routing as backup route because HQ and SiteB uses G729 as inter region codec. I tried to change the g711u to g729r8 in mtp configuration for FS and under outbound FS enable drop down option. Then in that case it is working for all the calls. Any other solution anybody can think of for this FS issue..?? Regards, Piyush Jain From: ccie_voice-requ...@onlinestudylist.com ccie_voice-requ...@onlinestudylist.com To: ccie_voice
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 56
Hi experts, i am also searching for that solution thanks i will also try that and let u know, which command i need to run on SB gateway to get the output,do i need call start fast under voice service voip in SB gateway ? Thanks On Tue, May 21, 2013 at 11:41 AM, ccie_voice-requ...@onlinestudylist.comwrote: 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. Re: CCIE_Voice Digest, Vol 87, Issue 55 (Mohammed Ameenullah) -- Message: 1 Date: Tue, 21 May 2013 11:41:39 +0300 From: Mohammed Ameenullah ameen...@gmail.com To: ccie_voice@onlinestudylist.com Cc: jainpiyush2...@ymail.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 55 Message-ID: caau-xfjcw1d08ac3hydp2npcgy_jzq_au8kcsulzeeh549z...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi Piyush, i have tried with with G711ulaw on SB gateway its working fine for me with redundant to HQ call routing here what i have done I have created MTP and Xcode on Site B router sccp ccm group 1 ass ccm 1 prio 1 ass ccm 2 prio 2 ass pro 1 reg SB-XCODE ass pro 2 reg SB-MTP dspfarm pro 1 trans max sess 4 ass app sccp no shut dspfram pro 2 mtp codec g711ulaw max sess soft 8 ass app sccp no shut and on CUCM u have to create MRG n MRGL and assign ths MRGL to SB Gateway and check MTP required in CUCM Gateway page you can try ths configuration and let me know ur feedback ... On Tue, May 21, 2013 at 10:08 AM, 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. (no subject) (ie ravindra) 2. (MGCP Teardown) (ie ravindra) 3. Re: (no subject) (Shabeeb Mohammed) 4. h323 Fast start configuration (Piyush Jain) -- Message: 1 Date: Tue, 21 May 2013 04:26:42 +0530 From: ie ravindra ieravin...@gmail.com To: CCIE Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] (no subject) Message-ID: cabadenywv08supms0meguj29ihtrbt+enrvpouse56vcyxu...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Dear All, Whats is the real meaning of MGCP tear down. Is it means dropping a call or , What ? thanks for your valuable input. Ravi, -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130521/91fde8a8/attachment-0001.html -- Message: 2 Date: Tue, 21 May 2013 04:41:48 +0530 From: ie ravindra ieravin...@gmail.com To: CCIE Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] (MGCP Teardown) Message-ID: CABADEnz40hKrB=bWpHQA8eNomj3v6nU= 57ufmvq+hnvmmm-...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Dear All, Whats is the real meaning of MGCP tear down. Is it means dropping a call or , What ? thanks for your valuable input. Ravi, -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130521/c6e48219/attachment-0001.html -- Message: 3 Date: Tue, 21 May 2013 10:36:49 +0530 From: Shabeeb Mohammed shabeebc...@gmail.com To: ie ravindra ieravin...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] (no subject) Message-ID: caojzbky0gwx7s_2udiduvw6euuqsfxfuwzm6638mtg1a+pn...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hey ravi, I believe it means that mgcp packets was disrupted in between transmission resulting in packet loss etc. This error implies that there is a bug in the ios. Try upgrading the ios and check Regards Shabeeb On 21 May 2013 04:28, ie ravindra ieravin...@gmail.com wrote: Dear All, Whats is the real meaning of MGCP tear
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 87, Issue 44
Hi Drake, Regarding your 1st question, when you upload any audio source/prompt to call manager it converts/translate to G711, G729 codecs by itself. So even if you upload a g711 codec, it will be available for both g729 and g711 moh stream. Regarding your 2nd question, ringback while user is in queue, i tried the CVL sol (putting the ringback prompt in network on hold and using the script same as of lab7) but it did not work for me. Then i tried below which worked for me. 1. upload the ringback prompt to CUCM. 2. Enabled g729 and g711 in IPVMA service parameters in call manager. 3. on UCCX, select G729 codec and restart node manager. 4. Assigned the prompt to USER ON HOLD in UCCX CTI/JTAPI Ports. 5. In the script, under queuing section, I added call hold after waiting in queue prompt is played. And also after delay, I added call unhold. Logic is- i am manually putting the call on hold/unhold using script while in queue and it uses user on hold prompt which is ringback. Let me know if this works for you. However i have come across several forums which says if you do as per CVL solutions you will get 100% in uccx section however i don't feel this since its solution didn't work for me. Thanks and Regards, Piyush Jain Message: 3 Date: Sun, 19 May 2013 19:53:15 +0530 From: Drake J jdrake...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX questions / G729 / ringback Message-ID: cafobmhqsxmsboya0mdoxh6bhy428rpjhxugo4rkbg8b6r2u...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 hi All, I have 2 questions here... 1) If UCCX was supposed to use g729 codec . Then if we are using unity connection or uccx to record prompts it would record this prompts in g711ulaw. Therefore the prompts played from the script will be in g711ulaw when the uccx is setup for g729 .What is the way around? 2) How do we make callers hear ringback with they wait in a UCCX queue for their call to be answered by agents? Please assist. Regards, Drake -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130519/5c3160f1/attachment-0001.html -- Message: 4 Date: Sun, 19 May 2013 09:30:25 -0500 From: Bill whl...@gmail.com To: Ravindra Lakpriya lakpr...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Voice translation issue Message-ID: 4c0778d6-2497-4e32-bbd7-57f3e4d30...@gmail.com Content-Type: text/plain; charset=us-ascii dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 forward-digits 3 Sent from my iPad On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote: dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130519/6ac2a592/attachment-0001.html -- Message: 5 Date: Sun, 19 May 2013 09:33:02 -0500 From: Bill whl...@gmail.com To: Ravindra Lakpriya lakpr...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Voice translation issue Message-ID: 9ae17c84-7792-49b3-952d-b18632a6e...@gmail.com Content-Type: text/plain; charset=us-ascii voice translation-rule 8 rule 1 // // type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn Sent from my iPad On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote: In the dial peer configure no digit strip. :) On Sun, May 19, 2013 at 5:47 PM, Martin Sloan martinsloa...@gmail.com wrote: I have a voice translation rule in place for '9911' calls on BR1 during SRST. I'm running into some odd behavior (from my perspective) and I'm hoping it's a config issue I'm just not spotting. I have the translation profile applied to the dial peer and the only other translation that would be in the calling path is on the voice port but even that one is applied to inbound calls for stripping down to 4 digits. Here's the config related to this dial-peer: voice translation-rule 8 rule 1 /^9911$/ /911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn voice translation-profile 9911 translate calling 8 translate called 8 dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 BR1-RTR#test voice translation-rule 8 9911 Matched with rule 1 Original number: 9911 Translated number: 911 Original number type: none Translated number type: unknown Original number plan: none Translated number plan: isdn -Debug ISDN q931- Calling Party Number i = 0x4181, '6173941002' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0x81,
Re: [OSL | CCIE_Voice] UCCX questions / G729 / ringback
Hi Drake, Regarding your 1st question, when you upload any audio source/prompt to call manager it converts/translate to G711, G729 codecs by itself. So even if you upload a g711 codec, it will be available for both g729 and g711 moh stream. Regarding your 2nd question, ringback while user is in queue, i tried the CVL sol (putting the ringback prompt in network on hold and using the script same as of lab7) but it did not work for me. Then i tried below which worked for me. 1. upload the ringback prompt to CUCM. 2. Enabled g729 and g711 in IPVMA service parameters in call manager. 3. on UCCX, select G729 codec and restart node manager. 4. Assigned the prompt to USER ON HOLD in UCCX CTI/JTAPI Ports. 5. In the script, under queuing section, I added call hold after waiting in queue prompt is played. And also after delay, I added call unhold. Logic is- i am manually putting the call on hold/unhold using script while in queue and it uses user on hold prompt which is ringback. Let me know if this works for you. However i have come across several forums which says if you do as per CVL solutions you will get 100% in uccx section however i don't feel this since its solution didn't work for me. Thanks and Regards, Piyush Jain Message: 3 Date: Sun, 19 May 2013 19:53:15 +0530 From: Drake J jdrake...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX questions / G729 / ringback Message-ID: cafobmhqsxmsboya0mdoxh6bhy428rpjhxugo4rkbg8b6r2u...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 hi All, I have 2 questions here... 1) If UCCX was supposed to use g729 codec . Then if we are using unity connection or uccx to record prompts it would record this prompts in g711ulaw. Therefore the prompts played from the script will be in g711ulaw when the uccx is setup for g729 .What is the way around? 2) How do we make callers hear ringback with they wait in a UCCX queue for their call to be answered by agents? Please assist. Regards, Drake -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130519/5c3160f1/attachment-0001.html -- Message: 4 Date: Sun, 19 May 2013 09:30:25 -0500 From: Bill whl...@gmail.com To: Ravindra Lakpriya lakpr...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Voice translation issue Message-ID: 4c0778d6-2497-4e32-bbd7-57f3e4d30...@gmail.com Content-Type: text/plain; charset=us-ascii dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 forward-digits 3 Sent from my iPad On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote: dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20130519/6ac2a592/attachment-0001.html -- Message: 5 Date: Sun, 19 May 2013 09:33:02 -0500 From: Bill whl...@gmail.com To: Ravindra Lakpriya lakpr...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Voice translation issue Message-ID: 9ae17c84-7792-49b3-952d-b18632a6e...@gmail.com Content-Type: text/plain; charset=us-ascii voice translation-rule 8 rule 1 // // type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn Sent from my iPad On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote: In the dial peer configure no digit strip. :) On Sun, May 19, 2013 at 5:47 PM, Martin Sloan martinsloa...@gmail.com wrote: I have a voice translation rule in place for '9911' calls on BR1 during SRST. I'm running into some odd behavior (from my perspective) and I'm hoping it's a config issue I'm just not spotting. I have the translation profile applied to the dial peer and the only other translation that would be in the calling path is on the voice port but even that one is applied to inbound calls for stripping down to 4 digits. Here's the config related to this dial-peer: voice translation-rule 8 rule 1 /^9911$/ /911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn voice translation-profile 9911 translate calling 8 translate called 8 dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 BR1-RTR#test voice translation-rule 8 9911 Matched with rule 1 Original number: 9911 Translated number: 911 Original number type: none Translated number type: unknown Original number plan: none Translated number plan: isdn -Debug ISDN q931- Calling Party Number i = 0x4181, '6173941002' Plan:ISDN, Type:Subscriber(local) Called Party Number i =
[OSL | CCIE_Voice] CUE/UCCX Integration
Hey Guys. I had some trouble this weekend with my CUE integration with CUCM and I would like to know how to do it correctly while doing it fast ... (no best practices here I know :) ) This is how I see things. -Create CTI RP in the none partition (accessible to anyone) with a CSS that can see CTI ports (let's say CSS_CTI_RP that includes PT_CTI_PORTS) - Create 2 CTI Ports that are in the PT_CTI_PORTS with a CSS that can see directly phones (CSS_INTERNAL that include PT_INTERNAL) -Create a user that has CTI standard enabled and controlled devices : CTI-RP and CTI_Ports -Create a voicemail pilot that is equal to the CTI RP in the none partition -Create a Voicemail profile that include the newly created voicemail pilot for BR2 -Apply Voicemail profile to the phones *reset them* Then on the CUE. -Check the license and upload new license if not OK -Go Offline and restore factory default - Wizard install: Configure CUCM IP/CTI user/ etc etc and imports users . Check if the CTI ports are registered on the CUCM Check if the CTI RP is registered on the CUCM Check by pressing the voicemail button on the phone... My main concern is about the CSS/PT configuration ... Can someone please enlighten me ? That would be awesome ! Thanks Nic ___ 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] CUE/UCCX Integration
This is what I follow For ccie lab, i keep all the phones in none partition so that everything has access to the phones. Also I keep uccx/cue route point and CTI port in none partition and assign none CSS to cue/ uccx route point and CTI port. This works for me very well. Thanks and regards, Piyush Jain Sent from my android device. -Original Message- From: ccie_voice-requ...@onlinestudylist.com To: ccie_voice@onlinestudylist.com Sent: Tue, 21 May 2013 9:37 PM Subject: CCIE_Voice Digest, Vol 87, Issue 65 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. CUE/UCCX Integration (Nicolas MICHEL) -- Message: 1 Date: Tue, 21 May 2013 14:23:31 +0200 From: Nicolas MICHEL mcl.nico...@gmail.com To: OSL Voice ccie_voice@onlinestudylist.com, Nicolas MICHEL mcl.nico...@gmail.com Subject: [OSL | CCIE_Voice] CUE/UCCX Integration Message-ID: 519b6743.2010...@gmail.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hey Guys. I had some trouble this weekend with my CUE integration with CUCM and I would like to know how to do it correctly while doing it fast ... (no best practices here I know :) ) This is how I see things. -Create CTI RP in the none partition (accessible to anyone) with a CSS that can see CTI ports (let's say CSS_CTI_RP that includes PT_CTI_PORTS) - Create 2 CTI Ports that are in the PT_CTI_PORTS with a CSS that can see directly phones (CSS_INTERNAL that include PT_INTERNAL) -Create a user that has CTI standard enabled and controlled devices : CTI-RP and CTI_Ports -Create a voicemail pilot that is equal to the CTI RP in the none partition -Create a Voicemail profile that include the newly created voicemail pilot for BR2 -Apply Voicemail profile to the phones *reset them* Then on the CUE. -Check the license and upload new license if not OK -Go Offline and restore factory default - Wizard install: Configure CUCM IP/CTI user/ etc etc and imports users . Check if the CTI ports are registered on the CUCM Check if the CTI RP is registered on the CUCM Check by pressing the voicemail button on the phone... My main concern is about the CSS/PT configuration ... Can someone please enlighten me ? That would be awesome ! Thanks Nic -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 87, Issue 65 ** ___ 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
