Re: [OSL | CCIE_Voice] CME: assign blf-speed-dial to a specific button?
I think you'll have to configure 3 place holder ephone-dn's with no number say ephone-dn 5 dual-line ephone-dn 6 dual-line ephone-dn 7 dual-line then button 2:5 3:6 4:7 this will force the blf speed dial to be pushed to button 5 On 28 March 2011 07:31, Michael Luo hout...@gmail.com wrote: I'm using CME 7.1 and 7965 phone. Let say, I want to configure a blf-speed-dial on the 5th button on the phone. How should I configure that? I know the blf-speed-dial command. But I couldn't figure out how to assign it to a specific button. Thanks! Michael ___ 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] MVA partial match
It is ..yes confusing Partial is number of digits selected from left ( decide the number based on your deb isdn q931 ) not displayed on phone screen. Complete match is the number in the MVA configuration. If you select complete match debug isdn q931 should show exact number. Look for page help for more info. Thanks Shrini On 3/27/2011 10:20 PM, Erwan Erwan wrote: hi all, can someone explain bit on MVA logic or flow ? I config MVA Complete match service parameter in UCM , but still I called from any number , it prompt for pin (as if it match) Thanks in advyou ___ 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] MVA using sip trunk
Just uncheck Redirection Header Delivery at SIP trunk configuration 27.03.2011 23:08 пользователь Erwan Erwan e_er...@yahoo.com написал: no just play around w sip trunk --- On Sun, 3/27/11, mihal caro mihalc...@gmail.com wrote: From: mihal caro mihalc...@gmail.com Subject: Re: [OSL | CCIE_Voice] MVA using sip trunk To: Erwan Erwan e_er...@yahoo.com Cc: ccie_voice@onlinestudylist.com Received: Sunday, March 27, 2011, 6:14 PM hi there I had the same problem SIp trunk MVA is support on CUCM 7.13 . If is for the CCIE lab you need to configure it as H323. regards. On 27 March 2011 07:24, Erwan Erwan e_er...@yahoo.com wrote: hi all, I tried to config MVA in HQ site, Scenario - I dial in from PSTN to 5999 (MVA #) it gave me prompt for pin and press 1 to dial any outside number (911) 1. If I use SIP trunk It will gave me error saying (below is the complete debug ccsip message Sent: SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field' 2. However if I use, H323 GW , call out by MVA to outside (ie :911 ) work fine Question - - wondering any parameter in SIP\ trunk I need to check ? tks in adv Complete debug === PSTN-WAN# Mar 27 06:07:26.026: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field' Via: SIP/2.0/TCP 10.10.210.10:5060;branch=z9hG4bK1861aabffa From: sip:??5001@10.10.210.10 ;tag=ad370c0e-63ed-49fa-adb9-1bd65232089c-20651851 To: sip:911@10.10.110.4;tag=1343344-12F6 Call-ID: 7bcce880-d8e1d41e-10-ad20a0a@10.10.210.10 CSeq: 101 INVITE Reason: Q.850;cause=100 Content-Length: 0 ___ 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] Voiceview error
did you assign the phone to jtapi user? On Mon, Mar 28, 2011 at 6:35 AM, Erwan Erwan e_er...@yahoo.com wrote: hi all, When I press the Voiceview Listen , it said :Unknown Error ,report to System Administrator all other Voiceview work fine, wondering what i miss here? tks ___ 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] WB II, Lab 10, Calling Party Localization
I was just trying to help. Worked for me no problem On Mon, Mar 28, 2011 at 5:44 AM, George Goglidze gogli...@gmail.com wrote: Hi Adam, I would really think that if I had that things that you mentioned not configured correctly, then even if I stopped Subscriber Call Manager service, it wouldn't have worked correctly once all registered to Publisher... config error but it does as I mentioned in my e-mails. Thanks, On Mon, Mar 28, 2011 at 3:51 AM, adam compton com...@gmail.com wrote: I'm actually doing that lab right now :) Make sure that use device pool calling transformation css is not checked and assign the css to the phone. Reboot the phone. Might require a no mgcp and mgcp on the gateways. On Sun, Mar 27, 2011 at 10:50 AM, George Goglidze gogli...@gmail.comwrote: Hi Roger, I checked the replication on a cli on both, PUB/SUB: admin:show perf query class Number of Replicates Created and State of Replication ==query class : - Perf class (Number of Replicates Created and State of Replication) has instances and values: ReplicateCount - Number of Replicates Created = 412 ReplicateCount - Replicate_State= 2 I know the config is correct, because if I stop Call manager service on the Sub, or remove it from the Call Manager group, everything works fine... So it has to be good. I don't want to loose much time with this, as I'm with proctorlabs online rack rental, so I refer to finish the lab. at the end if I finish it much earlier, then I might do it, and reset the replication between sub/pub. Thanks for anwering Roger! Cheers, On Sun, Mar 27, 2011 at 3:45 PM, Roger Carpio roger.car...@gmail.comwrote: Hello George, Very rarely I've seen replication status 2 and weird behaviors happen. Have you checked replication anywhere else? CLI / Unified Reporting? If you've done this many times and you're sure the configuration has been done correctly; drop this replication and reset it cluster wide. I hope it helps. Regards, Roger Carpio. On Sun, Mar 27, 2011 at 8:28 AM, George Goglidze gogli...@gmail.comwrote: Hi all, I was wondering if anyone has seen this before. There is a requirement to show in Missed/Received directory numbers in E164 format, but in alerting/connected state it should be localized for user. Subscriber to 7digit, national to 10digit, international to 011 I have configured correct prefixes on the incoming gateways, so Missed/Received directory is fine! It shows correct values with the +sign E164 format. And I have configured pt-norm-hq-ani, pt-norm-br1-ani, pt-norm-br2-ani, which are in their corresponding css-norm-hq-ani, css-norm-br1-ani, css-norm-br2-ani and I've applied them to the corresponding Device Pools, in Calling Party Transformation CSS. But when the call comes in, I still get in alerting/connected state E164 format. All my devices phones and gateways are registered to Sub first, and in failed to Publisher. If I change this and I register all the devices to the Publisher, instead of a subscriber, changing it in Callmanager group configuration, then it all works as expected. Therefore I thought maybe Sub is not replicating correctly, but I've checked and the replication Status is 2! which means it's replicating fine. I have even restarted the Subscriber but it didn't fix the issue. I am out of ideas, and I've done it many times in prior Labs, and it has always worked fine! I've searched through CCM Service Parameters, and couldn't find anything that could affect this behaviour. If anyone has seen this, please let me know... Regards, ___ 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] Multicast MoH to CUBE
Anyone can enlighten me? On Sun, Mar 27, 2011 at 8:23 PM, Alex Goh ncsalex@gmail.com wrote: Hi All, When trying to practice some MoH lab today, I notice that whenever HQ/BR1 phone make a call to BR2 Phone via H323 gateway, which BR2 setup as CME (telephony-service), when HQ/BR1 phone press hold, the MoH will not work (BR2 phone hear silence). But when BR2 phone press Hold, HQ/BR1 phone can hear MoH. Same for when HQ/BR1 tried to send call out to PSTN via the H323 gateway, the PSTN hear no MoH. BR2 Phone call PSTN the MoH is ok though. My HQ is set to send Unicast MoH, while BR1 is set to send Multicast MoH. MoH servers have different DP and Region to all site is g711. I did tried to turn the BR2 into a H323 gateway only (no CUBE) and configured the following commands: ccm-manager music-on-hold, moh music-on-hold.au multicast moh 239.1.1.5 port 16384 Then MoH is working fine when HQ/BR1 put the call to PSTN on hold via the H323 gateway, also when HQ/BR1 calling BR2 Phone which on different DP/Region, MoH is working fine for both direction. Is this normal that Multicast MoH or Unicast MoH is not supporting CUBE in this case? also, why is BR2 CME phone calling HQ/BR1 phone yet MoH is working fine? Regards, Alex ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] WB II, Lab 10, Calling Party Localization
Hi Adam, Just a question, which vRack are you using? I was on rack 15, maybe that one in particular has some problem? On Mon, Mar 28, 2011 at 1:12 PM, adam compton com...@gmail.com wrote: I was just trying to help. Worked for me no problem On Mon, Mar 28, 2011 at 5:44 AM, George Goglidze gogli...@gmail.comwrote: Hi Adam, I would really think that if I had that things that you mentioned not configured correctly, then even if I stopped Subscriber Call Manager service, it wouldn't have worked correctly once all registered to Publisher... config error but it does as I mentioned in my e-mails. Thanks, On Mon, Mar 28, 2011 at 3:51 AM, adam compton com...@gmail.com wrote: I'm actually doing that lab right now :) Make sure that use device pool calling transformation css is not checked and assign the css to the phone. Reboot the phone. Might require a no mgcp and mgcp on the gateways. On Sun, Mar 27, 2011 at 10:50 AM, George Goglidze gogli...@gmail.comwrote: Hi Roger, I checked the replication on a cli on both, PUB/SUB: admin:show perf query class Number of Replicates Created and State of Replication ==query class : - Perf class (Number of Replicates Created and State of Replication) has instances and values: ReplicateCount - Number of Replicates Created = 412 ReplicateCount - Replicate_State= 2 I know the config is correct, because if I stop Call manager service on the Sub, or remove it from the Call Manager group, everything works fine... So it has to be good. I don't want to loose much time with this, as I'm with proctorlabs online rack rental, so I refer to finish the lab. at the end if I finish it much earlier, then I might do it, and reset the replication between sub/pub. Thanks for anwering Roger! Cheers, On Sun, Mar 27, 2011 at 3:45 PM, Roger Carpio roger.car...@gmail.comwrote: Hello George, Very rarely I've seen replication status 2 and weird behaviors happen. Have you checked replication anywhere else? CLI / Unified Reporting? If you've done this many times and you're sure the configuration has been done correctly; drop this replication and reset it cluster wide. I hope it helps. Regards, Roger Carpio. On Sun, Mar 27, 2011 at 8:28 AM, George Goglidze gogli...@gmail.com wrote: Hi all, I was wondering if anyone has seen this before. There is a requirement to show in Missed/Received directory numbers in E164 format, but in alerting/connected state it should be localized for user. Subscriber to 7digit, national to 10digit, international to 011 I have configured correct prefixes on the incoming gateways, so Missed/Received directory is fine! It shows correct values with the +sign E164 format. And I have configured pt-norm-hq-ani, pt-norm-br1-ani, pt-norm-br2-ani, which are in their corresponding css-norm-hq-ani, css-norm-br1-ani, css-norm-br2-ani and I've applied them to the corresponding Device Pools, in Calling Party Transformation CSS. But when the call comes in, I still get in alerting/connected state E164 format. All my devices phones and gateways are registered to Sub first, and in failed to Publisher. If I change this and I register all the devices to the Publisher, instead of a subscriber, changing it in Callmanager group configuration, then it all works as expected. Therefore I thought maybe Sub is not replicating correctly, but I've checked and the replication Status is 2! which means it's replicating fine. I have even restarted the Subscriber but it didn't fix the issue. I am out of ideas, and I've done it many times in prior Labs, and it has always worked fine! I've searched through CCM Service Parameters, and couldn't find anything that could affect this behaviour. If anyone has seen this, please let me know... Regards, ___ 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] Multicast MoH to CUBE
Hi Alex - see this excellent post from Shingei re Multicast outwith cucm cluster http://onlinestudylist.com/archives/ccie_voice/2011-March/073393.html Your unicast issue sounds as if it may be codec / xcoder related. hth Brian http://onlinestudylist.com/archives/ccie_voice/2011-March/073393.html On Mon, Mar 28, 2011 at 1:14 PM, Alex Goh ncsalex@gmail.com wrote: Anyone can enlighten me? On Sun, Mar 27, 2011 at 8:23 PM, Alex Goh ncsalex@gmail.com wrote: Hi All, When trying to practice some MoH lab today, I notice that whenever HQ/BR1 phone make a call to BR2 Phone via H323 gateway, which BR2 setup as CME (telephony-service), when HQ/BR1 phone press hold, the MoH will not work (BR2 phone hear silence). But when BR2 phone press Hold, HQ/BR1 phone can hear MoH. Same for when HQ/BR1 tried to send call out to PSTN via the H323 gateway, the PSTN hear no MoH. BR2 Phone call PSTN the MoH is ok though. My HQ is set to send Unicast MoH, while BR1 is set to send Multicast MoH. MoH servers have different DP and Region to all site is g711. I did tried to turn the BR2 into a H323 gateway only (no CUBE) and configured the following commands: ccm-manager music-on-hold, moh music-on-hold.au multicast moh 239.1.1.5 port 16384 Then MoH is working fine when HQ/BR1 put the call to PSTN on hold via the H323 gateway, also when HQ/BR1 calling BR2 Phone which on different DP/Region, MoH is working fine for both direction. Is this normal that Multicast MoH or Unicast MoH is not supporting CUBE in this case? also, why is BR2 CME phone calling HQ/BR1 phone yet MoH is working fine? Regards, Alex ___ 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] Hardware conference is stuck in state TCP_CONN_ERROR
Hi all , I am trying to register my hardware conference to cucm/cme but it is stuck in TCP_CONN_ERROR error state. sh sccp Conferencing Oper State: ACTIVE_IN_PROGRESS - Cause Code: TCP_CONN_ERROR Active Call Manager: NONE TCP Link Status: CONNECT_PENDING, Profile Identifier: 1 Reported Max Streams: 8, Reported Max OOS Streams: 0 ... ... reloaded the router but did not help! Please suggest. Thx, Rahul ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Cbarge in SRST nt working !
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
Re: [OSL | CCIE_Voice] Cbarge in SRST nt working !
Rahul: You may need a 'no huntstop' under your ephone-dn 10 Example: ephone-dn 5 octo-linec number 4300 no-reg primary conference ad-hoc no huntstop If that does not work do a 'sh sccp' and post that output. On Mon, Mar 28, 2011 at 10:01 AM, Rahul Kapor rahul.kapo...@gmail.comwrote: 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
Re: [OSL | CCIE_Voice] Hardware conference is stuck in state TCP_CONN_ERROR
Is this for integration with UCM or CME? If CME, what version did you specify? Earl Hough CCIE #16508 (RS/Security/Voice) From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rahul Kapor Sent: Monday, March 28, 2011 11:58 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Hardware conference is stuck in state TCP_CONN_ERROR Hi all , I am trying to register my hardware conference to cucm/cme but it is stuck in TCP_CONN_ERROR error state. sh sccp Conferencing Oper State: ACTIVE_IN_PROGRESS - Cause Code: TCP_CONN_ERROR Active Call Manager: NONE TCP Link Status: CONNECT_PENDING, Profile Identifier: 1 Reported Max Streams: 8, Reported Max OOS Streams: 0 ... ... reloaded the router but did not help! Please suggest. Thx, Rahul _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Hardware conference is stuck in state TCP_CONN_ERROR
What is in your telephony-service section of your config? -Chase Date: Mon, 28 Mar 2011 21:27:30 +0530 From: rahul.kapo...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Hardware conference is stuck in state TCP_CONN_ERROR Hi all , I am trying to register my hardware conference to cucm/cme but it is stuck in TCP_CONN_ERROR error state. sh sccp Conferencing Oper State: ACTIVE_IN_PROGRESS - Cause Code: TCP_CONN_ERROR Active Call Manager: NONE TCP Link Status: CONNECT_PENDING, Profile Identifier: 1 Reported Max Streams: 8, Reported Max OOS Streams: 0 ... ... reloaded the router but did not help! Please suggest. 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
Re: [OSL | CCIE_Voice] Cbarge in SRST nt working !
Hi, You might want to add ephone type under the ephone. Hope it helps, Bo On Mon, Mar 28, 2011 at 9:01 AM, Rahul Kapor rahul.kapo...@gmail.comwrote: 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
Re: [OSL | CCIE_Voice] Hardware conference is stuck in state TCP_CONN_ERROR
Hi Rahul, I have seen this before and the solution was a router reboot. HTH Prashant On Mon, Mar 28, 2011 at 10:57 AM, Rahul Kapor rahul.kapo...@gmail.comwrote: Hi all , I am trying to register my hardware conference to cucm/cme but it is stuck in TCP_CONN_ERROR error state. sh sccp Conferencing Oper State: ACTIVE_IN_PROGRESS - Cause Code: TCP_CONN_ERROR Active Call Manager: NONE TCP Link Status: CONNECT_PENDING, Profile Identifier: 1 Reported Max Streams: 8, Reported Max OOS Streams: 0 ... ... reloaded the router but did not help! Please suggest. 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
[OSL | CCIE_Voice] A 2nd Corporate Directory
Hi all, Is it possible to set up a Secondary corporate directory as an IP Phone service that coexist with the primary corporate directory in CUCM 7 or CUCM 8? Thank you, Bo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] how to configure cucme to support phoneview
hello guys, great news Unified FX release their lab version. how to configure cucme to support phoneview? i can not find any tutorial on their website. could someone help？ Best Regards, bruno___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP SRST does not work
Looks like the calls are going through dial-peer 9 as soon a you hit the 1. What do you have on that dial-peer ? Another option would be to configure SIP dial rules so the phone will wait to get all 4 digits before sending the invite to CME. El lun, 28-03-2011 a las 20:11 +0200, Roig Borrell, Francesc Xavier escribió: Hi all, I have problems with SIP SRST. The SIP phone registers ok in SRST mode, but I can’t make or receive any type of calls (internal or external). Analyzing Internal calls I’ve seen that when PhoneA (1001) dials Phone B (1002). Only the first number of the called number is sent. The debug ccsip messages shows the first invite with the 1 but there are not any notify messages for kpml with all the other digits Have you found any problem with SRST? Is there anything I am missing in the config? Thanks in advance! voice service voip allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip sip bind control source-interface FastEthernet0/0.240 bind media source-interface FastEthernet0/0.240 registrar server voice register global system message SRST max-dn 4 max-pool 4 ! voice register pool 1 id network 192.168.21.0 mask 255.255.255.0 dtmf-relay rtp-nte codec g711ulaw BRANCH1#sh voice register dial-peers dial-peer voice 40001 voip destination-pattern 1002 redirect ip2ip session target ipv4:192.168.21.69:5060 session protocol sipv2 dtmf-relay rtp-nte digit collect kpml codec g711ulaw bytes 160 after-hours-exempt FALSE dial-peer voice 40002 voip destination-pattern 1001 redirect ip2ip session target ipv4:192.168.21.68:5060 session protocol sipv2 dtmf-relay rtp-nte digit collect kpml codec g711ulaw bytes 160 after-hours-exempt FALSE BRANCH1# *Mar 28 18:58:38.216: //-1//DPM/dpMatchPeersCore: Calling Number=1, Called Number=1, Peer Info Type=DIALPEER_INFO_SPEECH *Mar 28 18:58:38.216: //-1//DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1 *Mar 28 18:58:38.216: //-1//DPM/dpMatchPeersCore: Result=Success(0) after DP_MATCH_DEST *Mar 28 18:58:38.216: //-1//DPM/dpMatchPeersMoreArg: Result=SUCCESS(0) List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=9 *Mar 28 18:58:38.216: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=1001, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Mar 28 18:58:38.216: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40002 *Mar 28 18:58:38.220: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=1001, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Mar 28 18:58:38.220: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40002 *Mar 28 18:58:38.220: //-1/3968292780B0/DPM/dpAssociateIncomingPeerCore: Calling Number=1001, Called Number=1, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH *Mar 28 18:58:38.220: //-1/3968292780B0/DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40002 *Mar 28 18:58:38.228: //-1/3968292780B0/DPM/dpMatchPeersCore: Calling Number=, Called Number=1, Peer Info Type=DIALPEER_INFO_SPEECH *Mar 28 18:58:38.228: //-1/3968292780B0/DPM/dpMatchPeersCore: Match Rule=DP_MATCH_DEST; Called Number=1 *Mar 28 18:58:38.228: //-1/3968292780B0/DPM/dpMatchPeersCore: Result=Partial Matches(1) after DP_MATCH_DEST *Mar 28 18:58:38.228: //-1/3968292780B0/DPM/dpMatchPeersMoreArg: Result=MORE_DIGITS_NEEDED(1) Francesc Xavier Roig Borrell Network Senior Consultant Integración de Sistemas y Tecnología C/Santander, 49-51 Barcelona 08020 Tel. Fijo: (+34) 934965108 ext 64697 Tel. Móvil / Fax: (+34) 647 32 21 55 (24697) / (+34) 934965161 email: francesc.ro...@tecnocom.es http://www.tecnocom.es Por favor, antes de imprimir este mensaje, asegúrate de que es necesario. Ayudemos a cuidar el medio ambiente Este mensaje puede contener información confidencial o privilegiada. Si le ha llegado por error, rogamos no haga uso del mismo, avise al remitente y bórrelo. Consulte aviso legal This message may contain confidential or privileged information. If it has been sent to
