[OSL | CCIE_Voice] Antw: RE: Voicemail Access when AAR kicks in
Many thanks Scott, it did it!!! I forgot the AAR Field and the Ext Num Mask on the Hunt-Pilot. RObert Hardesty, Scott[EMAIL PROTECTED] schrieb am Montag, 10. November 2008 um 01:54 in Nachricht f6c9a5ae0fe45e837f5e2d8c6feeccd6: Check your voicemail hunt pilot configuration. At the VERY bottom of the VM Hunt pilot configuration page you need to assign the hunt pilot to an AAR group and external number mask. It is easy to miss because you have to scroll down to see it! Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Schuknecht Sent: Sunday, November 09, 2008 7:06 PM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] Voicemail Access when AAR kicks in Hi, should Voicemail Access, from BR1, work when AAR kicks in? In my LAB it is currently not working and i don´t know if it works like desinged or if i made an silly mistake. Normal Calls from BR1 to HQ are rerouted over PSTN, only Voicemail calls are not. Regards /Robert
[OSL | CCIE_Voice] VMPI problem.stucked for hours and no progress.who knows VMPI and exchange 2K well may help.
Hi, I setup vpmi between cue and unity and I get this trace output. I cannot find msg which was sent by ipphone in exchange 2000 message tracking center. and of course mwi would not turned on. Here is trace out put. I wasted 3 hours and still cannot understand why exchange does not get the msg. After Queued mail for delivery at the end of the trace I got nothing. I cannot see any msg arriving in exhange 2000. Any idea? 5977 03/02 19:25:45.757 netw smtp 6 250 2.6.0 JAE071504FG-AIM-FOC11131AVA-1015036523816 Queued mail for delivery 4649 03/02 19:28:39.033 netw smtp 2 5997 03/02 19:28:39.048 netw dns 1 ccie 4649 03/02 19:28:39.054 netw smtp 1 5997 03/02 19:28:39.068 netw dns 2 MX: unity.ccie priority: 10 5997 03/02 19:28:39.069 netw dns 1 unity.ccie 5997 03/02 19:28:39.082 netw dns 2 A: 142.4.64.13 5997 03/02 19:28:39.094 netw smtp 3 unity.ccie 5997 03/02 19:28:39.110 netw smtp 4 5997 03/02 19:28:39.112 netw smtp 6 220 unity.CCIE Microsoft ESMTP MAIL Service, Version: 5.0.2195.6713 ready at Mon, 8 Nov 2004 12:15:40 +0200 5997 03/02 19:28:39.116 netw smtp 5 EHLO 5997 03/02 19:28:39.123 netw smtp 6 250-unity.CCIE Hello [200.0.0.100] 5997 03/02 19:28:39.153 netw smtp 6 250-TURN 5997 03/02 19:28:39.155 netw smtp 6 250-ATRN 5997 03/02 19:28:39.155 netw smtp 6 250-SIZE 5997 03/02 19:28:39.156 netw smtp 6 250-ETRN 5997 03/02 19:28:39.157 netw smtp 6 250-PIPELINING 5997 03/02 19:28:39.158 netw smtp 6 250-DSN 5997 03/02 19:28:39.159 netw smtp 6 250-ENHANCEDSTATUSCODES 5997 03/02 19:28:39.160 netw smtp 6 250-8bitmime 5997 03/02 19:28:39.161 netw smtp 6 250-BINARYMIME 5997 03/02 19:28:39.162 netw smtp 6 250-CHUNKING 5997 03/02 19:28:39.163 netw smtp 6 250-VRFY 5997 03/02 19:28:39.164 netw smtp 6 250-X-EXPS GSSAPI NTLM LOGIN 5997 03/02 19:28:39.165 netw smtp 6 250-X-EXPS=LOGIN 5997 03/02 19:28:39.166 netw smtp 6 250-AUTH GSSAPI NTLM LOGIN 5997 03/02 19:28:39.167 netw smtp 6 250-AUTH=LOGIN 5997 03/02 19:28:39.168 netw smtp 6 250-X-LINK2STATE 5997 03/02 19:28:39.169 netw smtp 6 250-XEXCH50 5997 03/02 19:28:39.170 netw smtp 6 250 OK 5997 03/02 19:28:39.664 netw smtp 5 MAIL FROM [EMAIL PROTECTED] 5997 03/02 19:28:39.668 netw smtp 6 250 2.1.0 [EMAIL PROTECTED] OK 5997 03/02 19:28:39.669 netw smtp 5 RCPT TO [EMAIL PROTECTED] 5997 03/02 19:28:39.674 netw smtp 6 250 2.1.5 [EMAIL PROTECTED] 5997 03/02 19:28:39.675 netw smtp 5 DATA 5997 03/02 19:28:39.714 netw smtp 6 354 Start mail input; end with CRLF.CRLF 5997 03/02 19:28:39.718 netw vpim 3 VPIM 5997 03/02 19:28:39.771 netw vpim 3 VPIM: To: [EMAIL PROTECTED] 5997 03/02 19:28:39.779 netw vpim 3 VPIM: From: ph[EMAIL PROTECTED] 5997 03/02 19:28:39.788 netw vpim 3 VPIM: Date: Sat, 02 Mar 2002 08:28:38 + (GMT) 5997 03/02 19:28:39.789 netw vpim 3 VPIM: MIME-Version: 1.0 (Voice 2.0) 5997 03/02 19:28:39.790 netw vpim 3 VPIM: Content-Type: Multipart/Voice-Message; Version=2.0; 5997 03/02 19:28:39.792 netw vpim 3 VPIM: Boundary===VpimMsg==1015057719715 5997 03/02 19:28:39.793 netw vpim 3 VPIM: Content-Transfer-Encoding: 7bit 5997 03/02 19:28:39.795 netw vpim 3 VPIM: Message-ID: JAE071504FG-AIM-FOC11131AVA-1015036523817 5997 03/02 19:28:39.796 netw vpim 3 VPIM: 5997 03/02 19:28:39.797 netw vpim 3 VPIM: --==VpimMsg==1015057719715 5997 03/02 19:28:39.799 netw vpim 3 VPIM: Content-Type: text/directory; charset=us-ascii; profile=vCard 5997 03/02 19:28:39.800 netw vpim 3 VPIM: Content-Transfer-Encoding: 7bit 5997 03/02 19:28:39.801 netw vpim 3 VPIM: Content-Disposition: attachment; filename=ph.vcf 5997 03/02 19:28:39.802 netw vpim 3 VPIM: 5997 03/02 19:28:39.819 netw vpim 3 VPIM: BEGIN:vCard 5997 03/02 19:28:39.820 netw vpim 3 VPIM: FN:ph 5997 03/02 19:28:39.821 netw vpim 3 VPIM: EMAIL;TYPE=INTERNET;TYPE= VPIM:[EMAIL PROTECTED] [EMAIL PROTECTED] 5997 03/02 19:28:39.823 netw vpim 3 VPIM: TEL:4003 5997 03/02 19:28:39.824 netw vpim 3 VPIM: VERSION: 3.0 5997 03/02 19:28:39.825 netw vpim 3 VPIM: END:vCard 5997 03/02 19:28:39.826 netw vpim 3 VPIM: 5997 03/02 19:28:39.864 netw vpim 3 VPIM: --==VpimMsg==1015057719715 5997 03/02 19:28:39.865 netw vpim 3 VPIM: Content-Type: Audio/32KADPCM 5997 03/02 19:28:39.866 netw vpim 3 VPIM: Content-Transfer-Encoding: Base64 5997 03/02 19:28:39.867 netw vpim 3 VPIM: Content-Description: VPIM Message 5997 03/02 19:28:39.868 netw vpim 3 VPIM: Content-Disposition: inline; voice=Voice-Message 5997 03/02 19:28:39.870 netw vpim 3 VPIM: Content-ID: JAE071504FG-AIM-FOC11131AVA-1015036523817 5997 03/02 19:28:39.871 netw vpim 3 VPIM: 5997 03/02 19:28:39.887 netw vpim 7 5997 03/02 19:28:40.804 netw vpim 3 VPIMAUDIO:
[OSL | CCIE_Voice] Antw: VMPI problem.stucked for hours and noprogress.who knows VMPI and exchange 2K well may help.
Jeremy, if i understand your debug right, then are not using Fully Qualified Domain Names. In the trace i saw the following line: netw smtp 5 RCPT TO [EMAIL PROTECTED] . Are you using ccie as FQDN? If yes, try to change it to ccie.lab or any other Top-Level Domain which your DNS Server is using. HTH /Robert jeremy co[EMAIL PROTECTED] schrieb am Montag, 10. November 2008 um 13:50 in Nachricht ad3bb766b6890c8b5237ab4cc0713bec: Hi, I setup vpmi between cue and unity and I get this trace output. I cannot find msg which was sent by ipphone in exchange 2000 message tracking center. and of course mwi would not turned on. Here is trace out put. I wasted 3 hours and still cannot understand why exchange does not get the msg. After Queued mail for delivery at the end of the trace I got nothing. I cannot see any msg arriving in exhange 2000. Any idea? 5977 03/02 19:25:45.757 netw smtp 6 250 2.6.0 JAE071504FG-AIM-FOC11131AVA-1015036523816 Queued mail for delivery 4649 03/02 19:28:39.033 netw smtp 2 5997 03/02 19:28:39.048 netw dns 1 ccie 4649 03/02 19:28:39.054 netw smtp 1 5997 03/02 19:28:39.068 netw dns 2 MX: unity.ccie priority: 10 5997 03/02 19:28:39.069 netw dns 1 unity.ccie 5997 03/02 19:28:39.082 netw dns 2 A: 142.4.64.13 5997 03/02 19:28:39.094 netw smtp 3 unity.ccie 5997 03/02 19:28:39.110 netw smtp 4 5997 03/02 19:28:39.112 netw smtp 6 220 unity.CCIE Microsoft ESMTP MAIL Service, Version: 5.0.2195.6713 ready at Mon, 8 Nov 2004 12:15:40 +0200 5997 03/02 19:28:39.116 netw smtp 5 EHLO 5997 03/02 19:28:39.123 netw smtp 6 250-unity.CCIE Hello [200.0.0.100] 5997 03/02 19:28:39.153 netw smtp 6 250-TURN 5997 03/02 19:28:39.155 netw smtp 6 250-ATRN 5997 03/02 19:28:39.155 netw smtp 6 250-SIZE 5997 03/02 19:28:39.156 netw smtp 6 250-ETRN 5997 03/02 19:28:39.157 netw smtp 6 250-PIPELINING 5997 03/02 19:28:39.158 netw smtp 6 250-DSN 5997 03/02 19:28:39.159 netw smtp 6 250-ENHANCEDSTATUSCODES 5997 03/02 19:28:39.160 netw smtp 6 250-8bitmime 5997 03/02 19:28:39.161 netw smtp 6 250-BINARYMIME 5997 03/02 19:28:39.162 netw smtp 6 250-CHUNKING 5997 03/02 19:28:39.163 netw smtp 6 250-VRFY 5997 03/02 19:28:39.164 netw smtp 6 250-X-EXPS GSSAPI NTLM LOGIN 5997 03/02 19:28:39.165 netw smtp 6 250-X-EXPS=LOGIN 5997 03/02 19:28:39.166 netw smtp 6 250-AUTH GSSAPI NTLM LOGIN 5997 03/02 19:28:39.167 netw smtp 6 250-AUTH=LOGIN 5997 03/02 19:28:39.168 netw smtp 6 250-X-LINK2STATE 5997 03/02 19:28:39.169 netw smtp 6 250-XEXCH50 5997 03/02 19:28:39.170 netw smtp 6 250 OK 5997 03/02 19:28:39.664 netw smtp 5 MAIL FROM [EMAIL PROTECTED] 5997 03/02 19:28:39.668 netw smtp 6 250 2.1.0 [EMAIL PROTECTED] OK 5997 03/02 19:28:39.669 netw smtp 5 RCPT TO [EMAIL PROTECTED] 5997 03/02 19:28:39.674 netw smtp 6 250 2.1.5 [EMAIL PROTECTED] 5997 03/02 19:28:39.675 netw smtp 5 DATA 5997 03/02 19:28:39.714 netw smtp 6 354 Start mail input; end with CRLF.CRLF 5997 03/02 19:28:39.718 netw vpim 3 VPIM 5997 03/02 19:28:39.771 netw vpim 3 VPIM: To: [EMAIL PROTECTED] 5997 03/02 19:28:39.779 netw vpim 3 VPIM: From: ph[EMAIL PROTECTED] 5997 03/02 19:28:39.788 netw vpim 3 VPIM: Date: Sat, 02 Mar 2002 08:28:38 + (GMT) 5997 03/02 19:28:39.789 netw vpim 3 VPIM: MIME-Version: 1.0 (Voice 2.0) 5997 03/02 19:28:39.790 netw vpim 3 VPIM: Content-Type: Multipart/Voice-Message; Version=2.0; 5997 03/02 19:28:39.792 netw vpim 3 VPIM: Boundary===VpimMsg==1015057719715 5997 03/02 19:28:39.793 netw vpim 3 VPIM: Content-Transfer-Encoding: 7bit 5997 03/02 19:28:39.795 netw vpim 3 VPIM: Message-ID: JAE071504FG-AIM-FOC11131AVA-1015036523817 5997 03/02 19:28:39.796 netw vpim 3 VPIM: 5997 03/02 19:28:39.797 netw vpim 3 VPIM: --==VpimMsg==1015057719715 5997 03/02 19:28:39.799 netw vpim 3 VPIM: Content-Type: text/directory; charset=us-ascii; profile=vCard 5997 03/02 19:28:39.800 netw vpim 3 VPIM: Content-Transfer-Encoding: 7bit 5997 03/02 19:28:39.801 netw vpim 3 VPIM: Content-Disposition: attachment; filename=ph.vcf 5997 03/02 19:28:39.802 netw vpim 3 VPIM: 5997 03/02 19:28:39.819 netw vpim 3 VPIM: BEGIN:vCard 5997 03/02 19:28:39.820 netw vpim 3 VPIM: FN:ph 5997 03/02 19:28:39.821 netw vpim 3 VPIM: EMAIL;TYPE=INTERNET;TYPE= VPIM:[EMAIL PROTECTED] [EMAIL PROTECTED] 5997 03/02 19:28:39.823 netw vpim 3 VPIM: TEL:4003 5997 03/02 19:28:39.824 netw vpim 3 VPIM: VERSION: 3.0 5997 03/02 19:28:39.825 netw vpim 3 VPIM: END:vCard 5997 03/02 19:28:39.826 netw vpim 3 VPIM: 5997 03/02 19:28:39.864 netw vpim 3 VPIM: --==VpimMsg==1015057719715 5997 03/02 19:28:39.865 netw vpim 3 VPIM: Content-Type: Audio/32KADPCM 5997 03/02 19:28:39.866 netw vpim 3 VPIM: Content-Transfer-Encoding: Base64 5997 03/02 19:28:39.867 netw vpim 3 VPIM: Content-Description: VPIM Message 5997 03/02 19:28:39.868 netw vpim 3 VPIM: Content-Disposition: inline; voice=Voice-Message 5997 03/02 19:28:39.870 netw vpim 3 VPIM: Content-ID:
Re: [OSL | CCIE_Voice] IPCCX ring back tone to caller
Then try mucic on hold.. file as ring back tone... --- On Mon, 10/11/08, Erick Pineda [EMAIL PROTECTED] wrote: From: Erick Pineda [EMAIL PROTECTED] Subject: [OSL | CCIE_Voice] IPCCX ring back tone to caller To: OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Date: Monday, 10 November, 2008, 6:57 PM an ipccx agent gets a call, when he makes the tranfer the callers hears a ring back tone. does any boby has an idea how to do it, because right now when i make the tranfer the caller hear mucic on hold.. Regards Erick Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
[OSL | CCIE_Voice] Only a few days left.... Need help on redundant IP-Phone Services!!!!
Hi List, i have only a few days left until my first real LAB-Exam (its on Friday). And i don´t get the IP.Phone Services run in a redundant fashion. I would like to use DNS SRV Records, for services like IPMA, EM,FastDials, Addressbook...etc. But i can´t get it to work. Could anybody please explain how to setup CCM and DNS Server for using DNS SRV Records? Or, which are the other solutions to make IP-Phone Services redundant, other than configure 2 Service URLs (1 to PUB, 1 to SUB) pointing to the CCM-Cluster. Any input is very welcome! /Robert P.S.: My method for configuring and using DSN SRV Records was: - configure 2 SRV Records for http; 1 with prio 10 and weight 10, pointing to Publisher; 1 with prio 10 and weight 5 poiunting to Subscriber - configured phones with DNS-Server and domain - configured Phone Service like: http://voip.lab/rest of service url This did not work. I got always an HTTP Error 404 on the Phones. I changed the weight and Priority Parameter or i used only the Priority Parameter, for the SRV Records, but without luck.. :-(
[OSL | CCIE_Voice] Antw: IPCCX ring back tone to caller
Erick, i did not try the following, but what if you make the MOH Audio Source, of the IPCC CTI Ports to play a *.wav file with Ring-Tone. Search on Publisher/IPCCx under: c:\program files\wfavvid..., for *.wav files i think you will find some adequat soundfiles. HTH /Robert Erick Pineda[EMAIL PROTECTED] 10.11.2008 14:27 an ipccx agent gets a call, when he makes the tranfer the callers hears a ring back tone. does any boby has an idea how to do it, because right now when i make the tranfer the caller hear mucic on hold.. Regards Erick
Re: [OSL | CCIE_Voice] H323 GW PSTN - BR1 call.
You only need to add the H323-gateway voip bind srcaddr ip-addr This command is for H323 gateway, not gatekeeper. Pete Olson [EMAIL PROTECTED] 425-965-2577 From: James Jung [mailto:[EMAIL PROTECTED] Sent: Sunday, November 09, 2008 8:29 PM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] H323 GW PSTN - BR1 call. Hi all, I tested PSTN call to BR1 phone and the call was disconnected right away. But after I added below config, the call went through. Interface loop0 H323-gateway voip interface H323-gateway voip bind srcaddr ip-addr Is this normal? The problem is if I configure the above, this BR1 gateway registers to the gatekeeper. Is there any way I can make the call get through without registering the gateway to gatekeeper (without adding any additional config to the gatekeeper)? JJ
Re: [OSL | CCIE_Voice] Infrastructure Question
Is this a real-life question or a lab task? Anyway, I think it can be achieved using Private VLANs to separate Server+Data+Voice traffic. Make the router port promiscuous+configure secondary IP addresses on the router port if you are using a separate subnet per VLAN http://www.cisco.com/en/US/products/hw/switches/ps700/products_tech_note09186a008013565f.shtml What I don't know is whether Auxiliary VLAN and Private VLAN features can interoperate, i.e. whether Aux VLAN can be also configured as isolated PVLAN. IPExpert guys, can anyone comment on this? Rgds Alex - Original Message - From: Kumar, Narinder [EMAIL PROTECTED] To: OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Sent: Monday, November 10, 2008 11:04 AM Subject: [OSL | CCIE_Voice] Infrastructure Question The question is not to use any trunk between the router and switch. If I have only 2 vlans on HQ than this can be achieved easily. But if I have more than 2 vlans e.g server,data and voice, how can I achieve this without using trunk. Thanks 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] Infrastructure Question
The question is not to use any trunk between the router and switch. If I have only 2 vlans on HQ than this can be achieved easily. But if I have more than 2 vlans e.g server,data and voice, how can I achieve this without using trunk. Thanks 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] Section 28 Tasks 23 and 28, LAB Prep Workbook v4.0, call via GateKeeper and IPIPGW problem
Hello, I have problem with calls from CCM to BR2 via GK and IPIPGW. If I configure HQ-RTR GK without IPIPGW (no invia/outvia) like this: ! ! gatekeeper zone local HQ-RTR sldx.lab 172.1.100.1 zone local VGK sldx.lab zone remote PSTN-WAN sldx.lab 10.1.200.2 zone prefix PSTN-WAN 011* bandwidth remote 144 no shutdown ! Everything works fine, I mean 1st call (from 1001 to 3001) uses G711 and establishes without any problems, second call (from 1002 to 3002) uses G729 and establishes without any problems. Third call (1003 to 3003) fails as there is no more bandwidth available. Great. For me this behavior means that all CCM stuff is configured correctly (trunks, patterns, , PTs/CSS, RGs, RLs, etc.). The problem is that if I want to use IPIPGW and change GK config to: ! ! gatekeeper zone local HQ-RTR sldx.lab 172.1.100.1 zone local VGK sldx.lab zone remote PSTN-WAN sldx.lab 10.1.200.2 1719 invia VGK outvia VGK zone prefix PSTN-WAN 011* bandwidth remote 144 no shutdown ! only G711 calls works. On CCM I removed from RG trunk with G711 DP/Region and tried G729 call. The setup works, BR2 phone rings but if I pickup the BR2 phone no RTP is exchanged between the phones and after a while there is a fast busy signal. I can see that CCM phone tries to use G711 instead of G729. Any one has any idea what is going on? I tried some debugs (see below) and I see that I'm hitting dial-peer 0 for incoming call but I have no idea why. I'll be glad for any suggestion what to change in config in order to make it work as expected. Solution described in Proctor Guide book doesn't work form me :( My config and debugs: - HQ RTR - ! voice service voip allow-connections h323 to h323 ! ! interface Loopback0 ip address 172.1.100.1 255.255.255.0 h323-gateway voip interface h323-gateway voip id VGK ipaddr 172.1.100.1 1719 h323-gateway voip h323-id dupa h323-gateway voip bind srcaddr 172.1.100.1 ! ! dial-peer voice 2 voip destination-pattern 011T session target ras codec g711ulaw ! dial-peer voice 3 voip session target ras incoming called-number [12]... ! dial-peer voice 1 voip session target ras incoming called-number [12]... codec g711ulaw ! gateway ! ! gatekeeper zone local HQ-RTR sldx.lab 172.1.100.1 zone local VGK sldx.lab zone remote PSTN-WAN sldx.lab 10.1.200.2 1719 invia VGK outvia VGK zone prefix PSTN-WAN 011* bandwidth remote 144 no shutdown ! - deb gatek main 5 -- *Nov 10 16:47:06.795: gk_rassrv_arq: arqp=0x6490FCB0, crv=0x5, answerCall=0 *Nov 10 16:47:06.795: gk_dns_query: No Name servers *Nov 10 16:47:06.795: rassrv_get_addrinfo: (0113313213001) Tech-prefix match failed. *Nov 10 16:47:06.795: rassrv_get_addrinfo: (0113313213001) Matched zone prefix 011 and remainder 3313213001 *Nov 10 16:47:06.795: rassrv_arq_select_viazone: about to check the source side, src_zonep=0x6595C894 *Nov 10 16:47:06.795: rassrv_arq_select_viazone: matched zone is HQ-RTR, and z_invianamelen=0 *Nov 10 16:47:06.795: rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x708B971C *Nov 10 16:47:06.795: rassrv_arq_select_viazone: matched zone is PSTN-WAN, and z_outvianamelen=3 *Nov 10 16:47:06.795: rassrv_arq_select_viazone and z_outvianamep=VGK *Nov 10 16:47:06.795: rassrv_arq_select_viazone: Received ARQ for a zone (PSTN-WAN) that has an outviazone (VGK) specified. Pick an IP-IP gateway in that viazone. *Nov 10 16:47:06.795: gk_gw_select_ipipgw: zonep: 0x65BA4734, tpp: 0x0, current_endpt: 1 *Nov 10 16:47:06.795: gk_gw_select_ipipgw: Selecting any IPIPGW. qelemp.head=0x7093C848, use_count=1, current_endpt=1 *Nov 10 16:47:06.795: gk_gw_select_ipipgw: Gateway selection will start at the top of the linked list. use_count=1, current_endpt=0 *Nov 10 16:47:06.795: gk_gw_select_ipipgw: qelemp=0x7093C848, loop_count=0 *Nov 10 16:47:06.795: gk_gw_select_ipipgw: Examining tgwp 0x65C4BB00, g_supp_prots: 0x50 qelemp: 0x7093C848, loop_count: 1 *Nov 10 16:47:06.795: gk_gw_select_ipipgw: Found an IPIPGW. tgwp: 0x65C4BB00, endptsigIP: 172.1.100.1, endptrasIP: 172.1.100.1, zone: VGK *Nov 10 16:47:06.795: gk_gw_select_ipipgw: Selected an IPIPGW. *Nov 10 16:47:06.795: rassrv_get_addrinfo: (0113313213001) successfully resolved IPIPGW and returning with return code 0 *Nov 10 16:47:06.799: gk_rassrv_arq: arqp=0x7095C3BC, crv=0x23, answerCall=1 *Nov 10 16:47:06.803: gk_rassrv_arq: arqp=0x64919270, crv=0x24, answerCall=0 *Nov 10 16:47:06.803: gk_dns_query: No Name servers *Nov 10 16:47:06.803: rassrv_get_addrinfo: (0113313213001) Tech-prefix match failed. *Nov 10 16:47:06.803: rassrv_get_addrinfo: (0113313213001) Matched zone prefix 011 and remainder 3313213001 *Nov 10 16:47:06.803: rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x708B971C *Nov 10 16:47:06.803: rassrv_arq_select_viazone: matched zone is PSTN-WAN, and z_outvianamelen=3 *Nov 10 16:47:06.803: rassrv_arq_select_viazone and z_outvianamep=VGK *Nov
Re: [OSL | CCIE_Voice] Section 28 Tasks 23 and 28, LAB Prep Workbook v4.0, call via GateKeeper and IPIPGW problem
Hello, Not sure I understand your question, but I think you are asking why you can't get more than 1 G711 call to the remote zone? You have: bandwidth remote 144 This will allow only one G711 call to your remote zone. Each G711 call uses 128K of bandwidth (64K in each direction) To have more than 1 G711 call you need at least 256K of bandwidth Chris Bartosz Sokołowski wrote: Hello, I have problem with calls from CCM to BR2 via GK and IPIPGW. If I configure HQ-RTR GK without IPIPGW (no invia/outvia) like this: ! ! gatekeeper zone local HQ-RTR sldx.lab 172.1.100.1 zone local VGK sldx.lab zone remote PSTN-WAN sldx.lab 10.1.200.2 zone prefix PSTN-WAN 011* bandwidth remote 144 no shutdown ! Everything works fine, I mean 1st call (from 1001 to 3001) uses G711 and establishes without any problems, second call (from 1002 to 3002) uses G729 and establishes without any problems. Third call (1003 to 3003) fails as there is no more bandwidth available. Great. For me this behavior means that all CCM stuff is configured correctly (trunks, patterns, , PTs/CSS, RGs, RLs, etc.). The problem is that if I want to use IPIPGW and change GK config to: ! ! gatekeeper zone local HQ-RTR sldx.lab 172.1.100.1 zone local VGK sldx.lab zone remote PSTN-WAN sldx.lab 10.1.200.2 1719 invia VGK outvia VGK zone prefix PSTN-WAN 011* bandwidth remote 144 no shutdown ! only G711 calls works. On CCM I removed from RG trunk with G711 DP/Region and tried G729 call. The setup works, BR2 phone rings but if I pickup the BR2 phone no RTP is exchanged between the phones and after a while there is a fast busy signal. I can see that CCM phone tries to use G711 instead of G729. Any one has any idea what is going on? I tried some debugs (see below) and I see that I'm hitting dial-peer 0 for incoming call but I have no idea why. I'll be glad for any suggestion what to change in config in order to make it work as expected. Solution described in Proctor Guide book doesn't work form me :( My config and debugs: - HQ RTR - ! voice service voip allow-connections h323 to h323 ! ! interface Loopback0 ip address 172.1.100.1 255.255.255.0 h323-gateway voip interface h323-gateway voip id VGK ipaddr 172.1.100.1 1719 h323-gateway voip h323-id dupa h323-gateway voip bind srcaddr 172.1.100.1 ! ! dial-peer voice 2 voip destination-pattern 011T session target ras codec g711ulaw ! dial-peer voice 3 voip session target ras incoming called-number [12]... ! dial-peer voice 1 voip session target ras incoming called-number [12]... codec g711ulaw ! gateway ! ! gatekeeper zone local HQ-RTR sldx.lab 172.1.100.1 zone local VGK sldx.lab zone remote PSTN-WAN sldx.lab 10.1.200.2 1719 invia VGK outvia VGK zone prefix PSTN-WAN 011* bandwidth remote 144 no shutdown ! - deb gatek main 5 -- *Nov 10 16:47:06.795: gk_rassrv_arq: arqp=0x6490FCB0, crv=0x5, answerCall=0 *Nov 10 16:47:06.795: gk_dns_query: No Name servers *Nov 10 16:47:06.795: rassrv_get_addrinfo: (0113313213001) Tech-prefix match failed. *Nov 10 16:47:06.795: rassrv_get_addrinfo: (0113313213001) Matched zone prefix 011 and remainder 3313213001 *Nov 10 16:47:06.795: rassrv_arq_select_viazone: about to check the source side, src_zonep=0x6595C894 *Nov 10 16:47:06.795: rassrv_arq_select_viazone: matched zone is HQ-RTR, and z_invianamelen=0 *Nov 10 16:47:06.795: rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x708B971C *Nov 10 16:47:06.795: rassrv_arq_select_viazone: matched zone is PSTN-WAN, and z_outvianamelen=3 *Nov 10 16:47:06.795: rassrv_arq_select_viazone and z_outvianamep=VGK *Nov 10 16:47:06.795: rassrv_arq_select_viazone: Received ARQ for a zone (PSTN-WAN) that has an outviazone (VGK) specified. Pick an IP-IP gateway in that viazone. *Nov 10 16:47:06.795: gk_gw_select_ipipgw: zonep: 0x65BA4734, tpp: 0x0, current_endpt: 1 *Nov 10 16:47:06.795: gk_gw_select_ipipgw: Selecting any IPIPGW. qelemp.head=0x7093C848, use_count=1, current_endpt=1 *Nov 10 16:47:06.795: gk_gw_select_ipipgw: Gateway selection will start at the top of the linked list. use_count=1, current_endpt=0 *Nov 10 16:47:06.795: gk_gw_select_ipipgw: qelemp=0x7093C848, loop_count=0 *Nov 10 16:47:06.795: gk_gw_select_ipipgw: Examining tgwp 0x65C4BB00, g_supp_prots: 0x50 qelemp: 0x7093C848, loop_count: 1 *Nov 10 16:47:06.795: gk_gw_select_ipipgw: Found an IPIPGW. tgwp: 0x65C4BB00, endptsigIP: 172.1.100.1, endptrasIP: 172.1.100.1, zone: VGK *Nov 10 16:47:06.795: gk_gw_select_ipipgw: Selected an IPIPGW. *Nov 10 16:47:06.795: rassrv_get_addrinfo: (0113313213001) successfully resolved IPIPGW and returning with return code 0 *Nov 10 16:47:06.799: gk_rassrv_arq: arqp=0x7095C3BC, crv=0x23, answerCall=1 *Nov 10 16:47:06.803: gk_rassrv_arq: arqp=0x64919270, crv=0x24, answerCall=0 *Nov 10 16:47:06.803: gk_dns_query: No Name servers *Nov 10 16:47:06.803: rassrv_get_addrinfo: (0113313213001) Tech-prefix match failed.
[OSL | CCIE_Voice] Antw: H323 GW PSTN - BR1 call.
James, in my opinion, you have two options: 1) Configure your H323 Gateway, with the IP-Address of the Interface pointing to Callmanager, in the Gteway Config Page on Callmanager 2) Don´t issue the gateway command on the RTR. Without the gateway command the RTR won´t register to any gatekeeper. I think your problem is, that you configured the H323 Gateway with the Loopback-IP on Callmanager but you don´t defined an H323 Interface on the Router. So the Router took the IP-Adress of your Serial/Virtual-Template Interface for the H323 Signaling. HTH /Robert James Jung[EMAIL PROTECTED] schrieb am Montag, 10. November 2008 um 05:28 in Nachricht 25cf73e10328426318da1a386056a458: Hi all, I tested PSTN call to BR1 phone and the call was disconnected right away. But after I added below config, the call went through. Interface loop0 H323-gateway voip interface H323-gateway voip bind srcaddr ip-addr Is this normal? The problem is if I configure the above, this BR1 gateway registers to the gatekeeper. Is there any way I can make the call get through without registering the gateway to gatekeeper (without adding any additional config to the gatekeeper)? JJ
Re: [OSL | CCIE_Voice] Section 28 Tasks 23 and 28, LAB Prep Workbook v4.0, call via GateKeeper and IPIPGW problem
Hello, Not sure I understand your question, but I think you are asking why you can't get more than 1 G711 call to the remote zone? No, the question is - why G729 calls do not work with IPIPGW but they do work if GK without IPIPGW is used. Or in other words - why G711 calls work with IPIPGW but G729 calls do not? -- Best regards, Bartosz -- SOLIDEX S.A. Tel: +48 12 638 04 80 Fax: +48 12 638 04 70 http://www.SOLIDEX.com.pl http://www.SOLIDnySerwis.pl Niniejsza wiadomość zawiera informacje zastrzeżone i stanowiące tajemnicę przedsiębiorstwa firm grupy SOLIDEX. Ujawnianie tych informacji osobom trzecim lub nieuprawnione wykorzystanie ich do wlasnych celów jest zabronione. Jeżeli otrzymaliście Państwo niniejszą wiadomość omyłkowo, prosimy o niezwłoczne skontaktowanie się z nadawcą oraz usunięcie wszelkich kopii niniejszej wiadomości. This message contains proprietary information and trade secrets of SOLIDEX group companies. Unauthorized use or disclosure of this information to any third party is prohibited. If you received this message by mistake, please contact the sender immediately and delete all copies of this message.
Re: [OSL | CCIE_Voice] Antw: H323 GW PSTN - BR1 call.
Hi Robert Thank you for your comment. And than you all for the help. JJ -Original Message- From: Robert Schuknecht [mailto:[EMAIL PROTECTED] Sent: Monday, 10 November 2008 9:11 p.m. To: James Jung; OSL CCIE Voice Lab Exam Subject: Antw: [OSL | CCIE_Voice] H323 GW PSTN - BR1 call. James, in my opinion, you have two options: 1) Configure your H323 Gateway, with the IP-Address of the Interface pointing to Callmanager, in the Gteway Config Page on Callmanager 2) Don´t issue the gateway command on the RTR. Without the gateway command the RTR won´t register to any gatekeeper. I think your problem is, that you configured the H323 Gateway with the Loopback-IP on Callmanager but you don´t defined an H323 Interface on the Router. So the Router took the IP-Adress of your Serial/Virtual-Template Interface for the H323 Signaling. HTH /Robert James Jung[EMAIL PROTECTED] schrieb am Montag, 10. November 2008 um 05:28 in Nachricht 25cf73e10328426318da1a386056a458: Hi all, I tested PSTN call to BR1 phone and the call was disconnected right away. But after I added below config, the call went through. Interface loop0 H323-gateway voip interface H323-gateway voip bind srcaddr ip-addr Is this normal? The problem is if I configure the above, this BR1 gateway registers to the gatekeeper. Is there any way I can make the call get through without registering the gateway to gatekeeper (without adding any additional config to the gatekeeper)? JJ
[OSL | CCIE_Voice] Please remove me from this list
Can the moderator of this list please remove me?. I've requested this before but still continue to receive emails. Please remove me a.s.a.p. Thank you.
Re: [OSL | CCIE_Voice] Can't call to CUE pilot number from HQ
Pardeep, What u get from sh call active voice history brief? What is the reason for call disconnect in the output? On Tue, Nov 11, 2008 at 4:40 AM, Pardeep Singh (pardsing) [EMAIL PROTECTED] wrote: Team, I am having an issues when I call from my HQ site to CME using Gatekeeper Trunk over to CUE pilot number. Here is the call flow: HQ Phone--dial 4001GK TrunkCME = Call is successful HQ Phone---dial 4111GK TrunkCMECUE PILOT NUMBER = Call give this error back once I hang up the call over at CUCM Phone. This is the error I get on CME router: %SDP-3-SDP_PTR_ERROR: Recieved invalid SDP pointer form application. Unable to process. -Traceback= 0x41032A60 0x40EC8B58 on and on My configs: voice service voip allow-connections h323 to sip sip bind control source-interface Loopback0 bind media source-interface Loopback0 sccp local FastEthernet0/0.101 sccp ccm 142.33.66.1 identifier 1 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register mtp123456789 ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 8 associate application SCCP dial-peer voice 100 voip translation-profile incoming FROMGK destination-pattern 2...$ session target ras incoming called-number . not using the default one tech-prefix 5# dtmf-relay h245-alphanumeric no vad dial-peer voice 200 voip destination-pattern 4111$ session protocol sipv2 session target ipv4:142.707.77.253 incoming called-number 800[01] dtmf-relay sip-notify codec g711ulaw no vad telephony-service ip source-address 142.33.66.1 port 2000 system message Your current options sdspfarm units 1 sdspfarm transcode sessions 8 sdspfarm tag 1 mtp123456789 -- Sirus Moghadasian CCIE #21862 (RS)
Re: [OSL | CCIE_Voice] Only a few days left.... Need help on redundantIP-Phone Services!!!!
Robert, Have you seen this link http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a008097cd93.shtml ? I would imagine DNS A record method is the simplest - it all boils down to configuring 2 IP addresses in DNS for a given name+short TTL. If you want IP phones using services in a deterministic way, disable DNS round-robin default behaviour in M$ DNS http://technet.microsoft.com/en-us/library/cc787484.aspx You can try IOS SLB as well, and it will sure work for BR1 phones. IOS SLB could be difficult for HQ phones which are on the same subnet as CCM. Rgds Alex - Original Message - From: Robert Schuknecht [EMAIL PROTECTED] To: OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Sent: Monday, November 10, 2008 1:46 PM Subject: [OSL | CCIE_Voice] Only a few days left Need help on redundantIP-Phone Services Hi List, i have only a few days left until my first real LAB-Exam (its on Friday). And i don´t get the IP.Phone Services run in a redundant fashion. I would like to use DNS SRV Records, for services like IPMA, EM,FastDials, Addressbook...etc. But i can´t get it to work. Could anybody please explain how to setup CCM and DNS Server for using DNS SRV Records? Or, which are the other solutions to make IP-Phone Services redundant, other than configure 2 Service URLs (1 to PUB, 1 to SUB) pointing to the CCM-Cluster. Any input is very welcome! /Robert P.S.: My method for configuring and using DSN SRV Records was: - configure 2 SRV Records for http; 1 with prio 10 and weight 10, pointing to Publisher; 1 with prio 10 and weight 5 poiunting to Subscriber - configured phones with DNS-Server and domain - configured Phone Service like: http://voip.lab/rest of service url This did not work. I got always an HTTP Error 404 on the Phones. I changed the weight and Priority Parameter or i used only the Priority Parameter, for the SRV Records, but without luck.. :-(
Re: [OSL | CCIE_Voice] Configuring a frame relay switch for lab
You seem to have forgotten your frame-relay interface type which would be DCE on the FR switch side. (frame-relay intf-type dce) Please see the following link for more info on setting up FR switch and setting DCE via the interface type: http://www.cisco.com/en/US/docs/ios/12_2/wan/configuration/guide/wcffrely_ps1835_TSD_Products_Configuration_Guide_Chapter.html#wp1065161 Thank you. greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Kagadis (kagadis.com) Sent: Saturday, November 08, 2008 2:41 PM To: OSL CCIE Voice Lab Exam Subject: [OSL | CCIE_Voice] Configuring a frame relay switch for lab I've been trying to set up a frame switch for my home lab and would like to know if anyone may be able to help. None of my frame-relay PVCs are active and I'm wondering what I am doing wrong. 1) For the frame switch I am using a 2610 with c2600-ik9o3s3-mz.123-21.bin and three WIC-1DSU-T1 cards (one for each branch) 2) The branch routers are 2811 with WIC-1DSU-T1-V2 3) The connections are all crossover T1 and layer 1 is working I can't seem to find any good documentation on setting this up properly, any wisdom is very much appreciated. On Frame Relay switch interface Serial0/0 (connected to HQ router) description HQ no ip address encapsulation frame-relay IETF no fair-queue service-module t1 clock source internal service-module t1 timeslots 1-24 frame-relay lmi-type ansi frame-relay route 101 interface Serial1/0 103 frame-relay route 102 interface Serial1/1 104 ! interface Serial1/0 (connected to BR1 router) description BR1 no ip address encapsulation frame-relay IETF no fair-queue service-module t1 clock source internal service-module t1 timeslots 1-24 frame-relay lmi-type ansi frame-relay route 103 interface Serial0/0 101 ! interface Serial1/1 (connected to BR2 router) description BR2 no ip address encapsulation frame-relay IETF no fair-queue service-module t1 clock source internal service-module t1 timeslots 1-24 frame-relay lmi-type ansi frame-relay route 104 interface Serial0/0 102 On HQ router interface Serial0/1/0 no ip address encapsulation frame-relay IETF no keepalive service-module t1 clock source internal ! interface Serial0/1/0.1 point-to-point ip address 172.16.100.1 255.255.255.252 snmp trap link-status frame-relay interface-dlci 101 ! interface Serial0/1/0.2 point-to-point ip address 172.16.100.5 255.255.255.252 snmp trap link-status frame-relay interface-dlci 102 On BR1 router interface Serial0/1/0 no ip address encapsulation frame-relay IETF no keepalive ! interface Serial0/1/0.1 point-to-point ip address 172.16.100.2 255.255.255.252 snmp trap link-status frame-relay interface-dlci 103 On BR2 router interface Serial0/0/0 no ip address encapsulation frame-relay IETF no keepalive ! interface Serial0/0/0.1 point-to-point ip address 172.16.100.6 255.255.255.252 frame-relay interface-dlci 104 PVC Statistics for interface Serial0/0 (Frame Relay DTE) Active Inactive Deleted Static Local 0000 Switched 0200 Unused 0000 DLCI = 101, DLCI USAGE = SWITCHED, PVC STATUS = INACTIVE, INTERFACE = Serial0/0 input pkts 564 output pkts 0in bytes 65930 out bytes 0 dropped pkts 6 in pkts dropped 6 out pkts dropped 0out bytes dropped 0 in FECN pkts 0 in BECN pkts 0 out FECN pkts 0 out BECN pkts 0 in DE pkts 0 out DE pkts 0 out bcast pkts 0 out bcast bytes 0 30 second input rate 0 bits/sec, 0 packets/sec 30 second output rate 0 bits/sec, 0 packets/sec switched pkts 1 Detailed packet drop counters: no out intf 0out intf down 0 no out PVC 0 in PVC down 0out PVC down 6 pkt too big 0 shaping Q full 0 pkt above DE 0 policing drop 0 pvc create time 12:18:46, last time pvc status changed 00:30:22 DLCI = 102, DLCI USAGE = SWITCHED, PVC STATUS = INACTIVE, INTERFACE = Serial0/0 input pkts 533 output pkts 0in bytes 62730 out bytes 0 dropped pkts 10 in pkts dropped 10 out pkts dropped 0out bytes dropped 0 in FECN pkts 0 in BECN pkts 0 out FECN pkts 0 out BECN pkts 0 in DE pkts 0 out DE pkts 0 out bcast pkts 0 out bcast bytes 0 30 second input rate 0 bits/sec, 0 packets/sec 30 second output rate 0 bits/sec, 0 packets/sec switched pkts 0 Detailed packet drop counters: no out intf 0out intf down 0 no out PVC 0 in PVC down 0out PVC down 10 pkt too big 0 shaping Q full 0 pkt above DE 0 policing drop 0 pvc create time 12:18:31, last time pvc status changed
[OSL | CCIE_Voice] POD:28 - gatekeeper section -- task5
Hi, I'm trying to call BR2 CME user 0113313283003 From PSTN-WAN Router traces(below) I can see the called number but when I checked from BR2 traces (as attaached to the e-mail) Called Number=33132433313 is changing. I attached BR2 config (this time E1 is up state) What can be the problem? Thanks, Nov 10 18:50:02.954: gk_process: QUEUE_EVENT (minor 0) wakeup Nov 10 18:50:02.954: gk_rassrv_lrq: (0113313283003) Tech-prefix match failed. Nov 10 18:50:02.954: gk_rassrv_lrq: (0113313283003) Matched zone-prefix 011 Nov 10 18:50:02.954: gk_rassrv_lrq: checking the source of the LRQ. source_endptp=0x0 Nov 10 18:50:02.954: gk_rassrv_lrq: srcvia found gkname of source zone. looking up HQ-RTR in zone list Nov 10 18:50:02.954: gk_rassrv_lrq: about to check the source side, src_zonep=0x452E041C Nov 10 18:50:02.954: gk_rassrv_lrq: matched zone is HQ-RTR Nov 10 18:50:02.954: gk_rassrv_lrq and z_invianamelen=0 Nov 10 18:50:02.954: gk_rassrv_lrq: about to check the destination side, zonep=0x452E01C4 Nov 10 18:50:02.954: gk_rassrv_lrq:matched zone is PSTN-WAN Nov 10 18:50:02.954: gk_rassrv_lrq and z_outvianamelen=0 Nov 10 18:50:02.958: gk_zone_get_proxy_usage: local zone= PSTN-WAN, remote zone= HQ-RTR, call direction= 0, eptype= 2114 be_entry= 0 Nov 10 18:50:02.958: gk_zone_get_proxy_usage: returns proxied = 0 Nov 10 18:50:02.966: gk_process: QUEUE_EVENT (minor 0) wakeup Nov 10 18:50:02.970: gk_rassrv_arq: arqp=0x4432A860, crv=0x19, answerCall=1 Nov 10 18:50:02.970: gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Nov 10 18:50:07.702: gk_process: got a TIMER event Nov 10 18:50:07.702: gk_handle_timers Nov 10 18:50:07.702: gk_handle_timers: managed timer expired 0x43E9CA30 Nov 10 18:50:07.702: gk_handle_timers: managed timer expired 0x43E9C880 Nov 10 18:50:10.342: gk_process: QUEUE_EVENT (minor 0) wakeup Nov 10 18:50:10.350: gk_process: QUEUE_EVENT (minor 0) wakeup Nov 10 18:50:10.354: gk_rassrv_lrq: (0113313283003) Tech-prefix match failed. Nov 10 18:50:10.354: gk_rassrv_lrq: (0113313283003) Matched zone-prefix 011 Nov 10 18:50:10.354: gk_rassrv_lrq: checking the source of the LRQ. source_endptp=0x0 Nov 10 18:50:10.354: gk_rassrv_lrq: srcvia found gkname of source zone. looking up HQ-RTR in zone list Nov 10 18:50:10.354: gk_rassrv_lrq: about to check the source side, src_zonep=0x452E041C Nov 10 18:50:10.354: gk_rassrv_lrq: matched zone is HQ-RTR Nov 10 18:50:10.354: gk_rassrv_lrq and z_invianamelen=0 Nov 10 18:50:10.354: gk_rassrv_lrq: about to check the destination side, zonep=0x452E01C4 Nov 10 18:50:10.354: gk_rassrv_lrq:matched zone is PSTN-WAN Nov 10 18:50:10.354: gk_rassrv_lrq and z_outvianamelen=0 Nov 10 18:50:10.354: gk_zone_get_proxy_usage: local zone= PSTN-WAN, remote zone= HQ-RTR, call direction= 0, eptype= 2114 be_entry= 0 Nov 10 18:50:10.354: gk_zone_get_proxy_usage: returns proxied = 0 Nov 10 18:50:10.366: gk_process: QUEUE_EVENT (minor 0) wakeup Nov 10 18:50:10.366: gk_rassrv_arq: arqp=0x453DB0AC, crv=0x1A, answerCall=1 Nov 10 18:50:10.366: gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Nov 10 18:50:12.946: gk_process: QUEUE_EVENT (minor 0) wakeup Nov 10 18:50:17.738: gk_process: QUEUE_EVENT (minor 0) wakeup Nov 10 18:50:22.702: gk_process: got a TIMER event % Unrecognized command Pod28-BR2-RTR# Nov 10 18:47:11.931: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=, Called Number=, Voice-Interface=0x4777EE04, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Nov 10 18:47:11.931: //-1//DPM/dpAssociateIncomingPeerCore: Result=NO_MATCH(-1) After All Match Rules Attempt Pod28-BR2-RTR# Nov 10 18:47:16.959: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=1004, Called Number=33132433313, Voice-Interface=0x4777EE04, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Nov 10 18:47:16.959: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1 Nov 10 18:47:16.959: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=1004, Called Number=33132433313, Voice-Interface=0x4777EE04, Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Nov 10 18:47:16.959: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1 Nov 10 18:47:16.963: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=1004, Called Number=33132433313, Voice-Interface=0x4777EE04, T Pod28-BR2-RTR#imeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Nov 10 18:47:16.963: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1 Nov 10 18:47:16.971: //-1/D4CA39D88037/DPM/dpMatchPeersCore:
Re: [OSL | CCIE_Voice] Section 28 Tasks 23 and 28, LAB Prep Workbook v4.0, call via GateKeeper and IPIPGW problem
Bartosz, Try using 2 sets of dialpeers on IPIPGW: one set hardcoded to G711 with max-conn 1 and second set hardcoded to G729 with max-conn 1+priority 1. Please report your results if/when you have a chance of labbing this up:-) Cheers Alex - Original Message - From: Bartosz Sokołowski [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Sent: Monday, November 10, 2008 6:37 PM Subject: Re: [OSL | CCIE_Voice] Section 28 Tasks 23 and 28,LAB Prep Workbook v4.0, call via GateKeeper and IPIPGW problem Hello Alex, IPIPGW won't negotiate the codec with HQR, it has to be statically nailed down for all 4 legs. If you wish to have G729 call from CCM to CME via IPIPGW, then it should be 4 dialpeers with default G729 codecs (2 with incoming called-number, 2 with destination-pattern): That was it! Thanks! Now G729 works. As I understand - without a transcoder on HQ-RTR there is no codec flexibility - I mean a scenario when 1st call from CCM uses G711 and the second uses G729 (due to bandwidth limit on GK) is impossible? Now I have RG with both trunks (711/729) and I also added another four dial-peers hardcoded for G711 but it doesn't work. You can also change codec to 'codec transparent' on every dial-peer and then you have a choice on CCM - if you put 729-trunk 1st on RG list then it will use 729. If you put 711-trunk 1st on the list then it will use 711. The question is how to achieve a flexibility - 1st call 711 and 2nd 729 due to lack of bandwidth. -- Best regards, Bartosz dial-peer voice 100 voip incoming called-number [12]... # Inbound DP for CCM-IPIPGW call leg dtmf-relay h245-alpha ! dial-peer voice 101 voip incoming called-number 3313213... # Inbound DP for BR2-IPIPGW call leg dtmf-relay h245-alpha ! dial-peer voice 102 voip destination-pattern 011T # Outbound DP for IPIPGW-BR2 call leg session target ras dtmf-relay h245-alpha ! dial-peer voice 103 voip destination-pattern [12]... # Outbound DP for BR2-IPIPGW call leg session target ras dtmf-relay h245-alpha ! It always worked for me this way. If you want to transcode G711-G729 on IPIPGW, configure SRST/CCME instance, SDSP farm and register SDSP farm to SRST/CCME instance. Rgds Alex - Original Message - From: Bartosz Sokołowski [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Sent: Monday, November 10, 2008 4:54 PM Subject: [OSL | CCIE_Voice] Section 28 Tasks 23 and 28,LAB Prep Workbook v4.0, call via GateKeeper and IPIPGW problem Hello, I have problem with calls from CCM to BR2 via GK and IPIPGW. If I configure HQ-RTR GK without IPIPGW (no invia/outvia) like this: ! ! gatekeeper zone local HQ-RTR sldx.lab 172.1.100.1 zone local VGK sldx.lab zone remote PSTN-WAN sldx.lab 10.1.200.2 zone prefix PSTN-WAN 011* bandwidth remote 144 no shutdown ! Everything works fine, I mean 1st call (from 1001 to 3001) uses G711 and establishes without any problems, second call (from 1002 to 3002) uses G729 and establishes without any problems. Third call (1003 to 3003) fails as there is no more bandwidth available. Great. For me this behavior means that all CCM stuff is configured correctly (trunks, patterns, , PTs/CSS, RGs, RLs, etc.). The problem is that if I want to use IPIPGW and change GK config to: ! ! gatekeeper zone local HQ-RTR sldx.lab 172.1.100.1 zone local VGK sldx.lab zone remote PSTN-WAN sldx.lab 10.1.200.2 1719 invia VGK outvia VGK zone prefix PSTN-WAN 011* bandwidth remote 144 no shutdown ! only G711 calls works. On CCM I removed from RG trunk with G711 DP/Region and tried G729 call. The setup works, BR2 phone rings but if I pickup the BR2 phone no RTP is exchanged between the phones and after a while there is a fast busy signal. I can see that CCM phone tries to use G711 instead of G729. Any one has any idea what is going on? I tried some debugs (see below) and I see that I'm hitting dial-peer 0 for incoming call but I have no idea why. I'll be glad for any suggestion what to change in config in order to make it work as expected. Solution described in Proctor Guide book doesn't work form me :( My config and debugs: - HQ RTR - ! voice service voip allow-connections h323 to h323 ! ! interface Loopback0 ip address 172.1.100.1 255.255.255.0 h323-gateway voip interface h323-gateway voip id VGK ipaddr 172.1.100.1 1719 h323-gateway voip h323-id dupa h323-gateway voip bind srcaddr 172.1.100.1 ! ! dial-peer voice 2 voip destination-pattern 011T session target ras codec g711ulaw ! dial-peer voice 3 voip session target ras incoming called-number [12]... ! dial-peer voice 1 voip session target ras incoming called-number [12]... codec g711ulaw ! gateway ! ! gatekeeper zone local HQ-RTR sldx.lab 172.1.100.1 zone local VGK sldx.lab zone remote PSTN-WAN sldx.lab 10.1.200.2 1719 invia VGK outvia VGK zone prefix PSTN-WAN 011* bandwidth remote 144 no shutdown ! -- SOLIDEX S.A. Tel: +48 12 638 04 80 Fax: +48 12 638 04 70 http://www.SOLIDEX.com.pl http://www.SOLIDnySerwis.pl Niniejsza
[OSL | CCIE_Voice] IPCC Agent on BR2/CME
Is it possible to have an agent logged into a phone on the CME router? Since the phone is not registered to CCM, you can't associate the phone to the user agent. But could you associate a CTI route point to the agent user and then have the CTI route point forward all to the CME?
Re: [OSL | CCIE_Voice] CME Park
Wow- this is news- I didn't know there was a limitation with the # of softkeys...seems like a special feature to me. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Olson, Pete [EMAIL PROTECTED] Date: Sat, 8 Nov 2008 18:21:49 -0800 To: Olson, Pete [EMAIL PROTECTED], Hardesty, Scott [EMAIL PROTECTED], Vik Malhi [EMAIL PROTECTED], Ryan Trauernicht [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] CME Park I had park as the last item in the list. I changed it to third and now it works. The new last on the list is grayed out. Is there a setting to allow all 6 to be available or is the last on wasted? Pete Olson [EMAIL PROTECTED] 425-965-2577 -Original Message- From: Olson, Pete Sent: Saturday, November 08, 2008 1:02 PM To: Hardesty, Scott; Vik Malhi; Ryan Trauernicht Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME Park I tried that today and it didn't help. Reloaded the router and that didn't work either. Pete Olson [EMAIL PROTECTED] 425-965-2577 -Original Message- From: Hardesty, Scott [mailto:[EMAIL PROTECTED] Sent: Friday, November 07, 2008 10:42 AM To: Olson, Pete; Vik Malhi; Ryan Trauernicht Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] CME Park Pete, did you try create cnf files under telephony service after you added the call park functionality? That may do the trick... Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: Olson, Pete [EMAIL PROTECTED] Sent: Friday, November 07, 2008 12:41 PM To: Vik Malhi [EMAIL PROTECTED]; Ryan Trauernicht [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME Park Yes I do have an ephone-dn assigned to a park slot. Here is my configs. ephone-template 1 softkeys connected Hold Endcall Trnsfer Confrn Acct Park ! ephone-dn 7 number 3400 no-reg primary park-slot timeout 10 limit 6 ! phone 1 ephone-template 1 username BR2-Ph1 password cisco mac-address 000D.BCCC.6DF0 type 7940 button 1:1 ! ephone 3 ephone-template 1 username BR2-Ph3 password cisco mac-address 000D.ED40.9FDD type 7960 button 1:2 ! ephone 4 ephone-template 1 username BR2-Ph4 password cisco mac-address 0006.D781.C4A2 type 7960 button 1:3 Pete Olson [EMAIL PROTECTED] 425-965-2577 From: Vik Malhi [mailto:[EMAIL PROTECTED] Sent: Thursday, November 06, 2008 10:56 PM To: Ryan Trauernicht; Olson, Pete Cc: Jacob Owen; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME Park Nice call Ryan. I think that's it. Pete- post your config if this does not fix it. Thanks. -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Ryan Trauernicht [EMAIL PROTECTED] Date: Thu, 6 Nov 2008 23:14:30 -0600 To: Olson, Pete [EMAIL PROTECTED] Cc: Vik Malhi [EMAIL PROTECTED], Jacob Owen [EMAIL PROTECTED], ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME Park Do you have a ephone-dn assigned to a park slot? On Thu, Nov 6, 2008 at 3:07 PM, Olson, Pete [EMAIL PROTECTED] wrote: Yes, it is in the connected state when the park is grayed out. Pete Olson [EMAIL PROTECTED] 425-965-2577 From: Vik Malhi [mailto:[EMAIL PROTECTED] Sent: Thursday, November 06, 2008 11:24 AM To: Olson, Pete; Jacob Owen Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME Park Park is only applicable when you have an active call- make sure you are in the connected state before attempting to park. What state is the phone in when you see park grayed out? -- Vik Malhi - CCIE #13890 Senior Technical Instructor - IPexpert,
[OSL | CCIE_Voice] Unity MWI on SRST phone
Hi, Can anyone tell me how to configure to turn on mwi for phones in SRST mode when someone left a message? Is it possible? JamesJ
Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast
The failover should happen transparently. To test you should stop the IP Voice Media Streaming App service on the sub. Unicast MOH is dependent on Locations CAC being available whereas multicast is not. Check the DP of the PUB MOH server and make sure that there is enough bandwidth to the BR1 site for support of the negotiated codec. You could just set the Location at BR1 to be unlimited, resync bandwidth- then test again. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Ryan Trauernicht [EMAIL PROTECTED] Date: Sun, 9 Nov 2008 16:14:39 -0600 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MOH Multicast failover to Unicast I have phone A at the headquarters that has a MRG1 of Sub (multicast) and MRG2 of Pub (Unicast). Then in my MRGL I have MRG1 then MRG2. If I shut down the Subscriber I lose all MOH. Am I missing something to allow a Multicast MOH server failover to a Unicast MOH server?
Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast
Hi Questions about this moh. If you want to allow 2 g729 calls and moh(first multicast and later unicast), how much bandwidth do we need to configure? With 48 Kbps in the location bandwidth, it allows 2 g729 calls and multicast. But when the multicast moh server is down, the moh cannot be heard on BR1 site. To allow unicast moh, you need to add 80kbps to the 48kbps. The problem is this would allow 7 g729 calls when the moh is not being used. Any suggestion? JamesJ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vik Malhi Sent: Tuesday, 11 November 2008 9:38 a.m. To: Ryan Trauernicht; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast The failover should happen transparently. To test you should stop the IP Voice Media Streaming App service on the sub. Unicast MOH is dependent on Locations CAC being available whereas multicast is not. Check the DP of the PUB MOH server and make sure that there is enough bandwidth to the BR1 site for support of the negotiated codec. You could just set the Location at BR1 to be unlimited, resync bandwidth- then test again. -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Ryan Trauernicht [EMAIL PROTECTED] Date: Sun, 9 Nov 2008 16:14:39 -0600 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MOH Multicast failover to Unicast I have phone A at the headquarters that has a MRG1 of Sub (multicast) and MRG2 of Pub (Unicast). Then in my MRGL I have MRG1 then MRG2. If I shut down the Subscriber I lose all MOH. Am I missing something to allow a Multicast MOH server failover to a Unicast MOH server?
Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast
Sorry, typo, 5 g729 calls. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Jung Sent: Tuesday, 11 November 2008 9:46 a.m. To: Vik Malhi; Ryan Trauernicht; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast Hi Questions about this moh. If you want to allow 2 g729 calls and moh(first multicast and later unicast), how much bandwidth do we need to configure? With 48 Kbps in the location bandwidth, it allows 2 g729 calls and multicast. But when the multicast moh server is down, the moh cannot be heard on BR1 site. To allow unicast moh, you need to add 80kbps to the 48kbps. The problem is this would allow 7 g729 calls when the moh is not being used. Any suggestion? JamesJ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vik Malhi Sent: Tuesday, 11 November 2008 9:38 a.m. To: Ryan Trauernicht; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast The failover should happen transparently. To test you should stop the IP Voice Media Streaming App service on the sub. Unicast MOH is dependent on Locations CAC being available whereas multicast is not. Check the DP of the PUB MOH server and make sure that there is enough bandwidth to the BR1 site for support of the negotiated codec. You could just set the Location at BR1 to be unlimited, resync bandwidth- then test again. -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Ryan Trauernicht [EMAIL PROTECTED] Date: Sun, 9 Nov 2008 16:14:39 -0600 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MOH Multicast failover to Unicast I have phone A at the headquarters that has a MRG1 of Sub (multicast) and MRG2 of Pub (Unicast). Then in my MRGL I have MRG1 then MRG2. If I shut down the Subscriber I lose all MOH. Am I missing something to allow a Multicast MOH server failover to a Unicast MOH server?
Re: [OSL | CCIE_Voice] Unity MWI on SRST phone
Thanks Alex JamesJ From: Alex [mailto:[EMAIL PROTECTED] Sent: Tuesday, 11 November 2008 10:10 a.m. To: James Jung; OSL CCIE Voice Lab Exam Subject: Re: [OSL | CCIE_Voice] Unity MWI on SRST phone Q. What is the support for voicemail integration with the Cisco Unity server through analog or DTMF? A. SRST uses the same in-band analog or DTMF voicemail integration method that Cisco Unified Communications Manager Express uses to allow call forward busy, call forward no answer, or call forward all to the Cisco Unity server through analog or DTMF through the PSTN. An incoming call can be forwarded to the Cisco Unity voicemail server when call forward busy, call forward no answer, or call forward all is configured in the SRST router. However, MWI integration is not yet supported in SRST. You can rely on the Missed calls shown on the phone display to check for your voicemail. Note that FXO hairpin forwarded calls to voicemail must have disconnect supervision from the central office. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps216 9/prod_qas0900aecd8028d113.html Cheers Alex - Original Message - From: James Jung mailto:[EMAIL PROTECTED] To: OSL CCIE Voice Lab Exam mailto:ccie_voice@onlinestudylist.com Sent: Monday, November 10, 2008 8:12 PM Subject: [OSL | CCIE_Voice] Unity MWI on SRST phone Hi, Can anyone tell me how to configure to turn on mwi for phones in SRST mode when someone left a message? Is it possible? JamesJ
Re: [OSL | CCIE_Voice] Unity MWI on SRST phone
Not possible. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: James Jung [EMAIL PROTECTED] Date: Tue, 11 Nov 2008 09:12:47 +1300 To: OSL CCIE Voice Lab Exam ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Unity MWI on SRST phone Hi, Can anyone tell me how to configure to turn on mwi for phones in SRST mode when someone left a message? Is it possible? JamesJ
Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast
Re: [OSL | CCIE_Voice] MOH Multicast failover to UnicastJamesJ, You can configure unicast MOH server to be in a separate location. Then for this location allow whatever x 24kbps BW for whatever number of unicast MOH G729 streams you need to activate when multicast MOH server fails. Rgds Alex - Original Message - From: James Jung To: Vik Malhi ; Ryan Trauernicht ; ccie_voice@onlinestudylist.com Sent: Monday, November 10, 2008 8:53 PM Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast Sorry, typo, 5 g729 calls. -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Jung Sent: Tuesday, 11 November 2008 9:46 a.m. To: Vik Malhi; Ryan Trauernicht; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast Hi Questions about this moh. If you want to allow 2 g729 calls and moh(first multicast and later unicast), how much bandwidth do we need to configure? With 48 Kbps in the location bandwidth, it allows 2 g729 calls and multicast. But when the multicast moh server is down, the moh cannot be heard on BR1 site. To allow unicast moh, you need to add 80kbps to the 48kbps. The problem is this would allow 7 g729 calls when the moh is not being used. Any suggestion? JamesJ -- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vik Malhi Sent: Tuesday, 11 November 2008 9:38 a.m. To: Ryan Trauernicht; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast The failover should happen transparently. To test you should stop the IP Voice Media Streaming App service on the sub. Unicast MOH is dependent on Locations CAC being available whereas multicast is not. Check the DP of the PUB MOH server and make sure that there is enough bandwidth to the BR1 site for support of the negotiated codec. You could just set the Location at BR1 to be unlimited, resync bandwidth- then test again. -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Ryan Trauernicht [EMAIL PROTECTED] Date: Sun, 9 Nov 2008 16:14:39 -0600 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MOH Multicast failover to Unicast I have phone A at the headquarters that has a MRG1 of Sub (multicast) and MRG2 of Pub (Unicast). Then in my MRGL I have MRG1 then MRG2. If I shut down the Subscriber I lose all MOH. Am I missing something to allow a Multicast MOH server failover to a Unicast MOH server?
Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast
Perfection. Thanks. From: Alex [mailto:[EMAIL PROTECTED] Sent: Tuesday, 11 November 2008 10:17 a.m. To: James Jung; Vik Malhi; Ryan Trauernicht; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast JamesJ, You can configure unicast MOH server to be in a separate location. Then for this location allow whatever x 24kbps BW for whatever number of unicast MOH G729 streams you need to activate when multicast MOH server fails. Rgds Alex - Original Message - From: James Jung mailto:[EMAIL PROTECTED] To: Vik Malhi mailto:[EMAIL PROTECTED] ; Ryan Trauernicht mailto:[EMAIL PROTECTED] ; ccie_voice@onlinestudylist.com Sent: Monday, November 10, 2008 8:53 PM Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast Sorry, typo, 5 g729 calls. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Jung Sent: Tuesday, 11 November 2008 9:46 a.m. To: Vik Malhi; Ryan Trauernicht; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast Hi Questions about this moh. If you want to allow 2 g729 calls and moh(first multicast and later unicast), how much bandwidth do we need to configure? With 48 Kbps in the location bandwidth, it allows 2 g729 calls and multicast. But when the multicast moh server is down, the moh cannot be heard on BR1 site. To allow unicast moh, you need to add 80kbps to the 48kbps. The problem is this would allow 7 g729 calls when the moh is not being used. Any suggestion? JamesJ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vik Malhi Sent: Tuesday, 11 November 2008 9:38 a.m. To: Ryan Trauernicht; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MOH Multicast failover to Unicast The failover should happen transparently. To test you should stop the IP Voice Media Streaming App service on the sub. Unicast MOH is dependent on Locations CAC being available whereas multicast is not. Check the DP of the PUB MOH server and make sure that there is enough bandwidth to the BR1 site for support of the negotiated codec. You could just set the Location at BR1 to be unlimited, resync bandwidth- then test again. -- Vik Malhi - CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Ryan Trauernicht [EMAIL PROTECTED] Date: Sun, 9 Nov 2008 16:14:39 -0600 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MOH Multicast failover to Unicast I have phone A at the headquarters that has a MRG1 of Sub (multicast) and MRG2 of Pub (Unicast). Then in my MRGL I have MRG1 then MRG2. If I shut down the Subscriber I lose all MOH. Am I missing something to allow a Multicast MOH server failover to a Unicast MOH server?
Re: [OSL | CCIE_Voice] BACD Script Failing Across GK - 3rd Day in a Row !
Please confirm that this is not the transcoder. Set the DP on CCM to use g711. Set the inbound voip dial-peer to use g711. Confirm the results of this test. If the call works try cleaning up the codecs in your dspfarm profile. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Mike Brooks [EMAIL PROTECTED] Date: Mon, 10 Nov 2008 20:08:01 -0500 To: Norma Exel [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Script Failing Across GK - 3rd Day in a Row ! This the 3rd day in a row that I have had this problem. It seems to be a bug when using the ipipgw functionality of the CME. It happens regardless of codec. Has anyone been able to call from HQ or BR1 to the BACD script with G729 across the WAN ? Thx, Mike Brooks CCIE# 16027 (RS) On Sun, Nov 9, 2008 at 11:53 PM, Norma Exel [EMAIL PROTECTED] wrote: so incoming leg to dial-peer 2 to outgoing leg dial-peer 3301 to incoming leg dial-peer 3301 does not work but... incoming call leg dial-peer 3301 works? (experiment 2) I don't think it's a codec thing either. maybe a code thing. do you need multiple call legs? especially since what you're trying to do sounds like IPIPGW though it seems to work since debug shows it to be the case. curious to see what you find out. - Original Message - From: Mike Brooks [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Script Failing Across GK Date: Sun, 9 Nov 2008 13:38:52 -0500 I am having issues when calling across the GK from BR1 and HQ to the BACD script (g729). I can hit the BACD script from the PSTN and the CME phones. I also can call into CUE from BR1/HQ. My incoming dial-peer is dial-peer 2 which is stripping the tech-prefix of 1# and then hitting my outgoing dial-peer 3301 which sends it to the loopback. Then it hits the same incoming dial-peer 3301 which has the service aa command. All I see on the phone is call proceed for about 5 seconds and then a fast busy. I have had this problem 2 days in a row. So I must be doing something wrong. I have tried the following: 1. allowed the codec to be G711 across the GK (modified trunk region and dial-peer 2) - call fails. 2. removed the tech prefix 1# from the call and allowed it to hit a different incoming dial-peer on CME (DP 3301) with G711 - call works. * Also my calls to CUE from HQ work fine and do invoke the transcoder. * It seems regardless of codec if the call hits dial-peer 2 first the call fails. * I have also tried rebooting :-) From the debugs I can see the script is being called: Pod25-BR2-RTR#debug voice application script voip application script debugging is on Pod25-BR2-RTR# Pod25-BR2-RTR# Pod25-BR2-RTR# Pod25-BR2-RTR# Nov 9 17:44:48.613: //141//TCL :/tcl_PutsObjCmd: proc init_perCallvars Nov 9 17:44:48.613: Nov 9 17:44:48.617: //141//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing Welcome Prompt and options menu ++ Pod25-BR2-RTR# Pod25-BR2-RTR#debug voice dialpeer inout voip dialpeer inout debugging is on Pod25-BR2-RTR# Pod25-BR2-RTR# Pod25-BR2-RTR# Pod25-BR2-RTR# Nov 9 17:41:37.788: //-1/0095BC421000/DPM/dpAssociateIncomingPeerCore: Calling Number=2122251003, Called Number=1#3300, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Nov 9 17:41:37.788: //-1/0095BC421000/DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=2 Nov 9 17:41:37.788: //-1/0095BC421000/DPM/dpAssociateIncomingPeerCore: Calling Number=2122251003, Called Number=1#3300, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Nov 9 17:41:37.788: //-1/0095BC421000/DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=2 Nov 9 17:41:37.800: //-1/0095BC421000/DPM/dpAssociateIncomingPeerCore: Calling Number=2122251003, Called Number=1#3300, Voice-Interface=0x0, Timeout=TRUE, Peer En Pod25-BR2-RTR#cap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Nov 9 17:41:37.800: //-1/0095BC421000/DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=2 Nov 9 17:41:37.800: //-1/0095BC421000/DPM/dpAssociateIncomingPeerCore: Calling Number=2122251003, Called Number=1#3300, Voice-Interface=0x0, Timeout=TRUE,
Re: [OSL | CCIE_Voice] BACD Script Failing Across GK - 3rd Day in a Row !
By cleaning up the codecs- only have g711u, g711a and g729r8. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Vik Malhi [EMAIL PROTECTED] Date: Mon, 10 Nov 2008 17:28:47 -0800 To: Mike Brooks [EMAIL PROTECTED], Norma Exel [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Script Failing Across GK - 3rd Day in a Row ! Please confirm that this is not the transcoder. Set the DP on CCM to use g711. Set the inbound voip dial-peer to use g711. Confirm the results of this test. If the call works try cleaning up the codecs in your dspfarm profile. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Mike Brooks [EMAIL PROTECTED] Date: Mon, 10 Nov 2008 20:08:01 -0500 To: Norma Exel [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Script Failing Across GK - 3rd Day in a Row ! This the 3rd day in a row that I have had this problem. It seems to be a bug when using the ipipgw functionality of the CME. It happens regardless of codec. Has anyone been able to call from HQ or BR1 to the BACD script with G729 across the WAN ? Thx, Mike Brooks CCIE# 16027 (RS) On Sun, Nov 9, 2008 at 11:53 PM, Norma Exel [EMAIL PROTECTED] wrote: so incoming leg to dial-peer 2 to outgoing leg dial-peer 3301 to incoming leg dial-peer 3301 does not work but... incoming call leg dial-peer 3301 works? (experiment 2) I don't think it's a codec thing either. maybe a code thing. do you need multiple call legs? especially since what you're trying to do sounds like IPIPGW though it seems to work since debug shows it to be the case. curious to see what you find out. - Original Message - From: Mike Brooks [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Script Failing Across GK Date: Sun, 9 Nov 2008 13:38:52 -0500 I am having issues when calling across the GK from BR1 and HQ to the BACD script (g729). I can hit the BACD script from the PSTN and the CME phones. I also can call into CUE from BR1/HQ. My incoming dial-peer is dial-peer 2 which is stripping the tech-prefix of 1# and then hitting my outgoing dial-peer 3301 which sends it to the loopback. Then it hits the same incoming dial-peer 3301 which has the service aa command. All I see on the phone is call proceed for about 5 seconds and then a fast busy. I have had this problem 2 days in a row. So I must be doing something wrong. I have tried the following: 1. allowed the codec to be G711 across the GK (modified trunk region and dial-peer 2) - call fails. 2. removed the tech prefix 1# from the call and allowed it to hit a different incoming dial-peer on CME (DP 3301) with G711 - call works. * Also my calls to CUE from HQ work fine and do invoke the transcoder. * It seems regardless of codec if the call hits dial-peer 2 first the call fails. * I have also tried rebooting :-) From the debugs I can see the script is being called: Pod25-BR2-RTR#debug voice application script voip application script debugging is on Pod25-BR2-RTR# Pod25-BR2-RTR# Pod25-BR2-RTR# Pod25-BR2-RTR# Nov 9 17:44:48.613: //141//TCL :/tcl_PutsObjCmd: proc init_perCallvars Nov 9 17:44:48.613: Nov 9 17:44:48.617: //141//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing Welcome Prompt and options menu ++ Pod25-BR2-RTR# Pod25-BR2-RTR#debug voice dialpeer inout voip dialpeer inout debugging is on Pod25-BR2-RTR# Pod25-BR2-RTR# Pod25-BR2-RTR# Pod25-BR2-RTR# Nov 9 17:41:37.788: //-1/0095BC421000/DPM/dpAssociateIncomingPeerCore: Calling Number=2122251003, Called Number=1#3300, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Nov 9 17:41:37.788: //-1/0095BC421000/DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=2 Nov 9 17:41:37.788: //-1/0095BC421000/DPM/dpAssociateIncomingPeerCore: Calling
Re: [OSL | CCIE_Voice] IPCC custom prompt
ops, you have to upload the Prompt w/ Prompt Management Menu. Sergio. From: [EMAIL PROTECTED] To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com Date: Tue, 11 Nov 2008 02:25:53 + Subject: Re: [OSL | CCIE_Voice] IPCC custom prompt James You have to upload the script with Script management menu Sergio- Mensagem Original - De: James Jung [EMAIL PROTECTED] Enviada: segunda-feira, 10 de novembro de 2008 20:27 Para: ccie_voice@onlinestudylist.com Assunto: [OSL | CCIE_Voice] IPCC custom prompt Hi all I want to use custom prompt which is recorded and formatted correctly. When I put it in the system prompt folder, I can use it. But I cannot invoke the prompt if it is in a different folder(for example c:\prompt\). What is the proper script command for this using P[]? JamesJ. _ Receba GRÁTIS as mensagens do Messenger no seu celular quando você estiver offline. Conheça o MSN Mobile! http://mobile.live.com/signup/signup2.aspx?lc=pt-br
[OSL | CCIE_Voice] PSTN switch
I would like to set up a PSTN switch in my home lab. I have three ISRs that I would like phones in each branch to be able to call any other branch via T1 PRI. Currently, each of my ISRs have a VWIC-1MFT-T1, and would like to have the cards terminate to VWIC-1MFT-T1 cards on the PSTN switch. Can someone recommend a low-cost solution for the PSTN switch? The gatekeeper is also part of the PSTN network as well, so it sounds like I would need a PSTN switch with 4 VWIC-1MFT-T1 cards; 1 for each branch, HQ, and gatekeeper router. Can someone recommend such a router that can be used as a PSTN switch in this manner? Can a 2610 be used? I've found the inexpensive 2610 to be very handy as a frame relay switch (and also very inexpensive). -- Chris Kagadis
[OSL | CCIE_Voice] Limiting number of international calls on CME...
Question... Let's say we asked to limit the number of international calls from CME to 2 calls maximum at any time. If we have the only one international dial-peer, it's simple - just use max-conn on the dial-peer. But what if we have multiple dial-peers, which eventually falls under the definition international? For example, TEHO dial-peer to HQ considered to be international from the SiteC perspective. Or PSTN backup to 4-digits interoffice calls will dial international number... What do we do? Theoretically we can have the only one international dial-peer, and simulate/force all other dial-peers with voice translations... But to me it looks very complicated... Will it work at all?
Re: [OSL | CCIE_Voice] PSTN switch
Chris, u can build up your pstn switch with 2610 with 1xNM-HD-2VE and 2xVIC-2MFT-T1 cards. Setup is very simple and just building up touting number plan by bunch of dial peers. it's possible by either ds0-group or PRI setup. On Tue, Nov 11, 2008 at 1:41 PM, Chris Kagadis (kagadis.com) [EMAIL PROTECTED] wrote: I would like to set up a PSTN switch in my home lab. I have three ISRs that I would like phones in each branch to be able to call any other branch via T1 PRI. Currently, each of my ISRs have a VWIC-1MFT-T1, and would like to have the cards terminate to VWIC-1MFT-T1 cards on the PSTN switch. Can someone recommend a low-cost solution for the PSTN switch? The gatekeeper is also part of the PSTN network as well, so it sounds like I would need a PSTN switch with 4 VWIC-1MFT-T1 cards; 1 for each branch, HQ, and gatekeeper router. Can someone recommend such a router that can be used as a PSTN switch in this manner? Can a 2610 be used? I've found the inexpensive 2610 to be very handy as a frame relay switch (and also very inexpensive). -- Chris Kagadis -- Sirus Moghadasian CCIE #21862 (RS)
Re: [OSL | CCIE_Voice] Limiting number of international calls on CME...
Michael, following setup will work, however it's rather complicated solution. intl1 dial-peer | | intl1 dial-peer | ---translation Rules intermediate dial-peer ---translation Rules | intl2 dial-peer | | intl2 dial-peer Then u can apply max-connection under intermediate dial-peer. HTH, On Tue, Nov 11, 2008 at 1:53 PM, Michael Shavrov [EMAIL PROTECTED]wrote: Question... Let's say we asked to limit the number of international calls from CME to 2 calls maximum at any time. If we have the only one international dial-peer, it's simple - just use max-conn on the dial-peer. But what if we have multiple dial-peers, which eventually falls under the definition international? For example, TEHO dial-peer to HQ considered to be international from the SiteC perspective. Or PSTN backup to 4-digits interoffice calls will dial international number... What do we do? Theoretically we can have the only one international dial-peer, and simulate/force all other dial-peers with voice translations... But to me it looks very complicated... Will it work at all? -- Sirus Moghadasian CCIE #21862 (RS)
Re: [OSL | CCIE_Voice] Limiting number of international calls on CME...
I forgot to mention that, u should provide some sort of loop prevention mechanism to prevent loop between intl1 and intl2 dialpeers ( Input and output on following scenario) If I were u, I will manipulate calling number in first translation rule and then try to match that on last intl1 and intl2 dialpeers (they should be different than former intl1 and intl2) It may be simpler ways to accomplish that. let me know if u find one. HTH, On Tue, Nov 11, 2008 at 1:53 PM, Michael Shavrov [EMAIL PROTECTED]wrote: Question... Let's say we asked to limit the number of international calls from CME to 2 calls maximum at any time. If we have the only one international dial-peer, it's simple - just use max-conn on the dial-peer. But what if we have multiple dial-peers, which eventually falls under the definition international? For example, TEHO dial-peer to HQ considered to be international from the SiteC perspective. Or PSTN backup to 4-digits interoffice calls will dial international number... What do we do? Theoretically we can have the only one international dial-peer, and simulate/force all other dial-peers with voice translations... But to me it looks very complicated... Will it work at all? -- Sirus Moghadasian CCIE #21862 (RS)
Re: [OSL | CCIE_Voice] BACD Script Failing Across GK - 3rd Day in a Row !
Hi Vik, Yes I have set the DP/Region for G711 and the incoming DP on CME for G711 and the call still fails. Also when the DP/Region is set for G729 with the incoming DP on CME set for G729 I have cleaned up the dspfarm profile to only use g711u, g711a and g729r8 codecs. The call still fails (5 seconds of silence followed by fast busy). It seems that anytime the call hits 2 incoming dialpeers on the CME the call fails. If it hits only 1 incoming dialpeer the call works. CM (G729) IN DP (G729) -- OUT DP (G711) --- Loop0 -- IN DP (G711) -- serv. aa [FAILS] CM (G711) IN DP (G711) -- OUT DP (G711) --- Loop0 -- IN DP (G711) -- serv. aa [FAILS] CM (G711) IN DP (G711) -- serv. aa [SUCCESS] - CME and PSTN phones can call the BACD script. - CUE is reachable from all phones (HQ,BR1,CME,PSTN) So it seems it fails only when 2 incoming dial-peers are hit. Therefore it seems that there needs to be a command under the ipipgw configurations such as allow-connections h323 to h323 to h323. Not sure. Has anyone got this to work when hitting 2 incoming dial-peers ? Thanks, Mike Brooks CCIE#16027 (RS) On Mon, Nov 10, 2008 at 8:35 PM, Vik Malhi [EMAIL PROTECTED] wrote: By cleaning up the codecs- only have g711u, g711a and g729r8. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Vik Malhi [EMAIL PROTECTED] Date: Mon, 10 Nov 2008 17:28:47 -0800 To: Mike Brooks [EMAIL PROTECTED], Norma Exel [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Script Failing Across GK - 3rd Day in a Row ! Please confirm that this is not the transcoder. Set the DP on CCM to use g711. Set the inbound voip dial-peer to use g711. Confirm the results of this test. If the call works try cleaning up the codecs in your dspfarm profile. -- Vik Malhi CCIE #13890, CCSI #31584 Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.810.454.0130 Mailto: [EMAIL PROTECTED] 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: Mike Brooks [EMAIL PROTECTED] Date: Mon, 10 Nov 2008 20:08:01 -0500 To: Norma Exel [EMAIL PROTECTED] Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] BACD Script Failing Across GK - 3rd Day in a Row ! This the 3rd day in a row that I have had this problem. It seems to be a bug when using the ipipgw functionality of the CME. It happens regardless of codec. Has anyone been able to call from HQ or BR1 to the BACD script with G729 across the WAN ? Thx, Mike Brooks CCIE# 16027 (RS) On Sun, Nov 9, 2008 at 11:53 PM, Norma Exel [EMAIL PROTECTED] wrote: so incoming leg to dial-peer 2 to outgoing leg dial-peer 3301 to incoming leg dial-peer 3301 does not work but... incoming call leg dial-peer 3301 works? (experiment 2) I don't think it's a codec thing either. maybe a code thing. do you need multiple call legs? especially since what you're trying to do sounds like IPIPGW though it seems to work since debug shows it to be the case. curious to see what you find out. - Original Message - From: Mike Brooks [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] BACD Script Failing Across GK Date: Sun, 9 Nov 2008 13:38:52 -0500 I am having issues when calling across the GK from BR1 and HQ to the BACD script (g729). I can hit the BACD script from the PSTN and the CME phones. I also can call into CUE from BR1/HQ. My incoming dial-peer is dial-peer 2 which is stripping the tech-prefix of 1# and then hitting my outgoing dial-peer 3301 which sends it to the loopback. Then it hits the same incoming dial-peer 3301 which has the service aa command. All I see on the phone is call proceed for about 5 seconds and then a fast busy. I have had this problem 2 days in a row. So I must be doing something wrong. I have tried the following: 1. allowed the codec to be G711 across the GK (modified trunk region and dial-peer 2) - call fails. 2. removed the tech prefix 1# from the call and allowed it to hit a different incoming dial-peer on CME (DP 3301) with G711 - call works. * Also my calls to CUE from HQ work fine and do
[OSL | CCIE_Voice] ip phones from two different ccm cluster register to same h323 srst gateway
Hi! Not sure whether any voice experts in this forum can help me on this problem. Due to some special requirements, we have a remote branch with two groups of ip phones, each group registering to different call-manager cluster. There is only one H323 (not MGCP) SRST gateway in this branch. My question is when SRST kicks in, will these two group of ip phones able to register to the same H323 srst gateway without any issues? Assume that there is no overlapping DNs for these two group of phones. Thanks! Best regards Goh
Re: [OSL | CCIE_Voice] ip phones from two different ccm cluster register to same h323 srst gateway
Sekchye, u can set SRST reference of device pools on each CCM cluster to ip source add of the SRST/CME router. They will register with GW. u can verify process with debug ephone register. HTH, On Tue, Nov 11, 2008 at 2:30 PM, sekchye goh [EMAIL PROTECTED] wrote: Hi! Not sure whether any voice experts in this forum can help me on this problem. Due to some special requirements, we have a remote branch with two groups of ip phones, each group registering to different call-manager cluster. There is only one H323 (not MGCP) SRST gateway in this branch. My question is when SRST kicks in, will these two group of ip phones able to register to the same H323 srst gateway without any issues? Assume that there is no overlapping DNs for these two group of phones. Thanks! Best regards Goh -- Sirus Moghadasian CCIE #21862 (RS)
Re: [OSL | CCIE_Voice] PSTN switch
Chris, If u can ,use 256 mem on your 2610, (ito upgrade to 256 ram, should upgarde to bootstarp ver 12.2.8r ), I use 128 mem and version is 12.4(17) I use same router for GK function. HTH On Tue, Nov 11, 2008 at 3:00 PM, Chris Kagadis (kagadis.com) [EMAIL PROTECTED] wrote: Thanks, Cyrus. Can you tell me how much mem you use in yours and which IOS you are using? Also, what are you using for a gatekeeper? On Mon, Nov 10, 2008 at 7:59 PM, Cyrus [EMAIL PROTECTED] wrote: Chris, u can build up your pstn switch with 2610 with 1xNM-HD-2VE and 2xVIC-2MFT-T1 cards. Setup is very simple and just building up touting number plan by bunch of dial peers. it's possible by either ds0-group or PRI setup. On Tue, Nov 11, 2008 at 1:41 PM, Chris Kagadis (kagadis.com) [EMAIL PROTECTED] wrote: I would like to set up a PSTN switch in my home lab. I have three ISRs that I would like phones in each branch to be able to call any other branch via T1 PRI. Currently, each of my ISRs have a VWIC-1MFT-T1, and would like to have the cards terminate to VWIC-1MFT-T1 cards on the PSTN switch. Can someone recommend a low-cost solution for the PSTN switch? The gatekeeper is also part of the PSTN network as well, so it sounds like I would need a PSTN switch with 4 VWIC-1MFT-T1 cards; 1 for each branch, HQ, and gatekeeper router. Can someone recommend such a router that can be used as a PSTN switch in this manner? Can a 2610 be used? I've found the inexpensive 2610 to be very handy as a frame relay switch (and also very inexpensive). -- Chris Kagadis -- Sirus Moghadasian CCIE #21862 (RS) -- Chris Kagadis -- Sirus Moghadasian CCIE #21862 (RS)
[OSL | CCIE_Voice] NTP IN CUCM
Hello Team, Is there any good procedure to setup NTP in CUCM because I tried the one where you change the ntp.conf file and restart the NTP service but doesn't seem to work for me. This is what my ntp.conf file look like: server 142.707.64.254 Thanks
[OSL | CCIE_Voice] VPIM, Delivery Location issue
Hello Team, Question on VPIM: I have an issue when I go to create a Delivery Location on Unity for CUE it won't add it. If I create any other profile using other than VPIM, such as SMTP it works. I have already check my license and it does allow VPIM Pardeep
