Because I don't care what ya trunk to me with (Avaya, Nortel, Cisco, Asterisk, etc), I tend to care less about the "end" user's equipment as I'm sure many here have had to do from time to time. (Interops with Avaya one day, Nortel the next, intertwined with CCM here, Asterisk there). Anyhow, was doing an interop with someone and they CUCM was sending 491's. Figured I'd ask here since I believe someone must have a "oh yea... that stupid thing..." response. The scenario: Caller (Cisco Equipment) --> Their Network --> The Interweb --> My network --> so on... Caller --> My phone --> I pick up --> Their phone disconnects, immediate CANCEL 491 SIP? Caller --> CCM --> Router --> My SBC --> SIP/200 OK At that EXACT MOMENT of the SIP/200 OK: CCM sees SIP/481 : *Aug 10 20:38:45.295: //238407/000000000000/SIP/Call/sipSPICallInfo: Disconnect Cause (CC) : 16 Disconnect Cause (SIP) : 481 Anyone have a generic Cisco router config they'd care to share? I was fortunate enough to have them send me their end (implemented VoIP related garbage): voice call send-alert voice call carrier capacity active voice rtp send-recv voice service voip allow-connections sip to sip h323 sip bind control source-interface GigabitEthernet0/0.3 bind media source-interface GigabitEthernet0/0.3 voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw voice class h323 1 h225 timeout tcp establish 3 ccm-manager fallback-mgcp ccm-manager redundant-host xx.yy.z.15 ccm-manager mgcp ccm-manager music-on-hold ccm-manager config server xx.yy.z.16 ccm-manager config fax interface-type fax-mail interface GigabitEthernet0/0.3 description Voice VLAN encapsulation dot1Q 3 ip address xx.yy.z.1 255.255.255.0 ip helper-address vv.vv.v.v4 ip pim sparse-dense-mode ip cgmp no cdp enable dial-peer voice 9000 voip description ** Outbound 10-digit ** preference 1 destination-pattern 1[2-9]......... session protocol sipv2 session target ipv4:this.is.me.at.33 session transport udp fax-relay ecm disable no vad sip-ua call-manager-fallback max-conferences 8 ip source-address xx.yy.z.1 port 2000 max-ephones 48 max-dn 48 transfer-pattern 503....... default-destination 2000 voicemail xxxxxxxxxx SIP Messages: (ala ngrep ;)) U this.is.my.ip.33:5060 -> this.is.their.ip.131:5060 SIP/2.0 200 OK v: SIP/2.0/UDP this.is.their.ip.131:5060;branch=z9hG4bK1E3E f: "SIP Test DID"<sip:1503xxxyyyy at this.is.their.ip.131>;tag=2633AAF8-1A7C t: <sip:1212aaabbbb at this.is.my.ip.33>;tag=4483814b840cda0de9102ed591a599fc i: 15E4B9A3-A3F611DF-91EEFF90-66FF34BF at this.is.their.ip.131 CSeq: 101 INVITE m: <sip:+1212aaabbbb at this.is.my.ip.33:5060;transport=udp> c: application/sdp l: 111 v=0 o=- 58501 5850100 IN IP4 this.is.my.ip.38 s=- c=IN IP4 this.is.my.ip.38 t=0 0 m=audio 18168 RTP/AVP 0 18 U this.is.their.ip.131:57731 -> this.is.my.ip.33:5060 CANCEL sip:1212aaabbbb at this.is.my.ip.33:5060 SIP/2.0 Via: SIP/2.0/UDP this.is.their.ip.131:5060;branch=z9hG4bK1E3E From: "SIP Test DID" <sip:1503xxxyyyy at this.is.their.ip.131>;tag=2633AAF8-1A7C To: <sip:1212aaabbbb at this.is.my.ip.33>;tag=4483814b840cda0de9102ed591a599fc Date: Tue, 10 Aug 2010 20:38:36 GMT Call-ID: 15E4B9A3-A3F611DF-91EEFF90-66FF34BF at this.is.their.ip.131 CSeq: 101 CANCEL Max-Forwards: 70 Timestamp: 1281472724 Reason: Q.850;cause=16 Content-Length: 0.. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT "It takes 20 years to build a reputation and five minutes to ruin it. If you think about that, you'll do things differently." - Warren Buffett 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E
I'd normally want to see the call logic debug from the cisco side, but the important thing I see is the cause code 16, indicating normal call clearing. This suggests the local router decided to end the call before the 200 OK was received/ACKed, probably based on some user action or a timeout (however I'd expect a timeout to produce a different cause code). So to answer the question you need to know what the router is thinking. -Scott -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of J. Oquendo Sent: Tuesday, August 10, 2010 5:05 PM To: voiceops at voiceops.org Subject: [VoiceOps] Ye ole Cisco Cube CM Because I don't care what ya trunk to me with (Avaya, Nortel, Cisco, Asterisk, etc), I tend to care less about the "end" user's equipment as I'm sure many here have had to do from time to time. (Interops with Avaya one day, Nortel the next, intertwined with CCM here, Asterisk there). Anyhow, was doing an interop with someone and they CUCM was sending 491's. Figured I'd ask here since I believe someone must have a "oh yea... that stupid thing..." response. The scenario: Caller (Cisco Equipment) --> Their Network --> The Interweb --> My network --> so on... Caller --> My phone --> I pick up --> Their phone disconnects, immediate CANCEL 491 SIP? Caller --> CCM --> Router --> My SBC --> SIP/200 OK At that EXACT MOMENT of the SIP/200 OK: CCM sees SIP/481 : *Aug 10 20:38:45.295: //238407/000000000000/SIP/Call/sipSPICallInfo: Disconnect Cause (CC) : 16 Disconnect Cause (SIP) : 481 Anyone have a generic Cisco router config they'd care to share? I was fortunate enough to have them send me their end (implemented VoIP related garbage): voice call send-alert voice call carrier capacity active voice rtp send-recv voice service voip allow-connections sip to sip h323 sip bind control source-interface GigabitEthernet0/0.3 bind media source-interface GigabitEthernet0/0.3 voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw voice class h323 1 h225 timeout tcp establish 3 ccm-manager fallback-mgcp ccm-manager redundant-host xx.yy.z.15 ccm-manager mgcp ccm-manager music-on-hold ccm-manager config server xx.yy.z.16 ccm-manager config fax interface-type fax-mail interface GigabitEthernet0/0.3 description Voice VLAN encapsulation dot1Q 3 ip address xx.yy.z.1 255.255.255.0 ip helper-address vv.vv.v.v4 ip pim sparse-dense-mode ip cgmp no cdp enable dial-peer voice 9000 voip description ** Outbound 10-digit ** preference 1 destination-pattern 1[2-9]......... session protocol sipv2 session target ipv4:this.is.me.at.33 session transport udp fax-relay ecm disable no vad sip-ua call-manager-fallback max-conferences 8 ip source-address xx.yy.z.1 port 2000 max-ephones 48 max-dn 48 transfer-pattern 503....... default-destination 2000 voicemail xxxxxxxxxx SIP Messages: (ala ngrep ;)) U this.is.my.ip.33:5060 -> this.is.their.ip.131:5060 SIP/2.0 200 OK v: SIP/2.0/UDP this.is.their.ip.131:5060;branch=z9hG4bK1E3E f: "SIP Test DID"<sip:1503xxxyyyy at this.is.their.ip.131>;tag=2633AAF8-1A7C t: <sip:1212aaabbbb at this.is.my.ip.33>;tag=4483814b840cda0de9102ed591a599fc i: 15E4B9A3-A3F611DF-91EEFF90-66FF34BF at this.is.their.ip.131 CSeq: 101 INVITE m: <sip:+1212aaabbbb at this.is.my.ip.33:5060;transport=udp> c: application/sdp l: 111 v=0 o=- 58501 5850100 IN IP4 this.is.my.ip.38 s=- c=IN IP4 this.is.my.ip.38 t=0 0 m=audio 18168 RTP/AVP 0 18 U this.is.their.ip.131:57731 -> this.is.my.ip.33:5060 CANCEL sip:1212aaabbbb at this.is.my.ip.33:5060 SIP/2.0 Via: SIP/2.0/UDP this.is.their.ip.131:5060;branch=z9hG4bK1E3E From: "SIP Test DID" <sip:1503xxxyyyy at this.is.their.ip.131>;tag=2633AAF8-1A7C To: <sip:1212aaabbbb at this.is.my.ip.33>;tag=4483814b840cda0de9102ed591a599fc Date: Tue, 10 Aug 2010 20:38:36 GMT Call-ID: 15E4B9A3-A3F611DF-91EEFF90-66FF34BF at this.is.their.ip.131 CSeq: 101 CANCEL Max-Forwards: 70 Timestamp: 1281472724 Reason: Q.850;cause=16 Content-Length: 0.. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT "It takes 20 years to build a reputation and five minutes to ruin it. If you think about that, you'll do things differently." - Warren Buffett 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
On 08/10/2010 05:05 PM, J. Oquendo wrote:
voice service voip allow-connections sip to sip
I'm curious: which specific devices and IOSs allow hairpinning of VoIP calls? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/
Cisco 2800, 3800, 2900, 3900, 2431/2432, 2435, AS53/4xx series, 7200, ASR 100x, are the ones that come to mind. On the non-IOS XE systems, CUBE worked in 12.4(22) and above acceptably well, but it's much better with 15.1T (specifically 15.1.2T where transcoding on SIP to SIP is supported using PVDMs). -- Jason Nesheim ----- Original Message ----- From: "Alex Balashov" <abalashov at evaristesys.com> To: voiceops at voiceops.org Sent: Tuesday, August 10, 2010 6:00:35 PM Subject: Re: [VoiceOps] Ye ole Cisco Cube CM On 08/10/2010 05:05 PM, J. Oquendo wrote:
voice service voip allow-connections sip to sip
I'm curious: which specific devices and IOSs allow hairpinning of VoIP calls? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
I had a similar challenge a while back and I solved it with asterisk running as a vm. I tried terminating the sip trunk on an mgcp router and was only getting signalling passing. I then moved the router to h323 and had one way issues. I realised having spoken to cisco that I need to get cube license for each anticipated b2bua call. A little more experimenting and I came up with a plan to move the sip trunk direct to Call manager. CUCM cant do user agent based sip trunks however as they only allow ip based sip trunks. This is how asterisks came in to the picture as now it have trunks set to my itsp from asterisk and then from asterisk to CUCM . Two years and counting and this solution has been working without skipping a beat. On Fri, Aug 13, 2010 at 6:13 AM, Jason L. Nesheim <jnesheim at cytek.biz>wrote:
Cisco 2800, 3800, 2900, 3900, 2431/2432, 2435, AS53/4xx series, 7200, ASR 100x, are the ones that come to mind. On the non-IOS XE systems, CUBE worked in 12.4(22) and above acceptably well, but it's much better with 15.1T (specifically 15.1.2T where transcoding on SIP to SIP is supported using PVDMs).
-- Jason Nesheim
----- Original Message ----- From: "Alex Balashov" <abalashov at evaristesys.com> To: voiceops at voiceops.org Sent: Tuesday, August 10, 2010 6:00:35 PM Subject: Re: [VoiceOps] Ye ole Cisco Cube CM
On 08/10/2010 05:05 PM, J. Oquendo wrote:
voice service voip allow-connections sip to sip
I'm curious: which specific devices and IOSs allow hairpinning of VoIP calls?
-- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
-- Andrew Dixon ARTJ Consultants Bds Phone:(246) 422-6788 US Phone: (954) 342-5665 UK Phone: +44 (0) 20 3298 1043 Fax: (206) 424-9037 http://www.artjconsultants.com
I don't quite follow you... what's on the wire in the sip packet, 481 or 491? Do you have ngrep for the whole mess, including the 4xx responses? David Hiers CCIE (R/S, V), CISSP ADP Dealer Services 2525 SW 1st Ave. Suite 300W Portland, OR 97201 o: 503-205-4467 f: 503-402-3277 -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of J. Oquendo Sent: Tuesday, August 10, 2010 2:05 PM To: voiceops at voiceops.org Subject: [VoiceOps] Ye ole Cisco Cube CM Because I don't care what ya trunk to me with (Avaya, Nortel, Cisco, Asterisk, etc), I tend to care less about the "end" user's equipment as I'm sure many here have had to do from time to time. (Interops with Avaya one day, Nortel the next, intertwined with CCM here, Asterisk there). Anyhow, was doing an interop with someone and they CUCM was sending 491's. Figured I'd ask here since I believe someone must have a "oh yea... that stupid thing..." response. The scenario: Caller (Cisco Equipment) --> Their Network --> The Interweb --> My network --> so on... Caller --> My phone --> I pick up --> Their phone disconnects, immediate CANCEL 491 SIP? Caller --> CCM --> Router --> My SBC --> SIP/200 OK At that EXACT MOMENT of the SIP/200 OK: CCM sees SIP/481 : *Aug 10 20:38:45.295: //238407/000000000000/SIP/Call/sipSPICallInfo: Disconnect Cause (CC) : 16 Disconnect Cause (SIP) : 481 Anyone have a generic Cisco router config they'd care to share? I was fortunate enough to have them send me their end (implemented VoIP related garbage): voice call send-alert voice call carrier capacity active voice rtp send-recv voice service voip allow-connections sip to sip h323 sip bind control source-interface GigabitEthernet0/0.3 bind media source-interface GigabitEthernet0/0.3 voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw voice class h323 1 h225 timeout tcp establish 3 ccm-manager fallback-mgcp ccm-manager redundant-host xx.yy.z.15 ccm-manager mgcp ccm-manager music-on-hold ccm-manager config server xx.yy.z.16 ccm-manager config fax interface-type fax-mail interface GigabitEthernet0/0.3 description Voice VLAN encapsulation dot1Q 3 ip address xx.yy.z.1 255.255.255.0 ip helper-address vv.vv.v.v4 ip pim sparse-dense-mode ip cgmp no cdp enable dial-peer voice 9000 voip description ** Outbound 10-digit ** preference 1 destination-pattern 1[2-9]......... session protocol sipv2 session target ipv4:this.is.me.at.33 session transport udp fax-relay ecm disable no vad sip-ua call-manager-fallback max-conferences 8 ip source-address xx.yy.z.1 port 2000 max-ephones 48 max-dn 48 transfer-pattern 503....... default-destination 2000 voicemail xxxxxxxxxx SIP Messages: (ala ngrep ;)) U this.is.my.ip.33:5060 -> this.is.their.ip.131:5060 SIP/2.0 200 OK v: SIP/2.0/UDP this.is.their.ip.131:5060;branch=z9hG4bK1E3E f: "SIP Test DID"<sip:1503xxxyyyy at this.is.their.ip.131>;tag=2633AAF8-1A7C t: <sip:1212aaabbbb at this.is.my.ip.33>;tag=4483814b840cda0de9102ed591a599fc i: 15E4B9A3-A3F611DF-91EEFF90-66FF34BF at this.is.their.ip.131 CSeq: 101 INVITE m: <sip:+1212aaabbbb at this.is.my.ip.33:5060;transport=udp> c: application/sdp l: 111 v=0 o=- 58501 5850100 IN IP4 this.is.my.ip.38 s=- c=IN IP4 this.is.my.ip.38 t=0 0 m=audio 18168 RTP/AVP 0 18 U this.is.their.ip.131:57731 -> this.is.my.ip.33:5060 CANCEL sip:1212aaabbbb at this.is.my.ip.33:5060 SIP/2.0 Via: SIP/2.0/UDP this.is.their.ip.131:5060;branch=z9hG4bK1E3E From: "SIP Test DID" <sip:1503xxxyyyy at this.is.their.ip.131>;tag=2633AAF8-1A7C To: <sip:1212aaabbbb at this.is.my.ip.33>;tag=4483814b840cda0de9102ed591a599fc Date: Tue, 10 Aug 2010 20:38:36 GMT Call-ID: 15E4B9A3-A3F611DF-91EEFF90-66FF34BF at this.is.their.ip.131 CSeq: 101 CANCEL Max-Forwards: 70 Timestamp: 1281472724 Reason: Q.850;cause=16 Content-Length: 0.. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT "It takes 20 years to build a reputation and five minutes to ruin it. If you think about that, you'll do things differently." - Warren Buffett 227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x5CCD6B5E _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
participants (6)
-
abalashov＠evaristesys.com -
adixon＠artjconsultants.com -
David_Hiers＠adp.com -
jnesheim＠cytek.biz -
scott＠sberkman.net -
sil＠infiltrated.net