[cisco-voip] Columbia CUBE to Claro outbound issue
Hank Keleher (AM)
hank.keleher at dimensiondata.com
Sun Apr 3 21:15:07 EDT 2016
Oh, I have. Several times. No real response. They don't speak English so I'm not sure if there is a loss in translation or not (using a translator who is semi-technical.)
Thanks!
Hank
On Apr 3, 2016, at 21:09, Ryan Huff <ryanhuff at outlook.com<mailto:ryanhuff at outlook.com>> wrote:
Hank,
I dealt with something similar about 4 months ago (different carrier) and, is in part, why I asked if you got the 404 response with every attempt. As you likely know, the 404 is probably the least ambiguous response you can get -very clear what it means.
In your statement, the carrier wants 91+7d and when you do that, you get a 404. So with this you are able to answer the what, you just need the carrier to answer the why since the messaging clearly shows the CPE signaling makes it to PE and the PE that cannot route back.
I would generate one of the failed inbound calls and then ask the carrier for the debugging voice sipmsg output trace of that call off their Huawei equipment (or for them to evaluate it and provide the analysis). This should show exactly what is happening on the PE for the call.
= Ryan =
Email: ryanhuff at outlook.com<mailto:ryanhuff at outlook.com>
Spark: ryanhuff at outlook.com<mailto:ryanhuff at outlook.com>
Twitter: @ryanthomashuff<http://twitter.com/ryanthomashuff>
LinkedIn: ryanthomashuff<http://linkedin.com/in/ryanthomashuff>
Web ryanthomashuff.com<http://ryanthomashuff.com>
________________________________
From: cisco-voip <cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net>> on behalf of Hank Keleher (AM) <hank.keleher at dimensiondata.com<mailto:hank.keleher at dimensiondata.com>>
Sent: Sunday, April 3, 2016 8:35 PM
To: Daniel Pagan; Horton, Jamin
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Columbia CUBE to Claro outbound issue
Yeah, I have. Everything points to the issue being on their side but of course they say everything is okay as long as we’re doing this, that and the other. Which we are. As well as tried many other configuration changes to find a way to make it work.
At some point hopefully the light goes off and they either fix their issue or tell us more specifically what we need to do. My hope was that someone might offer a thought on something we haven’t tried or have familiarity with this service provider to help point us in a better direction.
Thanks!
Hank
From: Daniel Pagan <dpagan at fidelus.com<mailto:dpagan at fidelus.com>>
Date: Sunday, April 3, 2016 at 10:52
To: "Hank.Keleher" <hank.keleher at us.didata.com<mailto:hank.keleher at us.didata.com>>, "Horton, Jamin" <jamin.horton at oneneck.com<mailto:jamin.horton at oneneck.com>>
Cc: "cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>" <cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>>
Subject: RE: [cisco-voip] Columbia CUBE to Claro outbound issue
Personally, I would push back to them and simply provide them with the request URI your transmitting, which they should be using for routing purposes.
INVITE sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>:5060 SIP/2.0
I say push back because a 404 is a very straight forward response and sent for one reason. If their equipment can’t route to 1343111 then I would expect this behavior, but they should be able to tell you why it’s not being routed based on that request URI, regardless if it’s a fault in your configuration or theirs. They’re definitely receiving the INVITE request because we see provisional responses with the same Call-ID header and From Tag, and then acting upon it because we see the Reason header with 404, so any experienced technician at their company should be able to tell you the reason for the 183 with 404 cause in Reason header.
As for the 487 final you see below, that’s expected as well considering the call flow. The standard signaling flow for this scenario is…
---->> INVITE w/SDP offer
<<---- 100
<<---- 183 w/SDP answer
[[ You now have bi-directional media]]
[[ Likely hearing failure recording ]]
====================
** Calling party hangs up **
====================
---->> CANCEL [[ The CANCEL requests terminates pending SIP requests. ]]
<<--- 200 OK [[ Acceptance of CANCEL through 200 final response ]]
<<--- 487 [[ The 487 is used to terminate the INVITE – all SIP requests require a final response – here’s the final response for the INVITE ]]
---->> ACK [[ You send ACK since this transaction was for an INVITE – no other requests require to be ACKed ]]
I personally never worked with Claro, but SIP is SIP (for the most part), and they should be sending you a 404 specifically because they 1) received the INVITE, 2) processed the user portion (1343111) of the request URI, and 3) could not find a route to the destination.
Hope you find this helpful.
- Dan
From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Hank Keleher (AM)
Sent: Saturday, April 02, 2016 11:10 PM
To: Horton, Jamin <jamin.horton at oneneck.com<mailto:jamin.horton at oneneck.com>>
Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] Columbia CUBE to Claro outbound issue
I tried hard coding both UDP and TCP. Setting it to UDP produced the same result where as TCP had the same CC disconnect of 53 and SIP disconnect cause was 503 (200 with UDP.)
Nice suggestion though!
Thanks!
Hank
From: "Horton, Jamin" <jamin.horton at oneneck.com<mailto:jamin.horton at oneneck.com>>
Date: Saturday, April 2, 2016 at 22:11
To: "Hank.Keleher" <hank.keleher at us.didata.com<mailto:hank.keleher at us.didata.com>>
Cc: Ryan Huff <ryanhuff at outlook.com<mailto:ryanhuff at outlook.com>>, "cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>" <cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>>
Subject: Re: [cisco-voip] Columbia CUBE to Claro outbound issue
I just had this same issue yesterday. 2 hours later after banging my head against they keyboard I figured out it was my session transport protocol. They were expecting UDP and I was using TCP.
Try playing around with it. I believe you can do it under "voice service / sip" as well as the individual dial peer.
Hope it helps!
Sent from my iPhone forgive any typos
On Apr 2, 2016, at 11:43 AM, Hank Keleher (AM) <hank.keleher at dimensiondata.com<mailto:hank.keleher at dimensiondata.com>> wrote:
I remember seeing that and I changed the outbound local calls to 9+7 digits instead of the 9+1+7 digits that Claro says we should use. When I dial 9+7 digits I don’t get the 404 error but I do when dialing 9+1+7 digits as well as 9+00+any for international (that’s all I tested so far.)
All calls still get CC disconnect cause of 38.
Inbound works fine.
Thanks!
Hank
From: Ryan Huff <ryanhuff at outlook.com<mailto:ryanhuff at outlook.com>>
Date: Saturday, April 2, 2016 at 13:30
To: "Hank.Keleher" <hank.keleher at us.didata.com<mailto:hank.keleher at us.didata.com>>
Cc: "cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>" <cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>>
Subject: Re: [cisco-voip] Columbia CUBE to Claro outbound issue
Are you always getting the same 404 "unallocated number" response?
Thanks,
Ryan
> On Apr 2, 2016, at 1:05 PM, Hank Keleher (AM) <hank.keleher at dimensiondata.com<mailto:hank.keleher at dimensiondata.com>> wrote:
>
> Greetings!
>
> Does anyone have a CUBE configuration that works with Claro based in Columbia you can share with me? I have inbound working just fine but oubound keeps getting rejected. Below is an example of the config and relevant debug. They are saying that it should work even though I’ve pointed out that we are getting network out of order and request terminated disconnect causes from them. They are using a Huawei switch on their side (which is why we are using payload type 97.)
>
> Thanks!
> Hank
>
> ———————————————
>
> voice service voip
> no ip address trusted authenticate
> dtmf-interworking rtp-nte
> mode border-element
> allow-connections h323 to h323
> allow-connections h323 to sip
> allow-connections sip to h323
> allow-connections sip to sip
> fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
> h323
> sip
> no silent-discard untrusted
> sip-profiles 1
> !
> voice class codec 1
> codec preference 1 g711alaw
> codec preference 2 g729r8
> codec preference 3 g711ulaw
> !
> voice class h323 1
> h225 timeout tcp establish 6
> !
> !
> voice class sip-profiles 1
> request INVITE sip-header From modify "From: (.*) <sip:(.*)@.*>" "From: <sip:13904707 at 10.11.44.10<mailto:13904707 at 10.11.44.10>>”
> !
> voice translation-rule 1
> rule 1 /^9\(.*\)/ /\1/
> !
> voice translation-rule 3
> rule 1 /^\+571390\(.*\)/ /13904707/
> !
> voice translation-profile STRIP9
> translate calling 3
> translate called 1
> !
> dial-peer voice 1 voip
> description Incoming dialpeer for outgoing calls from CCM to PSTN
> session protocol sipv2
> incoming called-number 9.*
> voice-class codec 1
> voice-class sip dtmf-relay force rtp-nte
> voice-class sip bind control source-interface GigabitEthernet0/0
> voice-class sip bind media source-interface GigabitEthernet0/0
> dtmf-relay rtp-nte
> no vad
> !
> dial-peer voice 1003 voip
> description 7-digit Local Calls
> translation-profile outgoing STRIP9
> preference 1
> destination-pattern 91[2-9]......
> rtp payload-type cisco-codec-fax-ack 98
> rtp payload-type nte 97
> session protocol sipv2
> session target ipv4:10.11.0.9
> voice-class codec 1
> voice-class sip dtmf-relay force rtp-nte
> voice-class sip bind control source-interface GigabitEthernet0/1
> voice-class sip bind media source-interface GigabitEthernet0/1
> dtmf-relay rtp-nte
> no vad
>
>
> ———————————————
>
>
> Apr 2 10:49:39.726 COT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> INVITE sip:913431111 at 172.20.177.11<mailto:913431111 at 172.20.177.11>:5060 SIP/2.0
> Via: SIP/2.0/TCP 172.20.148.10:5060;branch=z9hG4bKe2d861fdd64ab
> From: "Hank Keleher" <sip:+5713904722 at 172.20.148.10<mailto:+5713904722 at 172.20.148.10>>;tag=9597216~2991ed22-0c89-4455-87b6-f5426e7b3728-57616955
> To: <sip:913431111 at 172.20.177.11<mailto:913431111 at 172.20.177.11>>
> Date: Sat, 02 Apr 2016 15:51:16 GMT
> Call-ID: ba7e7780-6ff1ea74-8e42b-a9414ac at 172.20.148.10<mailto:ba7e7780-6ff1ea74-8e42b-a9414ac at 172.20.148.10>
> Supported: timer,resource-priority,replaces
> Min-SE: 1800
> User-Agent: Cisco-CUCM10.5
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
> CSeq: 101 INVITE
> Expires: 180
> Allow-Events: presence, kpml
> Supported: X-cisco-srtp-fallback,X-cisco-original-called
> Cisco-Guid: 3128850304-0000065536-0000000176-0177476780
> Session-Expires: 1800
> P-Asserted-Identity: "Hank Keleher" <sip:+5713904722 at 172.20.148.10<mailto:+5713904722 at 172.20.148.10>>
> Remote-Party-ID: "Hank Keleher" <sip:+5713904722 at 172.20.148.10<mailto:+5713904722 at 172.20.148.10>>;party=calling;screen=yes;privacy=off
> Contact: <sip:+5713904722 at 172.20.148.10<mailto:+5713904722 at 172.20.148.10>:5060;transport=tcp>
> Max-Forwards: 69
> Content-Type: application/sdp
> Content-Length: 202
>
> v=0
> o=CiscoSystemsCCM-SIP 9597216 1 IN IP4 172.20.148.10
> s=SIP Call
> c=IN IP4 172.20.177.11
> t=0 0
> m=audio 16438 RTP/AVP 8 97
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 telephone-event/8000
> a=fmtp:97 0-15
>
> Apr 2 10:49:39.738 COT: //35/BA7E77800000/SIP/Msg/ccsipDisplayMsg:
> Sent:
> INVITE sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.11.44.10:5060;branch=z9hG4bK1E22F6
> Remote-Party-ID: "Hank Keleher" <sip:13904707 at 10.11.44.10<mailto:13904707 at 10.11.44.10>>;party=calling;screen=yes;privacy=off
> From: <sip:13904707 at 10.11.44.10<mailto:13904707 at 10.11.44.10>>;tag=14BC38-1B1D
> To: <sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>>
> Date: Sat, 02 Apr 2016 15:49:39 GMT
> Call-ID: 57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10<mailto:57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10>
> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
> Min-SE: 1800
> Cisco-Guid: 3128850304-0000065536-0000000176-0177476780
> User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
> CSeq: 101 INVITE
> Timestamp: 1459612179
> Contact: <sip:13904707 at 10.11.44.10<mailto:13904707 at 10.11.44.10>:5060>
> Expires: 180
> Allow-Events: telephone-event
> Max-Forwards: 68
> Session-Expires: 1800
> Content-Type: application/sdp
> Content-Disposition: session;handling=required
> Content-Length: 241
>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 8255 8391 IN IP4 10.11.44.10
> s=SIP Call
> c=IN IP4 10.11.44.10
> t=0 0
> m=audio 16442 RTP/AVP 8 97
> c=IN IP4 10.11.44.10
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 telephone-event/8000
> a=fmtp:97 0-15
> a=ptime:20
>
> COCGW01#
> Apr 2 10:49:39.746 COT: //35/BA7E77800000/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.11.44.10:5060;branch=z9hG4bK1E22F6
> From: <sip:13904707 at 10.11.44.10<mailto:13904707 at 10.11.44.10>>;tag=14BC38-1B1D
> To: <sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>>
> Call-ID: 57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10<mailto:57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10>
> CSeq: 101 INVITE
> Timestamp: 1459612179
>
>
> Apr 2 10:49:39.902 COT: //35/BA7E77800000/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP 10.11.44.10:5060;branch=z9hG4bK1E22F6
> From: <sip:13904707 at 10.11.44.10<mailto:13904707 at 10.11.44.10>>;tag=14BC38-1B1D
> To: <sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>>;tag=qe7asa39-CC-6
> Call-ID: 57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10<mailto:57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10>
> CSeq: 101 INVITE
> Timestamp: 1459612179
> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,UNSUBSCRIBE,REFER,PUBLISH
> Contact: <sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>:5060;transport=udp>
> Require: 100rel
> RSeq: 1
> Reason: Q.850;cause=1;text="Unallocated number",SIP;cause=404
> P-Early-Media: sendrecv
> Content-Length: 206
> Content-Type: application/sdp
>
> v=0
> o=HuaweiATS9900 200100589 200100589 IN IP4 10.11.0.9
> s=Sip Call
> c=IN IP4 10.11.0.9
> t=0 0
> m=audio 34822 RTP/AVP 8 97
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 telephone-event/8000
> a=sendrecv
> a=ptime:20
>
> Apr 2 10:49:39.906 COT: //35/BA7E77800000/SIP/Msg/ccsipDisplayMsg:
> Sent:
> PRACK sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 10.11.44.10:5060;branch=z9hG4bK1F78F
> From: "Hank Keleher" <sip:13904707 at 172.20.177.11<mailto:13904707 at 172.20.177.11>>;tag=14BC38-1B1D
> To: <sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>>;tag=qe7asa39-CC-6
> Date: Sat, 02 Apr 2016 15:49:39 GMT
> Call-ID: 57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10<mailto:57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10>
> CSeq: 102 PRACK
> RAck: 1 101 INVITE
> Allow-Events: telephone-event
> Max-Forwards: 70
> Content-Length: 0
>
>
> Apr 2 10:49:39.946 COT: //35/BA7E77800000/SIP/Msg/ccsipDisplayMsg:
> Received:
>
> COCGW01#SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.11.44.10:5060;branch=z9hG4bK1F78F
> From: "Hank Keleher" <sip:13904707 at 172.20.177.11<mailto:13904707 at 172.20.177.11>>;tag=14BC38-1B1D
> To: <sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>>;tag=qe7asa39-CC-6
> Call-ID: 57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10<mailto:57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10>
> CSeq: 102 PRACK
> Content-Length: 0
>
>
> COCGW01#
> Apr 2 10:49:44.730 COT: //34/BA7E77800000/SIP/Call/sipSPICallInfo:
> The Call Setup Information is:
> Call Control Block (CCB) : 0x0xC29CACA8
> State of The Call : STATE_DEAD
> TCP Sockets Used : YES
> Calling Number : +5713904722
> Called Number : 913431111
> Source IP Address (Sig ): 10.11.44.10
> Destn SIP Req Addr:Port : 172.20.148.10:5060
> Destn SIP Resp Addr:Port : 172.20.148.10:54242
> Destination Name : 172.20.148.10
>
> Apr 2 10:49:44.730 COT: //34/BA7E77800000/SIP/Call/sipSPIMediaCallInfo:
> Number of Media Streams: 1
> Media Stream : 1
> Negotiated Codec : g711alaw
> Negotiated Codec Bytes : 160
> Nego. Codec payload : 8 (tx), 8 (rx)
> Negotiated Dtmf-relay : 6
> Dtmf-relay Payload : 97 (tx), 97 (rx)
> Source IP Address (Media): 10.11.44.10
> Source IP Port (Media): 16440
> Destn IP Address (Media): 172.20.177.11
> Destn IP Port (Media): 16438
> Orig Destn IP Address:Port (Media): [ - ]:0
>
> Apr 2 10:49:44.730 COT: //34/BA7E77800000/SIP/Call/sipSPICallInfo:
> Disconnect Cause (CC) : 38
> Disconnect Cause (SIP) : 200
>
> Apr 2 10:49:44.730 COT: //35/BA7E77800000/SIP/Msg/ccsipDisplayMsg:
> Sent:
> CANCEL sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.11.44.10:5060;branch=z9hG4bK1E22F6
> From: "Hank Keleher" <sip:13904707 at 172.20.177.11<mailto:13904707 at 172.20.177.11>>;tag=14BC38-1B1D
> To: <sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>>
> Date: Sat, 02 Apr 2016 15:49:39 GMT
> Call-ID: 57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10<mailto:57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10>
> CSeq: 101 CANCEL
> Max-Forwards: 70
> Timestamp: 1459612184
> Reason: Q.850;cause=38
> Content-Length: 0
>
>
> Apr 2 10:49:44.738 COT: //35/BA7E77800000/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.11.44.10:5060;branch=z9hG4bK1E22F6
> From: "Hank Keleher" <sip:13904707 at 172.20.177.11<mailto:13904707 at 172.20.177.11>>;tag=14BC38-1B1D
> To: <sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>>;tag=qe7asa39-CC-6
> Call-ID: 57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10<mailto:57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10>
> CSeq: 101 CANCEL
> Timestamp: 1459612184
>
>
>
>
>
>
> ***** Notice the error message here *****
> ***** Observe el mensaje de error aquÌ *****
>
>
>
>
>
> Apr 2 10:49:44.762 COT: //35/BA7E77800000/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 487 Request Terminated
> Via: SIP/2.0/UDP 10.11.44.10:5060;branch=z9hG4bK1E22F6
> From: <sip:13904707 at 10.11.44.10<mailto:13904707 at 10.11.44.10>>;tag=14BC38-1B1D
> To: <sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>>;tag=qe7asa39-CC-6
> Call-ID: 57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10<mailto:57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10>
> CSeq: 101 INVITE
> Timestamp: 1459612179
> Warning: 399 P.5.107.ims.claro.com.co<http://ims.claro.com.co> "SS130000F156L944[00000] Cancel received on initial invite"
> Content-Length: 0
>
>
> Apr 2 10:49:44.762 COT: //35/BA7E77800000/SIP/Call/sipSPICallInfo:
> The Call Setup Information is:
> Call Control Block (CCB) : 0x0xC29D1358
> State of The Call : STATE_DEAD
> TCP Sockets Used : NO
> Calling Number : 13904707
> Called Number : 13431111
> Source IP Address (Sig ): 10.11.44.10
> Destn SIP Req Addr:Port : 10.11.0.9:5060
> Destn SIP Resp Addr:Port : 10.11.0.9:5060
> Destination Name : 10.11.0.9
>
> Apr 2 10:49:44.762 COT: //35/BA7E77800000/SIP/Call/sipSPIMediaCallInfo:
> Number of Media Streams: 1
> Media Stream : 1
> Negotiated Codec : g711alaw
> Negotiated Codec Bytes : 160
> Nego. Codec payload : 8 (tx), 8 (rx)
> Negotiated Dtmf-relay : 6
> Dtmf-relay Payload : 97 (tx), 97 (rx)
> Source IP Address (Media): 10.11.44.10
> Source IP Port (Media): 16442
> Destn IP Address (Media): 10.11.0.9
> Destn IP Port (Media): 34822
> Orig Destn IP Address:Port (Media): [ - ]:0
>
> Apr 2 10:49:44.762 COT: //35/BA7E77800000/SIP/Call/sipSPICallInfo:
> Disconnect Cause (CC) : 38
> Disconnect Cause (SIP) : 487
>
> COCGW01#
> Apr 2 10:49:44.762 COT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Sent:
> ACK sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.11.44.10:5060;branch=z9hG4bK1E22F6
> From: "Hank Keleher" <sip:13904707 at 172.20.177.11<mailto:13904707 at 172.20.177.11>>;tag=14BC38-1B1D
> To: <sip:13431111 at 10.11.0.9<mailto:13431111 at 10.11.0.9>>;tag=qe7asa39-CC-6
> Date: Sat, 02 Apr 2016 15:49:39 GMT
> Call-ID: 57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10<mailto:57EC5E3B-F82111E5-804CBAE9-D4D92286 at 10.11.44.10>
> Max-Forwards: 70
> CSeq: 101 ACK
> Allow-Events: telephone-event
> Content-Length: 0
>
>
> COCGW01#
> Apr 2 10:51:32.787 COT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> NOTIFY sip:+5713904712 at 10.11.44.10<mailto:+5713904712 at 10.11.44.10>:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.11.0.9:5060;branch=z9hG4bK32cjlb30c8g47u9ijr30.1
> Call-ID: bisqs3a399i7adqf7raqef3b7sr72qi3 at 19500.0.ATS.ats-0.ims.claro.com.co.22<mailto:bisqs3a399i7adqf7raqef3b7sr72qi3 at 19500.0.ATS.ats-0.ims.claro.com.co.22>
> From: <sip:+5713904712 at 10.11.0.9<mailto:+5713904712 at 10.11.0.9>>;tag=3i3ssggs-CC-22
> To: <sip:+5713904712 at 10.11.44.10<mailto:+5713904712 at 10.11.44.10>>
> CSeq: 1 NOTIFY
> Contact: <sip:+5713904712 at 10.11.0.9<mailto:+5713904712 at 10.11.0.9>:5060;transport=udp>
> Max-Forwards: 68
> Event: ua-profile
> Subscription-State: active
> P-Called-Party-ID: <sip:+5713904712 at ims.claro.com.co<mailto:+5713904712 at ims.claro.com.co>>
> Content-Length: 242
> Content-Type: application/simservs+xml
>
> <?xml version="1.0"?>
> <simservs>
> <dial-tone-management>
> <dial-tone-pattern>standard-dial-tone</dial-tone-pattern>
> </dial-tone-management>
> <three-party-conference active="true"/>
> <explicit-call-transfer active="true"/>
> </simservs>
>
> COCGW01#
> Apr 2 10:52:15.123 COT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> NOTIFY sip:+5713904714 at 10.11.44.10<mailto:+5713904714 at 10.11.44.10>:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.11.0.9:5060;branch=z9hG4bKt8vva4302o98hm8pr770.1
> Call-ID: q7drr3s4qsse2arq4eids94283dafab2 at 19500.0.ATS.ats-0.ims.claro.com.co.6<mailto:q7drr3s4qsse2arq4eids94283dafab2 at 19500.0.ATS.ats-0.ims.claro.com.co.6>
> From: <sip:+5713904714 at 10.11.0.9<mailto:+5713904714 at 10.11.0.9>>;tag=qe87egi4-CC-6
> To: <sip:+5713904714 at 10.11.44.10<mailto:+5713904714 at 10.11.44.10>>
> CSeq: 1 NOTIFY
> Contact: <sip:+5713904714 at 10.11.0.9<mailto:+5713904714 at 10.11.0.9>:5060;transport=udp>
> Max-Forwards: 68
> Event: ua-profile
> Subscription-State: active
> P-Called-Party-ID: <sip:+5713904714 at ims.claro.com.co<mailto:+5713904714 at ims.claro.com.co>>
> Content-Length: 242
> Content-Typ
> COCGW01#e: application/simservs+xml
>
> <?xml version="1.0"?>
> <simservs>
> <dial-tone-management>
> <dial-tone-pattern>standard-dial-tone</dial-tone-pattern>
> </dial-tone-management>
> <three-party-conference active="true"/>
> <explicit-call-transfer active="true"/>
> </simservs>
>
> COCGW01#
> Apr 2 10:53:00.115 COT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> NOTIFY sip:+5713904717 at 10.11.44.10<mailto:+5713904717 at 10.11.44.10>:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.11.0.9:5060;branch=z9hG4bK6q31tf3018vnhu57ecb0.1
> Call-ID: b2q829brqe43gd42g432rrgi7igraq8a at 19500.0.ATS.ats-0.ims.claro.com.co.7<mailto:b2q829brqe43gd42g432rrgi7igraq8a at 19500.0.ATS.ats-0.ims.claro.com.co.7>
> From: <sip:+5713904717 at 10.11.0.9<mailto:+5713904717 at 10.11.0.9>>;tag=3si878re-CC-7
> To: <sip:+5713904717 at 10.11.44.10<mailto:+5713904717 at 10.11.44.10>>
> CSeq: 1 NOTIFY
> Contact: <sip:+5713904717 at 10.11.0.9<mailto:+5713904717 at 10.11.0.9>:5060;transport=udp>
> Max-Forwards: 68
> Event: ua-profile
> Subscription-State: active
> P-Called-Party-ID: <sip:+5713904717 at ims.claro.com.co<mailto:+5713904717 at ims.claro.com.co>>
> Content-Length: 242
> Content-Typ
> COCGW01#e: application/simservs+xml
>
> <?xml version="1.0"?>
> <simservs>
> <dial-tone-management>
> <dial-tone-pattern>standard-dial-tone</dial-tone-pattern>
> </dial-tone-management>
> <three-party-conference active="true"/>
> <explicit-call-transfer active="true"/>
> </simservs>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
> https://puck.nether.net/mailman/listinfo/cisco-voip
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