[cisco-voip] SIP CME-as-SRST transfer failing

Sat Nov 9 11:30:47 EST 2013

That's what I figured that he needed telephony-service because it does not matter if it's SCCP or not right? Without Telephony-Service you can't specify which phone loads to load.

Sent from my iPhone

> On Nov 9, 2013, at 11:10 AM, "Adrian Van Wyk (ZA)" <adrian.vanwyk at dimensiondata.com> wrote:
> 
> Is this a SRST site or CME. If it is CME you would still need to configure telephony service. This is where you add the transfer pattern.
> 
> On 09 Nov 2013, at 17:55, "Peter Slow" <peter.slow at gmail.com> wrote:
> 
>> i hate to say this but that's where I start considering a reload of the router, if you try that and you keep getting that same crazy error, THEN you can start picking apart your config, do you have that troubleshooting option (rebooting) available?
>> 
>> -Peter
>> 
>> 
>>> On Fri, Nov 8, 2013 at 6:39 PM, Kenneth Hayes <kennethwhayes at gmail.com> wrote:
>>> Jason,
>>> 
>>> Here's an example I got from this doc... I think the phones MAC needs
>>> to be registered.
>>> 
>>> See the example:
>>> voice register pool 2
>>>  id mac 0012.7F3B.9025 <-phone
>>>  number 1 2800
>>>  codec g711ulaw
>>> !
>>> voice register pool 3
>>>  id mac 0012.7F57.628F
>>>  number 1 2801
>>>  codec g711ulaw
>>> !
>>> 
>>> 
>>> On Fri, Nov 8, 2013 at 6:27 PM, Jason Aarons (AM)
>>> <jason.aarons at dimensiondata.com> wrote:
>>> > Here is the debug ccsip messages, notice the error down at 17:08:12.228 ?
>>> > Researching it.
>>> >
>>> >
>>> >
>>> > *Nov  8 17:08:12.224: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> >
>>> > Received:
>>> >
>>> > INVITE sip:7454 at 10.30.2.253:5060 SIP/2.0
>>> >
>>> > Via: SIP/2.0/UDP 10.30.2.253:5060;branch=z9hG4bKBAFA43B
>>> >
>>> > Remote-Party-ID: "7484"
>>> > <sip:7484 at 10.30.2.253>;party=calling;screen=yes;privacy=off
>>> >
>>> > From: "7484" <sip:7484 at 10.27.16.11>;tag=7C7715E4-81A
>>> >
>>> > To: <sip:7454 at 10.30.2.253>
>>> >
>>> > Date: Fri, 08 Nov 2013 23:08:12 GMT
>>> >
>>> > Call-ID: 79856A55-480111E3-AFB5CA0E-60C774C9 at 10.30.2.253
>>> >
>>> > Supported: 100rel,timer,resource-priority,replaces
>>> >
>>> > Min-SE:  1800
>>> >
>>> > Cisco-Guid: 2038626741-1208029667-2947533326-1623684297
>>> >
>>> > User-Agent: Cisco-SIPGateway/IOS-12.x
>>> >
>>> > Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
>>> > NOTIFY, INFO, REGISTER
>>> >
>>> > CSeq: 101 INVITE
>>> >
>>> > Timestamp: 1383952092
>>> >
>>> > Contact: <sip:7484 at 10.30.2.253:5060>
>>> >
>>> > Call-Info:
>>> > <sip:10.30.2.253:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
>>> >
>>> > Expires: 180
>>> >
>>> > Allow-Events: telephone-event
>>> >
>>> > Max-Forwards: 69
>>> >
>>> > Content-Type: application/sdp
>>> >
>>> > Content-Disposition: session;handling=required
>>> >
>>> > Content-Length: 409
>>> >
>>> >
>>> >
>>> > v=0
>>> >
>>> > o=CiscoSystemsSIP-GW-UserAgent 3729 2253 IN IP4 10.30.2.253
>>> >
>>> > s=SIP Call
>>> >
>>> > c=IN IP4 10.30.2.253
>>> >
>>> > t=0 0
>>> >
>>> > m=audio 17618 RTP/AVP 0 18 100 101 121 19
>>> >
>>> > c=IN IP4 10.30.2.253
>>> >
>>> > a=rtpmap:0 PCMU/8000
>>> >
>>> > a=rtpmap:18 G729/8000
>>> >
>>> > a=fmtp:18 annexb=no
>>> >
>>> > a=rtpmap:100 X-NSE/8000
>>> >
>>> > a=fmtp:100 192-194
>>> >
>>> > a=rtpmap:101 telephone-event/8000
>>> >
>>> > a=fmtp:101 0-15
>>> >
>>> > a=rtpmap:121 frf-dialed-digit/8000
>>> >
>>> > a=fmtp:121 0-15
>>> >
>>> > a=rtpmap:19 CN/8000
>>> >
>>> >
>>> >
>>> > *Nov  8 17:08:12.228: %CALL_CONTROL-6-CALL_LOOP: The incoming call has a
>>> > global identifier already present in the list of currently handled calls. It
>>> > is being refused.
>>> >
>>> >
>>> >
>>> > *Nov  8 17:08:12.244: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>> >
>>> > Sent:
>>> >
>>> > SIP/2.0 500 Internal Server Error
>>> >
>>> > Via: SIP/2.0/UDP 10.30.2.253:5060;branch=z9hG4bKBAFA43B
>>> >
>>> > From: "7484" <sip:7484 at 10.27.16.11>;tag=7C7715E4-81A
>>> >
>>> > To: <sip:7454 at 10.30.2.253>;tag=7C7715EC-B31
>>> >
>>> > Date: Fri, 08 Nov 2013 23:08:12 GMT
>>> >
>>> > Call-ID: 79856A55-480111E3-AFB5CA0E-60C774C9 at 10.30.2.253
>>> >
>>> > Timestamp: 1383952092
>>> >
>>> > CSeq: 101 INVITE
>>> >
>>> > Allow-Events: telephone-event
>>> >
>>> > Server: Cisco-SIPGateway/IOS-12.x
>>> >
>>> > Content-Length: 0
>>> >
>>> >
>>> >
>>> > From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
>>> > Jason Aarons (AM)
>>> > Sent: Friday, November 08, 2013 6:16 PM
>>> > To: Kenneth Hayes
>>> >
>>> >
>>> > Cc: cisco-voip (cisco-voip at puck.nether.net)
>>> > Subject: Re: [cisco-voip] SIP CME-as-SRST transfer failing
>>> >
>>> >
>>> >
>>> > This is 100% SIP. The 8961 doesn’t do SCCP.
>>> >
>>> >
>>> >
>>> > From: Kenneth Hayes [mailto:kennethwhayes at gmail.com]
>>> > Sent: Friday, November 08, 2013 6:13 PM
>>> > To: Jason Aarons (AM)
>>> > Cc: cisco-voip (cisco-voip at puck.nether.net)
>>> > Subject: Re: [cisco-voip] SIP CME-as-SRST transfer failing
>>> >
>>> >
>>> >
>>> >
>>> >
>>> > Where is your telephony-service?
>>> >
>>> > Sent from my iPhone
>>> >
>>> >
>>> > On Nov 8, 2013, at 6:00 PM, "Jason Aarons (AM)"
>>> > <jason.aarons at dimensiondata.com> wrote:
>>> >
>>> >
>>> >
>>> > Ok what am I missing.  SIP 8961s and transfer is failing whilst in SRST
>>> > mode.
>>> >
>>> >
>>> >
>>> >
>>> >
>>> > voice service voip
>>> >
>>> > ip address trusted list
>>> >
>>> >   ipv4 192.168.16.10
>>> >
>>> >   ipv4 192.168.16.11
>>> >
>>> > allow-connections h323 to h323
>>> >
>>> > allow-connections h323 to sip
>>> >
>>> > allow-connections sip to h323
>>> >
>>> > allow-connections sip to sip
>>> >
>>> > redirect ip2ip
>>> >
>>> > fax protocol none
>>> >
>>> >  modem passthrough nse codec g711ulaw
>>> >
>>> > sip
>>> >
>>> >   bind control source-interface GigabitEthernet0/1.30
>>> >
>>> >   bind media source-interface GigabitEthernet0/1.30
>>> >
>>> >   registrar server
>>> >
>>> > !
>>> >
>>> > voice class codec 99
>>> >
>>> > codec preference 1 g711ulaw
>>> >
>>> > codec preference 2 g729r8
>>> >
>>> > !
>>> >
>>> > voice class codec 1
>>> >
>>> > codec preference 1 g711ulaw
>>> >
>>> > codec preference 2 g729r8
>>> >
>>> > !
>>> >
>>> > voice class h323 1
>>> >
>>> >   h225 timeout tcp establish 3
>>> >
>>> > !
>>> >
>>> > !
>>> >
>>> > voice register global
>>> >
>>> > max-dn 480
>>> >
>>> > max-pool 1
>>> >
>>> > !
>>> >
>>> > voice register pool  1
>>> >
>>> > id network 192.168.2.0 mask 255.255.255.0
>>> >
>>> > preference 1
>>> >
>>> > incoming called-number
>>> >
>>> >  proxy 192.168.2.253 preference 1
>>> >
>>> > dtmf-relay rtp-nte cisco-rtp sip-notify
>>> >
>>> > voice-class codec 1
>>> >
>>> > _______________________________________________
>>> > cisco-voip mailing list
>>> > cisco-voip at puck.nether.net
>>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>>> >
>>> >
>>> >
>>> > itevomcid
>>> 
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