12-03-2010 07:52 AM - edited 03-16-2019 02:15 AM
Can someone help with an issue that I have?
I have created a sip trunk between a router running cme and the communicator manager.
I can make calls from the cucm to ccme but not vice versa. I get fast busy.
I have used the cucm configuration from this url , although I did not check the mtp box.
The cucm registered phones has a directory number of 2724 and ip 192.168.15.244. The communicator manager is 192.168.15.1 with a route pattern 5xxx
the cme has phones with 5xxx ip addresses of the cme router are 10.200.40.2 and 10.250.250.3. Running 15.1T
The cme configuration is below, dial-peer 272:
CME_LAB#sh run | beg dial
dial-peer voice 9 pots
translation-profile incoming AA_CLI
translation-profile outgoing AA_CLI
destination-pattern 9...........
direct-inward-dial
port 0/2/0:15
forward-digits all
!
dial-peer voice 2 voip
destination-pattern 2...
session protocol sipv2
session target ipv4:10.200.40.4
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 4 voip
description Local NM-CUE (CME) aa
destination-pattern 4000
session protocol sipv2
session target ipv4:10.200.40.4
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 272 voip
description CCME_2_CUCM
destination-pattern 2724
session protocol sipv2
session target ipv4:192.168.15.1
dtmf-relay sip-notify
codec g711alaw
no vad
!
!
!
I have used the command csim start 2724 and this is the result:
CME_LAB#csim start 2724
csim: called number = 2724, loop count = 1 ping count = 0
csim err csimDisconnected recvd DISC cid(62)
csim: loop = 1, failed = 1
csim: call attempted = 1, setup failed = 1, tone failed = 0
Also here is the output from debug ccsip all, the first being a call from a phone on call manager to a phone on cme (works) and vice versa (doesn't work)
g711ulawCME_LAB#
Dec 3 15:41:24.783: //38/9D6A4A04806C/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x677AE288
State of The Call : STATE_ACTIVE
TCP Sockets Used : YES
Calling Number : 2724
Called Number : 5001
Source IP Address (Sig ): 10.250.250.3
Destn SIP Req Addr:Port : 192.168.15.1:5060
Destn SIP Resp Addr:Port : 192.168.15.1:48293
Destination Name : 192.168.15.1
Dec 3 15:41:24.783: //38/9D6A4A04806C/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 : 8
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 10.250.250.3
Source IP Port (Media): 17656
Destn IP Address (Media): 192.168.15.244
Destn IP Port (Media): 27256
Orig Destn IP Address:Port (Media): [ - ]:0
Dec 3 15:41:24.783: //38/9D6A4A04806C/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x677AE288
State of The Call : STATE_ACTIVE
TCP Sockets Used : YES
Calling Number : 2724
Called Number : 5001
Source IP Address (Sig ): 10.250.250.3
Destn SIP Req Addr:Port : 192.168.15.1:5060
Destn SIP Resp Addr:Port : 192.168.15.1:48293
Destination Name : 192.168.15.1
Dec 3 15:41:24.783: //38/9D6A4A04806C/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 : 8
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 10.250.250.3
Source IP Port (Media): 17656
Destn IP Address (Media): 192.168.15.244
Destn IP Port (Media): 27256
Orig Destn IP Address:Port (Media): [ - ]:0
Dec 3 15:41:29.471: //38/9D6A4A04806C/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x677AE288
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 2724
Called Number : 5001
Source IP Address (Sig ): 10.250.250.3
Destn SIP Req Addr:Port : 192.168.15.1:5060
Destn SIP Resp Addr:Port : 192.168.15.1:48293
Destination Name : 192.168.15.1
Dec 3 15:41:29.471: //38/9D6A4A04806C/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 : 8
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 10.250.250.3
Source IP Port (Media): 17656
Destn IP Address (Media): 192.168.15.244
Destn IP Port (Media): 27256
Orig Destn IP Address:Port (Media): [ - ]:0
Dec 3 15:41:29.471: //38/9D6A4A04806C/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 200
Dec 3 15:41:35.899: //41/A4BD48418072/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x677AE288
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 5001
Called Number : 2724
Source IP Address (Sig ): 10.250.250.3
Destn SIP Req Addr:Port : 192.168.15.1:5060
Destn SIP Resp Addr:Port : 192.168.15.1:5060
Destination Name : 192.168.15.1
Dec 3 15:41:35.899: //41/A4BD48418072/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 10.250.250.3
Source IP Port (Media): 19342
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
Dec 3 15:41:35.899: //41/A4BD48418072/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 63
Disconnect Cause (SIP) : 503
Dec 3 15:41:35.915: //42/A4BD48418072/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x677AE288
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 5001
Called Number : 2724
Source IP Address (Sig ): 10.200.40.2
Destn SIP Req Addr:Port : 10.200.40.4:5060
Destn SIP Resp Addr:Port : 10.200.40.4:5060
Destination Name : 10.200.40.4
Dec 3 15:41:35.915: //42/A4BD48418072/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 10.200.40.2
Source IP Port (Media): 19254
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
Dec 3 15:41:35.915: //42/A4BD48418072/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 38
Disconnect Cause (SIP) : 503
See attached ccme config
Thanks
Feisal
12-03-2010 10:45 AM
Looks like a codec mismatch:
Media Stream : 1
Negotiated Codec : No Codec
Several issues here.
1. You need dial peers supporting BOTH directions of the call (to the caller and to the called party). With your setup, you are using the default dial-peer (which is a bad thing to do since you can't control it) to handle at least half of this. You would be able to see this if you debug the dial peer rather than
2. You have the codec on your dial peer set to G711Alaw.....If you're in the US, the correct codec is normally G711Ulaw.
3. CCM cannot negotiate codec. CME can. You MUST have your dial peer exactly match the codec configured on the trunk on the CCM.ccsip
12-06-2010 07:07 AM
What would be the correct configuration for the incoming dial-peer?
I am in the uk and using alaw for the sip trunk.
Thanks
Feisal
12-03-2010 09:12 PM
Hello Feisalb,
Would you mind trying this:
!
dial-peer voice 272 voip
description CCME_2_CUCM
destination-pattern 2724
session protocol sipv2
session target ipv4:192.168.15.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
From the guide you followed to this:
1. Uncheck the MTP required.
2. DTMF signal method* OOB and RFC 2833
And if that does not work, please attach a debug ccsip messages, and attach a screenshot of the route pattern on cucm
Best regards,
Luis Sandi
12-06-2010 07:54 AM
Ok, I am the div.
Got it working by changing the destination ip address on the sip cucm to point to the serial interface 10.250.250.3 instead of 10.200.40.2 fastethernet sub interface.
Experimented a bit more and found the commands to bind sip to a particular interface.
voice service voip
sip
bind control source
bind media source
It is now working with the original dest ip 10.200.40.2
I guess it must be that same as using a source interface in h323.
The rest of the config is as the original, I didn't change the dtfm-relay to rtp-nte.
Next up is to see if cue voicemail works from phones on the cucm (loving all these acronyms).
I guess I will need a transcoder somewhere on the network to trancode g711alaw (5xxx phones) to g711 ulaw (cue) and vice versa.
Thanks
Feisal
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