06-03-2013 11:45 PM - edited 03-16-2019 05:40 PM
Hello!
Could you please help me!
I have cisco ip phone 7905 it's registered ont the CUCM and I try to call from this phone to PSTN. When CUCM advertises the supported codecs it's only advertises G729 codec but my ITSP dosen't support this codec. As I know the ip phone support g711ulaw and g711 alaw. Why this codec dosen't advertise by CUCM. What may be the problem?
and also here the debug ccsip message
Jun 4 05:52:54.237: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:93.190.xx.xx SIP/2.0
Via: SIP/2.0/UDP 93.190.xx.xxx:5060;branch=z9hG4bK5b861b74;rport
From: "asterisk" <sip:asterisk@93.190.xx.xx>;tag=as0b1aa8f0
To: <sip:93.190.xx.xx>
Contact: <sip:asterisk@93.190.xx.xx>
Call-ID: 1e852b0e25f4ec48467e666c3101c23f@93.190.xx.xx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 04 Jun 2013 05:52:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Jun 4 05:52:54.257: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 93.190.xx.xx:5060;branch=z9hG4bK5b861b74;rport
From: "asterisk" <sip:asterisk@93.190.xx.xx>;tag=as0b1aa8f0
To: <sip:93.190.xx.xx>;tag=E79D320-4D7
Date: Tue, 04 Jun 2013 05:52:54 GMT
Call-ID: 1e852b0e25f4ec48467e666c3101c23f@93.190.xx.xx
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Accept: application/sdp
Allow-Events: telepho
sochi-gw#ne-event
Content-Length: 167
Content-Type: application/sdp
v=0
o=CiscoSystemsSIP-GW-UserAgent 9934 5563 IN IP4 93.190.xx.xx
s=SIP Call
c=IN IP4 93.190.xx.xx
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 93.190.xx.xx
sochi-gw#
Jun 4 05:52:56.333: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:89881634594@192.168.228.1:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.21.3:5060;branch=z9hG4bK1c83f425e92ac2
From: <sip:2965873@192.168.21.3>;tag=2471146~96ef6ccc-73f3-4efb-a1b9-3c908a715a62-28044735
To: <sip:89881634594@192.168.228.1>
Date: Tue, 04 Jun 2013 05:52:56 GMT
Call-ID: ffb57180-1ad180b8-127478-315a8c0@192.168.21.3
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
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
Call-Info: <sip:192.168.21.3:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 4290081152-0000065536-0000084914-0051751104
Session-Expires: 1800
P-Asserted-Identity: <sip:2965873@192.168.21.3>
Remote-Party-ID: <sip:2965873@192.168.21.3>;party=calling;screen=yes;privacy=off
Contact: <sip:2965873@192.168.21.3:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 238
v=0
o=CiscoSystemsCCM-SIP 2471146 1 IN IP4 192.168.21.3
s=SIP Call
c=IN IP4 192.168.21.3
b=TIAS:8000
b=AS:8
t=0 0
m=audio 25426 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Jun 4 05:52:56.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TCP 192.168.21.3:5060;branch=z9hG4bK1c83f425e92ac2
From: <sip:2965873@192.168.21.3>;tag=2471146~96ef6ccc-73f3-4efb-a1b9-3c908a715a62-28044735
To: <sip:89881634594@192.168.228.1>;tag=E79DB54-2191
Date: Tue, 04 Jun 2013 05:52:56 GMT
Call-ID: ffb57180-1ad180b8-127478-315a8c0@192.168.21.3
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Warning: 304 192.168.18.9 "Media Type(s) Unavailable"
Allow-Events: telephone-event
Content-Length: 0
sochi-gw#
Jun 4 05:52:56.449: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:89881634594@192.168.228.1:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.21.3:5060;branch=z9hG4bK1c83f425e92ac2
From: <sip:2965873@192.168.21.3>;tag=2471146~96ef6ccc-73f3-4efb-a1b9-3c908a715a62-28044735
To: <sip:89881634594@192.168.228.1>;tag=E79DB54-2191
Date: Tue, 04 Jun 2013 05:52:56 GMT
Call-ID: ffb57180-1ad180b8-127478-315a8c0@192.168.21.3
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
Could you please help me!
Thank you in advance!
06-04-2013 12:18 AM
Post the phone config and the dialpeer config.
You may want to force the codec, using the codec command under the phone config.
G.
Sent from Cisco Technical Support Android App
06-04-2013 01:03 AM
here the phone config
true SCCP 0 true 0 2012c tzupdater.jar 000000 Off Disabled false 0 DP-CPT-SOCH CMLocal D.M.Y Etc/GMT-4 192.168.15.10 Directed Broadcast 192.168.15.11 Directed Broadcast 192.168.21.7 Directed Broadcast CUCM-SPB-CLUSTER true cucm2 cucm2 2000 5060 5061 2427 2428 cucm2 cucm1 cucm1 2000 5060 5061 2427 2428 cucm1 SRST-SOCH-GW User Specific true 192.168.218.1 2000 2000 2000 5060 5060 5060 false 120 true 2 CP7905080003SCCP070409A false 0 1 0 1370242819-d90061cc-4e0f-4f5e-b558-3c9263535578 Russian_Russian_Federation 5 ru_RU 9.1.1.1000-1 windows-1251 Russian_Federation Russian_Federation 50 0 http://192.168.21.2:8080/ccmcip/authenticate.jsp http://192.168.21.2:8080/ccmcip/xmldirectory.jsp http://192.168.21.2:8080/ccmcip/GetTelecasterHelpText.jsp http://192.168.21.2:8080/ccmcip/getservicesmenu.jsp 96 0 96 1 5 1 0 0 0 false 0 0 0 3804 cucm1 false 0 *81 *82 *83 *84 *85 0 Missed Calls Application:Cisco/MissedCalls Voicemail Application:Cisco/Voicemail Received Calls Application:Cisco/ReceivedCalls Placed Calls Application:Cisco/PlacedCalls Personal Directory Application:Cisco/PersonalDirectory Corporate Directory Application:Cisco/CorporateDirectory
and here the dial-peer config
sochi-gw#show run | sec dial-peer
dial-peer voice 100 voip
description SOH_City_OUT
translation-profile outgoing SOH_OUT_9
destination-pattern 9T
session protocol sipv2
session target ipv4:93.190.xx.xx
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 101 voip
description SOH_MGMNDEF_OUT
translation-profile outgoing SOH_OUT_9
destination-pattern 8T
session protocol sipv2
session target ipv4:93.190.xx.xx
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 102 voip
description SOCH_Emergency0x_OUT
translation-profile outgoing SOH_OUT_9
destination-pattern 0[1-4]
session protocol sipv2
session target ipv4:93.190.xx.xx
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 103 voip
description SOCH_Emergency_OUT_9
destination-pattern 112
session protocol sipv2
session target ipv4:93.190.xx.xx
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 200 voip
description IN_2965873
translation-profile incoming SOH_IN_PSTN
session protocol sipv2
incoming called-number 2965873
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 201 voip
description IN_6402
preference 1
destination-pattern 6402
session protocol sipv2
session target ipv4:192.168.21.2:5060
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 202 voip
description IN_6402
preference 2
destination-pattern 6402
session protocol sipv2
session target ipv4:192.168.21.3:5060
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 204 voip
description IN_6405
preference 2
destination-pattern 6405
session protocol sipv2
session target ipv4:192.168.21.3:5060
codec g711ulaw
no vad
dial-peer voice 300 voip
description From_CUCM1and2
answer-address 2965873
session protocol sipv2
dtmf-relay rtp-nte
codec g711ulaw
Could you please tell the command under the phone config in order to force the codec?
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