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ip phone 7905 dosen't advertise codecs g711ulaw and g711 alaw

denis.moiseenko
Level 1
Level 1

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!

2 Replies 2

Gergely Szabo
VIP Alumni
VIP Alumni

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

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

false010
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?