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SIP-Incoming Call Fail

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ilana_ilana
Level 1
Level 1

‎09-04-2016 07:59 AM - edited ‎03-17-2019 08:01 AM

Hello all,

One of our client is having CUCM with H.323 GW.

Recently they got SIP trunk from service provider. SIP trunk will be used only for incoming calls. 

Currently Incoming calls are failing with error " SIP/2.0 488 Not Acceptable Media ".

From search I found, this error relates to codec mismatch, I have configured codecs but not able to find out where is the issue:

Following is the output of debug voice ccapi inout and ccsip messages output and attached is show run.

Router#
*Sep 4 13:47:00.115: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0135109595@192.168.3.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 94.77.211.70:5060;branch=z9hG4bKbjimje00do6g75jrv1p1.1
Call-ID: 5NAln2qVnY05QvIaesjSB2vthp2VQe9LKhQjfY5LiRn08lN2iarKjWM0ijac@zteims
From: <sip:0115111262@ims.go.net.sa;user=phone>;tag=ztesipERZbsFRE-t5K4dfbg.4
To: <sip:0135109595@172.26.32.40;user=phone>
CSeq: 1000 INVITE
P-Asserted-Identity: <sip:0115111262@ims.go.net.sa;user=phone>
Privacy: none
Allow: INVITE,PRACK,ACK,CANCEL,BYE,UPDATE,OPTIONS
Max-Forwards: 69
P-Charging-Vector: icid-value=004-18e5a9c8-923-172.26.32.130
Content-Type: application/sdp
Content-Length: 211
Content-Disposition: session
Supported: 100rel, early-session
Contact: <sip:0115111262@94.77.211.70:5060;zte-did=4-0-560-J;transport=udp>

v=0
o=- 526323468 526323468 IN IP4 94.77.211.70
s=-
c=IN IP4 94.77.211.70
t=0 0
m=audio 23388 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
a=ptime:20

*Sep 4 13:47:00.115: %VOICE_IEC-3-GW: SIP: Internal Error (INVITE, codec mismatch): IEC=1.1.278.7.110.0 on callID 59
*Sep 4 13:47:00.119: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 94.77.211.70:5060;branch=z9hG4bKbjimje00do6g75jrv1p1.1
From: <sip:0115111262@ims.go.net.sa;user=phone>;tag=ztesipERZbsFRE-t5K4dfbg.4
To: <sip:0135109595@172.26.32.40;user=phone>;tag=2FF708-2EA
Date: Sun, 04 Sep 2016 13:47:00 GMT
Call-ID: 5NAln2qVnY05QvIaesjSB2vthp2VQe9LKhQjfY5LiRn08lN2iarKjWM0ijac@zteims
CSeq: 1000 INVITE
Allow-Events: telephone-event
Warning: 304 192.168.3.2 "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-15.2.4.M4
Content-Length: 0


*Sep 4 13:47:00.131: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0135109595@192.168.3.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 94.77.211.70:5060;branch=z9hG4bKbjimje00do6g75jrv1p1.1
CSeq: 1000 ACK
Call-ID: 5NAln2qVnY05QvIaesjSB2vthp2VQe9LKhQjfY5LiRn08lN2iarKjWM0ijac@zteims
From: <sip:0115111262@ims.go.net.sa;user=phone>;tag=ztesipERZbsFRE-t5K4dfbg.4
To: <sip:0135109595@172.26.32.40;user=phone>;tag=2FF708-2EA
Max-Forwards: 69
Content-Length: 0

Router#

Can any body advise where is the issue ?

Regards,

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1 Accepted Solution

Accepted Solutions

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Manish Gogna
Cisco Employee
Cisco Employee

‎09-04-2016 08:22 AM

Hi Ilana,

You may try the following:

Set the codec on dial-peers for these incoming calls from the provider to G711  and on cucm, set the region setting between IP phones and this gateway to G711 as well. You may use G729 as well but please ensure the codec is the same throughout the call flow. Reset the SIP trunk after any changes and try again. If the call fails please post complete "debug ccsip messages" output for a failed call 

View solution in original post

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3 Replies 3

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Manish Gogna
Cisco Employee
Cisco Employee

‎09-04-2016 08:22 AM

Hi Ilana,

You may try the following:

Set the codec on dial-peers for these incoming calls from the provider to G711  and on cucm, set the region setting between IP phones and this gateway to G711 as well. You may use G729 as well but please ensure the codec is the same throughout the call flow. Reset the SIP trunk after any changes and try again. If the call fails please post complete "debug ccsip messages" output for a failed call 

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ilana_ilana
Level 1
Level 1
In response to Manish Gogna

‎09-04-2016 10:22 AM

Hi Manish,

How to reset SIP Trunk in VG? There is no SIP Trunk between CUCM & VG

Call Manager & GW are configured as H323. Gateway is having SIP - PRI from Telco provider.

Debugs shared above are complete output of "debug ccsip messages" and "debug voice ccapi inout".

From incoming call logs (PSTN to Call Manager) and show run shared above, can you confirm the incoming dial-peer and codecs are configured proper in voice gateway ?

************************************************************************************************************

*Sep 4 13:47:00.115: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0135109595@192.168.3.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 94.77.211.70:5060;branch=z9hG4bKbjimje00do6g75jrv1p1.1
Call-ID: 5NAln2qVnY05QvIaesjSB2vthp2VQe9LKhQjfY5LiRn08lN2iarKjWM0ijac@zteims
From: <sip:0115111262@ims.go.net.sa;user=phone>;tag=ztesipERZbsFRE-t5K4dfbg.4
To: <sip:0135109595@172.26.32.40;user=phone>
CSeq: 1000 INVITE
P-Asserted-Identity: <sip:0115111262@ims.go.net.sa;user=phone>
Privacy: none
Allow: INVITE,PRACK,ACK,CANCEL,BYE,UPDATE,OPTIONS
Max-Forwards: 69
P-Charging-Vector: icid-value=004-18e5a9c8-923-172.26.32.130
Content-Type: application/sdp
Content-Length: 211
Content-Disposition: session
Supported: 100rel, early-session
Contact: <sip:0115111262@94.77.211.70:5060;zte-did=4-0-560-J;transport=udp>

v=0
o=- 526323468 526323468 IN IP4 94.77.211.70
s=-
c=IN IP4 94.77.211.70
t=0 0
m=audio 23388 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
a=ptime:20

*Sep 4 13:47:00.115: %VOICE_IEC-3-GW: SIP: Internal Error (INVITE, codec mismatch): IEC=1.1.278.7.110.0 on callID 59
*Sep 4 13:47:00.119: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 94.77.211.70:5060;branch=z9hG4bKbjimje00do6g75jrv1p1.1
From: <sip:0115111262@ims.go.net.sa;user=phone>;tag=ztesipERZbsFRE-t5K4dfbg.4
To: <sip:0135109595@172.26.32.40;user=phone>;tag=2FF708-2EA
Date: Sun, 04 Sep 2016 13:47:00 GMT
Call-ID: 5NAln2qVnY05QvIaesjSB2vthp2VQe9LKhQjfY5LiRn08lN2iarKjWM0ijac@zteims
CSeq: 1000 INVITE
Allow-Events: telephone-event
Warning: 304 192.168.3.2 "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-15.2.4.M4
Content-Length: 0


*Sep 4 13:47:00.131: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0135109595@192.168.3.2:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 94.77.211.70:5060;branch=z9hG4bKbjimje00do6g75jrv1p1.1
CSeq: 1000 ACK
Call-ID: 5NAln2qVnY05QvIaesjSB2vthp2VQe9LKhQjfY5LiRn08lN2iarKjWM0ijac@zteims
From: <sip:0115111262@ims.go.net.sa;user=phone>;tag=ztesipERZbsFRE-t5K4dfbg.4
To: <sip:0135109595@172.26.32.40;user=phone>;tag=2FF708-2EA
Max-Forwards: 69
Content-Length: 0

Regards

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Manish Gogna
Cisco Employee
Cisco Employee
In response to ilana_ilana

‎09-04-2016 07:05 PM

If the gateway is added as H323 on cucm then reset the h323 gateway on cucm after making any changes. As per the SIP INVITE the call is being requested to be set up on G729 codec, so try setting the G729 codec on the dial-peers on gateway and on region settings between IP phones and H323 gateway on cucm. You may also add transcoders in the MRGL of IP phones and H323 gateway so that it can be allocated depending on which side requests the transcoder.

One more thing that you can check o gateway is if the following is added:

voice service voip

allow-connections sip to h323

allow-connections h323 to sip

allow-connections sip to sip

allow-connections h323 to h323

Manish

Manish

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