Found unknown media description format GSM-HR-08 for ID 96

Asterisk Asterisk SIP
1

Hello I’m having a hard time to connect my calls. Any help would be appreciated.

sip.conf

[GSM]
type=friend
host=127.0.0.1
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=gsm
context=gsmsubscriber
port=5069

extensions.conf

[gsmsubscriber]
exten=>_XXXX,1,Dial(SIP/GSM/${EXTEN})
exten=>_XXXX,n,Playback(vm-nobodyavail)
exten=>_XXXX,n,HangUp

SIP debug stream

<--- SIP read from UDP:127.0.0.1:5069 --->
INVITE sip:8888@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5069;rport;branch=z9hG4bKU3j8U27vDtHZp
Max-Forwards: 70
From: <sip:4444@127.0.0.1:5069>;tag=Urrrtp6yFQa8r
To: <sip:8888@127.0.0.1:5060>
Call-ID: 66b81c41-8ebd-1237-1ba1-00073242fed9
CSeq: 939007839 INVITE
Contact: <sip:127.0.0.1:5069>
User-Agent: sofia-sip/1.12.11devel
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Length: 130

v=0
o=Osmocom 0 0 IN IP4 127.0.0.1
s=GSM Call
c=IN IP4 127.0.0.1
t=0 0
m=audio 26434 RTP/AVP 96
a=rtpmap:96 GSM-HR-08/8000
<------------->
--- (13 headers 7 lines) ---
Sending to 127.0.0.1:5069 (no NAT)
Sending to 127.0.0.1:5069 (no NAT)
Using INVITE request as basis request - 66b81c41-8ebd-1237-1ba1-00073242fed9
Found peer 'GSM' for '4444' from 127.0.0.1:5069
Found RTP audio format 96
Found unknown media description format GSM-HR-08 for ID 96
[Jan  9 09:26:31] NOTICE[3683][C-00000000]: chan_sip.c:10881 process_sdp: No compatible codecs, not accepting this offer!

<--- Reliably Transmitting (no NAT) to 127.0.0.1:5069 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 127.0.0.1:5069;branch=z9hG4bKU3j8U27vDtHZp;received=127.0.0.1;rport=5069
From: <sip:4444@127.0.0.1:5069>;tag=Urrrtp6yFQa8r
To: <sip:8888@127.0.0.1:5060>;tag=as77950649
Call-ID: 66b81c41-8ebd-1237-1ba1-00073242fed9
CSeq: 939007839 INVITE
Server: Asterisk PBX 13.24.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '66b81c41-8ebd-1237-1ba1-00073242fed9' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:127.0.0.1:5069 --->
ACK sip:8888@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5069;rport;branch=z9hG4bKU3j8U27vDtHZp
Max-Forwards: 70
From: <sip:4444@127.0.0.1:5069>;tag=Urrrtp6yFQa8r
To: <sip:8888@127.0.0.1:5060>;tag=as77950649
Call-ID: 66b81c41-8ebd-1237-1ba1-00073242fed9
CSeq: 939007839 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '66b81c41-8ebd-1237-1ba1-00073242fed9' Method: ACK
2

Asterisk does not support “GSM-HR-08” as a codec. You would need to use a different supported one.

3

Sure. However, I had a similar installation working before where I was able to make some calls so I’m sure I can get this working. The only change I’m aware that I made between the original installation and my current installation is going from 13.24.0 to 13.24.1 however something else might be different as well…

Anyways where could I find a list of the different GSM codecs that are supported.

4

Only one is supported, which is in SDP as “GSM”.

There has never been support in the tree for “GSM-HR-08”. It is possible you used a third party patch or something.

5

No not that I know of. I could get you the make menu configs of both installations.

6

I also used fresh tarballs without changes to the source code.

7

Then I do not know how it could have possibly worked before, unless something else changed.

8

Thank you for your time.

9

I’ve got it working. If people are working with osmocom Asterisk doesn’t support half rate channels only full rate channels.