Why does Cisco Jabber hang up a call after going off-hook?

Question

SIP

Good afternoon!
There is CUCM 8.6, it was necessary to install jabbers (only telephony service) on mobile phones.
Installed, all Jabbers successfully registered in the system.
But when you call from Jabber to a landline phone, when you pick up the handset, it is reset.
I removed the trace, in the trace there is such an error Reason: SIP/2.0 503 Service Unavailable Q.850;cause=47
Google suggested that this is a codec negotiation problem.
I looked at the Device Pool settings, made sure that the g711 codec is available (Jabber and Phones are in the same pool).

Can you please tell me what can be done to solve the problem?

Here's what's in INVITE:

07/20/2020 16:27:27.759 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.0.0.101 on port 43269 index 41812 with 1895 bytes:
[14096621,NET]
INVITE sip:[email protected];user= the SIP Image phone / 2.0 We do to
Via: the SIP / 2.0 We do / the TCP 10.0.0.101:43269;branch=z9hG4bK57dd75c5 the
From: "1117"; tag = 34791644a70100ed578b5085-79f3defb the
To:
of Call-ID: [email protected]
Max- Forwards: 70
Session-ID: 08ec9d0500105000a00034791644a701;remote=00000000000000000000000000000000
Date: Mon, 20 2020 12:27:24 GMT CSeq
: 101 INVITE
User-Agent.device.cisco.contact
: Cisco-;uccm.namesip! com="BOTKONEV"
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "1117" ;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco- srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info : x-cisco-conference
Content-Length: 659
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 5524 0 IN IP4 10.0.0.101
s=SIP Call
b=AS: 1536
t=0 0
a=cisco-mari:v1
a=cisco-mari-rate
m=audio 17102 RTP/AVP 18 111 101
c=IN IP4 10.0.0.101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap :111 x-ulpfecuc/8000
a= extmap:14/sendrecv protocols.cisco.com/timestamp#100us
a=fmtp:111 max_esel=1420;m=8;max_n=32;FEC_ORDER=FEC_SRTP
a=rtpmap:101 telephone- event/8000
a=fmtp:101 0-16
a=sendrecv
m=application 58302 UDP/UDT/IX *
c=IN IP4 10.0.0.101
a=ixmap:11 xccp
a=setup:actpass
a=fingerprint:sha-256 5F :41:DF:50:7F:07:28:80:40:12:F9:DF:45:C8:6A:2B:3F:9F:15:63:D6:9D:54:2D:E7:5D :7C:F5:7C:48:76:A8
a=sendrecv

Timestamp: 3678107247759
UTC Timestamp:3678107247759
Source Filename: ccm00000145.txt.gz

Answer 1

You don't have a codec in INVITE 711.

a=rtpmap:18 G729/8000
a=rtpmap:111 x-ulpfecuc/8000
a=rtpmap:101 telephone-event/8000

Must be an entry by type
a=rtpmap:8 PCMA/8000