Can somebody tell me why this call fails? It’s FreePBX and in the SIP settings, I have the following codecs enabled: alaw, ulaw, gsm, g729, g722.
It’s the only trunk where i can’t get incoming calls. sipgate.de works with similar settings.
asterisk*CLI> sip set debug on
SIP Debugging enabled
<--- SIP read from UDP:217.10.79.23:5060 --->
INVITE sip:2345678e0@217.250.160.7:5070 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.23;branch=z9hG4bK4b16.3799d5247d92099df88583235ac6df9c.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK4b16.7f0ac0d93afc8a0a0fd46f4f6bf34340.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK4b16.ef88a5849653c90281a7d1edaaac10c4.0
Via: SIP/2.0/UDP 217.10.77.42:5060;branch=z9hG4bK09902388
Record-Route: <sip:217.10.79.23;lr;ftag=as4e754ee7>
Record-Route: <sip:172.20.40.7;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as4e754ee7>
From: "01234567890" <sip:01234567890@sipgate.co.uk>;tag=as4e754ee7
To: <sip:00445678901234@sipgate.co.uk>
Call-ID: 5c93b3f1685d281c1ec0a9d839b5811e@sipgate.co.uk
CSeq: 103 INVITE
Contact: <sip:01234567890@217.10.77.42:5060>
max-forwards: 66
supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Content-Type: application/sdp
Content-Length: 425
v=0
o=root 222132892 222132893 IN IP4 192.168.1.14
s=sipgate VoIP GW
c=IN IP4 192.168.1.14
t=0 0
m=audio 7072 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (18 headers 19 lines) ---
Sending to 217.10.79.23:5060 (NAT)
Sending to 217.10.79.23:5060 (NAT)
Using INVITE request as basis request - 5c93b3f1685d281c1ec0a9d839b5811e@sipgate.co.uk
Found peer 'OUT_SG_2345678e0' for '01234567890' from 217.10.79.23:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 112
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 112
Found audio description format telephone-event for ID 101
[2018-10-12 12:41:04] NOTICE[2156][C-00000273]: chan_sip.c:10888 process_sdp: No compatible codecs, not accepting this offer!
<--- Reliably Transmitting (NAT) to 217.10.79.23:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 217.10.79.23;branch=z9hG4bK4b16.3799d5247d92099df88583235ac6df9c.0;received=217.10.79.23;rport=5060
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK4b16.7f0ac0d93afc8a0a0fd46f4f6bf34340.0
Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK4b16.ef88a5849653c90281a7d1edaaac10c4.0
Via: SIP/2.0/UDP 217.10.77.42:5060;branch=z9hG4bK09902388
From: "01234567890" <sip:01234567890@sipgate.co.uk>;tag=as4e754ee7
To: <sip:00445678901234@sipgate.co.uk>;tag=as1ae5b7e9
Call-ID: 5c93b3f1685d281c1ec0a9d839b5811e@sipgate.co.uk
CSeq: 103 INVITE
Server: FPBX-14.0.3.18(15.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '5c93b3f1685d281c1ec0a9d839b5811e@sipgate.co.uk' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:217.10.79.23:5060 --->
ACK sip:2345678e0@217.250.160.7:5070 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.23;branch=z9hG4bK4b16.3799d5247d92099df88583235ac6df9c.0
Via: SIP/2.0/UDP 172.20.40.7;branch=z9hG4bK4b16.7f0ac0d93afc8a0a0fd46f4f6bf34340.0
From: "01234567890" <sip:01234567890@sipgate.co.uk>;tag=as4e754ee7
To: <sip:00445678901234@sipgate.co.uk>;tag=as1ae5b7e9
Call-ID: 5c93b3f1685d281c1ec0a9d839b5811e@sipgate.co.uk
CSeq: 103 ACK
max-forwards: 66
x-hint: rr-enforced
Content-Length: 0
...
---
Really destroying SIP dialog '6ae252b237497b301b35dbc3795e3a98@217.250.160.7:5070' Method: OPTIONS
asterisk*CLI> sip set debug off
SIP Debugging Disabled