vicidial.org

Hello List;

Hello;

vicidial 2.8-420a, Build 130605-0841 , asterisk 1.4.44-vici, Single Machine

I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I am getting the error:

<--- Reliably Transmitting (no NAT) to Provider_IP_Address:5083 ---> SIP/2.0 488 Not acceptable here

as shown in the following debug, can someone advise me about the solution:

<--- SIP read from Provider_IP_Address:5083 --->
INVITE sip:22021782@Asterisk_IP_Address:5060 SIP/2.0
Via: SIP/2.0/UDP Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1
From: "1828444" <sip:1828444@c4.gw>;tag=rrZpHF51Z7a6D
To: <sip:22021782@Asterisk_IP_Address:5060>
Call-ID: 6bba7f72-3874-1234-0b95-0090fb3d96e0-UASession-CX9lUh8LWc-UASession-kOSBFqloi5
CSeq: 1 INVITE
Max-Forwards: 68
Supported: timer
Unsupported: refer
Allow: INVITE,ACK,CANCEL,BYE,INFO,REGISTER,NOTIFY
Contact: <sip:1828444@Provider_IP_Address:5083;transport=udp>
Content-Length: 729
Content-Type: application/sdp
User-Agent: Netborder SS7 to VoIP Media Gateway 5.1
Allow-Events: talk
Accept: application/sdp
Privacy: none
X-IP-Info: 10.11.11.3

v=0
o=FreeSWITCH 1453083377 1453083378 IN IP4 Provider_IP_Address
s=FreeSWITCH
c=IN IP4 Provider_IP_Address
t=0 0
m=audio 28388 RTP/AVP 8 0 98 9 99 100 18 3 102 101 13
a=rtpmap:98 AMR/8000
a=rtpmap:99 G7221/16000
a=fmtp:99 bitrate=32000
a=rtpmap:100 G726-32/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
m=audio 29684 RTP/AVP 4 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:30
m=audio 21364 RTP/AVP 8 0 98 9 99 100 18 3 102 101 13
a=rtpmap:98 AMR/8000
a=rtpmap:99 G7221/16000
a=fmtp:99 bitrate=32000
a=rtpmap:100 G726-32/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] --- (18 headers 29 lines) ---
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Sending to Provider_IP_Address : 5083 (no NAT)
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Using INVITE request as basis request - 6bba7f72-3874-1234-0b95-0090fb3d96e0-UASession-CX9lUh8LWc-UASession-kOSBFqloi5
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found peer 'gulfnet'
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 8
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 0
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 98
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 9
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 99
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 100
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 18
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 3
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 102
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 101
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 13
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found unknown media description format AMR for ID 98
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found unknown media description format G7221 for ID 99
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format G726-32 for ID 100
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format iLBC for ID 102
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format telephone-event for ID 101
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 4
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 101
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 13
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format telephone-event for ID 101
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 8
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 0
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 98
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 9
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 99
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 100
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 18
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 3
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 102
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 101
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found RTP audio format 13
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found unknown media description format AMR for ID 98
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found unknown media description format G7221 for ID 99
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format G726-32 for ID 100
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format iLBC for ID 102
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37] Found audio description format telephone-event for ID 101
[Jan 18 10:52:37] VERBOSE[2421] logger.c: [Jan 18 10:52:37]

<--- Reliably Transmitting (no NAT) to Provider_IP_Address:5083 --->
SIP/2.0 488 Not acceptable here

Via: SIP/2.0/UDP Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1;received=Provider_IP_Address
From: "1828444" <sip:1828444@c4.gw>;tag=rrZpHF51Z7a6D
To: <sip:22021782@Asterisk_IP_Address:5060>;tag=as5d16dbaf
Call-ID: 6bba7f72-3874-1234-0b95-0090fb3d96e0-UASession-CX9lUh8LWc-UASession-kOSBFqloi5
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


Regards
Bilal

Port 5083? I think you'll need to ask the provider for a suitable sip.conf configuration example. Or ask them why they are refusing the calls.

Try change the codec and will work.