Hello List;
I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I
am getting 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:[email protected]>;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:[email protected]>;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
<------------>
RegardsBilal
--
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