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[bilmar_gh at yahoo.com]

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: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

[asterisk_list at earth...]

On Wednesday 20 Jan 2016, bilal ghayyad wrote:

[bilmar_gh at yahoo.com]

Hello;


Thanks a lot for your kindly reply.
Actually the alaw is enabled at asterisk but what I got to know from the other side that they only enabled ulaw. Below is my asterisk sip configuration for the sip trunk. Please advise.


[user_name]
type=peer
host=Provider_IP_Address
port=5083
context=trunkinbound
disallow=all
allow = ulaw,alaw,gsm
call-limit = 256
insecure = port,invite
trunkstyle = provider
transport = udp
dtmfmode = rfc2833
remoteregister = yes
cbcallerid = 22021782
qualify = yes

srtpcapable = no


Regards
Bilal



On Wednesday, January 20, 2016 2:50 PM, A J Stiles <asterisk_list@earthshod.co.uk> wrote:



On Wednesday 20 Jan 2016, bilal ghayyad wrote:> 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> ..... [stuff deleted] .....> [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=Pro> vider_IP_Address From: "1828444" <sip:1828444@c4.gw (1828444@c4.gw)>;tag=rrZpHF51Z7a6D To:> <sip:22021782@Asterisk_IP_Address ([email]22021782@Asterisk_IP_Address[/email]):5060>;tag=as5d16dbaf Call-ID:> 6bba7f72-3874-1234-0b95-0090fb3d96e0-UASession-CX9lUh8LWc-UASession-kOSBFq> loi5 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL,> OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported:> replaces Content-Length: 0 <------------>
"488 Not acceptable here" usually means that negotiation failed for want of any mutually-supported codec. Make sure that you have "alaw", which is the native format used by the PSTN in civilised countries (and therefore, there is little need to use anything else unless you know you will never want PSTN connectivity), enabled at your end.Can you run this command and post the output? (It should all be on one line, but my mail client or yours may have eaten it)$ awk '/[[]|allow/&&!/^[ \t]*;/{printf "%6d:%s\n",NR, $0}' /etc/asterisk/sip.confThis will look for [section headers] in square brackets and lines containing "allow" (which also will catch "disallow") that are not commented out, in your SIP configuration, and print them out with line numbers.-- AJSNote: Originating address only accepts e-mail from list! If replying off-list, change address to asterisk1list at earthshod dot co dot uk .