asterisk users - Jan 2016 - 488 Not acceptable here

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 at
Asterisk_IP_Address:5060 SIP/2.0?Via: SIP/2.0/UDP
Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1?From:
"1828444" <sip:1828444 at c4.gw>;tag=rrZpHF51Z7a6D?To:
<sip:22021782 at 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 at
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 at c4.gw>;tag=rrZpHF51Z7a6D?To:
<sip:22021782 at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160120/b57d0a15/attachment.html>

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 at
c4.gw>;tag=rrZpHF51Z7a6D To:
> <sip:22021782 at Asterisk_IP_Address: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.conf

This 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.


-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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=peerhost=Provider_IP_Addressport=5083context=trunkinbounddisallow=allallow
= ulaw,alaw,gsmcall-limit = 256 ?insecure = port,invitetrunkstyle =
providertransport = udp ?dtmfmode = rfc2833remoteregister = yescbcallerid =
22021782qualify = yessrtpcapable = no
RegardsBilal 

    On Wednesday, January 20, 2016 2:50 PM, A J Stiles <asterisk_list at
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 at
c4.gw>;tag=rrZpHF51Z7a6D To:
> <sip:22021782 at Asterisk_IP_Address: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.conf

This 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.


-- 
AJS

Note:? Originating address only accepts e-mail from list!? If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .


  
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160120/a0030085/attachment.html>