Re: [asterisk-users] 488 Not acceptable here

2016-01-20 Thread bilal ghayyad 
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 
 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" ;tag=rrZpHF51Z7a6D To:
> ;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.


-- 
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Note:  Originating address only accepts e-mail from list!  If replying off-
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Re: [asterisk-users] 488 Not acceptable here

2016-01-20 Thread A J Stiles 
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" ;tag=rrZpHF51Z7a6D To:
> ;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 .

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[asterisk-users] 488 Not acceptable here

2016-01-20 Thread bilal ghayyad 
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" 
;tag=rrZpHF51Z7a6D To: 
 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: 
 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] F

[asterisk-users] 488 Not acceptable here

2010-10-22 Thread Jian Gao 
I am helping a friend on one of his sip trunk and couldn't find the way 
to resolve his problem.

His asterisk's problem is like this:
0. When incoming call to one of his sip trunk, Asterisk reply with "488 
Not acceptable here". So the call get dropped.
1. Recently upgraded Elastix with Asterisk 1.4.33
2. Was working fine before the upgrade
3. There are total 4 SIP trunks connected to different providers. All 
others works fine.
4. All codecs are allowed.
5. I setup his account on my Asterisk as a SIP trunk, both incoming and 
outgoing call work fine. (So it is not his provider's problem)
6. I checked his FreePBX style multi sip*.conf files and all seem correct.

So what can I do to find out where went wrong on this sip trunk?

Thanks.

Jian


Hers is the debug out put:


<--- SIP read from 208.65.xxx.xxx:5060 --->
INVITE sip:160428xx...@192.168.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP 
208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-ad239907e0915d0b-1---d8754z-;rport
Via: SIP/2.0/UDP 
208.65.xxx.xxx:5061;branch=z9hG4bK-pcerhxpz5hr4addh;rport=5061
Max-Forwards: 69
Record-Route: 
Contact: "Anonymous"
To: 
From: "CID NAME";tag=kvspovbxperbwmfk.o
Call-ID: 12904...@208.xx.xx.xx~o
CSeq: 493 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
cisco-GUID: 4084071434-3712422367-2859401243-560159692
h323-conf-id: 4084071434-3712422367-2859401243-560159692
Content-Length: 109

v=0
o=Sippy 153068680 0 IN IP4 74.205.xxx.xxx
s=-
t=0 0
m=audio 34772 RTP/AVP 0
c=IN IP4 74.205.xxx.xxx

<->
--- (17 headers 6 lines) ---
Sending to 208.65.xxx.xxx : 5060 (NAT)
Using INVITE request as basis request - 12904...@208.xx.xx.xx~o
Found peer 'vsp06'
Found RTP audio format 0

<--- Reliably Transmitting (NAT) to 208.65.xxx.xxx:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 
208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-ad239907e0915d0b-1---d8754z-;received=208.65.xxx.xxx;rport=5060
Via: SIP/2.0/UDP 
208.65.xxx.xxx:5061;branch=z9hG4bK-pcerhxpz5hr4addh;rport=5061
From: "CID NAME";tag=kvspovbxperbwmfk.o
To: ;tag=as40501684
Call-ID: 12904...@208.xx.xx.xx~o
CSeq: 493 INVITE
User-Agent: 000e082e83c7Linksys/SPA2102-5.2.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<>
Scheduling destruction of SIP dialog '12904...@208.xx.xx.xx~o' in 6400 
ms (Method: INVITE)

<--- SIP read from 208.65.xxx.xxx:5060 --->
ACK sip:160428xx...@192.168.1.83:5060 SIP/2.0
Via: SIP/2.0/UDP 
208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-ad239907e0915d0b-1---d8754z-;rport
Max-Forwards: 70
To: ;tag=as40501684
From: "CID NAME";tag=kvspovbxperbwmfk.o
Call-ID: 12904...@208.xx.xx.xx~o
CSeq: 493 ACK
Content-Length: 0


<->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '12904...@208.xx.xx.xx~o' Method: ACK
Really destroying SIP dialog 
'2ded46615e78d7992c15bea726fae...@127.0.0.1' Method: REGISTER
astpbx*CLI> sip set debug off
SIP Debugging Disabled

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Re: [asterisk-users] 488 Not Acceptable Here

2010-07-23 Thread Jamie A. Stapleton 
A packet capture would be most useful.  Then, you could review your SDP with 
your provider.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Beak
Sent: Friday, July 23, 2010 7:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 488 Not Acceptable Here

Hi,

I'm having real difficulty in getting calls to go through with 
Asterisk.  I've managed to check that my SIP connection is made to my 
provider.  Below is an email I received from them:

snipsnipsnip
I am not certain of the reason for rejection but it has to do with the 
SDP,  it does not seem to be a codec issue, the response is as you have 
seen:

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 
192.168.0.14;received=172.28.20.106;branch=z9hG4bK42d2ea03;rport=60017
From: "Andy" ;tag=as5c784926
To: ;tag=SD24jn898-4C46B8A2-5688CB2-0ADE2C09
Call-ID: 32d506cd3489aa81031937f467ef6...@192.168.0.14
CSeq: 102 INVITE
Reason: Q.850 ;cause=127 ;text="Interworking, unspecified"
Content-Length: 0

There looks to be a non-standard element in your SDP that is not 
supported by any of the networks.
snipsnipsnip

Which configuration file is possibly incorrect in this scenario?

What dumps are likely to be useful to me?

Thanks,
  Andy

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[asterisk-users] 488 Not Acceptable Here

2010-07-23 Thread Andy Beak 
Hi,

I'm having real difficulty in getting calls to go through with 
Asterisk.  I've managed to check that my SIP connection is made to my 
provider.  Below is an email I received from them:

snipsnipsnip
I am not certain of the reason for rejection but it has to do with the 
SDP,  it does not seem to be a codec issue, the response is as you have 
seen:

SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 
192.168.0.14;received=172.28.20.106;branch=z9hG4bK42d2ea03;rport=60017
From: "Andy" ;tag=as5c784926
To: ;tag=SD24jn898-4C46B8A2-5688CB2-0ADE2C09
Call-ID: 32d506cd3489aa81031937f467ef6...@192.168.0.14
CSeq: 102 INVITE
Reason: Q.850 ;cause=127 ;text="Interworking, unspecified"
Content-Length: 0

There looks to be a non-standard element in your SDP that is not 
supported by any of the networks.
snipsnipsnip

Which configuration file is possibly incorrect in this scenario?

What dumps are likely to be useful to me?

Thanks,
  Andy

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Re: [asterisk-users] 488 Not acceptable here sent by Asterisk - SIPdebug follows

2006-09-21 Thread Dinesh Nair 




On 09/20/06 15:06 Dinesh Nair said the following:



On 09/19/06 16:59 Steve Langstaff said the following:


I wonder whether you are experiencing the following bug (since the SIP
INVITE has a multipart SDP body):
http://bugs.digium.com/view.php?id=7124&nbn=4



thanks for the link,

however, on 18th may 2006, kpfleming's note says, "This should be fixed 
in both 1.2 branch and trunk," and i'm using 1.2.12.1 which was just 
released this week. looking thru the current chan_sip.c code, it does 
seem like kevin's modified patch has been committed into the branch i'm 
using, so this isnt the problem.


[am cc'ing reply into -dev because a bug report was opened on this at 
http://bugs.digium.com/view.php?id=8010 with a patch provided]


i've managed to track this down to a loop which terminated prematurely in 
find_sdp() in chan_sip.c. this bug would have prevented proper handling of 
multipart/mixed content types due to the loop which searches for the end of 
the block ending prematurely and setting req->sdp_start > req->sdp_end.


i've provided patches for trunk and 1.2.x in the bug entry, as i think this 
should also be committed to 1.2.x.


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Re: [asterisk-users] 488 Not acceptable here sent by Asterisk - SIPdebug follows

2006-09-20 Thread Dinesh Nair 



On 09/19/06 16:59 Steve Langstaff said the following:

I wonder whether you are experiencing the following bug (since the SIP
INVITE has a multipart SDP body):
http://bugs.digium.com/view.php?id=7124&nbn=4


thanks for the link,

however, on 18th may 2006, kpfleming's note says, "This should be fixed in 
both 1.2 branch and trunk," and i'm using 1.2.12.1 which was just released 
this week. looking thru the current chan_sip.c code, it does seem like 
kevin's modified patch has been committed into the branch i'm using, so 
this isnt the problem.


--
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+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  |
| done; done  |
+=+
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RE: [asterisk-users] 488 Not acceptable here sent by Asterisk - SIPdebug follows

2006-09-19 Thread Steve Langstaff 
I wonder whether you are experiencing the following bug (since the SIP
INVITE has a multipart SDP body):
http://bugs.digium.com/view.php?id=7124&nbn=4


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Dinesh Nair
> Sent: 19 September 2006 06:54
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] 488 Not acceptable here sent by 
> Asterisk - SIPdebug follows
> 
> 
> the situation
> 
> Asterisk <-- SIP ---> SIPGW <--- SIP Phone
> 
> SIP Phone is trying to call asterisk dialplan:
> 
> exten => 0224577501,1,Answer()
> exten => 0224577501,2,Playback(demo-instruct)
> exten => 0224577501,3,Hangup()
> 
> however, asterisk 1.2.12.1 (on FreeBSD 6.1) sends back a "488 
> Not acceptable here" with a CLI message of
> 
> WARNING[162]: chan_sip.c:3529 process_sdp: Insufficient 
> information for SDP (m = '', c = '')
> 
> 
> it seems to be dropping out in process_sdp() because it can't 
> find the m= 
> or the c=. this is a little odd, so am wondering if this has 
> triggered some 
> edge case in find_sdp(), get_sdp() or get_sdp_iterate(). i've 
> been poring 
> thru the code (as the box is remote, and i cant duplicate it 
> locally), but 
> can't find exactly where in chan_sip.c its borking.
> 
> any advice would be much appreciated.
> 
> the SIP debug is attached below:
> 
> (10.14.32.179 is the SIPGW, 10.14.32.164 is Asterisk)
> 
>  >>> begin sip debug
> <-- SIP read from 10.14.32.179:5060:
> INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.14.32.179:5060
> Via: SIP/2.0/UDP 10.14.32.189:5060
> Record-Route: 
> Supported: replaces
> User-Agent: SIP201 (lp201_sip0423.bin)
> Contact: 
> From:  
> ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> History-Info: ;index 1
> Content-Type: multipart/mixed;boundary=unique-boundary
> Content-Length: 474
> 
> --unique-boundary
> Content-Type: application/sdp
> 
> v=0
> o=SIP201 12367 0 IN IP4 10.14.32.189
> s=SIP201 Session
> i=Audio Session
> c=IN IP4 10.14.32.189
> t=0 0
> m=audio 16384 RTP/AVP 4 18 0 8 18
> a=rtpmap:4 G723/8000/1
> a=rtpmap:18 G729/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:18 G729/8000/1
> 
> --unique-boundary
> Content-Type: application/isup;version=Indonesia
> Content-Transfer-Encoding: binary
> 
> 
> --- (14 headers 21 lines)---
> Using INVITE request as basis request - 
> [EMAIL PROTECTED]
> Sending to 10.14.32.179 : 5060 (non-NAT)
> Found peer 'RISTI'
> Sep 19 09:38:53 WARNING[162]: chan_sip.c:3529 process_sdp: 
> Insufficient 
> information for SDP (m = '',
>   c = '')
> Transmitting (no NAT) to 10.14.32.179:5060:
> SIP/2.0 488 Not acceptable here
> Via: SIP/2.0/UDP 10.14.32.179:5060;received=10.14.32.179
> Via: SIP/2.0/UDP 10.14.32.189:5060
> From:  
> ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
> To: ;tag=as5a7aa73d
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> User-Agent: QubeTalk ECS
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: 
> Content-Length: 0
> 
> 
> ---
> Destroying call '[EMAIL PROTECTED]'
> suria*CLI>
> <-- SIP read from 10.14.32.179:5060:
> ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.14.32.179:5060
> Via: SIP/2.0/UDP 10.14.32.189:5060
> Record-Route: 
> Contact: 
> User-Agent: SIP201 (lp201_sip0423.bin)
> From:  
> ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
> To:  ;tag=as5a7aa73d
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 ACK
> Content-Length:0
> 
> 
> --- (11 headers 0 lines)---
> Destroying call '[EMAIL PROTECTED]'
>  >>> end sip debug
> 
> 
> -- 
> Regards,   /\_/\   "All dogs go to heaven."
> [EMAIL PROTECTED](0 0)   
> http://www.openmalaysiablog.com/
> +==oOO--(_)--OOo==
> +
> | for a in past present future; do
> |
> |   for b in clients employers associates relatives 
> neighbours pets; do   |
> |   echo "The opinions here in no way reflect the opinions of 
> my $a $b."  |
> | done; done  
> |
> +=
> +
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[asterisk-users] 488 Not acceptable here sent by Asterisk - SIP debug follows

2006-09-18 Thread Dinesh Nair 


the situation

Asterisk <-- SIP ---> SIPGW <--- SIP Phone

SIP Phone is trying to call asterisk dialplan:

exten => 0224577501,1,Answer()
exten => 0224577501,2,Playback(demo-instruct)
exten => 0224577501,3,Hangup()

however, asterisk 1.2.12.1 (on FreeBSD 6.1) sends back a "488 Not 
acceptable here" with a CLI message of


WARNING[162]: chan_sip.c:3529 process_sdp: Insufficient information for SDP 
(m = '', c = '')



it seems to be dropping out in process_sdp() because it can't find the m= 
or the c=. this is a little odd, so am wondering if this has triggered some 
edge case in find_sdp(), get_sdp() or get_sdp_iterate(). i've been poring 
thru the code (as the box is remote, and i cant duplicate it locally), but 
can't find exactly where in chan_sip.c its borking.


any advice would be much appreciated.

the SIP debug is attached below:

(10.14.32.179 is the SIPGW, 10.14.32.164 is Asterisk)

>>> begin sip debug
<-- SIP read from 10.14.32.179:5060:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.14.32.179:5060
Via: SIP/2.0/UDP 10.14.32.189:5060
Record-Route: 
Supported: replaces
User-Agent: SIP201 (lp201_sip0423.bin)
Contact: 
From:  ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
History-Info: ;index 1
Content-Type: multipart/mixed;boundary=unique-boundary
Content-Length: 474

--unique-boundary
Content-Type: application/sdp

v=0
o=SIP201 12367 0 IN IP4 10.14.32.189
s=SIP201 Session
i=Audio Session
c=IN IP4 10.14.32.189
t=0 0
m=audio 16384 RTP/AVP 4 18 0 8 18
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1

--unique-boundary
Content-Type: application/isup;version=Indonesia
Content-Transfer-Encoding: binary


--- (14 headers 21 lines)---
Using INVITE request as basis request - 
[EMAIL PROTECTED]

Sending to 10.14.32.179 : 5060 (non-NAT)
Found peer 'RISTI'
Sep 19 09:38:53 WARNING[162]: chan_sip.c:3529 process_sdp: Insufficient 
information for SDP (m = '',

 c = '')
Transmitting (no NAT) to 10.14.32.179:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.14.32.179:5060;received=10.14.32.179
Via: SIP/2.0/UDP 10.14.32.189:5060
From:  ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
To: ;tag=as5a7aa73d
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: QubeTalk ECS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: 
Content-Length: 0


---
Destroying call '[EMAIL PROTECTED]'
suria*CLI>
<-- SIP read from 10.14.32.179:5060:
ACK sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.14.32.179:5060
Via: SIP/2.0/UDP 10.14.32.189:5060
Record-Route: 
Contact: 
User-Agent: SIP201 (lp201_sip0423.bin)
From:  ;tag=a0e20bd-13c4-132e6-4aed8ab-2ea8
To:  ;tag=as5a7aa73d
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
Content-Length:0


--- (11 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
>>> end sip debug


--
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)   http://www.openmalaysiablog.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  |
| done; done  |
+=+
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Re: [Asterisk-Users] 488 Not acceptable here

2005-10-14 Thread Eroc Wieling 

SIP DEBUG would show the information in the INVITE.

Obelix wrote:


Quoting Ray Van Dolson <[EMAIL PROTECTED]>:

How can you determine which codecs are acceptable to them?

Do they have a way of indicating it?


 


Perhaps they dont' like the codec you're offering in your INVITE message?
   



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Re: [Asterisk-Users] 488 Not acceptable here

2005-10-14 Thread Eroc Wieling 

Obelix wrote:


I have been receiving a lot these  488 "Not Acceptable Here" from a number of
providers. What could the problem be?

What is the most common cause of this message?
 



In my experience, that is caused by one side requireing a codec that the 
other side does not support.

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Re: [Asterisk-Users] 488 Not acceptable here

2005-10-14 Thread BJ Weschke 
 Yes. They tell you what's acceptable to them inside the SIP messages you're trading back and forth on the call setup. 
 
 You can use "sip debug" within Asterisk to get a closer look at those messages. 
On 10/14/05, Obelix <[EMAIL PROTECTED]> wrote:
Quoting Ray Van Dolson <[EMAIL PROTECTED]>:How can you determine which codecs are acceptable to them?
Do they have a way of indicating it?> Perhaps they dont' like the codec you're offering in your INVITE message?>> Ray>> On Fri, Oct 14, 2005 at 01:36:17PM +, Obelix wrote:
> >> > I have been receiving a lot these  488 "Not Acceptable Here" from a number> of> > providers. What could the problem be?> >> > What is the most common cause of this message?
> ___> --Bandwidth and Colocation sponsored by Easynews.com -->> Asterisk-Users mailing list> 
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Re: [Asterisk-Users] 488 Not acceptable here

2005-10-14 Thread Obelix 
Quoting Ray Van Dolson <[EMAIL PROTECTED]>:

How can you determine which codecs are acceptable to them?

Do they have a way of indicating it?


> Perhaps they dont' like the codec you're offering in your INVITE message?
>
> Ray
>
> On Fri, Oct 14, 2005 at 01:36:17PM +, Obelix wrote:
> >
> > I have been receiving a lot these  488 "Not Acceptable Here" from a number
> of
> > providers. What could the problem be?
> >
> > What is the most common cause of this message?
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Re: [Asterisk-Users] 488 Not acceptable here

2005-10-14 Thread Ray Van Dolson 
Perhaps they dont' like the codec you're offering in your INVITE message?

Ray

On Fri, Oct 14, 2005 at 01:36:17PM +, Obelix wrote:
> 
> I have been receiving a lot these  488 "Not Acceptable Here" from a number of
> providers. What could the problem be?
> 
> What is the most common cause of this message?
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[Asterisk-Users] 488 Not acceptable here

2005-10-14 Thread Obelix 

I have been receiving a lot these  488 "Not Acceptable Here" from a number of
providers. What could the problem be?

What is the most common cause of this message?



This message was sent using IMP, the Internet Messaging Program.

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RE: [Asterisk-Users] 488 Not Acceptable Here

2005-06-15 Thread Nabeel Jafferali 
> The message is generated directly by the called Sipura/PAP2.

No, if you read the sip debug carefully, you would see Asterisk is
transmitting 488 Not Acceptable Here. If you mean the destination device,
that's not possible since the user was calling an echo test.

> This message is very likely due to a codec issue (ie, the called unit
> was instructed to use G279 but it had already one call setup with G729),
> or the called unit was in the process of setting up a call and had no
> available G279 codec for the second call.  ( the Sipuras/PAP2 reserve
> G729 during call setup even though it might end up using G711).  The
> codec is only released once the call is set up.

I know that, which is why the sip.conf entry is set to allow=g729 and
allow=ulaw.

Any other ideas?

--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990

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Re: [Asterisk-Users] 488 Not Acceptable Here

2005-06-14 Thread Andres 



Nabeel Jafferali wrote:


I have a whole bunch of remote devices connected to my Asterisk box,
including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only
rolled out recently and I am having a problem that is intermittent and
inconsistent.

It happens to some users but not other users on the same ISP. It happens to
users in 2 different countries where the Internet setup (NAT issues) are
completely different. It happens to some users intermittently and some users
on every call. Note all the PAP2-NAs are running the same and latest
firmware.

Am I missing something completely obvious? Is there a way to see why
Asterisk is sending 488 (i.e. what is not acceptable?). sip debug peer and
sip.conf is below:
 


The message is generated directly by the called Sipura/PAP2.

This message is very likely due to a codec issue (ie, the called unit 
was instructed to use G279 but it had already one call setup with G729), 
or the called unit was in the process of setting up a call and had no 
available G279 codec for the second call.  ( the Sipuras/PAP2 reserve 
G729 during call setup even though it might end up using G711).  The 
codec is only released once the call is set up.


--
Andres
Network Admin
http://www.telesip.net


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[Asterisk-Users] 488 Not Acceptable Here

2005-06-14 Thread Nabeel Jafferali 
I have a whole bunch of remote devices connected to my Asterisk box,
including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only
rolled out recently and I am having a problem that is intermittent and
inconsistent.

It happens to some users but not other users on the same ISP. It happens to
users in 2 different countries where the Internet setup (NAT issues) are
completely different. It happens to some users intermittently and some users
on every call. Note all the PAP2-NAs are running the same and latest
firmware.

Am I missing something completely obvious? Is there a way to see why
Asterisk is sending 488 (i.e. what is not acceptable?). sip debug peer and
sip.conf is below:

Sip read: 
INVITE sip:[EMAIL PROTECTED] SIP/2.0
v: SIP/2.0/UDP 216.252.155.227:11288;branch=z9hG4bK-8079155a;rport
f: Bode Dar ;tag=faf9295d7d8c85e6o0
t: 
i: [EMAIL PROTECTED]
CSeq: 103 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="3255",realm="asterisk",nonce="",uri="sip:[EMAIL PROTECTED]",
algorithm=MD5,response=""
m: Bode Dar 
Expires: 240
User-Agent: Linksys/PAP2-2.0.14(LSVa)
l: 402
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
k: x-sipura
c: application/sdp

v=0
o=- 13600 13600 IN IP4 216.252.155.227
s=-
c=IN IP4 216.252.155.227
t=0 0
m=audio 14274 RTP/AVP 18 0 2 8 96 97 98 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

15 headers, 18 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 216.252.155.227:11288;branch=z9hG4bK-8079155a
From: Bode Dar ;tag=faf9295d7d8c85e6o0
To: ;tag=as0be177dc
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0

 to 216.252.155.227:11288


The user's sip.conf entry is:

[3255]
type=friend
username=3255  
secret=**  
accountcode=3255 
callerid="Bode Dar" <>
context=clients-int
host=dynamic 
qualify=yes
nat=yes
disallow=all
allow=g729
allow=ulaw


--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990



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Re: [Asterisk-Users] "488 not acceptable here" message

2003-11-12 Thread Anton L. Kapela 
[EMAIL PROTECTED] said:
> I'm creating a test environment for Asterisk.  I have Asterisk running
> on a
> PC with only a NIC card, No FXO, FXS, TDM cards.  I have two Cisco
> 7960
> phones setup for SIP.  Within Asterisk, the SIP SHOW PEERS, shows the
> phones.  They don't appear under SIP SHOW REGISTRY.  When I call phone
> 2
> from phone 1, I get a message stating it is from Phone 2, stating, Got
> SIP
> Response 488 "Not Acceptable Here" back from 167.131.14.26.  Can
> someone
> point me in the direction to look for the trouble?

You'll want to try forcing allowed codecs in your sip.conf to
something like:

disallow=all
allow=ulaw

Last I experienced any 4xx responses the issue for me was codec
selection (rather, problems with selection). In my own searches, I
didn't find any record of this issue being clearly explained or
related to codecs. Hopefully Google catches this.

-tk
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[Asterisk-Users] "488 not acceptable here" message

2003-11-12 Thread Robert . J . TESCH 
Title: "488 not acceptable here" message





I'm creating a test environment for Asterisk.  I have Asterisk running on a PC with only a NIC card, No FXO, FXS, TDM cards.  I have two Cisco 7960 phones setup for SIP.  Within Asterisk, the SIP SHOW PEERS, shows the phones.  They don't appear under SIP SHOW REGISTRY.  When I call phone 2 from phone 1, I get a message stating it is from Phone 2, stating, Got SIP Response 488 "Not Acceptable Here" back from 167.131.14.26.  Can someone point me in the direction to look for the trouble?