[asterisk-users] Asterisk Inbound Problem

Tue Feb 20 11:37:39 MST 2007

Well, could be the fact provider not pushing as g729 or someting else.

Can you set debug 999 and set verbose 999
then redump that ? you are missing the before the answer part also..

Also try G711 first then work your way to other codecs

On 2/20/07, Rajeev Natarajan <twogigbox at gmail.com> wrote:
>
> Am working with Arun on this project - here's a longer description of the
> problem:
>
> We've been fighting with our service provider on this issue - we seem to
> be getting a BYE just after we receive an ACK. They claim that it is an
> asterisk issue! The service provider provides only IP based authentication
> for inbound.
>
> We have used username-password based authentication with the same setup
> with *no problems*  whatsoever!
>
> If we configure an Audiocodes MEdia gateway to receive the calls, there is
> no issue - so there's something that asterisk is doing? or asterisk-Provider
> gateway combo?
>
> In our efforts to mask IP, I have used PROVIDER-IP for the IP of my
> service provider (host) and AsteriskIP to indicate my asterisk server
>
> sip.conf
> [PROVIDER]
> type=peer
> disallow=all
> allow=g729
> context=default
> host=xxxx
> fromuser=y.y.y.y
> port=5060
> insecure=very
> canreinvite=no
> nat=yes
> qualify=yes
>
> CLI output:
>
>    -- Executing Answer("SIP/PROVIDER-IP-b7a076a8", "") in new stack
> We're at 124.7.195.102 port 47698
> Adding codec 0x100 (g729) to SDP
> Reliably Transmitting (NAT) to PROVIDER-IP:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> PROVIDER-IP:5060;branch=z9hG4bK6bd3121243ee9f936c4aeb96d6785b7a;received=PROVIDER-IP
>
> From: <sip:919444072925 at PROVIDER-IP>;tag=3380976385-794612
> To: <sip:8009422419 at 192.168.11.2:5060>;tag=as52d36855
> Call-ID: 45018316-3380976385-794578 at nextone-msw.mydomain.com
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:8009422419@'AsteriskIP'>
> Content-Type: application/sdp
> Content-Length: 183
>
> v=0
> o=root 2172 2172 IN IP4 AsteriskIP
> s=session
> c=IN IP4 AsteriskIP
> t=0 0
> m=audio 47698 RTP/AVP 18
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=silenceSupp:off - - - -
>
> ---
>
>  -- Executing Playback("SIP/PROVIDER-IP-b7a076a8", "park") in new stack
>     -- Playing 'park' (language 'en')
> AstSQL*CLI>
> <-- SIP read from PROVIDER-IP:5060:
> ACK sip:8009422419 at AsteriskIP SIP/2.0
> Max-Forwards: 5
> To: <sip:8009422419 at 192.168.11.2:5060>;tag=as52d36855
> From: <sip:919444072925 at PROVIDER-IP>;tag=3380976385-794612
> Contact: <sip:919444072925 at PROVIDER-IP:5060>
> Call-ID: 45018316-3380976385-794578 at nextone-msw.mydomain.com
> CSeq: 1 ACK
> Via: SIP/2.0/UDP 221.135.102.100:5060;branch=z9hG4bK02505a1dcc5937d9a648eebc0052b422
> Content-Length: 0
>
>
> --- (9 headers 0 lines) ---
> AstSQL*CLI>
> <-- SIP read from PROVIDER-IP:5060:
> BYE sip:8009422419 at AsteriskIP SIP/2.0
> Max-Forwards: 5
> To: <sip:8009422419 at 192.168.11.2:5060>;tag=as52d36855
> From: <sip:919444072925 at PROVIDER-IP>;tag=3380976385-794612
> Contact: <sip:919444072925 at PROVIDER-IP:5060>
> Call-ID: 45018316-3380976385-794578 at nextone-msw.mydomain.com
> CSeq: 2 BYE
> Via: SIP/2.0/UDP 221.135.102.100:5060
> ;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f
> Content-Length: 0
>
>
> --- (9 headers 0 lines) ---
> Sending to PROVIDER-IP : 5060 (NAT)
> Transmitting (NAT) to PROVIDER-IP:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> PROVIDER-IP:5060;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f;received=PROVIDER-IP
>
> From: <sip:919444072925 at PROVIDER-IP>;tag=3380976385-794612
> To: <sip:8009422419 at 192.168.11.2:5060>;tag=as52d36855
> Call-ID: 45018316-3380976385-794578 at nextone-msw.mydomain.com
> CSeq: 2 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:8009422419 at AsteriskIP>
> Content-Length: 0
>
> ----------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> The following is an ngrep of the traffic for an inbound call - 'U' marks
> the begin of the packet grabbed.
>
>
> U <PROVIDER-IP>:5060 -> <AsteriskIP>:5060
>   INVITE sip:800942xxxx@<AsteriskIP> SIP/2.0..Max-Forwards:
> 5..Session-Expires: 3600;Refresher=uac..Suppor ted: timer..To: <
> sip:800942xxxx at 192.168.11.2:5060>..From:
> <sip:<PROVIDER-IP>>;tag=3380960452-790279..Co ntact:
> <sip:<PROVIDER-IP>:5060>..Remote-Party-Id:
> <sip:<PROVIDER-IP>>;party=calling;screen=no;privacy =off..Call-ID:
> 44786336-3380960452-790235 at nextone-msw.mydomain.com..CSeq: 1 INVITE..Via:
> SIP/2.0/UDP 221. 135.102.100:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4..Allow-Events:
> telephone-event..Content-T ype: application/sdp..Content-Length:
> 206....v=0..o=nextone-msw1 1774 4816 IN IP4 <PROVIDER-IP>..s=sip call..c=IN
> IP4 <PROV-IP-2>..t=0 0..m=audio 18932 RTP/AVP 18 19..a=ptime:20..a=rtpmap:19
> CN/8000..a=fm tp:18 annexb=yes..a=rtpmap:18 G729/8000..
>
>
> #
> U <AsteriskIP>:5060 -> <PROVIDER-IP>:5060
>   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> <PROVIDER-IP>:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4;
> received=<PROVIDER-IP>..From: <sip:<PROVIDER-IP>>;tag=3380960452-790279..To:
> < sip:800942xxxx at 192.168. 11.2:5060>..Call-ID:
> 44786336-3380960452-790235 at nextone-msw.mydomain.com..CSeq: 1
> INVITE..User-Agent: Ast erisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
> REFER, SUBSCRIBE, NOTIFY..Contact: < sip:800942xxxx at AsteriskIP>..Content-Length:
> 0....
>
>
> #
> U <AsteriskIP>:5060 -> <PROVIDER-IP>:5060
>   SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
> <PROVIDER-IP>:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4
> ;received=<PROVIDER-IP>..From:
> <sip:<PROVIDER-IP>>;tag=3380960452-790279..To: < sip:800942xxxx at 192.168.11.2:5060>;tag=as78bcde29..Call-ID:
> 44786336-3380960452-790235 at nextone-msw.mydomain.com ..CSeq: 1 INVITE.
> .User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
> SUBSCRIBE, NOTIFY..Contact: < sip:800942xxxx@<AsteriskIP>>..Content-Length:
> 0....
>
>
>
> #
> U <AsteriskIP>:5060 -> <PROVIDER-IP>:5060
>   SIP/2.0 200 OK..Via: SIP/2.0/UDP
> <PROVIDER-IP>:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4;rece
> ived=<PROVIDER-IP>..From: <sip:<PROVIDER-IP>>;tag=3380960452-790279..To: <
> sip:800942xxxx at 192.168.11.2 :5060>;tag=as78bcde29..Call-ID:
> 44786336-3380960452-790235 at nextone-msw.mydomain.com ..CSeq: 1 INVITE..User
> -Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
> SUBSCRIBE, NOTIFY..Contact: <sip: 800942xxxx@<AsteriskIP>>..Content-Type:
> application/sdp..Content-Length: 182....v=0..o=root 2156 2156 IN IP4
> <AsteriskIP>..s=session..c=IN IP4 <Asterisk>..t=0 0..m=audio 5676 RTP/AVP
> 18..a=rtpmap:18 G729/80 00..a=fmtp:18 annexb=no..a=silenceSupp:off - - - -..
>
>
>
>
> #
> U <PROVIDER-IP>:5060 -> <AsteriskIP>:5060
>   ACK sip:800942xxxx@<AsteriskIP> SIP/2.0..Max-Forwards: 5..To: <
> sip:800942xxxx at 192.168.11.2:5060>;tag=as7 8bcde29..From:
> <sip:<PROVIDER-IP>>;tag=3380960452-790279..Contact:
> <sip:<PROVIDER-IP>:5060>..Call-ID:
> 44786336-3380960452-790235 at nextone-msw.mydomain.com..CSeq: 1 ACK..Via:
> SIP/2.0/UDP <PROVIDER-IP>:5060;
> branch=z9hG4bK74ac10cb8c5d89375bf77d4aaa15fcea..Content-Length: 0....
>
>
> #
> U <PROVIDER-IP>:5060 -> <AsteriskIP>:5060
>   BYE sip:800942xxxx@<AsteriskIP> SIP/2.0..Max-Forwards: 5..To: <
> sip:800942xxxx at 192.168.11.2:5060>;tag=as7 8bcde29..From:
> <sip:<PROVIDER-IP>>;tag=3380960452-790279..Contact:
> <sip:<PROVIDER-IP>:5060>..Call-ID:
> 44786336-3380960452-790235 at nextone-msw.mydomain.com..CSeq: 2 BYE..Via:
> SIP/2.0/UDP <PROVIDER-IP>:5060;
> branch=z9hG4bK610e4f29ad9631a0065d4b23dc6c8c27..Content-Length: 0....
>
>
> #
> U <AsteriskIP>:5060 -> <PROVIDER-IP>:5060
>   SIP/2.0 200 OK..Via: SIP/2.0/UDP
> <PROVIDER-IP>:5060;branch=z9hG4bK610e4f29ad9631a0065d4b23dc6c8c27;rece
> ived=<PROVIDER-IP>..From: <sip:<PROVIDER-IP>>;tag=3380960452-790279..To: <sip:800942xxxx at 192.168.11.2:5060>;tag=as78bcde29..Call-ID:
> 44786336-3380960452-790235 at nextone-msw.mydomain.com..CSeq: 2 BYE..User-Ag
> ent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
> SUBSCRIBE, NOTIFY..Contact: <sip:8009 422419@<AsteriskIP>>..Content-Length:
> 0....
>
>
> --------------------------------------------------------------------------------------------------------------------------------
>
> Any help appreciated
> Thanks!
> Rajeev
>
> On 2/20/07, Arun Kumar < arunvoip at gmail.com> wrote:
> >
> > Instead of forwarding to IAX softphone if I'll play some music same
> > thing is happening in this case also.
> >
> > On 2/20/07, Mark Phillips < g7ltt at g7ltt.com> wrote:
> > >
> > > Without seeing your config files my guess would be that this is
> > > something to do with a bad codec negotiation.
> > >
> > > I'd bet that your IAX phone is using ulaw and your DID provider is
> > > using
> > > something else like G729.
> > >
> > > Mark
> > >
> > > On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote:
> > > > HI
> > > >
> > > > I've configred an Incoming DID in my asterisk and when I call from
> > > > outside I see call is coming to my Asterisk server and then from
> > > > asterisk it rings on a particulat exten but when I pickup the call
> > > the
> > > > call get disconnect immediate and on the other end it keep trying
> > > > (ringing).
> > > >
> > > > here is my exten.conf:
> > > >
> > > > exten => _80.,1,Answer
> > > > exten => _80.,2,Dial(IAX2/2001)
> > > >
> > > > did starts with 80 and any call comes for my number they are sending
> > > > to my asterisk IP.
> > > >
> > > > thanks
> > > >
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