[asterisk-users] Asterisk Inbound Problem
Arun Kumar
arunvoip at gmail.com
Wed Feb 21 03:19:54 MST 2007
My service provider only supports g729 and I tried what you have mentioned
here but same thing is happening here. Is there any why that I can see which
codec my service provider is pushing when I'm receiving call on my asterisk
server. When call comes comes to my server and then I type show g729 it
shows 0/0 out of 15 lic.
thanks
arun
On 2/21/07, Mike Lynchfield <theclubvoip at gmail.com> wrote:
>
> 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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>
> --
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