[asterisk-users] Asterisk Inbound Problem

Tue Feb 20 09:14:27 MST 2007

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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