[Asterisk-Users] Why can't sip/200 call sip/202
dbruce
dbruce at bananatel.ca
Sun Jul 24 12:39:30 MST 2005
Marc: My answer is not incorrect... it is incomplete.
The OP stipulated 2 extensions 200 and 202... and provided a sip debug
indicating a call from 200 to 777.
I pointed out the obvious.
If the OP is dialing 202 on the phone, and the phone is dialing 777, then he
needs to look at the dialplan configuration of the phone. If he is dialing
777 on the phone and expecting to reach 202, then he will need to have
translations in the asterisk dialplan. But, the question was "what should I
be looking at?"... Using just the information provided, and the fact that he
is new to asterisk... without any further information... the first thing he
should be looking at is why the phone is trying to reach 777 when he wants
to reach 202... Many new users do not realize the complexity of the SIP
protocol, and only really look at the trace in a general manner... such as:
INVITE
407 Proxy Authentication Required
ACK
INVITE
404 Not Found
ACK
The idea was to provide a clue... not to provide a complete working dialplan
and phone configuration. Providing new users with "the complete package" is
a dis-service to them. They will only learn from thier mistakes and
experiences.. providing clues allows them to expand their experience and
build their confidence... It requires them to look at the details and learn
to analyse them.
Regards,
Derek
----- Original Message -----
From: "Marc Storck" <marc.storck at msnetworks.lu>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Sunday, July 24, 2005 12:53 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
> Derek: you reply is uncorrect. If Angus has the extension 777 in his
> dialplan/extensions.conf which will dial 202. The name of the peer has
> absolutely nothing to do with which number/name he would have to dial.
> Without dialplan he will be unable to call any extension even 202, as
> 202 is only the name of the peer.
>
> Angus: please paste your extensions.conf to pastebin.ca
>
> Regards,
>
> Marc
>
> dbruce wrote:
> > It appears from the debug that extension 200 is trying to call 777, not
> > 202. Your Asterisk server can't find an extension 777 and returns "404
> > not found". That will explain why you can't call extension 777 from
> > extension 200. If you want to call extension 202, you will need to dial
> > 202 on extension 200, not 777.
> >
> > Regards,
> > Derek
> >
> >
> > ----- Original Message -----
> > *From:* Angus Comber <mailto:angus at iteloffice.com>
> > *To:* asterisk-users at lists.digium.com
> > <mailto:asterisk-users at lists.digium.com>
> > *Sent:* Sunday, July 24, 2005 11:51 AM
> > *Subject:* [Asterisk-Users] Why can't sip/200 call sip/202
> >
> > I have 2 sip accounts setup - 200 and 202. If I do sip show peers I
> > get:
> >
> > sip show peers
> > Name/username Host Dyn Nat ACL Mask
> > Port Status
> > 202/202 192.168.0.6 D 255.255.255.255
> > 5060 Unmonitored
> > 201/201 (Unspecified) D 255.255.255.255
> > 5060 Unmonitored
> > 200/200 192.168.0.3 D 255.255.255.255
> > 5060 Unmonitored
> >
> > 200 is a Grandstream GXP200 IP Phone and 202 is a Grandstream BT100
> > IP phone.
> >
> > relevant bit of sip.conf:
> >
> > [200]
> > username=200
> > type=friend
> > secret=1234
> > port=5060
> > nat=never
> > dtmfmode=rfc2833
> > context=default
> > callerid="Angus Comber" <200>
> > host=dynamic
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=g723.1
> > allow=g729
> >
> > [202]
> > username=202
> > type=friend
> > secret=1234
> > port=5060
> > nat=never
> > dtmfmode=rfc2833
> > context=default
> > callerid="Sam Comber" <202>
> > host=dynamic
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=g723.1
> > allow=g729
> >
> >
> > But whenever I try to dial between phones I get this:
> >
> >
> > Sip read:
> >
> > 0 headers, 0 lines
> >
> >
> > Sip read:
> > INVITE sip:777 at 192.168.0.13;user=phone SIP/2.0
> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
> > From: "Angus Comber"
> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > To: <sip:777 at 192.168.0.13;user=phone>
> > Contact: <sip:200 at 192.168.0.3;user=phone>
> > Supported: replaces, timer
> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > CSeq: 45925 INVITE
> > User-Agent: Grandstream GXP2000 1.0.1.9
> > Max-Forwards: 70
> > Allow:
> >
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> > Content-Type: application/sdp
> > Content-Length: 258
> >
> > v=0
> > o=200 8000 8000 IN IP4 192.168.0.3
> > s=SIP Call
> > c=IN IP4 192.168.0.3
> > t=0 0
> > m=audio 5004 RTP/AVP 18 0 8 101
> > a=sendrecv
> > a=rtpmap:18 G729/8000
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:8 PCMA/8000
> > a=ptime:20
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-11
> >
> > 13 headers, 13 lines
> > Using latest request as basis request
> > Sending to 192.168.0.3 : 5060 (non-NAT)
> > Reliably Transmitting (no NAT):
> > SIP/2.0 407 Proxy Authentication Required
> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
> > From: "Angus Comber"
> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > CSeq: 45925 INVITE
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > Contact: <sip:777 at 192.168.0.13>
> > Proxy-Authenticate: Digest realm="asterisk", nonce="0c555366"
> > Content-Length: 0
> >
> >
> > to 192.168.0.3:5060
> > Scheduling destruction of call '11e4ca07b25c9335 at 192.168.0.3'
> > <mailto:'11e4ca07b25c9335 at 192.168.0.3'> in 15000 ms
> > Found user '200'
> >
> >
> > Sip read:
> > ACK sip:777 at 192.168.0.13;user=phone SIP/2.0
> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
> > From: "Angus Comber"
> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
> > Contact: <sip:200 at 192.168.0.3;user=phone>
> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > CSeq: 45925 ACK
> > User-Agent: Grandstream GXP2000 1.0.1.9
> > Max-Forwards: 70
> > Allow:
> >
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> > Content-Length: 0
> >
> >
> > 11 headers, 0 lines
> >
> >
> > Sip read:
> > INVITE sip:777 at 192.168.0.13;user=phone SIP/2.0
> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
> > From: "Angus Comber"
> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > To: <sip:777 at 192.168.0.13;user=phone>
> > Contact: <sip:200 at 192.168.0.3;user=phone>
> > Supported: replaces, timer
> > Proxy-Authorization: Digest username="200", realm="asterisk",
> > algorithm=MD5, uri="sip:777 at 192.168.0.13;user=phone",
> > nonce="0c555366", response="ee6088fb4e50da5fe412913ae40dd45c"
> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > CSeq: 45926 INVITE
> > User-Agent: Grandstream GXP2000 1.0.1.9
> > Max-Forwards: 70
> > Allow:
> >
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> > Content-Type: application/sdp
> > Content-Length: 258
> >
> > v=0
> > o=200 8000 8001 IN IP4 192.168.0.3
> > s=SIP Call
> > c=IN IP4 192.168.0.3
> > t=0 0
> > m=audio 5004 RTP/AVP 18 0 8 101
> > a=sendrecv
> > a=rtpmap:18 G729/8000
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:8 PCMA/8000
> > a=ptime:20
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-11
> >
> > 14 headers, 13 lines
> > Using latest request as basis request
> > Sending to 192.168.0.3 : 5060 (non-NAT)
> > Found user '200'
> > Found RTP audio format 18
> > Found RTP audio format 0
> > Found RTP audio format 8
> > Found RTP audio format 101
> > Peer audio RTP is at port 192.168.0.3:5004
> > Found description format G729
> > Found description format PCMU
> > Found description format PCMA
> > Found description format telephone-event
> > Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10c
> > (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c
(ulaw|alaw|g729)
> > Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined
> > - 0x1 (g723)
> > Looking for 777 in default
> > Reliably Transmitting (no NAT):
> > SIP/2.0 404 Not Found
> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
> > From: "Angus Comber"
> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > CSeq: 45926 INVITE
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > Contact: <sip:777 at 192.168.0.13>
> > Content-Length: 0
> >
> >
> > to 192.168.0.3:5060
> >
> >
> > Sip read:
> > ACK sip:777 at 192.168.0.13;user=phone SIP/2.0
> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
> > From: "Angus Comber"
> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
> > Contact: <sip:200 at 192.168.0.3;user=phone>
> > Proxy-Authorization: Digest username="200", realm="asterisk",
> > algorithm=MD5, uri="sip:777 at 192.168.0.13;user=phone",
> > nonce="0c555366", response="7fcb1024a81b3ea3bcc56baeca4bac3e"
> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > CSeq: 45926 ACK
> > User-Agent: Grandstream GXP2000 1.0.1.9
> > Max-Forwards: 70
> > Allow:
> >
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> > Content-Length: 0
> >
> >
> > 12 headers, 0 lines
> > Destroying call '11e4ca07b25c9335 at 192.168.0.3'
> > <mailto:'11e4ca07b25c9335 at 192.168.0.3'>
> >
> >
> > How can I troubleshoot? What should I be looking at?
> >
> > Angus
> >
> >
>
------------------------------------------------------------------------
> >
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> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
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>
> --
> CTO Marc Storck
> MS Networks SA mstorck at msnetworks.lu
> IT Service Provider http://www.msnetworks.lu
> 15, route d'Esch Phone: +352 2727 3030
> L-4450 Belvaux Fax: +352 2727 3060
>
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