[Asterisk-Users] Why can't sip/200 call sip/202
Marc Storck
marc.storck at msnetworks.lu
Sun Jul 24 11:53:49 MST 2005
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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--
CTO Marc Storck
MS Networks SA mstorck at msnetworks.lu
IT Service Provider http://www.msnetworks.lu
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