[Asterisk-Users] Why can't sip/200 call sip/202
Rich Adamson
radamson at routers.com
Sun Jul 24 12:04:50 MST 2005
> 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
> 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
> 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' 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
> 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
> 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
> 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
> 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'
>
>
> How can I troubleshoot? What should I be looking at?
In the debug trace shown above, I see:
Looking for 777 in default
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
It would appear something is trying to dial "777" in the default context
(in extensions.conf), and that extension isn't defined, therefor you are
getting a "404 Not Found".
Without looking at your extensions.conf contents, can't guess any closer
on the problem.
Also, until you get your arms around diagnosing problems, I'd suggest
starting with a single codec, like:
disallow=all
allow=ulaw
; allow=alaw
; allow=g723.1
; allow=g729
When the basics are well understood, then go back and experiment with
various codecs.
I'm also a strong believer in _not_ using a context name such as
"default". Asterisk will frequently try to do something with a
specified context and if it fails, fall back to the "default" context
without you noticing. Unless you happen to see that in the CLI, you
form an opinion that your configuration is working fine, but it
really isn't.
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