[Asterisk-Users] SIP call doesn't execute the 's'-extension

[Chris Stinson]

You only have a 1 in the javaAgi context and you aren't point the 
javaAgi to any other contexts, pressing anyting else but 1 will get a 
not found error because you only have 1 defined. If you want the call to 
continue you need to send it to another context or add more to the 
javaAgi context.

Tobias Wolf wrote:
> Hi,
> 
> i have just started to configure access to the * over SIP-Phones. 
> Therefore I have defined this SIP-Phone in sip.conf:
> 
> [tobias]
> type=friend
> username=tobias
> secret=tobias
> auth=md5
> host=dynamic
> reinvite=no
> dtmfmode=inband
> callerid="Tobias" <1087006>
> allow=all
> context=javaAgi
> dtmfmode=rfc2833
> 
> 
> As you can see i am directing calls from this user to the context 
> [javaAgi] which is defined here in extension.conf:
> 
> [javaAgi]
> exten => s,1,Answer()
> exten => s,2,Playback(code1000)
> exten => s,3,Hangup()
> exten => 1,1,Answer()
> exten => 1,2,Playback(code1000)
> exten => 1,3,Hangup()
> 
> If i dial 1 on my SIP Phone everything works as suspected, the call is 
> answered and the gsm-file is played. My understanding of the 
> 's'-extension is, that it is executed then a call comes in an there is 
> no extension wich matches the called number. But if i dial a random 
> number i get an "404 Not found" error.
> 
> Here is an snippet of what * tells me on sip debug, but i can't get a 
> clue out of it:
> 
> 
> 12 headers, 13 lines
> Using latest request as basis request
> Sending to 10.3.4.98 : 5060 (non-NAT)
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 3
> Found RTP audio format 98
> Found RTP audio format 97
> Found RTP audio format 101
> Peer audio RTP is at port 10.3.4.98:8000
> Found description format pcmu
> Found description format pcma
> Found description format gsm
> Found description format iLBC
> Found description format speex
> Found description format telephone-event
> Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - 
> audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 
> 0xe(GSM|ULAW|ALAW)
> Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 
> 0x1(G723)
> Found user 'tobias'
> Looking for 2 in javaAgi
> Reliably Transmitting (no NAT):
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 
> 10.3.4.98:5060;branch=z9hG4bK102F7CD4855F4C4E927C3398E3C57BF4
> From: Tobias <sip:tobias at 10.3.1.6>;tag=2760968676
> To: <sip:2 at 10.3.1.6>;tag=as396962de
> Call-ID: 79A5523F-AFCA-4DBE-9AA2-F51377E8B5AE at 10.3.4.98
> CSeq: 58303 INVITE
> User-Agent: evision PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:2 at 10.3.1.6>
> Content-Length: 0
> 
> Perhaps anyone can point me to the right direction ??
> 
> Tobias
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-- 
-----

Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
noc at isdn.net