[Asterisk-Users] broadvoice config problem.

[Craig Simon]

This is the pieces of my extensions.conf.  All this has been sent to me 
from broadvoice, so I can't tell you if it's correct or not.  I do have 
an extension 100 created, and that is what I am logging on with my 
softphone as.  And testing my outgoing calling.

Thanks for the help!
Craig


[default]
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()


[from-broadvoice]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Dial(LOCAL/100 at default,25)
;exten => s,3,Goto(ivr,1,1)
exten => s,4,VoiceMail(u${EXTEN}@default)
exten => s,5,Hangup




Rich Adamson wrote:

>>I have been fighting with * for a couple of days now.  I have recieved 
>>some help from the list but have not been successful in receiving calls 
>>from broadvoice to my asterisk box yet.  I can place calls however, just 
>>not receive them.  I enabled sip debugging today and here is the output 
>>from an incoming call:
>>
>>asterisk*CLI> sip debug
>>SIP Debugging Enabled
>>asterisk*CLI>
>>
>>Sip read:
>>INVITE sip:100 at 207.145.49.194 SIP/2.0
>>Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK2imkag00d031s9k3e1k1.1sr
>>From: "Simon 
>>
>>    
>>
>Craig"<sip:9251234567 at 147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-171810273
>0-1113256175088
>  
>
>>To: "Craig Simon"<sip:9255582025 at sip.broadvoice.com;user=phone>
>>Call-ID: SD5oc1901-e89dc59f2c879f7144697a9486593537-js1h002
>>CSeq: 429822713 INVITE
>>Contact: 
>><sip:9251234567 at 147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp>
>>Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
>>Supported: 100rel,timer
>>Min-SE: 60
>>Accept: application/sdp,application/dtmf
>>Max-Forwards: 69
>>Content-Type: application/sdp
>>Content-Length: 289
>>
>>v=0
>>o=BroadWorks 3802511 1 IN IP4 147.135.8.128
>>s=-
>>c=IN IP4 147.135.8.128
>>t=0 0
>>m=audio 14022 RTP/AVP 0 8 96 18 97 101
>>a=ptime:20
>>a=rtpmap:0 PCMU/8000
>>a=rtpmap:8 PCMA/8000
>>a=rtpmap:96 G726-32/8000
>>a=rtpmap:18 G729/8000
>>a=rtpmap:97 iLBC/8000
>>a=rtpmap:101 telephone-event/8000
>>
>>14 headers, 13 lines
>>Using latest request as basis request
>>Sending to 147.135.8.128 : 5060 (NAT)
>>Found peer 'sip.broadvoice.com'
>>Found RTP audio format 0
>>Found RTP audio format 8
>>Found RTP audio format 96
>>Found RTP audio format 18
>>Found RTP audio format 97
>>Found RTP audio format 101
>>Peer audio RTP is at port 147.135.8.128:14022
>>Found description format PCMU
>>Found description format PCMA
>>Found description format G726-32
>>Found description format G729
>>Found description format iLBC
>>Found description format telephone-event
>>Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51c 
>>(ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
>>Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 
>>0x1 (g723)
>>Looking for 100 in from-broadvoice
>>Reliably Transmitting (no NAT):
>>SIP/2.0 404 Not Found
>>    
>>
>
>Looks to me like Broadvoice is trying to authenticate a call with
>your system and can only find g723 codec (which doesn't exist in *). 
>Then it looks like its trying to dial x100. Do you have a context 
>called "from-broadvoice" that includes something like:
> exten => 100,1,Dial(SIP/3001,15,r) 
>
>So, not sure whether your result is an incorrect codec, an exten=>100
>problem, or both.
>
>
>
>  
>
>>Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK2imkag00d031s9k3e1k1.1sr
>>From: "Simon 
>>
>>    
>>
>Craig"<sip:9251234567 at 147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-171810273
>0-1113256175088
>  
>
>>To: "Craig 
>>Simon"<sip:9255582025 at sip.broadvoice.com;user=phone>;tag=as1fe9ff98
>>Call-ID: SD5oc1901-e89dc59f2c879f7144697a9486593537-js1h002
>>CSeq: 429822713 INVITE
>>User-Agent: Asterisk PBX
>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>>Contact: <sip:100 at 207.145.49.194>
>>Content-Length: 0
>>
>>
>> to 147.135.8.128:5060
>>asterisk*CLI>
>>
>>Sip read:
>>ACK sip:100 at 207.145.49.194 SIP/2.0
>>Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK2imkag00d031s9k3e1k1.1sr
>>From: "Simon 
>>
>>    
>>
>Craig"<sip:9251234567 at 147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;tag=SD5oc1901-171810273
>0-1113256175088
>  
>
>>To: "Craig 
>>Simon"<sip:9255582025 at sip.broadvoice.com;user=phone>;tag=as1fe9ff98
>>Call-ID: SD5oc1901-e89dc59f2c879f7144697a9486593537-js1h002
>>CSeq: 429822713 ACK
>>
>>
>>6 headers, 0 lines
>>Destroying call 'SD5oc1901-e89dc59f2c879f7144697a9486593537-js1h002'
>>asterisk*CLI>
>>
>>
>>I see the 404 in the middle of the log, I just am not sure what it is 
>>looking for and not finding.  Any help would be great.
>>
>>Thanks
>>Craig
>>
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>    
>>
>
>---------------End of Original Message-----------------
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>