[Asterisk-Users] weird outbound problem through broadvoice (new)

[Paul P. Pongco]

Hello,

I changed my asterisk to the recently posted software on CVS (Asterisk
CVS-v1-0-03/15/05-12:11:02). Problem still persists.
What is weird here is I can dial certain numbers (broadvoice support
number works) but cant on others.
Checked the SIP call flow via ethereal and I can see Im sending and
receiving invites from the same broadvoice server (147.135.8.128) w/c is
what I have mapped sip.broadvoice.com to at /etc/hosts.
Any other way I can debug this? Thanks.


On Mon, 2005-03-14 at 17:40, Paul P. Pongco wrote:
> Hello,
> 
> Have a weird problem when using asterisk (1.0.6). There are certain
> numbers I cannot dial when using asterisk with my broadvoice account.
> No problems with inbound. With outbound calls, I can call some numbers
> (for example broadvoice customer support number) and unsuccessfully with
> some. However, when I configure my account directly on x-lite, I dont
> see these outbound problems.
> Here is a snapshot of my sip.conf
> 
> register => UUUUUUUUUU at sip.broadvoice.com:PPPPPPPPPP:UUUUUUUUUU at sip.broadvoice.com
>  
> 
> [sip.broadvoice.com]
> type=peer
> host=sip.broadvoice.com
> fromuser=UUUUUUUUUU
> fromdomain=sip.broadvoice.com
> secret=PPPPPPPPPP
> username=UUUUUUUUUU
> port=5060
> dtmfmode=inband
> dtmf=inband
> insecure=very
> context=incoming
> authname=UUUUUUUUUU
> canreinvite=no
> qualify=no
> nat=no
> 
> extensions.conf
> [outgoing]
> exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
> exten => _1NXXNXXXXXX, 2, congestion()
> exten => _1NXXNXXXXXX, 102, busy()
> 
> A portion of sip debug during successful calls (calling broadvoice
> support)
> 
> Sip read:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a
> From: "1001" <sip:UUUUUUUUUU at sip.broadvoice.com>;tag=as65b65920
> To: <sip:19784187300 at sip.broadvoice.com>
> Call-ID: 2007fca97e36e72b54818caa377e6dcc at sip.broadvoice.com
> CSeq: 103 INVITE
>   
> 6 headers, 0 lines
> CLI>
>   
> Sip read:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a
> From: "1001" <sip:UUUUUUUUUU at sip.broadvoice.com>;tag=as65b65920
> To:
> <sip:19784187300 at sip.broadvoice.com>;tag=SD58a8499-104694000-1110784950009
> Call-ID: 2007fca97e36e72b54818caa377e6dcc at sip.broadvoice.com
> CSeq: 103 INVITE
> Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
> Supported: 100rel,timer
> Contact:
> <sip:19784187300 at 147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp>
> Remote-Party-ID: "Auto Attendant
> PrimaryAttendant"<sip:9784187395 at 147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;screen=yes;party=called;privacy=off;id-type=subscriber
> Content-Length: 0
> 
> A portion of sip debug during unsuccessful calls, where TTTTTTTTT is the
> target phone number
> 
> Sip read:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18
> From: "1001" <sip:UUUUUUUUUU at sip.broadvoice.com>;tag=as6f6dba69
> To: <sip:1TTTTTTTTTT at sip.broadvoice.com>
> Call-ID: 095981b26d97329e4155ccd529617e5c at sip.broadvoice.com
> CSeq: 103 INVITE
>   
>   
> 6 headers, 0 lines
> Reliably Transmitting:
> CANCEL sip:1TTTTTTTTTT at sip.broadvoice.com SIP/2.0
> Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18
> From: "1001" <sip:UUUUUUUUUU at sip.broadvoice.com>;tag=as6f6dba69
> To: <sip:1TTTTTTTTTT at sip.broadvoice.com>
> Contact: <sip:UUUUUUUUUU at x.x.x.x>
> Call-ID: 095981b26d97329e4155ccd529617e5c at sip.broadvoice.com
> CSeq: 103 CANCEL
> User-Agent: Asterisk PBX
> Proxy-Authorization: Digest username="UUUUUUUUUU", realm="BroadWorks",
> algorithm=MD5,
> uri="sip:1TTTTTTTTTT at sip.broadvoice.com", nonce="1110785211206",
> response="f68a31735aec843b9ef68b7909fcf178", opaque=""
> Content-Length: 0
>   
>  (no NAT) to 147.135.8.128:5060
> Scheduling destruction of call
> '095981b26d97329e4155ccd529617e5c at sip.broadvoice.com' in 15000 ms
> Transmitting (no NAT):
> SIP/2.0 503 Service Unavailable
> Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK01853115f3033a3c
> From: <sip:1001 at x.x.x.x>;tag=9d9e03fd7b4508e9
> To: <sip:1TTTTTTTTTT at x.x.x.x>;tag=as79fd7936
> Call-ID: 3512f0bb5f5ebf20 at x.x.x.x
> CSeq: 7327 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:1TTTTTTTTT at x.x.x.x>
> Content-Length: 0
>    
> to x.x.x.x:5060
> 
> Asterisk box not behind firewall. No iptables filters either. It seems
> that asterisk is sending CANCEL due to call timeout after the 2nd 100
> Trying during INVITE message flow. I am not sure what is causing the
> timeout. Anyone experienced this before? Tried using ethereal to debug
> the problem deeply, but I can only see the same flow as the sip debug.
> Hoping for your assistance. Thanks.
> 
> 
> 
> 
> 
> 
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-- 
Cheers,

Paul P. Pongco
Mosaic Communications Inc.