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 => [EMAIL PROTECTED]:PPPPPPPPPP:[EMAIL PROTECTED] > > > [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/[EMAIL PROTECTED],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:[EMAIL PROTECTED]>;tag=as65b65920 > To: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > 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:[EMAIL PROTECTED]>;tag=as65b65920 > To: > <sip:[EMAIL PROTECTED]>;tag=SD58a8499-104694000-1110784950009 > Call-ID: [EMAIL PROTECTED] > CSeq: 103 INVITE > Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY > Supported: 100rel,timer > Contact: > <sip:[EMAIL > PROTECTED]:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp> > Remote-Party-ID: "Auto Attendant > PrimaryAttendant"<sip:[EMAIL > PROTECTED];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:[EMAIL PROTECTED]>;tag=as6f6dba69 > To: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 103 INVITE > > > 6 headers, 0 lines > Reliably Transmitting: > CANCEL sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18 > From: "1001" <sip:[EMAIL PROTECTED]>;tag=as6f6dba69 > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Proxy-Authorization: Digest username="UUUUUUUUUU", realm="BroadWorks", > algorithm=MD5, > uri="sip:[EMAIL PROTECTED]", nonce="1110785211206", > response="f68a31735aec843b9ef68b7909fcf178", opaque="" > Content-Length: 0 > > (no NAT) to 147.135.8.128:5060 > Scheduling destruction of call > '[EMAIL PROTECTED]' 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:[EMAIL PROTECTED]>;tag=9d9e03fd7b4508e9 > To: <sip:[EMAIL PROTECTED]>;tag=as79fd7936 > Call-ID: [EMAIL PROTECTED] > CSeq: 7327 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > 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. > > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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 -- Cheers, Paul P. Pongco Mosaic Communications Inc. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@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