Paul P. Pongco
2005-Mar-14 02:40 UTC
[Asterisk-Users] weird outbound problem through broadvoice (new)
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@sip.broadvoice.com:PPPPPPPPPP:UUUUUUUUUU@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@sip.broadvoice.com>;tag=as65b65920 To: <sip:19784187300@sip.broadvoice.com> Call-ID: 2007fca97e36e72b54818caa377e6dcc@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@sip.broadvoice.com>;tag=as65b65920 To: <sip:19784187300@sip.broadvoice.com>;tag=SD58a8499-104694000-1110784950009 Call-ID: 2007fca97e36e72b54818caa377e6dcc@sip.broadvoice.com CSeq: 103 INVITE Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY Supported: 100rel,timer Contact: <sip:19784187300@147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp> Remote-Party-ID: "Auto Attendant PrimaryAttendant"<sip:9784187395@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@sip.broadvoice.com>;tag=as6f6dba69 To: <sip:1TTTTTTTTTT@sip.broadvoice.com> Call-ID: 095981b26d97329e4155ccd529617e5c@sip.broadvoice.com CSeq: 103 INVITE 6 headers, 0 lines Reliably Transmitting: CANCEL sip:1TTTTTTTTTT@sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18 From: "1001" <sip:UUUUUUUUUU@sip.broadvoice.com>;tag=as6f6dba69 To: <sip:1TTTTTTTTTT@sip.broadvoice.com> Contact: <sip:UUUUUUUUUU@x.x.x.x> Call-ID: 095981b26d97329e4155ccd529617e5c@sip.broadvoice.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Proxy-Authorization: Digest username="UUUUUUUUUU", realm="BroadWorks", algorithm=MD5, uri="sip:1TTTTTTTTTT@sip.broadvoice.com", nonce="1110785211206", response="f68a31735aec843b9ef68b7909fcf178", opaque="" Content-Length: 0 (no NAT) to 147.135.8.128:5060 Scheduling destruction of call '095981b26d97329e4155ccd529617e5c@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@x.x.x.x>;tag=9d9e03fd7b4508e9 To: <sip:1TTTTTTTTTT@x.x.x.x>;tag=as79fd7936 Call-ID: 3512f0bb5f5ebf20@x.x.x.x CSeq: 7327 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:1TTTTTTTTT@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.
Paul P. Pongco
2005-Mar-14 21:48 UTC
[Asterisk-Users] weird outbound problem through broadvoice (new)
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@sip.broadvoice.com:PPPPPPPPPP:UUUUUUUUUU@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@sip.broadvoice.com>;tag=as65b65920 > To: <sip:19784187300@sip.broadvoice.com> > Call-ID: 2007fca97e36e72b54818caa377e6dcc@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@sip.broadvoice.com>;tag=as65b65920 > To: > <sip:19784187300@sip.broadvoice.com>;tag=SD58a8499-104694000-1110784950009 > Call-ID: 2007fca97e36e72b54818caa377e6dcc@sip.broadvoice.com > CSeq: 103 INVITE > Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY > Supported: 100rel,timer > Contact: > <sip:19784187300@147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp> > Remote-Party-ID: "Auto Attendant > PrimaryAttendant"<sip:9784187395@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@sip.broadvoice.com>;tag=as6f6dba69 > To: <sip:1TTTTTTTTTT@sip.broadvoice.com> > Call-ID: 095981b26d97329e4155ccd529617e5c@sip.broadvoice.com > CSeq: 103 INVITE > > > 6 headers, 0 lines > Reliably Transmitting: > CANCEL sip:1TTTTTTTTTT@sip.broadvoice.com SIP/2.0 > Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18 > From: "1001" <sip:UUUUUUUUUU@sip.broadvoice.com>;tag=as6f6dba69 > To: <sip:1TTTTTTTTTT@sip.broadvoice.com> > Contact: <sip:UUUUUUUUUU@x.x.x.x> > Call-ID: 095981b26d97329e4155ccd529617e5c@sip.broadvoice.com > CSeq: 103 CANCEL > User-Agent: Asterisk PBX > Proxy-Authorization: Digest username="UUUUUUUUUU", realm="BroadWorks", > algorithm=MD5, > uri="sip:1TTTTTTTTTT@sip.broadvoice.com", nonce="1110785211206", > response="f68a31735aec843b9ef68b7909fcf178", opaque="" > Content-Length: 0 > > (no NAT) to 147.135.8.128:5060 > Scheduling destruction of call > '095981b26d97329e4155ccd529617e5c@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@x.x.x.x>;tag=9d9e03fd7b4508e9 > To: <sip:1TTTTTTTTTT@x.x.x.x>;tag=as79fd7936 > Call-ID: 3512f0bb5f5ebf20@x.x.x.x > CSeq: 7327 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:1TTTTTTTTT@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. > > > > > > > _______________________________________________ > 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.