Joe
2005-Mar-09 11:41 UTC
[Asterisk-Users] Broadvoice latest changes and still not working-An
I've tried everything with the * box after this weekend. I have read every document on the problems people are having with them after this weekend as well, but none of them address my problem. I checked my settings in my sips which I have below as well, I have changed the host file a few times, but this was new to me and I never had modified it before. I have and the same results happened. I have always used the CHI proxy until this past weekend. I get a 404 not found when the invite goes out. Below is my debug for broadvoice, which I think tells the whole story, but for the life of me, I can not figure out where the 404 is coming from. I have listed my sip file below as well. Inbound calls work and I am registered. Before we go into the debug, I get this message when I reload my configs files. Mar 9 13:23:48 NOTICE[3161]: chan_sip.c:6779 handle_response: Outbound Registration: Expiry for sip.broadvoice.com is 1948 sec (Scheduling reregistration in 1933000 ms) Below is the debug: -- Executing Dial("OSS/dsp", "SIP/xxxxxxxxxx@sip.broadvoice.com|30") in new stack We're at outsideIPaddress port 14842 Answering with preferred capability 0x4 (ulaw) 12 headers, 8 lines Reliably Transmitting: INVITE sip:xxxxxxxxxx@proxy.lax.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com> Contact: <sip:BBBBBBBBBB@outsideIPaddress> Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 09 Mar 2005 18:15:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 17647 17647 IN IP4 outsideIPaddress s=session c=IN IP4 outsideIPaddress t=0 0 m=audio 14842 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - (no NAT) to 147.135.8.128:5060 -- Called xxxxxxxxxx@sip.broadvoice.com asterisk1*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com> Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 INVITE 6 headers, 0 lines asterisk1*CLI> Sip read: SIP/2.0 404 Not Found Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99- Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 INVITE Content-Length: 0 7 headers, 0 lines -- Got SIP response 404 "Not Found" back from 147.135.8.128 Transmitting: ACK sip:xxxxxxxxxx@proxy.lax.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99- Contact: <sip:BBBBBBBBBB@outsideIPaddress> Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 147.135.8.128:5060 -- SIP/sip.broadvoice.com-2a2c is circuit-busy == Everyone is busy/congested at this time -- Executing Busy("OSS/dsp", "") in new stack Destroying call '0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com' asterisk1*CLI> hangup == Spawn extension (default, 509, 102) exited non-zero on 'OSS/dsp' << Hangup on console >> [sip.broadvoice.com] type=peer host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com fromuser= BBBBBBBBBB username= BBBBBBBBBB ;authuser= BBBBBBBBBB secret= secret context=sip nat=no insecure=very dtmfmode=inband -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/6dce8b51/attachment.htm
Scott Wolfe
2005-Mar-09 11:44 UTC
[Asterisk-Users] Broadvoice latest changes and still not working-An
Just wondering. How are you getting this debug. I am having problems to and I cant seem to track it down. ----- Original Message ----- From: Joe To: asterisk-users@lists.digium.com Sent: Wednesday, March 09, 2005 10:41 AM Subject: [Asterisk-Users] Broadvoice latest changes and still not working-An I've tried everything with the * box after this weekend. I have read every document on the problems people are having with them after this weekend as well, but none of them address my problem. I checked my settings in my sips which I have below as well, I have changed the host file a few times, but this was new to me and I never had modified it before. I have and the same results happened. I have always used the CHI proxy until this past weekend. I get a 404 not found when the invite goes out. Below is my debug for broadvoice, which I think tells the whole story, but for the life of me, I can not figure out where the 404 is coming from. I have listed my sip file below as well. Inbound calls work and I am registered. Before we go into the debug, I get this message when I reload my configs files. Mar 9 13:23:48 NOTICE[3161]: chan_sip.c:6779 handle_response: Outbound Registration: Expiry for sip.broadvoice.com is 1948 sec (Scheduling reregistration in 1933000 ms) Below is the debug: -- Executing Dial("OSS/dsp", "SIP/xxxxxxxxxx@sip.broadvoice.com|30") in new stack We're at outsideIPaddress port 14842 Answering with preferred capability 0x4 (ulaw) 12 headers, 8 lines Reliably Transmitting: INVITE sip:xxxxxxxxxx@proxy.lax.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com> Contact: <sip:BBBBBBBBBB@outsideIPaddress> Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 09 Mar 2005 18:15:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 17647 17647 IN IP4 outsideIPaddress s=session c=IN IP4 outsideIPaddress t=0 0 m=audio 14842 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - (no NAT) to 147.135.8.128:5060 -- Called xxxxxxxxxx@sip.broadvoice.com asterisk1*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com> Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 INVITE 6 headers, 0 lines asterisk1*CLI> Sip read: SIP/2.0 404 Not Found Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99- Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 INVITE Content-Length: 0 7 headers, 0 lines -- Got SIP response 404 "Not Found" back from 147.135.8.128 Transmitting: ACK sip:xxxxxxxxxx@proxy.lax.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99- Contact: <sip:BBBBBBBBBB@outsideIPaddress> Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 147.135.8.128:5060 -- SIP/sip.broadvoice.com-2a2c is circuit-busy == Everyone is busy/congested at this time -- Executing Busy("OSS/dsp", "") in new stack Destroying call '0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com' asterisk1*CLI> hangup == Spawn extension (default, 509, 102) exited non-zero on 'OSS/dsp' << Hangup on console >> [sip.broadvoice.com] type=peer host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com fromuser= BBBBBBBBBB username= BBBBBBBBBB ;authuser= BBBBBBBBBB secret= secret context=sip nat=no insecure=very dtmfmode=inband ------------------------------------------------------------------------------ _______________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/d04f78e0/attachment.htm
Marios Andreou
2005-Mar-09 12:45 UTC
[Asterisk-Users] Broadvoice latest changes and still not working-An
The problem that you have it was the one that I stabled across the very first time tried to setup BV. The 404 not found that you are getting is because there is no such phone number xxxxx@proxy.lax.broadvoice.com But there is a xxxxx@sip.broadvoice.com. This is like saying xxxxx@yourownlocaldomain.tld (you are going to get a 404) The chi worked because it was a test server (beta/debug) that I read somewhere in this list. So the fix for you will be to change the host=proxy.lax.broadvoice.com to host=sip.broadvoice.com Now if you are getting better responses from lax then change your host file to 147.135.8.128 sip.broadvoice.com This is because sip.broadvoice.com resolves to proxy.dca.broadvoice.com. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Joe Sent: Wednesday, March 09, 2005 1:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Broadvoice latest changes and still not working-An I've tried everything with the * box after this weekend. I have read every document on the problems people are having with them after this weekend as well, but none of them address my problem. I checked my settings in my sips which I have below as well, I have changed the host file a few times, but this was new to me and I never had modified it before. I have and the same results happened. I have always used the CHI proxy until this past weekend. I get a 404 not found when the invite goes out. Below is my debug for broadvoice, which I think tells the whole story, but for the life of me, I can not figure out where the 404 is coming from. I have listed my sip file below as well. Inbound calls work and I am registered. Before we go into the debug, I get this message when I reload my configs files. Mar 9 13:23:48 NOTICE[3161]: chan_sip.c:6779 handle_response: Outbound Registration: Expiry for sip.broadvoice.com is 1948 sec (Scheduling reregistration in 1933000 ms) Below is the debug: -- Executing Dial("OSS/dsp", "SIP/xxxxxxxxxx@sip.broadvoice.com|30") in new stack We're at outsideIPaddress port 14842 Answering with preferred capability 0x4 (ulaw) 12 headers, 8 lines Reliably Transmitting: INVITE sip:xxxxxxxxxx@proxy.lax.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com> Contact: <sip:BBBBBBBBBB@outsideIPaddress> Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 09 Mar 2005 18:15:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 17647 17647 IN IP4 outsideIPaddress s=session c=IN IP4 outsideIPaddress t=0 0 m=audio 14842 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - (no NAT) to 147.135.8.128:5060 -- Called xxxxxxxxxx@sip.broadvoice.com asterisk1*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com> Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 INVITE 6 headers, 0 lines asterisk1*CLI> Sip read: SIP/2.0 404 Not Found Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99- Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 INVITE Content-Length: 0 7 headers, 0 lines -- Got SIP response 404 "Not Found" back from 147.135.8.128 Transmitting: ACK sip:xxxxxxxxxx@proxy.lax.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc From: "asterisk" <sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9 To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99- Contact: <sip:BBBBBBBBBB@outsideIPaddress> Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 147.135.8.128:5060 -- SIP/sip.broadvoice.com-2a2c is circuit-busy == Everyone is busy/congested at this time -- Executing Busy("OSS/dsp", "") in new stack Destroying call '0674f2a33bfee57a7c9232e10282b5ab@sip.broadvoice.com' asterisk1*CLI> hangup == Spawn extension (default, 509, 102) exited non-zero on 'OSS/dsp' << Hangup on console >> [sip.broadvoice.com] type=peer host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com fromuser= BBBBBBBBBB username= BBBBBBBBBB ;authuser= BBBBBBBBBB secret= secret context=sip nat=no insecure=very dtmfmode=inband -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/1ee5be25/attachment.htm