Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****Solved*****!
I concur. I rebuilt today and now I seem to be able to dial out. MARK. Chris Nibeck wrote: thank you everyone! It does not seen that it was configuration problems at all. It appears it was the CVS that I was using from yesterday. I decided to start over, downloaded the latest CVS, recompiled, and voila! * started working Indeed even a Cisco ATA that was never working before started working! Thanks to everyone! Chris On Mar 9, 2005, at 10:28 AM, Chris Nibeck wrote: there are two of us with the same problem so I will answer for me. Yes I tried the below instructions. The current thinking by multiple people is * never tries authenticating so removing the FQDN will force * to go to the related section named by either a phone number or a non Fully Qualified Domain Name. But I still don't have it working so who knows. Anyone that wishes to call me via BV my number is 8475100139 and it is up. Chris On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote: Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? ___ 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 ___ 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 ___ 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 ___ 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
***SOLVED*** [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****Solved*****!
thank you everyone! It does not seen that it was configuration problems at all. It appears it was the CVS that I was using from yesterday. I decided to start over, downloaded the latest CVS, recompiled, and voila! * started working Indeed even a Cisco ATA that was never working before started working! Thanks to everyone! Chris On Mar 9, 2005, at 10:28 AM, Chris Nibeck wrote: there are two of us with the same problem so I will answer for me. Yes I tried the below instructions. The current thinking by multiple people is * never tries authenticating so removing the FQDN will force * to go to the related section named by either a phone number or a non Fully Qualified Domain Name. But I still don't have it working so who knows. Anyone that wishes to call me via BV my number is 8475100139 and it is up. Chris On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote: Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? ___ 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 ___ 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 ___ 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
Fwd: Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****SOLVED****
This configuration solved my problem. I could have sworn I tried this before. I guess not. I did not need to apply the patch. Also, I am using a regular Registration setup in my sip.conf not broadvoice's funky one... The only thing I can surmise is that order of the variables matters. This is what worked for me: [PP] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=PP secret=XX username=PP insecure=very context=sip authname=PP dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no Thank you On Wednesday 09 March 2005 08:23 am, Mike Matthews wrote: > Have you tried this: > > http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup > > Zanzamar Majere wrote: > >Thank you for the response. I still have the errors mentioned below, sip > >response and Failed to authenticate on INVITE > > > >[PP] > >type=peer > >username=PP > >fromuser=PP > >authuser=PP > >fromdomain=sip.broadvoice.com > >secret=XX > >host=sip.broadvoice.com > >dtmfmode=inband > >insecure=very > >context=sip > >qualify=yes > >disallow=all > >allow=ulaw > >allow=gsm > >;Disable canreinvite if you are behind a NAT > >;canreinvite=no > >nat=no > > > >Does anyone else have any other suggestions? > > ___ > 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 --- ___ 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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
there are two of us with the same problem so I will answer for me. Yes I tried the below instructions. The current thinking by multiple people is * never tries authenticating so removing the FQDN will force * to go to the related section named by either a phone number or a non Fully Qualified Domain Name. But I still don't have it working so who knows. Anyone that wishes to call me via BV my number is 8475100139 and it is up. Chris On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote: Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? ___ 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 ___ 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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Thanks MF, Yes that was me that sent my PW :-) It is changed now. Same error... Mar 9 10:12:46 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '"Chris Nibeck" ;tag=as0cefa74c' Sip.conf... [*8475100139*] type=peer ;user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=8475100139 secret=x username=8475100139 insecure=very context=default authname=8475100139 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no nat=no extensions.conf... exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice for 30 seconds exten => _8X.,2, congestion() ; No answer, nothing exten => _8X., 102, busy() ; On Mar 9, 2005, at 7:56 AM, MF Hulber wrote: Try changing the extension from Broadvoice1 to the actual phone number (and don't send your secret in a public email or maybe that's Chris'): [*8475100139*] type=peer ;user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=8475100139 secret=XXX username=8475100139 Zanzamar Majere wrote: I have made all the changes to sip.conf for my broadvoice peer friend(and I have tried it as peer) and I am still seeing this response (on call out). Any suggestions? I don't think it is a problem with the phones themselves authenticating, as Asterisk takes care of all the authentication from my understanding. Free world does work for calling out however. So I know at least that works. -- Got SIP response 400 "Bad request" back from 147.135.0.128 Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed to authenticate on INVITE to '"PP" ;tag=as5b80cade' On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote: First off... please cancel previous amplification request. I have implemented your ideas with the same errored result. I am not sure that we are not making it thru authentication. From my digging and comparing packet dumps comparing the soft phone to asterisk they have identical transactions through the ACK reply (the last one on the debug below). The softphone seems to be authenticated after the ACK. I am a newbie to debugging this stuff. I just want to get it working. Thanks everyone in advance for your help. I am certainly very very happy to try anything. Based on Luki's suggestions I... Changed sip.conf... [broadvoice1] type=peer ;user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=8475100139 secret=DELETED username=8475100139 insecure=very context=default authname=8475100139 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no nat=no Changed extensions.conf... exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice for 30 seconds exten => _8X.,2, congestion() ; No answer, nothing exten => _8X., 102, busy() ; End result... Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '"6050" ;tag=as545ccba3' SIP debug... -- Executing Dial("SIP/6050-132b", "SIP/[EMAIL PROTECTED]|30") in new stack We're at xxx.xxx.xxx.xxx port 18212 Answering with capability 2 Answering with capability 4 Answering with capability 8 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 From: "6050" ;tag=as545ccba3 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 09 Mar 2005 07:30:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 205 v=0 o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18212 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - (no NAT) to 147.135.8.128:5060 -- Called [EMAIL PROTECTED] com*CLI> Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221 From: 6050 ;tag=7e2776985d5a0891o0 To: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: [EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a 10c 129dd4fb5f97ec47" Contact: 6050 Expires: 240 User-Agent: Sipura/SPA3000-2.0.10(GWf) Content-Length: 241 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 1138990026 1138990026 IN IP4 64.4.192.110 s=- c=IN IP4 64.4.192.110 t=0 0 m=audio 16388 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv 15 headers, 12 lines Ignoring this request Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221 From: 6050 ;tag=7e2776985d5a0891o0 To: ;tag=as2f065f18 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
Re: [Asterisk-Users] Broadvoice latest changes and still not working-An Additional Server
Jerry- Thank you I accidently sent my password on the LISTSERV last night so I just changed (pasted) the new one in. Still the same problem... Mar 9 09:51:13 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '"Chris Nibeck" ;tag=as4b70f2e7' Incoming works fine still. Anyone can call me at that number. Please do. It is a free call from another BV account. Chris On Mar 9, 2005, at 7:42 AM, Jerry Geis wrote: CHris, I had the exact same problem with the exact same error. My password was entered incorrectly in context section. The register line had the correct password. That is why you get incoming calls. and not outgoing. Jerry ___ 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 ___ 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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? ___ 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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? On Wednesday 09 March 2005 06:56 am, MF Hulber wrote: > Try changing the extension from Broadvoice1 to the actual phone number > (and don't send your secret in a public email or maybe that's Chris'): > > [*8475100139*] > type=peer > ;user=phone > host=sip.broadvoice.com > fromdomain=sip.broadvoice.com > fromuser=8475100139 > secret=XXX > username=8475100139 > > Zanzamar Majere wrote: > >I have made all the changes to sip.conf for my broadvoice peer > >friend(and I have tried it as peer) and I am still seeing this response > >(on call out). Any suggestions? I don't think it is a problem with the > >phones themselves authenticating, as Asterisk takes care of all the > >authentication from my understanding. > > > >Free world does work for calling out however. So I know at least that > >works. > > > > > > > >-- Got SIP response 400 "Bad request" back from 147.135.0.128 > >Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed > >to authenticate on INVITE to '"PP" > >;tag=as5b80cade' > > > >On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote: > >>First off... please cancel previous amplification request. I have > >>implemented your ideas with the same errored result. > >> > >>I am not sure that we are not making it thru authentication. From my > >>digging and comparing packet dumps comparing the soft phone to asterisk > >>they have identical transactions through the ACK reply (the last one > >>on the debug below). The softphone seems to be authenticated after the > >>ACK. I am a newbie to debugging this stuff. I just want to get it > >>working. > >> > >>Thanks everyone in advance for your help. I am certainly very very > >>happy to try anything. > >> > >>Based on Luki's suggestions I... > >> > >>Changed sip.conf... > >> > >>[broadvoice1] > >>type=peer > >>;user=phone > >>host=sip.broadvoice.com > >>fromdomain=sip.broadvoice.com > >>fromuser=8475100139 > >>secret=DELETED > >>username=8475100139 > >>insecure=very > >>context=default > >>authname=8475100139 > >>dtmfmode=inband > >>dtmf=inband > >>;Disable canreinvite if you are behind a NAT > >>canreinvite=no > >>nat=no > >> > >>Changed extensions.conf... > >> > >>exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice > >>for 30 seconds > >>exten => _8X.,2, congestion() ; No answer, nothing > >>exten => _8X., 102, busy() ; > >> > >>End result... > >> > >>Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed > >>to authenticate on INVITE to '"6050" > >>;tag=as545ccba3' > >> > >> > >>SIP debug... > >> > >> -- Executing Dial("SIP/6050-132b", > >>"SIP/[EMAIL PROTECTED]|30") in new stack > >>We're at xxx.xxx.xxx.xxx port 18212 > >>Answering with capability 2 > >>Answering with capability 4 > >>Answering with capability 8 > >>12 headers, 10 lines > >>Reliably Transmitting: > >>INVITE sip:[EMAIL PROTECTED] SIP/2.0 > >>Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 > >>From: "6050" ;tag=as545ccba3 > >>To: > >>Contact: > >>Call-ID: [EMAIL PROTECTED] > >>CSeq: 102 INVITE > >>User-Agent: Asterisk PBX > >>Date: Wed, 09 Mar 2005 07:30:41 GMT > >>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > >>Content-Type: application/sdp > >>Content-Length: 205 > >> > >>v=0 > >>o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx > >>s=session > >>c=IN IP4 xxx.xxx.xxx.xxx > >>t=0 0 > >>m=audio 18212 RTP/AVP 3 0 8 > >>a=rtpmap:3 GSM/8000 > >>a=rtpmap:0 PCMU/8000 > >>a=rtpmap:8 PCMA/8000 > >>a=silenceSupp:off - - - - > >> (no NAT) to 147.135.8.128:5060 > >> -- Called [EMAIL PROTECTED] > >>com*CLI> > >> > >>Sip read: > >>INVITE sip:[EMAIL PROTECTED] SIP/2.0 > >>Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221 > >>From: 6050 ;tag=7e2776985d5a0891o0 > >>To: > >>Call-ID: [EMAIL PROTECTED] > >>CSeq: 102 INVITE > >>Max-Forwards: 70 > >>Proxy-Authorization: Digest > >>username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: > >>[EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a10c > >>129dd4fb5f97ec47" > >>Contact: 6050 > >>Expires: 240 > >>User-Agent: Sipura/SPA3000-2.0.10(GWf) > >>Content-Length: 241 > >>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > >>Supported: x-sipura > >>Content-Type: application/sdp > >> > >>v=0 > >>o=- 1138990026 1138990026 IN IP4 64.4.192.110 > >>s=- > >>c=IN IP4 64.4.192.110 > >>t=0 0 > >>m=audio 16388 RTP/AVP 0 100 101 > >>a=rtpmap:0 PCMU/8000 > >>a=rtpmap:100 NSE/8000 > >>a=rtpmap:101 telephone-event/8000 > >>a=fmtp:101 0-15 > >>a=ptime:30 > >>a=sendrecv > >> > >>15 header
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Try changing the extension from Broadvoice1 to the actual phone number (and don't send your secret in a public email or maybe that's Chris'): [*8475100139*] type=peer ;user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=8475100139 secret=XXX username=8475100139 Zanzamar Majere wrote: I have made all the changes to sip.conf for my broadvoice peer friend(and I have tried it as peer) and I am still seeing this response (on call out). Any suggestions? I don't think it is a problem with the phones themselves authenticating, as Asterisk takes care of all the authentication from my understanding. Free world does work for calling out however. So I know at least that works. -- Got SIP response 400 "Bad request" back from 147.135.0.128 Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed to authenticate on INVITE to '"PP" ;tag=as5b80cade' On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote: First off... please cancel previous amplification request. I have implemented your ideas with the same errored result. I am not sure that we are not making it thru authentication. From my digging and comparing packet dumps comparing the soft phone to asterisk they have identical transactions through the ACK reply (the last one on the debug below). The softphone seems to be authenticated after the ACK. I am a newbie to debugging this stuff. I just want to get it working. Thanks everyone in advance for your help. I am certainly very very happy to try anything. Based on Luki's suggestions I... Changed sip.conf... [broadvoice1] type=peer ;user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=8475100139 secret=DELETED username=8475100139 insecure=very context=default authname=8475100139 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no nat=no Changed extensions.conf... exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice for 30 seconds exten => _8X.,2, congestion() ; No answer, nothing exten => _8X., 102, busy() ; End result... Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '"6050" ;tag=as545ccba3' SIP debug... -- Executing Dial("SIP/6050-132b", "SIP/[EMAIL PROTECTED]|30") in new stack We're at xxx.xxx.xxx.xxx port 18212 Answering with capability 2 Answering with capability 4 Answering with capability 8 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 From: "6050" ;tag=as545ccba3 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 09 Mar 2005 07:30:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 205 v=0 o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18212 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - (no NAT) to 147.135.8.128:5060 -- Called [EMAIL PROTECTED] com*CLI> Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221 From: 6050 ;tag=7e2776985d5a0891o0 To: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: [EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a10c 129dd4fb5f97ec47" Contact: 6050 Expires: 240 User-Agent: Sipura/SPA3000-2.0.10(GWf) Content-Length: 241 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 1138990026 1138990026 IN IP4 64.4.192.110 s=- c=IN IP4 64.4.192.110 t=0 0 m=audio 16388 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv 15 headers, 12 lines Ignoring this request Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221 From: 6050 ;tag=7e2776985d5a0891o0 To: ;tag=as2f065f18 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 64.4.192.110:5060 com*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 From: "6050" ;tag=as545ccba3 To: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE 6 headers, 0 lines com*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 From: "6050" ;tag=as545ccba3 To: ;tag=SD38rq699- Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE WWW-Authenticate: DIGEST realm="BroadWorks",algorithm=MD5,nonce="1110353299563" Content-Length: 0 8 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 From: "6050" ;tag=as545ccba3 To: ;tag=SD38rq699- Contact: Call-ID: [EMAIL PROTECTED] CSeq:
[Asterisk-Users] Broadvoice latest changes and still not working-An Additional Server
CHris, I had the exact same problem with the exact same error. My password was entered incorrectly in context section. The register line had the correct password. That is why you get incoming calls. and not outgoing. Jerry ___ 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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
I have made all the changes to sip.conf for my broadvoice peer friend(and I have tried it as peer) and I am still seeing this response (on call out). Any suggestions? I don't think it is a problem with the phones themselves authenticating, as Asterisk takes care of all the authentication from my understanding. Free world does work for calling out however. So I know at least that works. -- Got SIP response 400 "Bad request" back from 147.135.0.128 Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed to authenticate on INVITE to '"PP" ;tag=as5b80cade' On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote: > First off... please cancel previous amplification request. I have > implemented your ideas with the same errored result. > > I am not sure that we are not making it thru authentication. From my > digging and comparing packet dumps comparing the soft phone to asterisk > they have identical transactions through the ACK reply (the last one > on the debug below). The softphone seems to be authenticated after the > ACK. I am a newbie to debugging this stuff. I just want to get it > working. > > Thanks everyone in advance for your help. I am certainly very very > happy to try anything. > > Based on Luki's suggestions I... > > Changed sip.conf... > > [broadvoice1] > type=peer > ;user=phone > host=sip.broadvoice.com > fromdomain=sip.broadvoice.com > fromuser=8475100139 > secret=zjh018g8f8 > username=8475100139 > insecure=very > context=default > authname=8475100139 > dtmfmode=inband > dtmf=inband > ;Disable canreinvite if you are behind a NAT > canreinvite=no > nat=no > > Changed extensions.conf... > > exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice > for 30 seconds > exten => _8X.,2, congestion() ; No answer, nothing > exten => _8X., 102, busy() ; > > End result... > > Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed > to authenticate on INVITE to '"6050" > ;tag=as545ccba3' > > > SIP debug... > > -- Executing Dial("SIP/6050-132b", > "SIP/[EMAIL PROTECTED]|30") in new stack > We're at xxx.xxx.xxx.xxx port 18212 > Answering with capability 2 > Answering with capability 4 > Answering with capability 8 > 12 headers, 10 lines > Reliably Transmitting: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 > From: "6050" ;tag=as545ccba3 > To: > Contact: > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Date: Wed, 09 Mar 2005 07:30:41 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Content-Type: application/sdp > Content-Length: 205 > > v=0 > o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx > s=session > c=IN IP4 xxx.xxx.xxx.xxx > t=0 0 > m=audio 18212 RTP/AVP 3 0 8 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=silenceSupp:off - - - - > (no NAT) to 147.135.8.128:5060 > -- Called [EMAIL PROTECTED] > com*CLI> > > Sip read: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221 > From: 6050 ;tag=7e2776985d5a0891o0 > To: > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > Max-Forwards: 70 > Proxy-Authorization: Digest > username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: > [EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a10c > 129dd4fb5f97ec47" > Contact: 6050 > Expires: 240 > User-Agent: Sipura/SPA3000-2.0.10(GWf) > Content-Length: 241 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura > Content-Type: application/sdp > > v=0 > o=- 1138990026 1138990026 IN IP4 64.4.192.110 > s=- > c=IN IP4 64.4.192.110 > t=0 0 > m=audio 16388 RTP/AVP 0 100 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:100 NSE/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:30 > a=sendrecv > > 15 headers, 12 lines > Ignoring this request > Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221 > From: 6050 ;tag=7e2776985d5a0891o0 > To: ;tag=as2f065f18 > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Content-Length: 0 > > > to 64.4.192.110:5060 > com*CLI> > > Sip read: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 > From: "6050" ;tag=as545ccba3 > To: > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > > > 6 headers, 0 lines > com*CLI> > > Sip read: > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 > From: "6050" ;tag=as545ccba3 > To: ;tag=SD38rq699- > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > WWW-Authenticate: DIGEST > realm="BroadWorks",algorithm=MD5,nonce="1110353299563" > Content-Length: 0 > > > 8 headers, 0 lines > Transmitting: > ACK sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 > From: "6050" ;tag=as545ccba3 > To: ;tag=SD38rq699- > Con
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
First off... please cancel previous amplification request. I have implemented your ideas with the same errored result. I am not sure that we are not making it thru authentication. From my digging and comparing packet dumps comparing the soft phone to asterisk they have identical transactions through the ACK reply (the last one on the debug below). The softphone seems to be authenticated after the ACK. I am a newbie to debugging this stuff. I just want to get it working. Thanks everyone in advance for your help. I am certainly very very happy to try anything. Based on Luki's suggestions I... Changed sip.conf... [broadvoice1] type=peer ;user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=8475100139 secret=zjh018g8f8 username=8475100139 insecure=very context=default authname=8475100139 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no nat=no Changed extensions.conf... exten => _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice for 30 seconds exten => _8X.,2, congestion() ; No answer, nothing exten => _8X., 102, busy() ; End result... Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '"6050" ;tag=as545ccba3' SIP debug... -- Executing Dial("SIP/6050-132b", "SIP/[EMAIL PROTECTED]|30") in new stack We're at xxx.xxx.xxx.xxx port 18212 Answering with capability 2 Answering with capability 4 Answering with capability 8 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 From: "6050" ;tag=as545ccba3 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 09 Mar 2005 07:30:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 205 v=0 o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 18212 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - (no NAT) to 147.135.8.128:5060 -- Called [EMAIL PROTECTED] com*CLI> Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221 From: 6050 ;tag=7e2776985d5a0891o0 To: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: [EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a10c 129dd4fb5f97ec47" Contact: 6050 Expires: 240 User-Agent: Sipura/SPA3000-2.0.10(GWf) Content-Length: 241 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 1138990026 1138990026 IN IP4 64.4.192.110 s=- c=IN IP4 64.4.192.110 t=0 0 m=audio 16388 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv 15 headers, 12 lines Ignoring this request Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221 From: 6050 ;tag=7e2776985d5a0891o0 To: ;tag=as2f065f18 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 64.4.192.110:5060 com*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 From: "6050" ;tag=as545ccba3 To: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE 6 headers, 0 lines com*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 From: "6050" ;tag=as545ccba3 To: ;tag=SD38rq699- Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE WWW-Authenticate: DIGEST realm="BroadWorks",algorithm=MD5,nonce="1110353299563" Content-Length: 0 8 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 From: "6050" ;tag=as545ccba3 To: ;tag=SD38rq699- Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 147.135.8.128:5060 Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '"6050" ;tag=as545ccba3' On Mar 9, 2005, at 12:08 AM, Luki wrote: Chris, first of all, if your server has been up for 200 days, I suggest you update the kernel -- you don't say if it's Linux, but chances are that yes... and there have been some security bugs patched recently. That aside. I'm not sure, but it's possible that since you are using a valid host name ('sip.broadvoice.com') in your dial statement, perhaps * tried to talk to it directly and does not consider the section in sip.conf. Just a guess. You will notice from the the sip debug output that * does not even try to authenticate, as if it didn't know about the user/secret. I use the BV number as the section name, so the dial statement essentially looks like: Dial([EMAIL PROTECTED]) Try changing
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
First, thanks for your help. I have been changing these to different values but not getting it. Could you further amplify your statement... Try changing yours to say "broadvoice" and then the corresponding section in sip.conf. Thanks! Chris On Mar 9, 2005, at 12:08 AM, Luki wrote: Chris, first of all, if your server has been up for 200 days, I suggest you update the kernel -- you don't say if it's Linux, but chances are that yes... and there have been some security bugs patched recently. That aside. I'm not sure, but it's possible that since you are using a valid host name ('sip.broadvoice.com') in your dial statement, perhaps * tried to talk to it directly and does not consider the section in sip.conf. Just a guess. You will notice from the the sip debug output that * does not even try to authenticate, as if it didn't know about the user/secret. I use the BV number as the section name, so the dial statement essentially looks like: Dial([EMAIL PROTECTED]) Try changing yours to say "broadvoice" and then the corresponding section in sip.conf. I'm using the DCA server, and didn't have an issue at all when they introduced INVITE authentication on the weekend. This is how my section looks like: [360350] type=peer dtmfmode=inband username=360350 fromuser=360350 secret=XX host=sip.broadvoice.com fromdomain=sip.broadvoice.com canreinvite=no nat=no insecure=very context=incoming outgoinglimit=2 In /etc/hosts I have: 147.135.0.128 sip.broadvoice.com It's the proxy.dca.broadvoice.com server. Hope this helps... --Luki ___ 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 ___ 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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Chris, first of all, if your server has been up for 200 days, I suggest you update the kernel -- you don't say if it's Linux, but chances are that yes... and there have been some security bugs patched recently. That aside. I'm not sure, but it's possible that since you are using a valid host name ('sip.broadvoice.com') in your dial statement, perhaps * tried to talk to it directly and does not consider the section in sip.conf. Just a guess. You will notice from the the sip debug output that * does not even try to authenticate, as if it didn't know about the user/secret. I use the BV number as the section name, so the dial statement essentially looks like: Dial([EMAIL PROTECTED]) Try changing yours to say "broadvoice" and then the corresponding section in sip.conf. I'm using the DCA server, and didn't have an issue at all when they introduced INVITE authentication on the weekend. This is how my section looks like: [360350] type=peer dtmfmode=inband username=360350 fromuser=360350 secret=XX host=sip.broadvoice.com fromdomain=sip.broadvoice.com canreinvite=no nat=no insecure=very context=incoming outgoinglimit=2 In /etc/hosts I have: 147.135.0.128 sip.broadvoice.com It's the proxy.dca.broadvoice.com server. Hope this helps... --Luki ___ 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
[Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
I have been going crazy with this also since Sat. Our server was working perfectly with BV but will now not place calls to BV. Incoming from BV works fine. I felt sad rebooting it today, it had been running for almost 200 days! Here is my error message from the console... Notice I am running today's CVS Connected to Asterisk CVS-03/08/05-14:32:39 currently running on com (pid = 1624) -- Executing Dial("SIP/6050-5bc9", "SIP/[EMAIL PROTECTED]|30") in new stack -- Called [EMAIL PROTECTED] Mar 8 23:11:55 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '"6050" ;tag=as20911f6e' I have tried many versions of sip.conf, here is the current... [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=8475100139 secret=blah username=8475100139 insecure=very context=default authname=8475100139 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no I have tried the different proxies proxy.dca.broadvoice.com, lax, mia, and was originally using chi when the system worked. BV told me Mon that chi is considered a test server that should not be used for production, it is expected to go up and down. My hosts file points to one of the working ones. I verified my account through a softphone. It works fine to BV. There is something wrong with the authentication. Here is the SIP debug... -- Executing Dial("SIP/6050-019c", "SIP/[EMAIL PROTECTED]|30") in new stack We're at xxx.xxx.xxx.xxx port 16776 Answering with capability 2 Answering with capability 4 Answering with capability 8 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c From: "6050" ;tag=as292b9469 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 09 Mar 2005 05:35:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 205 v=0 o=root 3501 3501 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 16776 RTP/AVP 3 0 8 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - (no NAT) to 147.135.8.128:5060 -- Called [EMAIL PROTECTED] com*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c From: "6050" ;tag=as292b9469 To: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE 6 headers, 0 lines com*CLI> Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c From: "6050" ;tag=as292b9469 To: ;tag=SD38ad399- Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE WWW-Authenticate: DIGEST realm="BroadWorks",algorithm=MD5,nonce="1110346372627" Content-Length: 0 8 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c From: "6050" ;tag=as292b9469 To: ;tag=SD38ad399- Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 147.135.8.128:5060 Mar 8 23:35:15 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '"6050" ;tag=as292b9469' TIA Chris ___ 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