Re: [asterisk-users] sipgate outgoing calls
Probably worth noting that sipgate will close (at least in the U.S.) on Oct. 31: http://www.besttechie.com/2013/09/13/voip-provider-sipgate-will-close-oct-31/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten Wemheuer Sent: Thursday, September 19, 2013 10:54 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sipgate outgoing calls Hi, Am Mittwoch, den 18.09.2013, 14:29 +0100 schrieb gpxctawjc...@irational.org: Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line What Sipgate product are You using? At least in Germany there are different configurations for the different products necessary. For Sipgate trunking and Sipgate team You have to configure an outboundproxy (which differs between both products). For Sipgate Basic you don't need an outboundproxy. As far as I remember there was an issue with some asterisk versions and the outboundproxy for Sipdate team. Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
On Thu, 19 Sep 2013, David Duffett wrote: i am getting these errors in asterisk cli -- Executing [01179553708@default:1] Set(SIP/-015b, CALLERID(num)=xx) in new stack -- Executing [01179553708@default:2] Dial(SIP/-015b, SIP/01179553708@sipgate,30,trg) in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Called 01179553708@sipgate [Sep 19 10:08:03] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to ' sip:xx...@sipgate.co.uk;tag=as055d9532' -- SIP/sipgate-015c is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) any further ideas ? many thanks I believe registration is in place, otherwise inbound calls would not work. Also, registration is not required for outbound calls to work. I would suggest cutting down your sip.conf profile to this minimal configuration: host=sipgate.co.uk username=xxx fromuser=xxx insecure=invite,port secret=xxx context=my-inbound-context type=peer If outbound calls still do not with this, I would suggest that there may be an issue in the general section of your sip.conf Assuming calls do work, you can then add any other configuration lines you feel are necessary - but remember, as with all Asterisk configuration files, less is more :-) On 18 Sep 2013 22:06, Administrator TOOTAI ad...@tootai.net wrote: Le 18/09/2013 15:29, gpxctawjc...@irational.org a écrit : Hello Hi i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615@sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '01179553708 sip:sip...@sipgate.co.uk;tag=as30eb9dd1' -- SIP/sipgate-014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) here is my sip.conf file [general] port = 5060 bindaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register = SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk qualify=yes disallow=all allow=alaw dtmfmode=rfc2833 SIP-ID:SIP-Password obviously, i replace these with my login details but, are these the same thing ? SIP-Password SIP_PASSWORD the sipgate guides are contradictory http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri sk any suggestions ? Many thanks My setup with sipgate.de [sipgate] type=peer secret=MY-PASSWORD defaultuser=SIP-ID host=217.10.79.9 fromuser=SIP-ID fromdomain=sipgate.de context=incoming-sipgate ;qualify=900 dtmfmode=info directmedia=yes insecure=port,invite disallow=all allow=ulaw,alaw accountcode=MY-ACCOUNTCODE What you forget is to register with them: ; Sipgate register = SIP-ID:my-passw...@sipgate.de/SIP-ID ;don't accept to register without FQDN Hope that help -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
It looks like the challenge response after INVITE is not been accepted. Provide more detail. $ sip set debug peer sipgate -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 9/19/2013 7:10 PM, gpxctawjc...@irational.org wrote: On Thu, 19 Sep 2013, David Duffett wrote: i am getting these errors in asterisk cli -- Executing [01179553708@default:1] Set(SIP/-015b, CALLERID(num)=xx) in new stack -- Executing [01179553708@default:2] Dial(SIP/-015b, SIP/01179553708@sipgate,30,trg) in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Called 01179553708@sipgate [Sep 19 10:08:03] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to ' sip:xx...@sipgate.co.uk;tag=as055d9532' -- SIP/sipgate-015c is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) any further ideas ? many thanks I believe registration is in place, otherwise inbound calls would not work. Also, registration is not required for outbound calls to work. I would suggest cutting down your sip.conf profile to this minimal configuration: host=sipgate.co.uk username=xxx fromuser=xxx insecure=invite,port secret=xxx context=my-inbound-context type=peer If outbound calls still do not with this, I would suggest that there may be an issue in the general section of your sip.conf Assuming calls do work, you can then add any other configuration lines you feel are necessary - but remember, as with all Asterisk configuration files, less is more :-) On 18 Sep 2013 22:06, Administrator TOOTAI ad...@tootai.net wrote: Le 18/09/2013 15:29, gpxctawjc...@irational.org a écrit : Hello Hi i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615@sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '01179553708 sip:sip...@sipgate.co.uk;tag=as30eb9dd1' -- SIP/sipgate-014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) here is my sip.conf file [general] port = 5060 bindaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register = SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk qualify=yes disallow=all allow=alaw dtmfmode=rfc2833 SIP-ID:SIP-Password obviously, i replace these with my login details but, are these the same thing ? SIP-Password SIP_PASSWORD the sipgate guides are contradictory http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri sk any suggestions ? Many thanks My setup with sipgate.de [sipgate] type=peer secret=MY-PASSWORD defaultuser=SIP-ID host=217.10.79.9 fromuser=SIP-ID fromdomain=sipgate.de context=incoming-sipgate ;qualify=900 dtmfmode=info directmedia=yes insecure=port,invite disallow=all allow=ulaw,alaw accountcode=MY-ACCOUNTCODE What you forget is to register with them: ; Sipgate register = SIP-ID:my-passw...@sipgate.de/SIP-ID ;don't accept to register without FQDN Hope that help -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
Re: [asterisk-users] sipgate outgoing calls
It looks like the challenge response after INVITE is not been accepted. Provide more detail. $ sip set debug peer sipgate server*CLI sip set debug peer sipgate SIP Debugging Enabled for IP: 217.10.79.23:5060 Really destroying SIP dialog '3ef8ff1a6ec360626af409b112b860ee@127.0.1.1' Method: REGISTER -- Registered SIP 'x' at 86.140.115.135 port 5060 == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Executing [01179553708@default:1] Set(SIP/x-015d, CALLERID(num)=x) in new stack -- Executing [01179553708@default:2] Dial(SIP/x-015d, SIP/01179553708@sipgate,30,trg) in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Called 01179553708@sipgate [Sep 19 10:50:15] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to 'x sip:xx...@sipgate.co.uk;tag=as629ee6f8' -- SIP/sipgate-015e is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [01179553708@default:3] Hangup(SIP/x-015d, ) in new stack == Spawn extension (default, 01179553708, 3) exited non-zero on 'SIP/x-015d' --- server*CLI --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK73a1638c;rport=5060 From: asterisk sip:asterisk@92.63.131.3;tag=as5dcb32d8 To: sip:sipgate.co.uk;tag=99199803810c7e807ea44745826d9aa4.df2d Call-ID: 65cb07675eefaaef5f655e8a0be6b2f6@92.63.131.3 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Content-Length: 0 - --- (11 headers 0 lines) --- Really destroying SIP dialog '65cb07675eefaaef5f655e8a0be6b2f6@92.63.131.3' Method: OPTIONS [Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:11588 sip_reregister:-- Re-registration for xxx...@sipgate.co.uk REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 217.10.79.23:5060: REGISTER sip:sipgate.co.uk SIP/2.0 Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK13b202d7;rport Max-Forwards: 70 From: sip:x...@sipgate.co.uk;tag=as19513575 To: sip:xx...@sipgate.co.uk Call-ID: 3ef8ff1a6ec360626af409b112b860ee@127.0.1.1 CSeq: 182 REGISTER User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4 Authorization: Digest username=xx, realm=sipgate.co.uk, algorithm=MD5, uri=sip:sipgate.co.uk, nonce=523ac9531b1cc7962e07bce6a76683ee24da44d0, response=c82fac231a41085c275899ad84f73317 Expires: 120 Contact: sip:xx@92.63.131.3 Content-Length: 0 --- server*CLI --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.3:5060;received=92.63.131.3;branch=z9hG4bK13b202d7;rport=5060 From: sip:xx...@sipgate.co.uk;tag=as19513575 To: sip:xx...@sipgate.co.uk;tag=c3e497ecaece77a8e244e564b4212178.3e46 Call-ID: 3ef8ff1a6ec360626af409b112b860ee@127.0.1.1 CSeq: 182 REGISTER Contact: sip:xx@92.63.131.3;expires=120 Content-Length: 0 - --- (8 headers 0 lines) --- Scheduling destruction of SIP dialog '3ef8ff1a6ec360626af409b112b860ee@127.0.1.1' in 32000 ms (Method: REGISTER) [Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:18301 handle_response_register: Outbound Registration: Expiry for sipgate.co.uk is 120 sec (Scheduling reregistration in 105 s) Reliably Transmitting (no NAT) to 217.10.79.23:5060: OPTIONS sip:sipgate.co.uk SIP/2.0 Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK2d73294a;rport Max-Forwards: 70 From: asterisk sip:asterisk@92.63.131.3;tag=as5afd24b2 To: sip:sipgate.co.uk Contact: sip:asterisk@92.63.131.3 Call-ID: 1becd4dc336869c4692fc4e55e109562@92.63.131.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4 Date: Thu, 19 Sep 2013 09:51:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- server*CLI --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK2d73294a;rport=5060 From: asterisk sip:asterisk@92.63.131.3;tag=as5afd24b2 To: sip:sipgate.co.uk;tag=99199803810c7e807ea44745826d9aa4.c753 Call-ID: 1becd4dc336869c4692fc4e55e109562@92.63.131.3 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Content-Length: 0 - --- (11 headers 0 lines) --- Really destroying SIP dialog '1becd4dc336869c4692fc4e55e109562@92.63.131.3' Method: OPTIONS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
Challenge authentication look good. --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Are you sure this number format 01179553708 is accepted in that SIP trunk? Some VOIP providers only accept international format. -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 9/19/2013 7:55 PM, gpxctawjc...@irational.org wrote: It looks like the challenge response after INVITE is not been accepted. Provide more detail. $ sip set debug peer sipgate server*CLI sip set debug peer sipgate SIP Debugging Enabled for IP: 217.10.79.23:5060 Really destroying SIP dialog '3ef8ff1a6ec360626af409b112b860ee@127.0.1.1' Method: REGISTER -- Registered SIP 'x' at 86.140.115.135 port 5060 == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Executing [01179553708@default:1] Set(SIP/x-015d, CALLERID(num)=x) in new stack -- Executing [01179553708@default:2] Dial(SIP/x-015d, SIP/01179553708@sipgate,30,trg) in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Called 01179553708@sipgate [Sep 19 10:50:15] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to 'x sip:xx...@sipgate.co.uk;tag=as629ee6f8' -- SIP/sipgate-015e is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [01179553708@default:3] Hangup(SIP/x-015d, ) in new stack == Spawn extension (default, 01179553708, 3) exited non-zero on 'SIP/x-015d' --- server*CLI --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK73a1638c;rport=5060 From: asterisk sip:asterisk@92.63.131.3;tag=as5dcb32d8 To: sip:sipgate.co.uk;tag=99199803810c7e807ea44745826d9aa4.df2d Call-ID: 65cb07675eefaaef5f655e8a0be6b2f6@92.63.131.3 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Content-Length: 0 - --- (11 headers 0 lines) --- Really destroying SIP dialog '65cb07675eefaaef5f655e8a0be6b2f6@92.63.131.3' Method: OPTIONS [Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:11588 sip_reregister: -- Re-registration for xxx...@sipgate.co.uk REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 217.10.79.23:5060: REGISTER sip:sipgate.co.uk SIP/2.0 Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK13b202d7;rport Max-Forwards: 70 From: sip:x...@sipgate.co.uk;tag=as19513575 To: sip:xx...@sipgate.co.uk Call-ID: 3ef8ff1a6ec360626af409b112b860ee@127.0.1.1 CSeq: 182 REGISTER User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4 Authorization: Digest username=xx, realm=sipgate.co.uk, algorithm=MD5, uri=sip:sipgate.co.uk, nonce=523ac9531b1cc7962e07bce6a76683ee24da44d0, response=c82fac231a41085c275899ad84f73317 Expires: 120 Contact: sip:xx@92.63.131.3 Content-Length: 0 --- server*CLI --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.3:5060;received=92.63.131.3;branch=z9hG4bK13b202d7;rport=5060 From: sip:xx...@sipgate.co.uk;tag=as19513575 To: sip:xx...@sipgate.co.uk;tag=c3e497ecaece77a8e244e564b4212178.3e46 Call-ID: 3ef8ff1a6ec360626af409b112b860ee@127.0.1.1 CSeq: 182 REGISTER Contact: sip:xx@92.63.131.3;expires=120 Content-Length: 0 - --- (8 headers 0 lines) --- Scheduling destruction of SIP dialog '3ef8ff1a6ec360626af409b112b860ee@127.0.1.1' in 32000 ms (Method: REGISTER) [Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:18301 handle_response_register: Outbound Registration: Expiry for sipgate.co.uk is 120 sec (Scheduling reregistration in 105 s) Reliably Transmitting (no NAT) to 217.10.79.23:5060: OPTIONS sip:sipgate.co.uk SIP/2.0 Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK2d73294a;rport Max-Forwards: 70 From: asterisk sip:asterisk@92.63.131.3;tag=as5afd24b2 To: sip:sipgate.co.uk Contact: sip:asterisk@92.63.131.3 Call-ID: 1becd4dc336869c4692fc4e55e109562@92.63.131.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4 Date: Thu, 19 Sep 2013 09:51:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- server*CLI --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK2d73294a;rport=5060 From: asterisk sip:asterisk@92.63.131.3;tag=as5afd24b2 To: sip:sipgate.co.uk;tag=99199803810c7e807ea44745826d9aa4.c753 Call-ID: 1becd4dc336869c4692fc4e55e109562@92.63.131.3 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Content-Length: 0 - --- (11 headers 0 lines) --- Really destroying SIP dialog '1becd4dc336869c4692fc4e55e109562@92.63.131.3' Method: OPTIONS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] sipgate outgoing calls
On Thu, 19 Sep 2013, Miguel Oyarzo wrote: Challenge authentication look good. --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Are you sure this number format 01179553708 is accepted in that SIP trunk? Some VOIP providers only accept international format. when i use a softphone client to connect directly to sipgate i can dial 01179553708 and get through -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
you have insecure=port,invite in sipgate peer? On Thu, Sep 19, 2013 at 12:26 PM, gpxctawjc...@irational.org wrote: On Thu, 19 Sep 2013, Miguel Oyarzo wrote: Challenge authentication look good. --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Are you sure this number format 01179553708 is accepted in that SIP trunk? Some VOIP providers only accept international format. when i use a softphone client to connect directly to sipgate i can dial 01179553708 and get through -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
What you don't have mentioned yet is whether your outbound call reaches the destination. -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 9/19/2013 8:26 PM, gpxctawjc...@irational.org wrote: On Thu, 19 Sep 2013, Miguel Oyarzo wrote: Challenge authentication look good. --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Are you sure this number format 01179553708 is accepted in that SIP trunk? Some VOIP providers only accept international format. when i use a softphone client to connect directly to sipgate i can dial 01179553708 and get through -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
Le 19/09/2013 05:01, David Duffett a écrit : I believe registration is in place, otherwise inbound calls would not work. Yes, I didn't read carefully the original message, sorry. [...] -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
Hi, Am Mittwoch, den 18.09.2013, 14:29 +0100 schrieb gpxctawjc...@irational.org: Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line What Sipgate product are You using? At least in Germany there are different configurations for the different products necessary. For Sipgate trunking and Sipgate team You have to configure an outboundproxy (which differs between both products). For Sipgate Basic you don't need an outboundproxy. As far as I remember there was an issue with some asterisk versions and the outboundproxy for Sipdate team. Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
Le 18/09/2013 15:29, gpxctawjc...@irational.org a écrit : Hello Hi i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615@sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '01179553708 sip:sip...@sipgate.co.uk;tag=as30eb9dd1' -- SIP/sipgate-014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) here is my sip.conf file [general] port = 5060 bindaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register = SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk qualify=yes disallow=all allow=alaw dtmfmode=rfc2833 SIP-ID:SIP-Password obviously, i replace these with my login details but, are these the same thing ? SIP-Password SIP_PASSWORD the sipgate guides are contradictory http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri sk any suggestions ? Many thanks My setup with sipgate.de [sipgate] type=peer secret=MY-PASSWORD defaultuser=SIP-ID host=217.10.79.9 fromuser=SIP-ID fromdomain=sipgate.de context=incoming-sipgate ;qualify=900 dtmfmode=info directmedia=yes insecure=port,invite disallow=all allow=ulaw,alaw accountcode=MY-ACCOUNTCODE What you forget is to register with them: ; Sipgate register = SIP-ID:my-passw...@sipgate.de/SIP-ID ;don't accept to register without FQDN Hope that help -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sipgate outgoing calls
I believe registration is in place, otherwise inbound calls would not work. Also, registration is not required for outbound calls to work. I would suggest cutting down your sip.conf profile to this minimal configuration: host=sipgate.co.uk username=xxx fromuser=xxx insecure=invite,port secret=xxx context=my-inbound-context type=peer If outbound calls still do not with this, I would suggest that there may be an issue in the general section of your sip.conf Assuming calls do work, you can then add any other configuration lines you feel are necessary - but remember, as with all Asterisk configuration files, less is more :-) On 18 Sep 2013 22:06, Administrator TOOTAI ad...@tootai.net wrote: Le 18/09/2013 15:29, gpxctawjc...@irational.org a écrit : Hello Hi i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615@sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '01179553708 sip:sip...@sipgate.co.uk;**tag=as30eb9dd1' -- SIP/sipgate-014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) here is my sip.conf file [general] port = 5060 bindaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register = SIP-ID:SIP-Password@sipgate.**co.uk/SIP-IDhttp://SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.**sipgate.co.ukhttp://proxy.live.sipgate.co.uk qualify=yes disallow=all allow=alaw dtmfmode=rfc2833 SIP-ID:SIP-Password obviously, i replace these with my login details but, are these the same thing ? SIP-Password SIP_PASSWORD the sipgate guides are contradictory http://www.sipgate.com/faq/**article/394/How_do_I_**configure_Asteriskhttp://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk http://www.live.sipgate.co.uk/**faq/article/508/How_do_I_** configure_Asterihttp://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri sk any suggestions ? Many thanks My setup with sipgate.de [sipgate] type=peer secret=MY-PASSWORD defaultuser=SIP-ID host=217.10.79.9 fromuser=SIP-ID fromdomain=sipgate.de context=incoming-sipgate ;qualify=900 dtmfmode=info directmedia=yes insecure=port,invite disallow=all allow=ulaw,alaw accountcode=MY-ACCOUNTCODE What you forget is to register with them: ; Sipgate register = SIP-ID:my-passw...@sipgate.de/**SIP-IDhttp://SIP-ID:my-passw...@sipgate.de/SIP-ID;don't accept to register without FQDN Hope that help -- Daniel -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
