Re: [asterisk-users] Can't connect to Asterisk cli
Hello, No, another installation haven't solved the problem! It looks more like something related to the configuration in setting the running environment! Thanks. -- _ -- 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
[asterisk-users] The call is established but without exchanged voice packets
Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see thischan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration[general]context=internalallowguest=noallowoverlap=nobindport=5060bindaddr=0.0.0.0srvlookup=nodisallow=allallow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=IP[7001]type=friendhost=dynamicsecret=123context=internal[7002]type=friendhost=dynamicsecret=456context=internal A snoop capture for my call is uploaded in the following link. I wonder if there is any missing configuration or plugin need to be set here!http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 Thanks. -- _ -- 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] The call is established but without exchanged voice packets
Choose suitable NAT settings from sip.conf turn direct media in sip.conf or per peer off On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.comwrote: Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine. The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see this chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse externip=IP [7001] type=friend host=dynamic secret=123 context=internal [7002] type=friend host=dynamic secret=456 context=internal A snoop capture for my call is uploaded in the following link. I wonder if there is any missing configuration or plugin need to be set here! http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 Thanks. -- _ -- 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 -- Regards ** Muhammad Salman *** -- _ -- 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] Can't connect to Asterisk cli
remove content of /var/log/asterisk/messages /var/log/asterisk/messages run asterisk and post content of /var/log/asterisk/messages to pastebin. On Thu, Sep 19, 2013 at 9:39 AM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, No, another installation haven't solved the problem! It looks more like something related to the configuration in setting the running environment! Thanks. -- _ -- 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] MeetMe and setting conference timeout
exten = 123,1,Set(TIMEOUT(absolute)=3600) exten = 123,n,MeetMe(blah,d) if you are using freepbx and you want to set timeout for all conference rooms go here -http://dn.forceit.ru/asterisk-conference-timeout -- _ -- 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
[asterisk-users] Looking for Asterisk+Pacemaker+Corosync+DRBD example
I'm trying to setup a pair of FreePBX-4.211.64 boxes using Pacemaker, Corosync, and DRBD. All the examples I've found so far use Heartbeat, but Heartbeat is not in the repositories and doesn't want to compile from source. Does anyone have a working configuration they can share or a tutorial they can point me to? Also, what does drbdlinks bring to the party? Isn't just linking the 'top level' directories (/etc/asterisk/, /var/lib/asterisk/, /var/lib/mysql, etc) sufficient? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
[asterisk-users] How to customize CDR(src) value ?
Hi, Asterisk 11 doc says CDR(src) value is read-only (see https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR). For various reasons, I would appreciate to change its value so that it my own presentation rules instead of telco rules. Very often, I'm connected to telcos through DAHDI (and ISDN). For instance, telco presents calls with 123456789 while I would prefer a normalized +34123456789. Whenever I change CallerID presentation, the updated value persists in CDR(callerid) which matches my needs. Unfortunately, for CDR(dst), I'm still looking for an appropriate function or application. Looking at Asterisk doc, I saw NoCDR and ForkCDR apps but couldn't link those to what I'm after. How can I (re-)set CDR(src) value ? Regards -- _ -- 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] How to customize CDR(src) value ?
On Thu, Sep 19, 2013 at 9:02 AM, Olivier oza_4...@yahoo.fr wrote: Hi, Asterisk 11 doc says CDR(src) value is read-only (see https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR). For various reasons, I would appreciate to change its value so that it my own presentation rules instead of telco rules. Very often, I'm connected to telcos through DAHDI (and ISDN). For instance, telco presents calls with 123456789 while I would prefer a normalized +34123456789. Whenever I change CallerID presentation, the updated value persists in CDR(callerid) which matches my needs. Unfortunately, for CDR(dst), I'm still looking for an appropriate function or application. Looking at Asterisk doc, I saw NoCDR and ForkCDR apps but couldn't link those to what I'm after. How can I (re-)set CDR(src) value ? You can't. It is a read-only property. If you want a custom value - my-src or something like that - you can add a new value to your CDR record by using the CDR function, i.e., Set(CDR(my-src)=+34123456789). Certain CDR backends - such as cdr_custom or cdr_adpative_odbc - have the ability to store custom values. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Looking for Asterisk+Pacemaker+Corosync+DRBD example
Be careful with DRDB singe failing drive/corruption on one peers takes down the other too... Check out haast as well (at www.generationd.com) for a commercial asterisk clustering solution. Michelle (GenerationD Systems) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Bakko [asannu...@gmail.com] Sent: Thursday, September 19, 2013 10:24 AM To: Asterisk Users List Subject: Re: [asterisk-users] Looking for Asterisk+Pacemaker+Corosync+DRBD example Hello Edwards you can install fedora repositories and the HeartBeat from those repositories. If the failover is only for two servers, this is a good solution. In the directory list, you have to add /etc/dahdi (is you use dahdi) and /var/spool/asterisk Regards El 19/09/2013 08:58, Steve Edwards escribió: I'm trying to setup a pair of FreePBX-4.211.64 boxes using Pacemaker, Corosync, and DRBD. All the examples I've found so far use Heartbeat, but Heartbeat is not in the repositories and doesn't want to compile from source. Does anyone have a working configuration they can share or a tutorial they can point me to? Also, what does drbdlinks bring to the party? Isn't just linking the 'top level' directories (/etc/asterisk/, /var/lib/asterisk/, /var/lib/mysql, etc) sufficient? -- _ -- 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] Looking for Asterisk+Pacemaker+Corosync+DRBD example
Hello Edwards you can install fedora repositories and the HeartBeat from those repositories. If the failover is only for two servers, this is a good solution. In the directory list, you have to add /etc/dahdi (is you use dahdi) and /var/spool/asterisk Regards El 19/09/2013 08:58, Steve Edwards escribió: I'm trying to setup a pair of FreePBX-4.211.64 boxes using Pacemaker, Corosync, and DRBD. All the examples I've found so far use Heartbeat, but Heartbeat is not in the repositories and doesn't want to compile from source. Does anyone have a working configuration they can share or a tutorial they can point me to? Also, what does drbdlinks bring to the party? Isn't just linking the 'top level' directories (/etc/asterisk/, /var/lib/asterisk/, /var/lib/mysql, etc) sufficient? -- _ -- 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] The call is established but without exchanged voice packets
Asmaa Ahmed wrote: I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine. The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, a nd lead the call to be disconnected after! By checking the logs, I can see this chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) The SIP trace you provided breaks down as follows: X-Lite Asterisk --- --- INVITE(No Auth) --- --- 401 Unauthorized ACK --- INVITE(Auth)--- --- 100 Trying --- 200 OK --- 200 OK (Retransmitted 10 Times) --- BYE OK --- This shows that the three-way handshake (INVITE/200 OK/ACK) used to establish SIP sessions is not completed because Asterisk never receives an ACK from X-Lite. After retransmitting the 200 OK 10 times Asterisk gives up and disconnects the call. Here's my simple sip configuration [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse externip=IP From the SIP trace, I believe 'externip=41.46.164.96' is set. If that is the case, try changing it to 'externip=54.241.129.14'. You should also set localnet as follows: ; RFC 1918 addresses localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 If that doesn't work you can also try setting 'nat=force_rport' instead of 'nat=yes'. [7001] type=friend host=dynamic secret=123 context=internal [7002] type=friend host=dynamic secret=456 context=internal A snoop capture for my call is uploaded in the following link. I wonder if there is any missing configuration or plugin need to be set here! http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 At this point, you should be able to establish a call between the two X-Lite phones that won't get disconnected due to failing to complete the three-way handshake. There may still not be voice because the firewall(s) between Asterisk and the X-Lite phones may block the RTP traffic. The phones appear to be on the same network, so you can try setting 'canreinvite=yes' to workaround this problem until the firewall(s) are configured to allow RTP traffic on the UDP port range specified in 'rtp.conf' (the default range is 1-2). Good luck and please report your progress back to the list. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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
[asterisk-users] AstDB Partial Replication?
Is anyone aware of a way to replicate parts of the AstDB to another Asterisk install? For example, to export all CF entries on a FreePBX based system to another system running FreePBX, I might do: asterisk -rx 'database show' | grep CF This gives me a list of data, which I can rsync to another host to reimport using 'database put'. BUT, the problem comes in when I want to sync CF entries to/from both Asterisk systems. I seem to be having race conditions where an entry is removed on system A, but before that removal can sync to system B, we've already imported that to system A again. Does this make sense? TLDR; How do I sync AstDB entries between two hosts, in both directions, while maintaining data integrity? Thanks --Tim -- _ -- 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] How to customize CDR(src) value ? [SOLVED]
2013/9/19 Matthew Jordan mjor...@digium.com On Thu, Sep 19, 2013 at 9:02 AM, Olivier oza_4...@yahoo.fr wrote: Hi, Asterisk 11 doc says CDR(src) value is read-only (see https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR). For various reasons, I would appreciate to change its value so that it my own presentation rules instead of telco rules. Very often, I'm connected to telcos through DAHDI (and ISDN). For instance, telco presents calls with 123456789 while I would prefer a normalized +34123456789. Whenever I change CallerID presentation, the updated value persists in CDR(callerid) which matches my needs. Unfortunately, for CDR(dst), I'm still looking for an appropriate function or application. Looking at Asterisk doc, I saw NoCDR and ForkCDR apps but couldn't link those to what I'm after. How can I (re-)set CDR(src) value ? You can't. It is a read-only property. If you want a custom value - my-src or something like that - you can add a new value to your CDR record by using the CDR function, i.e., Set(CDR(my-src)=+34123456789). Certain CDR backends - such as cdr_custom or cdr_adpative_odbc - have the ability to store custom values. To me, your suggestion is a very acceptable work around. Thank you very much for it. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
[asterisk-users] iax packet loss again.
I saw this thread which is very similar to my issue, though I cannot solve mine with iptables. http://lists.digium.com/pipermail/asterisk-users/2013-September/280429.html Using asterisk 11.5, IAX does not seem to be able to receive any packets. My IP tables looks like this: root@dlaptop:/home/darryl# iptables -L Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination could it be any simpler with IAX debugging on in asterisk I see this in the console: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00015ms SCall: 00525 DCall: 0 [184.75.215.106:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00014ms SCall: 00890 DCall: 0 [67.205.74.184:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00010ms SCall: 05381 DCall: 0 [99.245.204.155:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00015ms SCall: 00525 DCall: 0 [184.75.215.106:4569] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00014ms SCall: 00890 DCall: 0 [67.205.74.184:4569] Notice there are no Rx-Frames, and my peer table looks like this: dlaptop*CLI iax2 show peers Name/UsernameHost Mask Port Status Description voipms/121322_i 184.75.215.106 (S) 255.255.255.255 4569 UNREACHABLE voipms2/121322_ 67.205.74.184 (S) 255.255.255.255 4569 UNREACHABLE /99.245.204.155 (S) 255.255.255.255 4569 UNREACHABLE 3 iax2 peers [0 online, 3 offline, 0 unmonitored] tcpdump can see all the packets though: 17:23:35.840421 IP 184-75-215-106.amanah.com.iax dlaptop-2.local.iax: UDP, length 12 17:23:35.872904 IP 67.205.74.184.iax dlaptop-2.local.iax: UDP, length 12 17:23:36.790984 IP dlaptop-2.local.iax CPEc47d4f8848e7-CM0026f30cc55d.cpe.net.cable.rogers.com.iax: UDP, length 14 17:23:36.792680 IP CPEc47d4f8848e7-CM0026f30cc55d.cpe.net.cable.rogers.com.iax dlaptop-2.local.iax: UDP, length 12 17:23:36.814493 IP dlaptop-2.local.iax 184-75-215-106.amanah.com.iax: UDP, length 14 17:23:36.834119 IP dlaptop-2.local.iax 67.205.74.184.iax: UDP, length 14 17:23:36.842537 IP 184-75-215-106.amanah.com.iax dlaptop-2.local.iax: UDP, length 12 17:23:36.877078 IP 67.205.74.184.iax dlaptop-2.local.iax: UDP, length 12 17:23:43.836844 IP dlaptop-2.local.iax CPEc47d4f8848e7-CM0026f30cc55d.cpe.net.cable.rogers.com.iax: UDP, length 24 17:23:43.838705 IP CPEc47d4f8848e7-CM0026f30cc55d.cpe.net.cable.rogers.com.iax dlaptop-2.local.iax: UDP, length 65 but my socket buffers are backing up horribly: root@dlaptop:/home/darryl# lsof -n -P -Tq | grep UDP | grep 4569 lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system /home/darryl/.gvfs Output information may be incomplete. asterisk 2575root8u IPv4 334734592 0t0 UDP *:4569 (QR=163904 QS=0) Am i crazy? Is there something as simple as iptables that I missed? Or is there some kind of bug in Asterisk which is being missed? I've only had this issue on two machines which I've compiled 11.5 on. Generally all my production machines are using the stock version 1.8 which is in the Ubuntu 12.04 repository. Unloading and reloading the chan_iax module only has the effect of resetting the receive queue size in lsof. Anyone have any ideas what I could possibly be missing here? Sip works fine by the way. Thanks Darryl -- _ -- 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
[asterisk-users] proper use of Internal Timing
Hi All, Could anyone tell me the real use of internal_ timing=yes option on asterisk.conf file? I am using asterisk 1.4.22. As per my understanding if we don't have any TDM card installed with appropriate driver, we use internal_timing = yes to get the timing from ztdummy /ztDahdi. Is there any advantage on enabling internal_timing=yes even if we are proving timing from TDM card? I would really appreciate your feedback. ThanksSam -- _ -- 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] proper use of Internal Timing
And here I thought I was back in the dark ages using 1.4.44!! You had better consider moving up to a more current version before you get bit real hard! John Novack Comp Aholic wrote: Hi All, Could anyone tell me the real use of internal_ timing=yes option on asterisk.conf file? I am using asterisk 1.4.22. As per my understanding if we don't have any TDM card installed with appropriate driver, we use internal_timing = yes to get the timing from ztdummy /ztDahdi. Is there any advantage on enabling internal_timing=yes even if we are proving timing from TDM card? I would really appreciate your feedback. Thanks Sam -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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
