Re: [asterisk-users] Asterisk SIP TCP
;ignoreregexpire=yes; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether ; it has expired or not; if it expires while the realtime peer ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage Also remove all qualify related parameters and keepalive if set Hope it will solve your problem Regards, Bharat Lalcheta On Tue, Apr 16, 2013 at 11:26 AM, Zohair Raza engineerzuhairr...@gmail.comwrote: Here is what I have, also attached sip show settings output and part of sip.conf in issues [general] udpbindaddr=172.20.255.40 transport=udp,tcp tcpenable=yes tlsenable=no tcpbindaddr=172.20.255.40 directrtpsetup=no directmedia=yes allowguest=no match_auth_username=yes tos_sip=AF31 tos_audio=ef tos=0xB8 tos_video=af41 ; Sets TOS for RTP video packets. tos_text=af41 ; Sets TOS for RTP text packets. trustrpid = yes ; If Remote-Party-ID should be trusted sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no) disallow=all allow=alaw allow=ulaw allow=g729 maxforwards=70 relaxdtmf=yes rpid_update = yes maxexpiry=400 minexpiry=60 defaultexpiry=300 qualify=yes ; notifycid = yes ; Control whether caller ID information is sent along with dialog-info+xml notifications (supported by snom phones) qualifyfreq=300 qualifypeers=1 qualifygap=2000 registertimeout=20 registerattempts=10 progressinband=never ignoreregexpire=yes On Tue, Apr 16, 2013 at 9:44 AM, Bharat Lalcheta bharatlalch...@gmail.com wrote: Can you give sip.conf ? I am using asterisk 1.8.15 on both udp and tcp and not able to generate this scenario. Regards, Bharat Lalcheta On Tue, Apr 16, 2013 at 11:03 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Backtrace and logs attached here : https://issues.asterisk.org/jira/browse/ASTERISK-21447 Regards, Zohair Raza On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry markhenry...@gmail.comwrote: this is my secondary email Regards Zohair On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.comwrote: Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com wrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza --
Re: [asterisk-users] Asterisk SIP TCP
On Tue, Apr 16, 2013 at 10:12 AM, Bharat Lalcheta bharatlalch...@gmail.comwrote: ;ignoreregexpire=yes; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether ; it has expired or not; if it expires while the realtime peer ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage I tried setting it to no already, but asterisk was keep trying to establish connection at old ip and port Also remove all qualify related parameters and keepalive if set when qualify is set to no, does qualifyfreq have an effect? because I tried qualify=no bu the qualifyfreq was set at that time, I set qualifyfreq=300 but requests were going every few seconds (around 30 secs) One thing I doubt is Insecure field, it is set to no at the moment. By name it is for security only but setting it insecure=port may effect? Hope it will solve your problem Regards, Bharat Lalcheta On Tue, Apr 16, 2013 at 11:26 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Here is what I have, also attached sip show settings output and part of sip.conf in issues [general] udpbindaddr=172.20.255.40 transport=udp,tcp tcpenable=yes tlsenable=no tcpbindaddr=172.20.255.40 directrtpsetup=no directmedia=yes allowguest=no match_auth_username=yes tos_sip=AF31 tos_audio=ef tos=0xB8 tos_video=af41 ; Sets TOS for RTP video packets. tos_text=af41 ; Sets TOS for RTP text packets. trustrpid = yes ; If Remote-Party-ID should be trusted sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no) disallow=all allow=alaw allow=ulaw allow=g729 maxforwards=70 relaxdtmf=yes rpid_update = yes maxexpiry=400 minexpiry=60 defaultexpiry=300 qualify=yes ; notifycid = yes ; Control whether caller ID information is sent along with dialog-info+xml notifications (supported by snom phones) qualifyfreq=300 qualifypeers=1 qualifygap=2000 registertimeout=20 registerattempts=10 progressinband=never ignoreregexpire=yes On Tue, Apr 16, 2013 at 9:44 AM, Bharat Lalcheta bharatlalch...@gmail.com wrote: Can you give sip.conf ? I am using asterisk 1.8.15 on both udp and tcp and not able to generate this scenario. Regards, Bharat Lalcheta On Tue, Apr 16, 2013 at 11:03 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Backtrace and logs attached here : https://issues.asterisk.org/jira/browse/ASTERISK-21447 Regards, Zohair Raza On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry markhenry...@gmail.comwrote: this is my secondary email Regards Zohair On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.comwrote: Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com wrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len
[asterisk-users] Asterisk SIP TCP
Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- 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] Asterisk SIP TCP
I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.comwrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- 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] Asterisk SIP TCP
Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.comwrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- 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 -- _ -- 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] Asterisk SIP TCP
this is my secondary email Regards Zohair On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.com wrote: Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com wrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- 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 -- _ -- 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] Asterisk SIP TCP
Backtrace and logs attached here : https://issues.asterisk.org/jira/browse/ASTERISK-21447 Regards, Zohair Raza On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry markhenry...@gmail.com wrote: this is my secondary email Regards Zohair On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.comwrote: Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com wrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- 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 -- _ -- 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] Asterisk SIP TCP
Can you give sip.conf ? I am using asterisk 1.8.15 on both udp and tcp and not able to generate this scenario. Regards, Bharat Lalcheta On Tue, Apr 16, 2013 at 11:03 AM, Zohair Raza engineerzuhairr...@gmail.comwrote: Backtrace and logs attached here : https://issues.asterisk.org/jira/browse/ASTERISK-21447 Regards, Zohair Raza On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry markhenry...@gmail.comwrote: this is my secondary email Regards Zohair On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.comwrote: Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com wrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- 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 -- _ -- 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 -- Bharat Lalcheta -- _ -- 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] Asterisk SIP TCP
Here is what I have, also attached sip show settings output and part of sip.conf in issues [general] udpbindaddr=172.20.255.40 transport=udp,tcp tcpenable=yes tlsenable=no tcpbindaddr=172.20.255.40 directrtpsetup=no directmedia=yes allowguest=no match_auth_username=yes tos_sip=AF31 tos_audio=ef tos=0xB8 tos_video=af41 ; Sets TOS for RTP video packets. tos_text=af41 ; Sets TOS for RTP text packets. trustrpid = yes ; If Remote-Party-ID should be trusted sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no) disallow=all allow=alaw allow=ulaw allow=g729 maxforwards=70 relaxdtmf=yes rpid_update = yes maxexpiry=400 minexpiry=60 defaultexpiry=300 qualify=yes ; notifycid = yes ; Control whether caller ID information is sent along with dialog-info+xml notifications (supported by snom phones) qualifyfreq=300 qualifypeers=1 qualifygap=2000 registertimeout=20 registerattempts=10 progressinband=never ignoreregexpire=yes On Tue, Apr 16, 2013 at 9:44 AM, Bharat Lalcheta bharatlalch...@gmail.comwrote: Can you give sip.conf ? I am using asterisk 1.8.15 on both udp and tcp and not able to generate this scenario. Regards, Bharat Lalcheta On Tue, Apr 16, 2013 at 11:03 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Backtrace and logs attached here : https://issues.asterisk.org/jira/browse/ASTERISK-21447 Regards, Zohair Raza On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry markhenry...@gmail.comwrote: this is my secondary email Regards Zohair On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.comwrote: Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com wrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- 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 -- _ -- 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 -- Bharat Lalcheta --
Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging
Hi Jim, Thanks for your kind offer - I may well need to pick your knowledge at some point. I've not long got 2007 up and running and am trying to convert a few people back at the office that this could be something useful to look at (generally the NBX phone system we have currently doesn't impress as much as it should, so interesting days ahead :)) Thanks again Wayne. Sigma Networks wrote: I have a production PBX (1.6.0.9) 150+ phones with MS Exchange 2007 and OCS working very well out of the box. We're using SIP/TCP support in 1.6.x; Believe it or not the most challenging part is to get MWI signaling back from Exchange. Let me know if I can help. Jim j...@sigma-networks.com; 408-701-9929 - Original Message - From: Wayne wa...@planetwayne.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 11, 2009 10:10:05 AM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging David Backeberg wrote: I would ask the question the other way around. Are there any plans for Microsoft to release a unified messaging product that will comply with SIP over UDP? I do see your point in a potential (ok who are kidding - real) risk of a system crash with using MS having full control over your phone system but, I was thinking along the lines of using exchange really only as a messaging system - ie voice mail, email reader. From what I can make out MS are even going along the lines of doing speech to text with 2010 version (I think it has text to speech already). I would have to agree that the PBX side of things is held still by Asterisk and I don't see my view on that changing yet, but, I would imagine MS would dig their heels in rather than changing exchange. The Asterisk community, being more open minded to change, could easily(?) make this work. Thanks Wayne. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging
I have a production PBX (1.6.0.9) 150+ phones with MS Exchange 2007 and OCS working very well out of the box. We're using SIP/TCP support in 1.6.x; Believe it or not the most challenging part is to get MWI signaling back from Exchange. Let me know if I can help. Jim j...@sigma-networks.com; 408-701-9929 - Original Message - From: Wayne wa...@planetwayne.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 11, 2009 10:10:05 AM GMT -08:00 US/Canada Pacific Subject: Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging David Backeberg wrote: I would ask the question the other way around. Are there any plans for Microsoft to release a unified messaging product that will comply with SIP over UDP? I do see your point in a potential (ok who are kidding - real) risk of a system crash with using MS having full control over your phone system but, I was thinking along the lines of using exchange really only as a messaging system - ie voice mail, email reader. From what I can make out MS are even going along the lines of doing speech to text with 2010 version (I think it has text to speech already). I would have to agree that the PBX side of things is held still by Asterisk and I don't see my view on that changing yet, but, I would imagine MS would dig their heels in rather than changing exchange. The Asterisk community, being more open minded to change, could easily(?) make this work. Thanks Wayne. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging
David Backeberg wrote: I would ask the question the other way around. Are there any plans for Microsoft to release a unified messaging product that will comply with SIP over UDP? I do see your point in a potential (ok who are kidding - real) risk of a system crash with using MS having full control over your phone system but, I was thinking along the lines of using exchange really only as a messaging system - ie voice mail, email reader. From what I can make out MS are even going along the lines of doing speech to text with 2010 version (I think it has text to speech already). I would have to agree that the PBX side of things is held still by Asterisk and I don't see my view on that changing yet, but, I would imagine MS would dig their heels in rather than changing exchange. The Asterisk community, being more open minded to change, could easily(?) make this work. Thanks Wayne. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging
On Wed, 2009-06-10 at 23:00 +0100, Wayne wrote: I was wondering what the current development plans / patches etc are to allow Asterisk to talk to Exchange 2007 Unified Messaging with respect to adding SIP over TCP support? There is experimental support for SIP over TCP in Asterisk 1.6.0 and later. It's probably not perfect yet, but we'd be happy to hear how it works for you, and that will help the Asterisk developers make it better. As far as other things related to the vague notion of unified communications, there's the code that Terry Wilson just added on being able to read Exchange calendars (iCal/CalDAV are supported as well) from the Asterisk dialplan, there's plenty of Jabber work being done on the IM side, and Asterisk can already store voicemail in an IMAP mail server. (I've long since let go of my Windows skills, but I'm assuming that modern versions of Exchange still let you communicate via IMAP, right?) In short, there are a lot of exciting things happening in the world of Asterisk with regards to unified communications. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging
Hi all, I was wondering what the current development plans / patches etc are to allow Asterisk to talk to Exchange 2007 Unified Messaging with respect to adding SIP over TCP support? I've been googleing and looking through various posts on the wiki and all seem to suggest that it could be happening in later versions but all seem to be dated quite old now and nothing suggesting that its now done. Thanks Wayne. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - SIP - TCP and Exchange 2007 Unified Messaging
On Wed, Jun 10, 2009 at 6:00 PM, Waynewa...@planetwayne.com wrote: Hi all, I was wondering what the current development plans / patches etc are to allow Asterisk to talk to Exchange 2007 Unified Messaging with respect to adding SIP over TCP support? I would ask the question the other way around. Are there any plans for Microsoft to release a unified messaging product that will comply with SIP over UDP? I've been wondering whether people are going to trust Microsoft to handle their phone calls. Do you really not want to be able to call 911 because you just got a blue screen of death? The claims I've seen about why I want Microsoft to handle my phones is that I'll save a lot of money by eliminating desk phones. Sounds like a scary, scary place to work when something goes wrong with the Microsoft OS, but that never happens, right? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
