Re: [asterisk-users] SIP call drops after 32 seconds, but only when....
On 22.11.2014, at 13:40, Yves A. yves...@gmx.de wrote: I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Do a 'sip set debug on' and see what they (Asterisk and the registrar) are talking about just before the call drops. -- marie -- _ -- 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] SIP call drops after 32 seconds, but only when....
Call drop after 30+sec happens if RTP is not received by asterisk for 30 seconds (RTP Timeout). You should look for media IP address in SDP. If there is firewall, apart from port UDP/5060, you also need to open port UDP/1-UDP/2 (standard RTP ports) You should try with RTP debug. It should show bidirectional traffic. If not, you surely have an issue with media IP or ports. *Thanks Regards,* Amit Patkar On 11/27/2014 10:01 AM, Marie Fischer wrote: On 22.11.2014, at 13:40, Yves A. yves...@gmx.de wrote: I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Do a 'sip set debug on' and see what they (Asterisk and the registrar) are talking about just before the call drops. -- _ -- 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] SIP call drops after 32 seconds, but only when....
Hi, I know this Bug,,, at least when you´re talking about x-lite 3... quite annoying, but if you know it... so no... its not the phone... tested with zoiper and 3cx ... both work...but the problem occurs ONLY, as soon as I register at more than one registrar... yves Am 22.11.2014 um 19:19 schrieb Ron Wheeler: You might check your phones as well. We had this problem early on with a softphone and it was a setting in the phone that was set to hang up after 30 seconds of inactivity in case of network disruption. For some reason it was detecting network disruption in every call even when the calls were proceeding normally. Unchecking this box solved the problem. It may not be related to your problem but if it is the cause, you will spend a lot of time trying to fix this in Asterisk. :-D At least I did! On the bright side, it does force people to get point in a hurry! Ron On 22/11/2014 12:50 PM, Eric Wieling wrote: Try setting directmedia=no in sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Saturday, November 22, 2014 8:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Are both your servers behind the same NAT router? thanks for taking part... I don´t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- 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] SIP call drops after 32 seconds, but only when....
Hi, the useragents nothing to do with the problem... i tried numeric, alpha and alphanumeric... no difference. they work all as long as I only use ONE registrar... as soon as I register at more than one registrar... the line drops after 32 seconds really strange. yves Am 22.11.2014 um 19:01 schrieb Rafael Visser: Hi Yves.. This may be silly... but what is the useragent of your sip configuration? In the case that useragent has some special characters like (., please remove it and tell us if there is any change!!. Regards. rv 2014-11-22 14:50 GMT-03:00 Eric Wieling ewiel...@nyigc.com mailto:ewiel...@nyigc.com: Try setting directmedia=no in sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Saturday, November 22, 2014 8:06 AM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Are both your servers behind the same NAT router? thanks for taking part... I don´t know... one is siptrunk.ovh.net http://siptrunk.ovh.net and the other one is sip.ovh.fr http://sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- 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 --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- 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] SIP call drops after 32 seconds, but only when....
hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in between... I tried to work around this by increasing the settings for timerb... but I realized that asterisk does not care at all, what I set this value to... sip show settings always gives me 32000ms, and it does not make any difference if I configure timerb in the general context or in the phone context... any ideas? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- 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] SIP call drops after 32 seconds, but only when....
but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Are both your servers behind the same NAT router? -- Andreas Sikkema -- _ -- 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] SIP call drops after 32 seconds, but only when....
Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Are both your servers behind the same NAT router? thanks for taking part... I don´t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- 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] SIP call drops after 32 seconds, but only when....
Try setting directmedia=no in sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Saturday, November 22, 2014 8:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Are both your servers behind the same NAT router? thanks for taking part... I don´t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- 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] SIP call drops after 32 seconds, but only when....
Hi Yves.. This may be silly... but what is the useragent of your sip configuration? In the case that useragent has some special characters like (., please remove it and tell us if there is any change!!. Regards. rv 2014-11-22 14:50 GMT-03:00 Eric Wieling ewiel...@nyigc.com: Try setting directmedia=no in sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Saturday, November 22, 2014 8:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Are both your servers behind the same NAT router? thanks for taking part... I don´t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- 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] SIP call drops after 32 seconds, but only when....
You might check your phones as well. We had this problem early on with a softphone and it was a setting in the phone that was set to hang up after 30 seconds of inactivity in case of network disruption. For some reason it was detecting network disruption in every call even when the calls were proceeding normally. Unchecking this box solved the problem. It may not be related to your problem but if it is the cause, you will spend a lot of time trying to fix this in Asterisk. :-D At least I did! On the bright side, it does force people to get point in a hurry! Ron On 22/11/2014 12:50 PM, Eric Wieling wrote: Try setting directmedia=no in sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Saturday, November 22, 2014 8:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Are both your servers behind the same NAT router? thanks for taking part... I don´t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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
