Robert La Ferla
2006-Oct-18 22:30 UTC
[asterisk-users] Asterisk hangs up on incoming analog calls after a while
I have been experiencing a problem where after someone calls me from an analog line, the phone call is terminated after a period of time (anywhere from 15 seconds to 15 minutes) The phone that I use to answer the call is an Aastra 9133i SIP phone. There are several other SIP extensions on the network as well as a few analog extensions on a shared FXS line. When a call comes in the analog line on the FXO, * dials all the extensions (SIP and analog.) I have a Digium card with 1 FXO and 1 FXS. How can I diagnose this problem? Has anyone experienced anything similar?
Eric "ManxPower" Wieling
2006-Oct-19 07:33 UTC
[asterisk-users] Asterisk hangs up on incoming analog calls after a while
Robert La Ferla wrote:> I have been experiencing a problem where after someone calls me from an > analog line, the phone call is terminated after a period of time > (anywhere from 15 seconds to 15 minutes) The phone that I use to answer > the call is an Aastra 9133i SIP phone. There are several other SIP > extensions on the network as well as a few analog extensions on a shared > FXS line. When a call comes in the analog line on the FXO, * dials all > the extensions (SIP and analog.) I have a Digium card with 1 FXO and 1 > FXS.Do you have callprogress=yes or busydetect=yes in your /etc/asterisk/zapata.conf ?
Robert La Ferla
2006-Oct-20 12:40 UTC
[asterisk-users] Asterisk hangs up on incoming analog calls after a while
On Oct 19, 2006, at 3:00 PM, asterisk-users-request@lists.digium.com wrote:> Date: Thu, 19 Oct 2006 09:30:38 -0500 > From: "Eric \"ManxPower\" Wieling" <eric@fnords.org> > Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog > calls after a while > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <45378C0E.7070708@fnords.org> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > Do you have callprogress=yes or busydetect=yes in your > /etc/asterisk/zapata.conf ?No. They are not set. i.e. default -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061020/2110bc1a/attachment.htm
Steve Murphy
2006-Oct-21 08:49 UTC
[asterisk-users] Re: Asterisk hangs up on incoming analog calls after a while
On Fri, 2006-10-20 at 22:38 -0700, Robert La Ferla <robertlaferla@comcast.net> wrote:> On Oct 19, 2006, at 3:00 PM, asterisk-users-request@lists.digium.com > wrote: > > Date: Thu, 19 Oct 2006 09:30:38 -0500 > > > > From: "Eric \"ManxPower\" Wieling" <eric@fnords.org> > > > > Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog > > > > calls after a while > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > <asterisk-users@lists.digium.com> > > > > Message-ID: <45378C0E.7070708@fnords.org> > > > > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > > > > > > Do you have callprogress=yes or busydetect=yes in your > > > > /etc/asterisk/zapata.conf ? > > > > > > No. They are not set. i.e. defaultLet me guess: the incoming caller gets connected to the calling party via a Dial(Zap/1&Sip/what,...) type thing, and the Zap line answers and gets the call? and the Sip/what phone isn't even there? You merrily talk away and bang! you get disconnected not very long into your conversation? You should publish your console/log messages in those moments before and at the time of the hangup. I'll bet that some Sip phone in the Dial list has some event right when you hang up. See if you can narrow down your alternate dialing list to just the Zap and the Sip that are involved. Sounds like you may have a few different Sip phones involved. When you get it, then file a bug report. murf -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3239 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20061021/3b14a931/smime.bin