Re: [asterisk-users] Setting outgoing CALLERID without changing CDR(src)
On Wed, Mar 28, 2018 at 08:16:26PM -0600, Carlos Chavez wrote: > ?? I thought I had found and answer to this question by using > CALLERID(ani) but it seems that only works on versions prior to 12.?? On > Asterisk 13 setting CALLERID(num) before dialing to an external trunk always > changes CDR(src) to the number you set and the original extension number > that dialed is lost.?? How are you handling this??? Am I forced to use a > custom field to keep the original caller number??? My billing software uses > the src field to get the extension that dialed the call.?? Any tips? What does your external trunk use as sources for callerID? For example if your provider supports PAI and sets this headers as callerID, your problem is solved. Otherwise you could for example use the custom csv output and set a custom CDR field (that stores the original src) in the place of src. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip trunk config question + DNS related error messages
On Thu, Mar 29, 2018, at 4:38 AM, Kevin Long wrote: > Greetings, > > > I am getting the following error (below) continually in my asterisk log, > related to qualify_frequency I believe. I am trying to use sip trunking > with the company flowroute. > > 3 questions if I may: > > 1) Is using qualify_frequency with a sip trunk a common or recommended > practice? I figured it would function as a keep-alive and keep the > ‘pjsip show endpoints’ availability data up-to-date if I wanted to check > on the health of the trunk. Sound right? > > 2) Any idea what this error means? Googling showed almost nothing except > one other post to this list for a bug that was fixed in 14 , I’m on 15.x > > 3) Any other recommendations for this trunking config? > > > Thanks very much ! Especially jcolp and gtjoseph for answering my > queries in the past, sorry if I don’t always respond again as I haven’t > actually figured out a good way to do that unless I am subscribed to > receiving all mails from the list. > > > > > [Mar 28 23:17:43] ERROR[4812]: res_pjsip.c:3770 endpt_send_request: > Error 320047 'No answer record in the DNS resp > onse (PJLIB_UTIL_EDNSNOANSWERREC)' sending OPTIONS request to endpoint > flowroute > > > [flowroute] > type=auth > auth_type=userpass > password=** > username=** > [flowroute] > type=aor > contact=sip:sip.flowroute.com:5060 Is there any reason you aren't just using sip:flowroute.com here? PJSIP does SRV resolution so that'll use SRV instead which I know works. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Local channel Earlymedia
Hello, I noticed that Earlymedia isn't possible as soon as the Media Stream has to pass through Asterisk. In my Scenario I need Local Channels and Earlymedia at the same time. So the Media Stream needs to flow through Asterisk. Is there a Workaround to get Earlymedia with Local Channels working? Thanks Benjamin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pjsip trunk config question + DNS related error messages
Greetings, I am getting the following error (below) continually in my asterisk log, related to qualify_frequency I believe. I am trying to use sip trunking with the company flowroute. 3 questions if I may: 1) Is using qualify_frequency with a sip trunk a common or recommended practice? I figured it would function as a keep-alive and keep the ‘pjsip show endpoints’ availability data up-to-date if I wanted to check on the health of the trunk. Sound right? 2) Any idea what this error means? Googling showed almost nothing except one other post to this list for a bug that was fixed in 14 , I’m on 15.x 3) Any other recommendations for this trunking config? Thanks very much ! Especially jcolp and gtjoseph for answering my queries in the past, sorry if I don’t always respond again as I haven’t actually figured out a good way to do that unless I am subscribed to receiving all mails from the list. [Mar 28 23:17:43] ERROR[4812]: res_pjsip.c:3770 endpt_send_request: Error 320047 'No answer record in the DNS resp onse (PJLIB_UTIL_EDNSNOANSWERREC)' sending OPTIONS request to endpoint flowroute [flowroute] type=auth auth_type=userpass password=** username=** [flowroute] type=aor contact=sip:sip.flowroute.com:5060 qualify_frequency = 15 [flowroute] type=endpoint transport=transport-udp context=from-flowroute disallow=all allow=ulaw outbound_auth=flowroute aors=flowroute [flowroute] type=identify endpoint=flowroute match=216.115.69.144 match=70.167.153.130 1. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
