Re: [asterisk-users] MixMonitor fdiles
Hello, I am running Asterisk 1.4.20-1 and having the exact same problem. It looks like others are having issues as well according to this thread: http://www.trixbox.org/forums/trixbox-forums/help/recordings-out-sync-using-mixmonitor Anyone have any idea's? On Wed, Apr 9, 2008 at 7:16 AM, robert boardman [EMAIL PROTECTED] wrote: Hi, I have a load of files recorded with MixMonitor that are out of sync ie one leg of the call is 2-3 seconds behind the other, is this a bug in Asterisk 1.4.18, or am I possibly doing something wrong Is it possible to edit the file and re sync the a/b leg? Thanks for your help Robb ___ -- 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 -- Isaac McDonald Got VoIP? [EMAIL PROTECTED] Cell: +1 253-223-8673 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE205P
Hello, I am working on a project and have a few questions. I want to connect one port of the TE205p to the PSTN, and another port to the PRI port of a PBX. Basically Asterisk will sit between the PSTN and the existing PBX. Are there any gotcha's I need to be aware of? Will the existing PBX be able to dial through asterisk as if it was a PSTN connection? -- Isaac McDonald Got VoIP? [EMAIL PROTECTED] Cell: +1 253-223-8673 ___ -- 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] Festival patch
Anyone know where I can get the patch described here: http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html I am getting compile errors when trying to compile speechtools... Any help would be greatly appreciated, Isaac ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adit 600
Has anyone had any success using the Adit 600 with the CMG card talking MGCP to asterisk? I want to have a central asterisk server with 10 Adit 600's at various locations providing 24 FXS ports Thanks, Isaac ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware Transcoder
Does anyone know of a hardware transcoder? Or a software transcoder for that matter. I would consider using asterisk but it seems that Asterisk per the WIKI can only support at most 100 channels transcoding from g.711 to g.729. I would be transcoding from g.711 to g.723.1 or g.729. Thanks, IAM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Prepaid
As far as a low credit warning is concerned.look at bug id: 1353 on http://bugs.digium.com I have been using it just fine. usedcanon wrote: I have a requirement for a setup with prepaid call credits. I am aware of the two applications available (been researching for the past week), app_prepaid and app_rateengine. However neither of the two sound like exactly what I want. However I was wondering that someone who has used it might be able to say if they could be used in my scenario. Basically my scenario is pretty straight forward. Credit will be allocated to the ddi, I dont need any announcements etc (maybe low credit warning during call could be useful thoug). From the users prespective everything will be transparent. However the call should disconnect when the credit runs out. The CDR and the account DB need to be adjusted according to the call made. My guess is that app_prepaid could used with modification, I am assuming here that this is not possible as-is with configuration. Basically in case of the prepaid app, the card number can be replace transparently with the callerID. All help, guidence and comments will be extremelly appreciated. Umar. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: Interconnecting to an Altigen PBX?
Ian McLaughlin wrote: Has anyone got Asterisk talking successfully to an Altigen PBX using h323? I can successfully make calls from Asterisk to Altigen, but calls from Altigen to Asterisk get a fast busy. There seems to be a lack of h323 example files (or maybe I'm looking in the wrong places) as well as a severe lack of h323 documentation from Altigen. Any pointers would be greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have successfully got my Altigen talking h.323 to my asterisk server. There is an option when setting up h.323 connections in Altigen that essentally tells the Altigen to dial after so many digits have been entered.I had to disable that option on the Altigen side of things as it would not let me dial a full 11 digit number, it would send the first 4 digits or so. I am able to make calls from the * server to internal extensions on the Altigen just fiine, but when I try and grab an outside line it fails.have you been able to grab one of your Altigens outside lines via Asterisk? Isaac ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse Connect Problems
Robert Jackson wrote: Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls, through VoicePulse of course. It was working properly yesterday, and without changing anything it stopped working. Thanks in advance, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have two incoming DID's from Voicepulse. The first one I set up over 3 months ago and has been working fine, including incoming DTMF. The second one I ordered last week in the Birmingham, AL market, its a new rate center, DTMF does not work on this DID. I e-mailed support on this issue and have yet to get a response. Is your DID in a of of their new markets by chance? Isaac ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse Connect Problems
It works now! I did nothing on my end either. VP must monitor this list. Isaac Robert Jackson wrote: Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls, through VoicePulse of course. It was working properly yesterday, and without changing anything it stopped working. Thanks in advance, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free PSTN calls
I have set up my * box to provide free calling. You can access it by dialing 1-700-945-2475. Once you hear the prompt dial 1 the area code and number. I would also like to test some direct incoming IAX connections from some other * boxes to see if I can terminate PSTN calls that way. If you would like to help me testemail me: [EMAIL PROTECTED] Also, more information is available at http://www.freephoneproject.com/nexthop Isaac [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicepulse and IAX
That did the trick! Thank you Christopher! Isaac -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Christopher Stephens Sent: Wednesday, October 29, 2003 11:39 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicepulse and IAX The instructions they sent you (and me) are slightly faulty. in iax.conf: context=VPWS ;I'm not sure what VPWS stands for. would better be context=from-voicepulse ;(for example...this is what I use) then in extensions.conf: (this is the simplest example) ;snip [from-voicepulse] exten=3017275115,1,Dial(SIP/1234) ;obviously replacing 3017275115 with your voicepulse # (if it isn't already) ; and SIP/1234 with your registered SIP client or Zap channel or whatever. ; then this phone will ring when someone calls your number. You could, of course, leave the context as VPWS in iax.conf, just make sure there is a corresponding context in your extensions.conf. An incoming call that no context/extension covers is the source of the NOTICE you mentioned. Hope this helps! Chris - Original message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Wed, 29 Oct 2003 17:07:42 + Subject: [Asterisk-Users] Voicepulse and IAX I am trying to set up IAX with Voicepulse. When I turn on debugging I get the following message when I call my PSTN number: NOTICE[1142106560]: File chan_iax2.c, Line 4321 (socket_read): Rejected connect attempt from 66.234.228.132, request '[EMAIL PROTECTED]' does not exist Any help would be GREATLY appreciated. Thanks, Isaac [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
