Hi there i have some problems with some clients of my asterisk box, i have some cases when a client tried to make a call and there is no ring back only a silence and then the call hung up. I dont know why this is happening. I have the following stable asterisk version: CVS-v1-0-01/18/05-19:49:31 I did the an update a few days ago, the version that i had installed before was: CVS-HEAD-10/08/04-12:44:50, I had some troubles with that version related with IAX trunk=yes so i had to make the update. Any help would be useful. Thanks a lot for your help. Carlos Andres Medina
I am running on 1.2.7.1 and have an intermittent problem when making outgoing calls. Sometimes the calling party does not hear the ring tone in their handset, but the call goes through. From my extension I have only had 3 calls like this in the last couple of weeks, other people have had 3 or 4 calls in a single day and then not have a problem for a couple of days. The called phone number is not the problem because sometimes it works and sometimes not. We have both Aastra and Cisco phone sets and the problem occurs on both of them. We have SIP to PRI connections. I believe that this problem started after we upgraded from 1.0.9 but not 100 percent sure of that. Any help or suggestions that you have would be appreciated. Thank you, Tim
Here is a copy of indications.conf I have not included other country codes. I noticed that there are no spaces in "country=us" does that matter? Thanks [general] country=us [us] description = United States / North America ringcadance = 2000,4000 dial = 350+440 busy = 480+620/500,0/500 ring = 440+480/2000,0/4000 congestion = 480+620/250,0/250 callwaiting = 440/300,0/10000 dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440 record = 1400/500,0/15000 info = !950/330,!1400/330,!1800/330,0 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Eric "ManxPower" Wieling Sent: Wednesday, June 14, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No ring tone on outgoing calls Make sure you have /etc/asterisk/indications.conf set up. People that don't know any better might tell you to use the "r" option to Dial. Those people are confused. Don't do that until you have tried everything else. Tim Sharp wrote:> I am running on 1.2.7.1 and have an intermittent problem when making outgoing calls. Sometimes the calling party does not hear the ring tone in their handset, but the call goes through. From my extension I have only had 3 calls like this in the last couple of weeks, other people have had 3 or 4 calls in a single day and then not have a problem for a couple of days. The called phone number is not the problem because sometimes it works and sometimes not. We have both Aastra and Cisco phone sets and the problem occurs on both of them. We have SIP to PRI connections. I believe that this problem started after we upgraded from 1.0.9 but not 100 percent sure of that. Any help or suggestions that you have would be appreciated. Thank you, Tim-- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Arik Raffael Funke
2006-Jun-14 11:43 UTC
[Asterisk-Users] Re: No ring tone on outgoing calls
Tim Sharp wrote:> I am running on 1.2.7.1 and have an intermittent problem when making outgoing calls. Sometimes the calling party does not hear the ring tone in their handset, but the call goes through. From my extension I have only had 3 calls like this in the last couple of weeks, other people have had 3 or 4 calls in a single day and then not have a problem for a couple of days. The called phone number is not the problem because sometimes it works and sometimes not. We have both Aastra and Cisco phone sets and the problem occurs on both of them. We have SIP to PRI connections. I believe that this problem started after we upgraded from 1.0.9 but not 100 percent sure of that. Any help or suggestions that you have would be appreciated. Thank you, TimI have a similar problem with following setup: Idefix on external network -> IAX in -> Asterisk -> SIP out to VoIP-Stunt -> Landline Also, the first few seconds of the conversation are missing. I.e. the other party answers the phone, but I never hear it. Audio goes through however after a bit. With following config I have no problem what-so-ever: ISDN phone -> ZapHFC -> Asterisk -> Sip out to VoIP-Stunt -> Landline Thus the problem lies somehow with the IAX in. Has anybody seen this before? Cheers, Arik