[asterisk-users] incoming calls fall into echo test mode
Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has this problem: his outgoing calls are ok, but when someone dials his phone number (be it from our network or from any other place in the world), the caller ears the standard message signalling he has entered the echo test mode and must dial # to exit that mode. Most callers don't understand what's going on, then give up and hang up without dialling #. Very few dial # one or more times, then those few get our customer's phone ringing and are then able to reach our customer. I went through all the docs, wikis and discussions I found on the web, without finding any data on how to solve that problem. I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with another phone number would solve the problem (I tried it and it works), but this customer wants to keep his current phone number and when I tried deleting his extension then creating a new one with his current phone number, the new extension presented the same problem as the previous one... Anyone knows what could cause such a problem and/or how to solve it ? Thanks, Norman. ad...@csur.ca -- _ -- 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] incoming calls fall into echo test mode
You might get a better response on the FreePBX forum. (FreePBX adds pre-built dialplan elements onto standard asterisk. This forum is more for Asterisk) But some suggestions: SSH to your PBX enter the Asterisk CLI set verbose to 10 Call into the problematic number ...and watch where the call is being misrouted in the dialplan From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Norman Molhant ad...@csur.ca Sent: Saturday, July 19, 2014 10:43 AM To: Asterisk Users List Subject: [asterisk-users] incoming calls fall into echo test mode Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has this problem: his outgoing calls are ok, but when someone dials his phone number (be it from our network or from any other place in the world), the caller ears the standard message signalling he has entered the echo test mode and must dial # to exit that mode. Most callers don't understand what's going on, then give up and hang up without dialling #. Very few dial # one or more times, then those few get our customer's phone ringing and are then able to reach our customer. I went through all the docs, wikis and discussions I found on the web, without finding any data on how to solve that problem. I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with another phone number would solve the problem (I tried it and it works), but this customer wants to keep his current phone number and when I tried deleting his extension then creating a new one with his current phone number, the new extension presented the same problem as the previous one... Anyone knows what could cause such a problem and/or how to solve it ? Thanks, Norman. ad...@csur.ca -- _ -- 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] incoming calls fall into echo test mode
Perhaps assigned as a test number somewhere along the line? Are these ISDN, SIP, IAX calls? There are MANY smart people on this list. Maybe sharing the relevant configs and traces is a good place to start??? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant Sent: Saturday, July 19, 2014 10:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming calls fall into echo test mode Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has this problem: his outgoing calls are ok, but when someone dials his phone number (be it from our network or from any other place in the world), the caller ears the standard message signalling he has entered the echo test mode and must dial # to exit that mode. Most callers don't understand what's going on, then give up and hang up without dialling #. Very few dial # one or more times, then those few get our customer's phone ringing and are then able to reach our customer. I went through all the docs, wikis and discussions I found on the web, without finding any data on how to solve that problem. I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with another phone number would solve the problem (I tried it and it works), but this customer wants to keep his current phone number and when I tried deleting his extension then creating a new one with his current phone number, the new extension presented the same problem as the previous one... Anyone knows what could cause such a problem and/or how to solve it ? Thanks, Norman. ad...@csur.ca -- _ -- 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] incoming calls fall into echo test mode
check your logs /var/log/asterisk/full -- make sure your verbosity is set high enough to do you good and you wll probably find the answer. Pat Collins drdialt...@optonline.net wrote: Perhaps assigned as a test number somewhere along the line? Are these ISDN, SIP, IAX calls? There are MANY smart people on this list. Maybe sharing the relevant configs and traces is a good place to start??? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant Sent: Saturday, July 19, 2014 10:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming calls fall into echo test mode Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has this problem: his outgoing calls are ok, but when someone dials his phone number (be it from our network or from any other place in the world), the caller ears the standard message signalling he has entered the echo test mode and must dial # to exit that mode. Most callers don't understand what's going on, then give up and hang up without dialling #. Very few dial # one or more times, then those few get our customer's phone ringing and are then able to reach our customer. I went through all the docs, wikis and discussions I found on the web, without finding any data on how to solve that problem. I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with another phone number would solve the problem (I tried it and it works), but this customer wants to keep his current phone number and when I tried deleting his extension then creating a new one with his current phone number, the new extension presented the same problem as the previous one... Anyone knows what could cause such a problem and/or how to solve it ? Thanks, Norman. ad...@csur.ca -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] chan_motify / res_xmpp bind address?
Hi, I've been trying to talk xmpp with asterisk with ICE-UDP, but still does not work 2014-07-18 7:26 GMT-06:00 Daniel Pocock dan...@pocock.pro: I have a multi-homed machine (quite a few IP addresses on one of the interfaces) For SIP I found that using externaddr in sip.conf would make it much more reliable with ICE and RTP using the correct IP Is there an equivalent setting for XMPP / motif.conf? I saw bindaddr in gtalk.conf but it doesn't appear to be mentioned in the source code for chan_motif -- _ -- 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 -- rickygm http://gnuforever.homelinux.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
