Re: [asterisk-users] Crash asterisk res_odbc

February 29 2016 5:40 AM, "Maxime" wrote: > Hi, > > Thank you for the reply. > > My OS is : Debian 7. > > But i have more than 20 servers with the same features/resources (OS, > material, ... ) without the > issue. > Maybe could debug the crash getting a Backtrace

Re: [asterisk-users] How to recive Incoming calls in Chan Dongle ?

Humm yes, thanks very much ! Em 04/03/2016 18:00, "Ashish Gupta" escreveu: > Hi Vitor, > > The dongle.conf file contains your configuration setting related to your > particular dongle. There, set the "context=dongle" (or anything you > specified in extensions.conf), then

Re: [asterisk-users] Asterisk 13.5 and higher (asterisk 13.7.2) quitting

Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North Ninth Street Lafayette, IN 47905 ry...@oscarwinski.com (765) 742-1102 We're not the IT departmentWe're the I-TEAM department! > -Original Message- > From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] How to recive Incoming calls in Chan Dongle ?

Hi Vitor, The dongle.conf file contains your configuration setting related to your particular dongle. There, set the "context=dongle" (or anything you specified in extensions.conf), then provide the "exten=1234" (The extensions that will be called in the particular context). Also provide the imei

Re: [asterisk-users] PJSIP signaling question

I can’t quite figure it out , I went ahead and pulled everything yet again, and I made sure to delete everything related to pjproject from my system, all the PJ lib and include files that were in /usr/lib/ , I pulled pjproject from svn , pulled asterisk code from gerrit, recompiled

[asterisk-users] How to recive Incoming calls in Chan Dongle ?

Hi! How can I setup my Chan Dongle recived calls in my Asterisk? I have to setup in dongle.conf ? Or in extensions.conf? And the code for recive I found this site http://asterisk-service.com/page/chan-dongle-use I have to To save Subscriber Number before? See the error log in my Asterisk

[asterisk-users] Asterisk 13 Realtime MusicOnHold

I am having a problem trying to use the realtime database for musiconhold for Asterisk 13. Everything is setup and I can see the mapping: ===> musiconhold (db=general, table=musiconhold) Only what is in the musiconhold.conf file appears in Asterisk and the database is completely

Re: [asterisk-users] How to control host part of From: field content from the dialplan [SOLVED]

2016-03-04 18:59 GMT+01:00 Richard Mudgett : > > > On Fri, Mar 4, 2016 at 11:45 AM, Olivier wrote: > >> Hello, >> >> I've read SIP Connect 2.0 draft lately. >> >> It mentions specific use if either of the following values is present in >> the From: field

Re: [asterisk-users] How to control host part of From: field content from the dialplan

On Fri, Mar 4, 2016 at 11:45 AM, Olivier wrote: > Hello, > > I've read SIP Connect 2.0 draft lately. > > It mentions specific use if either of the following values is present in > the From: field of an INVITE message. > The values are: > sip:unavailable@unkown.invalid >

[asterisk-users] How to control host part of From: field content from the dialplan

Hello, I've read SIP Connect 2.0 draft lately. It mentions specific use if either of the following values is present in the From: field of an INVITE message. The values are: sip:unavailable@unkown.invalid sip:anonymous@anonymous.invalid I'm using Asterisk 13 and PJSIP. Which dialplan function

Re: [asterisk-users] Asterisk 13.5 and higher (asterisk 13.7.2) quitting

Hi Travis, On 04-03-16 15:23, Ryan, Travis wrote: I start asterisk 13.7.2 and it dies before I can rasterisk into it. I’ve tried getting a coredump, but it doesn’t coredump. I know there are a lot of errors in the log below, but most of those just say it’ll not load a module, and no big deal.

Re: [asterisk-users] PJSIP signaling question

On Fri, Mar 4, 2016 at 1:16 AM, Kevin Long wrote: > Hi George the patch was from here , you wrote it I believe . I pulled > asterisk 13 from git, apply this patch which fixed RTP issue , but I think > tla transport issue came back for me . > >

[asterisk-users] Asterisk 13.5 and higher (asterisk 13.7.2) quitting

I start asterisk 13.7.2 and it dies before I can rasterisk into it. I've tried getting a coredump, but it doesn't coredump. I know there are a lot of errors in the log below, but most of those just say it'll not load a module, and no big deal. When launching from commandline (not service

Re: [asterisk-users] Rasterisk freeze on 4G link [SOLVED]

Thank you all for pointing me in the right direction. Now I learned I have to care about MTU. Best regards 2016-03-03 21:27 GMT+01:00 Toufic Khreish (Gmail) : > Hello, > > > > > > You need to determine the correct MTU value by doing the following: > > > > ping

Re: [asterisk-users] PJSIP signaling question

Hi George the patch was from here , you wrote it I believe . I pulled asterisk 13 from git, apply this patch which fixed RTP issue , but I think tla transport issue came back for me . https://gerrit.asterisk.org/#/c/2346/ Thank you Sent from my iPhone > On Mar 4, 2016, at 12:01 AM, George

Re: [asterisk-users] PJSIP signaling question

On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long wrote: > > Thanks George I appreciate the info . Being able to see what codec is in > use for call in progress is very handy sometimes. > > As far as the RTP stats goes, I see there is some info with “rtp” and > “rtcp”