Some users have complained that their calls drop after about 30 seconds. Not all, just some. After looking at the log files the only difference I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-000000..
Author : Carlos Chavez
Home » Articles posted by Carlos Chavez (paged 2)
I am having a strange problem with a new provider. We already have a couple SIP trunks working fine. We are trying a new provider but we are having one way audio problems with outgoing calls. Incoming calls do have two way audio, only outgo..
I am trying to troubleshoot two Asterisk servers that have an IAX2 trunk between them. Calls come and go but there is no CallerID from the remote server either way. One of the servers is running Asterisk 16 and the other is an older 1.8 inst..
What is the best way to debug DTMF on a PJSIP trunk? I have cycled through all available options (rfc4733,inband,info,auto,auto_info) but my IVR does not recognize any options from the remote end. I have also tried changing codecs from g..
Since yesterday I have a stuck channel on my Asterisk server and I do not know how to eliminate it: Message/ast_msg_queu macro-dial-one s 59 Up Dial PJSIP/1218/sip:1218@19..
I know we should not be running an Asterisk so old but this customer really does not want to replace this particular installation. I am having a problem when calling Gosub from a macro. It seems that if I call gosub and return to the ma..
I am trying to send messages to asterisk-r2@lists.digium.com but I do not get an error or any messages back. In the archive I do not see any messages past November 2018. — Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez ..
Hi. I am having a problem when trying to receive calls via en E1 from Telmex using MFC/R2 (MX Variant). Outgoing calls are fine. We are using a PBXact system with a Digium TE420 (5th Gen) card. Here is a log from the call: [10:46:37:7..
It seems that app_swift does not work with Asterisk 15 or 16. I just get errors when trying to compile: [root@pbxoficina app_swift]# ./configure checking gcc… checking swift… checking asterisk… creating Makefile *****************************************************..
I just finished installing a brand new server with CentOS 7.5 and Asterisk 13.22.0 and the minute I a call from the PSTN (from a SIP trunk) bridges with any SIP phone Asterisk crashes: Jul 20 10:59:53 localhost kernel: asterisk[2819]: segfa..