similar to: Changing the sip port in sip.conf does not work

Hi I?m a new user and I do test with my hardware . I have a x100p and telephone vozip. And when I run this command asterisk ?vvvvc for to test it . My computer show it ?warning? [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory) Apr 2 07:45:12 ERROR[16384]:

Running 11.13.1 on Fedora. This is a new install, but a copy of a previous - working -install. module load chan_sip Unable to load module chan_sip Command 'module load chan_sip' failed. SIP channel loading... [Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to load config sip.conf I don't think it's permissions: ls -ld /etc/asterisk /etc/asterisk/sip*

Hi, I am new to asterisk , i am getting the following error,& the /etc/zaptel.conf file entry is as follows defaultzone=us loadzone=us span=1,1,0,esf,b8zs,yellow bchan=1-23 dchan=24 Parsing '/etc/asterisk/zapata.conf': Found Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664 zt_open: Unable to specify channel 1: No such device or address Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296

Good day all What is the file sip_notify.conf for Thanks Altus

This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. rom root@localhost.localdomain Wed Jun 15 18:42:00 2005 Date: Wed, 15 Jun 2005 18:42:00 -0400 From: root@localhost.localdomain (Cron Daemon)

If I fire up asterisk it connects to my MSSQL server via ODBC just fine. However, if I issue a reload it unloads the ODBC.. then loads it again and I get an error... and keep getting it until I On a fresh start: Aug 26 15:43:14 WARNING[13818] cdr_odbc.c: cdr_odbc: table not specified. Assuming cdr Aug 26 15:43:14 VERBOSE[13818] logger.c: -- cdr_odbc: dsn is MSSQL-asterisk Aug 26 15:43:14

Greetings. I'm having a bit of a problem using the sipdtmfmode app. I have two incoming paths to * from pstn via FWD that use differing dtmfmode. IPKall wants rfc2833, libretel wants inband. If I set dtmfmode= in the fwd peer config in sip.conf each works seperately, and I'm trying to use gotoif and sipdtmfmode to switch based on the CID calling. Output seems to indicate sipdtmfmode

I want to share some facts with the Asterisk community. I have been very successful providing a Vonage-type system based on Asterisk. For instance, one company that uses Asterisk and offers a similar service to Vonage is Voyze.com. The key concept is that Asterisk works like a Cisco, for all the intelligence is provided by SQL Server, outside Linux. I don't even save the CDR locally. The

All, Here's a cool one.. I was attempting to call a retarded conferencing service, and was having problems with it picking up my DTMF.. after trying all the settings my Sipura SPA2000 offers, I found inband actually works.. unfortunately, I can't get anything else to pick up my inband DTMF (including asterisk's builtin voicemail! It just times out and says I never entered a login!).

I've configured asterisk to connect a MySQL database for CDR, Voicemail and SIP/IAX2 peers. - CDR are reccorded - Voicemail config is readen directly in the database but SIP/IAX2 peers still have to be declared in sip/iax2.conf to make calls... However, when I restart Asterisk: [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found ==

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protect the server from unauthorized users or is there

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure because anyone can bypass the SER and register themselves as a peer with the asterisk. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of

Hi, If I have [myprovider] section with context=something. When I do an outgoing call by using Dial(SIP/myprovider/464646)", does context=... affect anything? As I understand it, it only affects incoming calls, but I might be wrong. Another thing, the setting of context=... on [default] section will affect all [provider] sections without context=..., right? What if I don't specify any

Hello everybody, I am having a problem with realtime SIP peers. On Asterisk 1.8, I had SIP peers for external SIP providers configured in database and additional register lines in sip.conf so they would register. Now I upgraded to Asterisk 11.3.0, partly because of the promised callbackextension feature for realtime peers (https://reviewboard.asterisk.org/r/1717/). Removed the 'register'

I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1 (see config below) and with a bit of messing about using sample config, have been able to make the test call to device 1000, and also through to the IAX test number at Digium. However, operation is extremely flaky - I cannot reliably startup and use the system on a regular basis. I have several problems listed below

All, I run a bunch of (well 20+ actually) Asterisk boxes at home, work, friends and the lie with our own dialplan in the form 8EEXXXX where 'EE' is the exchange number and 'XXXX' is the extension number. This arrangement has been in for 2+ years and worked well with a central box (asterisk.thorcom.net) acting as the routing hub and SIP exchange point with various public

I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session

Through the optware-package I have installed Asterisk on an external USB. Further I have a Linksys WRT610N with DD-WRT v24 mega. I start asterisk with the following command : /opt/sbin/asterisk -c I get the following WARNING : root at DD-WRT:/opt/etc/asterisk# /opt/sbin/asterisk -c Asterisk 1.4.22.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at

Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. ---------------------------------- [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for

Can't find a way to control the dtmf mode on a per session basis with pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any hints on how to do this?