similar to: Incoming PSTN Calls - Stumped

Just another bit of info which might help solve this: Looking at the Asterisk log messages - I notice when I start up Asterisk, I see the error: pbx_config.c: Can't use 'next' priority on the first entry! Could I be right that its something got to do with priorities? I changed the incomingpstn context to the following eliminating the 'n' field and still the same errors were

Hi all, I am having difficulty getting incoming PSTN calls working. I have set up an account with a third party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and

Hi, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don't work. - Strange Anyhow I was getting an error: Process_sdp: No compatible codecs! And from the SIP

Guess I got where your confusiion lies.... you DON'T set up the ALL contexts the users will have acces to in sip.conf, in this file you will only set 1 single context. All other contexts you want the users to have acces to, you will have to add them in the extensions.conf So what you have to do is the following: -user 2092, set it the createmenu context in sip .conf - in extensions.conf

Hi, I am getting the following error when I attempt to listen to voice messages by dialing 9999 (I can hear nothing): --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. I read in previous posts that this may be to do with the dtmf

Following on from my below email, things have taken another bizarre twist.. I have continued getting the error when 2092 tries to listen to messages by dialing 9999. --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. Then I

Hello, Hope somebody can help me - Asterisk is behaving very oddly and I'm totally stumped! I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a

I get the following errors when starting Asterisk. == Parsing '/etc/asterisk/zapata.conf': Found Feb 19 21:14:35 WARNING[10440]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device Feb 19 21:14:35 ERROR[10440]: chan_zap.c:6860 mkintf: Unable to open channel 1: No such device here = 0, tmp->channel = 1, channel = 1 Feb 19 21:14:35

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

Dears; The problem was related to something else. The Digium card has two PRI ports, actually to get it UP, I have to configure the two ports and both of those two ports to take the timing from span 1. Why this, I do not know ! Although I am using only one E1 connected to span 1, so why I have to configure the other span !! After configuring the second span, so now one D channel for span 1 is

Hello, i'm still trying to find out what is the right path for ctdb.conf (ubuntu 18.04, samba was compiled from source!!). When im trying to start CTDB without any config file, my log in /usr/local/samba/var/log/log.ctdb shows me: 2018/05/07 12:56:44.363513 ctdbd[4503]: Failed to read nodes file "/usr/local/samba/etc/ctdb/nodes" 2018/05/07 12:56:44.363546 ctdbd[4503]: Failed to

Hi , I am using the plot function for some data , and the plot is coming back pure black , with scales on the side . Regards Malcolm [[alternative HTML version deleted]]

Hello, at this time i want to install a CTDB Cluster with SAMBA 4.7.7 from SOURCE! I compiled samba as follow: |./configure| |--with-cluster-support ||--with-shared-modules=idmap_rid,idmap_tdb2,idmap_ad| The whole SAMBA enviroment is located in /usr/local/samba/. CTDB is located in /usr/local/samba/etc/ctdb. I guess right that the correct path of ctdbd.conf (node file, public address file

Hi, I'm wondering if anyone has experienced an issue with the XLite softphone and asterisk accepting dtmf? I can listen to my voicemail perfectly from my hardphone. However when I dial the voicemail number from my XLite softphone and enter the password at the voicemail prompt, an error appears vm-incorrect and I get an "Unable to read password" message on the asterisk console. Has

Hi All; My ISDN was working fine, and suddenly I start getting the below: sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway! There is a Yellow Alarm, so what it could be the problem? My configuration as following: system.conf span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 chan_dahi.conf context=IncomingPSTN group=0 signalling=pri_cpe switchtype=euroisdn

Hi All; Suddenly, we restarted the Asterisk machine and the echo appeared. The lines are analoge. At the consol, I see this message: [Aug 10 14:36:05] WARNING[3789]: chan_dahdi.c:4831 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) [Aug 10 14:36:07] WARNING[3789]: chan_dahdi.c:4831 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW:

Hi All, Does anyone know what the current status is of the G729 codec on Solaris? According to the following link: http://www.asteriskvoipnews.com/asterisk_releases/_digium_g729_codec_now_available_forsolarissparc.html there is a version available for SPARC processor's. However, I have just had a quick look around Digium's FTP server and cannot seem to find these codecs (supported or

I wrote a nice detailed post before, and then my mail program lost it for me... so here I go again... I've followed the same process with three different versions of asterisk, my local source copy from about 1 week ago CVS, current CVS from about 24 hours ago, and version 1.0.1, all three versions had identical results: I compiled/installed libpri, zaptel, asterisk I copied config file from