similar to: Asterisk won't listen on another port

Hello, 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 different port. It is my understanding that I just need to set the port in sip.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

Hi, Yes InternalExtension is the context and 2093 the extension. Just to explain something odd that?s happening (and I?m very stumped with this) .I think my contexts are definately the reason that I can?t interrupt the menu for incoming pstn calls to choose a submenu: My users register with my sip proxy (SER). Therefore when I create an entry for them in sip.conf I set only one context. Also to

Hello, I think I might have an inkling as to where the issue may be at. For some reason when I create a new context, a directory is not created in /var/spool/asterisk/voicemail. The default context resides there but new ones are not created. Has anyone ever experienced this or does anyone have any idea as to how I would solve this? Hope someone can shed light on this, Many thanks, Aisling.

I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like

Hi Friends, I want to use 5062 port for SIP protocol. I made the below modifications in my server to use 5062 port. Polycom phone: port=5062 Trunk settings: port=5062 sip.conf: bindaddr=5062 Extension configuration details: 5062 Our VoIP provider told me that they are allowing the SIP traffic through 5060 to 5064. I observed on my server console that my server is registered with our VoIP

Hello, I am doing some tests with asterisk on a dual-stack environment. I have some doubts regarding asterisk binding addresses on a server with 2 network cards. According to asterisk documentation: /; With the current situation, you can do one of four things:/ /; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1/ /; b) Listen on a specific IPv6 address.

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 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

Hi Folks, I am newbie to asterisk. Recentely I have installed asterisk on Linux Fedora 2 box. After reading some document, I tried to configure the server. When I connect to our server, SIP user-agent shows that I am logged in. But it doesn't show my system(client) IP when I issue command at astrisk CLI. The O/P is as below. *CLI> sip show peers Name/username Host

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

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:

hello, All AreskiCC users: I faced some problems in using AreskiCC. one is when I reload the asterisk server, the system display some errors such as execution 30 .. second one is there is no data display for admin added before. Does anyone know how to solve the problems, Please tell me! thanks in advance!

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'm trying to connect to a SIP carrier who never connected with Asterisk. I managed to connect with a sipura phone or a grandstream, no problem. When I configure asterisk, I'm able to send out calls to the carrier no problems, however, receiving calls doesn't work, and I keep getting the following messages: <-- SIP read from 69.xx.xx.xx:5060: INVITE

Good day all How do I get my asterisk and sip to use the password.I'm using x-lite.If I use just the username and no password it still logs on? Here is my sip.conf entry? [101] type=friend callerid="Test User" <101> context = test_1 ; Default context for incoming calls username=101 secret=123456 host=dynamic dtmfmode=inband ; Choices are inband, rfc2833, or info

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, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer after a designated amount of time, the request is forwarded to asterisk, and I can leave a message. Now, this may seem a ridiculous question but how can I listen to my message afterwards? I have read about a solution by Java

Hi All, I've been banging my head on a small dundi problem... I have two * servers setup, both have almost identical dundi.conf files: root@tsjonge:/opt/asterisk/etc# cat dundi.conf [general] department=thuis organization=pipsworld locality=Amsterdam stateprov=NH country=NL email=remco@pipsworld.nl phone=+31207508308 ;bindaddr=0.0.0.0 ;port=4520 entity=00:02:b3:49:69:5e ttl=16