similar to: add iax user

Hello, I'm trying to conect two 1.6.2.13 Asterisk server with IAX. This is my configuration: Asterisk A: iax.conf register => coiax:pass1 at 69.164.207.166 [smiax] type=friend host=dynamic trunk=yes secret=pass2 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.207.166/255.255.255.255 qualify=yes Console: iax2 registry 69.164.207.166:4569 N coiax 69.164.197.105:4569

Hello all! I am trying to load sip.conf from mysql database. I have followed the instructions at <http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers>. Seems that the authentication (user & psw) works fine but I would like to get more information from mysql and I don't know how to retrieve it. Could anybody help me? Any idea about how to do it? Regards, Victor.

Boy after really digging into this, I have discovered that there is more information about each of these topics than I previously realized. Strangely, searching the wiki on "iax" returns exactly nothing. But searching on iax2 does start to dig up some good stuff. Sorry for the hassle. Tough day. -brian -------- Original Message -------- Subject: Re: [Asterisk-Users] IAX config

Hi, I have a pentium 4 with Intel motherboard and one TDM400P (2 fxs, 2fxo) modprobe zaptel is Ok but When I execute modprobe wctdm never load the module, I can wait for 1 year but never response me (error or OK). I need to do ctrl+c Any idea? Edgardo >>> jan@irial.com 06/21/05 10:07 AM >>> i know that there are extensive rework on the transfer in SIP at the moment. --On

Hello, My question is.. How does cdr_manager work? Does it suppose to populate cdr-csv/Master.csv? What about the cdr table on the database? What is the event some people talk about? I have changed (and reloaded) my configuration of cdr_manager.conf to ; ; Asterisk Call Management CDR ; [general] enabled = yes and it doesn't seem to make any difference. After originate a call from the

Thank you for your advise, I'll find something with a lot of memory.... Adrien -- Adrien Laurent - CIO 514-284-2020 adrien@modulis.ca www.modulis.ca -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jason Walker Sent: Saturday, October 01, 2005 12:25 AM To: 'Asterisk Users Mailing List - Non-Commercial

It's pretty obvious from the wiki that realtime and Nat don't befriend quite well. As It is obvious the necesity of both of them, mainly have clients under nat talking to an asterisk server. The question I would like to throw away is.. What would you do to have both of them? I have two possible solutions in mind. 1. Use static configuration for sip users with nat=yes. 2. Buy iax

Hi, Monitor and soxmix (m option) work fine in CVS Head, not in Asterisk 1.0.9, as the Wiki says. http://www.voip-info.org/tiki-index.php?page=Monitor+setup+sample Anyway I am wondering why asterisk 1.0.9 console shows on Hang up: "monitor executing ( nice -n 19 soxmix "//var/spool/asterisk/monitor/45/47-20050812-113631-in.wav"

Hello, I'm trying to migrate my system to FreeBSD and the Makefile for asterisk-addons fails in the first make clean: bash-2.05b# make clean "Makefile", line 56: Missing dependency operator "Makefile", line 57: Could not find .depend "Makefile", line 58: Need an operator make: fatal errors encountered -- cannot continue I would like to think there is no

Hello, I can understand why asterisk is designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but I have to find a solution for this. My first option is use SER as an extension end of Asterisk, to allow more than one SIP endpoint to register with the same details http://www.voip-info.org/wiki-Asterisk+at+large. I

Hello: . I'm having troubles registering on nufone's IAX service . I'm really new to Asterisk Nufone provide me some config examples ... I can dialout but I can't register my * Box, eg. whe I do "iax show registry" I got only a "Request Sent" and later I have a "Timeout" My box is on a Public IP and no firewall. I'm using *0.5 on a

Hi, I'm trying to transfer an incoming call from the PSTN to another PSTN number through a SER - Asterisk system. SER doing only routing.. pstn call-> SER -> asterisk (call forward) -> SER -> pstn Logic for SER: If something comes from the pstn, send it to asterisk. If something comes from asterisk, send it to the pstn. Every time I am getting a "Got SIP response 481

Hello, with asterisk 1.6 i am trying to make a dialplan Which i have such entry in extensions.conf exten => _8XXX,1,Dial(SIP/${EXTEN}) But some of my clients have both IAX and SIP accounts, to use iax clients while outside of my Local Area, and SIP clients (or hardware phones) in local area. But with such rule, i can only dial SIP accounts. Is there a parameter to find how the user connected?

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

Hi All; Did anyone try to use IAX IP Phone behind NAT, and let it receive calls from Asterisk without doing port mapping at the router existed at the site where the IAX IP Phone existed? Is the need just to let the IAX IP Phone that is NATed to register on the Asterisk and at asterisk I set nat=yes for the IAX client configuration? Or it is impossible to let the NATed IAX to receive calls

Hi all, i have a problem with 2 peers conecting to an asterisk machine, both are conected behind nat without any port mapping in the router, and the * is conected behind other nat with the port 4569 mapped to it address, the problem is: when a peer register to the asterisk the other cant register and viceversa, only gets registration the first one, im using firefly and a hardphone from wuchuan,

Hello, I have asterisk setup with Broadvoice. It works great as PBX and Outgoing calling server for all local clients withing 192.168.1.0 network. My asterisk is running over NAT. I use linksys router. Now, I am trying to connect from outside to my asterisk server. I use Diax as iax client. For some reason I cannot connect to my server from outside. On my router I forward those ports to my

Is there any documentation on configuring IAX between * machines? I've noticed references to many topics in the config files, including: - dialplans - trunking - authentication - transfers But before I go and try to grok 8000 lines of source (in one file, no less) I was hoping that somewhere there exists even something like a man page that describes the configuration options.

Hi, I'm aware that incoming and outgoing calls are going fine when isdn channels are involved - caller id properly identifies calling party, so user can call back.... But how to properly handle this for iax, sip calls.... I have few questions : - BTW, what to type for instance in remote firefly to make standalone calls to Asterisk default context or particular extension ? - If I receive

Dear Sanjay, Sorry sanjay i miss to explain completely. My PC2PC mean is Dialer2Dialer i want to allow call between Dialer with out any registry and authentication through IAX. so i need to setup Asterisk accept calls from any user and users can call to each other without any password and registration. please help how can i configure Asterisk using IAX in this regards. thanks, Asif Message: 9