similar to: IAX client USB phone

Hi all, I've just being wondering if Dundi has the same purpose as ENUM. I don't know much (actually almost nothing) about these technologies. As far as I know they are a kind of DNS resolver used in the VoIP context. For example, user ronaldo at doamin.com has the extension namber 1001. This way nobody has to know the ronaldo's extension number. I'll appreciate if someone

Hi , > > My Xen has been working fine for a few days. Then today it suddenly > can not run. > > When it runs It give me this error : - Error starting domain: virDomainCreate() failed POST operation failed: (xend.err "Error creating domain: Boot loader didn''t return any data!") Details: - Traceback (most recent call last): File

Hi all, I just want to know how I can make sure that incoming calls to my asterisk server are being treated by [incoming] section of extension.conf file. Thanks in advance. Ronaldo.

Hi all, What does (T) mean on the output of "iax2 show peers"? The following my output. darkstar*CLI> iax2 show peers Name/Username Host Mask Port Status ronaldo (Unspecified) (D) 255.255.255.255 0 UNKNOWN sp/ata 201.26.67.102 (S) 255.255.255.255 4569 (T) UNKNOWN 2 iax2 peers [0

Hi all, I just started programming using AGI and I have a simple question about STDERR. If I understood it right, all the messages sent to "STDERR" should be shown in the Asterisk console, but using the following python code I just can't see anything. #!/usr/bin/python # # File: /var/lig/asterisk/agi-bin/agi-test.py # # Description: An AGI Script # import sys env =

Hi all, I have a IAX trunk between two asterisk servers, both with dynamic IP and both have a DNS name associated with it. In the iax.conf file I configure the "host" parameter with the DNS name of the servers. Everything works fine until one of these servers get a new IP, so the other can't find its peer (the one that has just gotten a new IP). If I manually issue a "iax2

Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. Ronaldo.

A friend of mine just sent me this email - he is looking for an IAX inbound service in Holland - any thoughts? Voip info only has Nadiz which seems to be SIP only. Hi Dean, I need a Dutch number with IAX support. Do you have any leads in that direction? It's been difficult for me to figure it out -- especially since most of their sites seem to be in Dutch... Regards,

Hi, I''m trying to configure QoS on my linux in the following manner: I have a main link with 64K, so I divided it in 3 classes of 18K, 14K and 9K with an excess (not used for classified traffic, just to be shared) of 23K. This excess should be distributed proportonally among the 3 classes, that is, the class that has more rate should borrow more bandwidth. What is happening is

Has anyone here implemented "Ring back when free" in Asterisk? The way it works in the UK is as follows: 1. A calls B. B is engaged (busy). 2. A hears "The number you called is busy. To use ringback, press 5" 3. A presses 5, and hears "Your ringback request has been accepted". 4. A hangs up. 5. Later, B hangs up. The system then calls A (if A is now busy, it

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998

I've added lines like this: speeddial = 123,test speeddial = 260,Bob in the [device] section for my 7921, but the speed dials do NOT appear on the menu (click right from the main screen). Am I missing something obvious here? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.

I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian

I would like to hear stories from people using TAPI, CTI or CallerID software with Asterisk. What are you guys using, setup examples, etc. Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do it. Are you running callerid software? Did you stumble into problems like using tapi and callerid software returned both the callerid and calledid? Hope you can help me out with

Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. Afer compiling mISDN into the kernel and building chan_misdn Asterisk stops loading with : [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables

Hi, Do you if a DECT-GAP (or DECT-CAP) compliant handset MUST or MAY support roaming and handover and are these functions transparent for handset (then, these functions are implemented in DECT base stations) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080110/4254f602/attachment.htm

I would like to have asterisk pass along the caller ID phone number to a database server on a my local network (the same network that the * server resides on ) so that our customer service app. can pull up customer data automatially. Asterisk passes along caller ID to the phones fine, can someone tell me how to make it pass this info to my database server? Any suggestions would be greatly

Greetings folks. I seem to be having a problem where calls made from an IAX device (three single-line phones attached to IAXys) do not play the ring tone when calling out. There's a dial tone when I pick up the phone, and the call goes through just fine, it just doesn't ring. All my SIP phones ring normally, however. Is there an option I need to enable that I'm missing?