similar to: Dynamic dialplan

I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem?

I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users

I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is

I'm currently writing some code to support conferencing in Asterisk without using the Meetme application. The conference runs in its own thread and every new inbound or outbound channel that is created is passed to it. This thread runs the conference loop reading and writing frames to each channel. I'm writing this as if it were a bridge with more than two channels, and I'm not

Is there anyone who is currently using Asterisk as a production H323 gateway? And using which combination of asterisk and H323 (chan_h323, chan_oh323?) The main issue is interoperability with other H323 parties (Cisco AS53xx, Nextone, etc). Searching the mailing list it seems that both h323 and oh323 are not so stable, is it only an impression or using h323 is really not so advisable?

Hello everybody, for all of you that have searched for a real fax solution, look no further. We now have T38 faxing. Please contact me for more information. Thanks Michael D. Schelin ShellTel 626-814-2354

Hi all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately. At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -------------- next part -------------- An HTML

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Hi, all. I am building a SER+asterisk PBX airming at around 10k persons' usage. For authentication purpose I am in favor of ldap storage, while I am not sure the current ldap module for asterisk(0.9.9.2) is stable enough? sorry I do not master the proper testing mechanisms to find out myself. Thanks in advance.

Is there any stable version of spandsp that works on a 64 bit Linux on an AMD64 machine. When compiling version 0.0.1k I get the following error: gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c testcpuid.c -MT testcpuid.lo -MD -MP -MF .deps/testcpuid.TPlo -fPIC -DPIC -o .libs/testcpuid.lo /tmp/ccXxGHg6.s: Assembler messages: /tmp/ccXxGHg6.s:8: Error: suffix or operands invalid for `pushf'

Hi, After reading this valuable forum and the voip-info wiki and follow all the steps , but my Cisco 12SP+ remains unregistered. These are my config files: skinny.conf [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 172.20.1.1 ; Address to bind to dateFormat = D-M-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 languaje=es allow = all ; disallow

Hi all, I was reading http://bugs.digium.com/view.php?id=2639 and it seems that anthm's great native MoH patch only works on HEAD. Does anyone have a version of the native MoH patch that works on 1.0.8? If so please point me to its location or email it off-list. Thanks and regards, Patrick

Hi Guys! I have a problems with some sipuras 841 and asterisk 1.0.9. Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with steve's unicall. Everything compiled fine and in fact I can make and receive calls but I have a problem with bad sound when the sipuras call the outside E1's lines. I can listen to the caller without problems but they heard me with a choppy

Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? Regards, Liangliang

I am a complete newbie, completed editing the extensions.conf file, having problem reloading my diaplan via asterisk console, tried to reload it with diaplan reload command, but it says command does not exist. Please help _________________________________________________________________ Hotmail: Powerful Free email with security by Microsoft.

Hello, When reloading a diaplan (asterisk 1.6.1.X), I can see in console : [Apr 2 09:02:00] WARNING[2217]: ael/pval.c:2522 check_pval_item: Warning: file /etc/asterisk/extensions.ael, line 621-621: application call to Gosub affects flow of control, and needs to be re-written using AEL if, while, goto, etc. keywords instead! What is then the recommended substitution for Gosub() application

Hi, For MGCP users. Is there any success stories with any MGCP IAD vendor. I?m trying to find an IAD which works with Asterisk. I?ve tried the Cisco IAD 2430 without success; but SIP on this IAD works but it?s limited (no authentication, no notify messages, etc) and with higher density IAD (16 or more ports) it?s nice to control using MGCP. Any information will be apreciated ! Thanks. --

Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"

Dear Sirs, I'm running an Asterisk 0.9.1 in a Fedora Core 2 box. I installed a X100P card on my box and when i try to load modules i am rejected. When i type modprobe zaptel my Fedora respond : "FATAL: Module zaptel not found." . The same uccurs when i type "modprobe wcfxo" May you help me. Thank you in advance Juanjo