similar to: Asterisk on VMWare 4.5, Error Ouch ... error while writing audio data

Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Jan 28

After installing the stable version of * and the Zaptel drivers with a TDM400 card using 1 FXO module on port 4, I start Asterisk and get this rolling up my screen thousands of times: Ouch... Error while writing audio data Ouch... Error while writing audio data Ouch... Error while writing audio data Ouch... Error while writing audio data Ouch... Error while writing audio data Ouch... Error while

I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the new box, I've installed a generic ebay X100P. I don't have my livevoip or voicepulse accounts set up yet on the new box (can both boxes be registered at the same time?). I've set up one IP phone (SPA841) with the new box. I have my SBC POTS line plugged into the fxo card. I set up everything in AMP.

I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything seems to be running fine but after some time asterisk just goes crazy (even withouth any incoming or outgoing call activity perviously). If I leave the box up for some time * goes haywire and the console is flooded with this message: Ouch ... error while writing audio data: : Broken pipe At that time I can see that there

Hi, I have the following situation: At a branch , there is a Cisco Call Manager with users all having Cisco phones. Now I put down a Asterisk 1.2.12 box at the branch, which talks H323 via chan_oh323 to the CCM. So calls go from the CCM, go H323 to the local Asterisk box, then I take it via SIP to another Asterisk box. From there I am hooked up to 2 different providers, for Local and

Hello, the bench that gives me a problem is the following: asterisk1 + spandsp ===>===PSTN===>=== asterisk2 + spandsp 1) i put a .call file inside Asterisk spool directory 2) the fax is sent successfully, I got this message in the sender's log: .... [Oct 4 17:57:02] DEBUG[17610] app_txfax.c: FLOW FAX Set tx type 8 [Oct 4 17:57:03] DEBUG[17610] app_txfax.c: FLOW FAX Set rx type 0

I have to re-compile also app_rxfax.so and app_txfax.so or just spandsp ? Thanks Giordano ________________________________ Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Colin Anderson Inviato: gioved? 12 gennaio 2006 17.20 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: RE: [Asterisk-Users]

Hi, I just installed spandsp 0.0.3pre6 with libtiff 3.7.1 and evrythinghs is ok. I recompiled asterisk and run ldocnfig, but whrn I start asterisk I get this error: [app_rxfax.so]Jan 12 16:40:42 WARNING[1569]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Jan 12 16:40:42 WARNING[1569]: loader.c:440 load_modules: Loading

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

Hello everybody, I?ve been trying to solve a problem for several weeks now but it really beats me. There are several hard phones connected to an Innovaphone 3000 VoIP gateway. On the other side I have a SIP softphone connected to Asterisk. The problem I have is that on incoming calls (hardphones to softphone) I only have outgoing audio (from soft to hardphone); everything is OK when I call the

Anyone run app_txfax and app_rxfax successfully on asterisk 1.2 beta1? I get an error when trying to run asterisk: [app_txfax.so]Oct 5 12:05:24 WARNING[14665]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/app_txfax.so: undefined symbol: fax_set_header_info Oct 5 12:05:24 WARNING[14665]: loader.c:543 load_modules: Loading module app_txfax.so failed! Ouch ... error while writing

Hi everybody! I've managed to compile the bristuff patch on asterisk from Junghanns.net. I want to run this on the quadBRI card built into the PC. As I said, driver and asterisk are compiling well, but "asterisk -vvvvc" bombs with this message: [pbx_dundi.so]Nov 24 15:57:53 WARNING[1076754432]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/pbx_dundi.so: undefined

Hi!I've a question: I've 2 asterisk, I want pull the ethernet wire and then reconnect it after 5 second, using the VRRP protocol, where must I set the IP for the connection goes on the second asterisk? I want this: I call to asterisk1, then I pull the ethernet wire down, vrrp makes up the other asterisk but not the audio streaming...the callers are always pointed to asterisk1, but for the

Hi, My configuration is SipPhone<----->asterisk1 <----->asterisk2. My asterisk version is 1.2.10. I installed chan_h323 according to 'http://astrecipes.net/?n=102'. When i call from asterisk1 to asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Regards, Jason. #------h323.conf for both------------------------ [general]

Hello, I'm trying to setup the revised Polycom remote reboot script as found on: http://www.voip-info.org/tiki-index.php?page=Polycom+reboot+hardphone+script I'm not sure how to use this script, it's just a perl script, so I tried creating an executable perl script and running it, but I get the following: [root@asterisk1 agi-bin]# ./polycom_reboot.pl 192.168.3.205 Checking ARP

Dear Asterisk developers, I run the same asterisk version on the home machine and on the work. On the home machine I have Slackware 10.0 (kernel 2.4.24) while on the work machine I have Mandrake 10.1 (kernel 2.6.8.1 <http://2.6.8.1>). When I run asterisk on the work machine, these warnings and error appear (there are no warnings or error at home): [ Booting......Oct 17 18:19:04

Ouch, part reset, quickly restoring reality (0) Power alarm on module 1, resetting! I have looked though a lot of email on this issue, and no one seems to have the answer. How many people are seeing this on a TDM400 card? Does anyone have a "REAL" answer to it. Yes, I do have the power connected to it! Regards Garry Taylor

Amir, I had a similar problem with A@H while troubleshooting a PRI circuit. I accidentally put this in my Zapata.conf file: signalling=cpe_pri The correct line is: signalling=pri_cpe The "broken pipe" error occurs quite frequently when there are bad configuration entries in Zapata.conf. Definitely double-check each line in Zapata.conf and make sure there aren't any subtle

Hi I'm trying to use ouch's hansen and brown functions but I get the error: > brown(logflatnodes,archotreeouch) Error in backsolve(l, x, k = k, upper.tri = upper.tri, transpose = transpose) : NA/NaN/Inf in foreign function call (arg 1) and with hansen also: Error in optim(par = c(sqrt.alpha, sigma), fn = function(par) { : function cannot be evaluated at initial parameters

Hi, I recently configured Linux HA for Asterisk service (using Asterisk resource agent downloaded from link: https://github.com/ClusterLabs/resource-agents/blob/master/heartbeat/asterisk ). As per configuration it is working good but when I include "monitor_sipuri=" sip:42 at 10.3.152.103" " parameter in primitive section it is giving me an errors like listed below; root at