similar to: asterisk: call failed error 408 timeout

Full_Name: Jose M. Perez Version: 0.99.0a OS: Linux - 2.0.38 Submission from: (NULL) (159.90.200.68) Hi, I'm having problems with the check of infinite recursion in the expression evaluation depth. The problem appears in a program with no recursions whatsoever. The layout of my program is a series of routines called by a big loop. I printed out depthsave in src/main/eval.c and it appears

We're experiencing an issue where calls disconnect after 15 minutes. It seems to happen just after Asterisk sends an update mesage. RTP is being set up directly. Asterisk is only in the SIP dialog. Has anyone experienced this issue? 4 PRIs inbound, 4 PRIs outbound, asterisk provides switching. SIP/2.0 200 OK Via: SIP/2.0/UDP 38.XXX.XXX.XXX:5060;branch=z9hG4bK1c4b524f From:

Dear puppet masters, I''m having frequent timeout errors reported from my puppet clients in syslog: Jun 27 01:23:41 nc82 puppetd[9089]: Could not call fileserver.describe: #<RuntimeError: HTTP-Error: 408 Request Timeout > Jun 27 01:23:41 nc82 puppetd[9089]:

Hello: Here is my sip and extentions configuration and the log of x-lite, because i don`t can call inside my LAN with asterisk PBX 1.2 and i don`t have NAT. i hope you can help me. SIP.conf [default] include=>anexos include=>anexos1 include=>anexos2 [anexos] exten=> 100,1,Dial(SIP/100,0) exten=> 100,2,Hangup [anexos1] exten=> 101,1,Dial(SIP/101,0) exten=> 101,2,Hangup

On our tests using asterisk, some calls have been terminated abruptely with exact 185 seconds. This is happening with all our incoming calls from a trunk from 1 of my DID providers ( other providers or trunks are fine) and I could reproduce it by calling a queue from my Wengophone Softphone and letting the MoH play for 185 secs. If I make the same call from my WRTP54G on the same place,

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

Hi there I am using an IAX2 softphone built from the IaxClient library dialing into Meetme conferences. The IaxClient seems to use silence suppression, and not sure if this can be disabled. The client works fine through most routers, but for some it disconnects the client after about 5 minutes and gives this error in the asterisk logs: Chan_iax2.c:1480 attempt_transmit: Max retries exceeded to

Is there a softphone that works with Asterisk?

Hello, Please forgive me if this has been addressed previously. I have been searching the archives and have not come across what I thought was a solution. My * server is behind a DSL router using a NAT IP address of 10.0.0.9. A colleague running XP and X-Lite can register with * from his home, specifying my public IP as the SIP proxy in X-Lite (however this is only true if I have the NAT flag

Tengo un archivo de texto donde cada línea es de la siguiente forma: "aa-mm-dd hh:mm:ss Nombre Apellido" ¿Hay alguna forma de usar read.table o algo similar para obtener directamente un data frame que tenga dos columnas donde una tenga la fecha y quede de tipo PosiX y la otra character con el nombre completo? Gracias, Sebastián.

hi guys, i am getting today my dev kit with fxo and fxs boards. i intend to do the following : 1) be able to route an incoming call from the pstn fxo port to an ip (answering with netmeeting or anyother sip softphone) 2) be able to call from netmeeting to my pstn fxo port to place calls. questions : how can i do this ? what are the commands for this simple setup ? how can i place calls

Has anyone had any luck getting the Avaya IP SoftPhone to work in wine? I get these errors on Ubuntu 8 Code: preloader: Warning: failed to reserve range 00000000-60000000 wine: creating configuration directory '/home/microchp/.wine'... preloader: Warning: failed to reserve range 00000000-60000000 preloader: Warning: failed to reserve range 00000000-60000000 err:dosmem:setup_dos_mem

Hello, ------------- -------- --- -------- |Sip Softphone|-------|Internet|--------|F.W|-----|Asterisk| ------------- -------- --- -------- IP addresses: a.b.c.d q.w.e.r The SIP softphone(x-lite) is configured to register with the asterisk server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as the

I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to softphone 2, I presume that Asterisk should do transcoding (canreinvite is set to no):

Hi, I'm having an issue with all my calls going out my SIP provider. I'm using a softphone registering to a local Asterisk PBX (I'm using Jitsi by the way - it's great and actively growing). I register as extension 4053 to asterisk server at 10.215.147.115 (alias IP - real IP addr. is 10.215.147.111) and dial a phone number that is routed via an Internet SIP provider. The call

Hi list, I am faced by a situation where I am trying to make a softphone and a Siemens C450IP talk to each other. Both are hooked up directly to the same asterisk, in the same IP net. - a softphone runs on 192.168.14.3 - the C450IP is at 192.168.14.30 - asterisk runs on the machine known as 192.168.14.1 I am running Asterisk 1.4.11, backported to Debian Etch by Xorcom. If I set

All, I have a simple question and a complicated reason for asking: Is it possible to change asterisk's source port for outbound IAX2 connections? I've tried using "sourceaddress" to no avail. I can set it to: proper.ip.of.box:4569 or 0.0.0.0:4569 and it works as expected. But if I try to set it to: proper.ip.of.box:5000 or 0.0.0.0:5000 it fails around line 8536 in

I have a software-only PBX set up. I can register various softphones and they will call each other -- but I've never succeeded in getting any voice routed from any of the softphones. Only the console will transmit audio. I am writing to ask if I have missed some obvious step in configuring the system. Conditions: (1) Softphones running on the same machine as the PBX: Only Kphone seems

Hi, Asterisk 1.2.5 serves as a pstn gateway to voip users (xlite clients) by means of a X100P (a small test system) The pstn side has clear sound but i can hardly hear the other person (voip side)talking. The voip side (xlite xten softphone) has crackling sounds and also a very low voice volume (i can hardly hear the pstn person). Xlite connects using GSM to the asterisk machine, it's on the

Hello, First sorry for my poor english. slack 9.1 dovecot-0.99.11 (thanks for this) compiled with openssl (0.9.7b) MUA : Mozilla 1.5 (win32) Under a ssl session, at first, it's work. But a later ago, i recieved alway, when i tried to contact the server imap : "connecting to <host> timed out ..." This fact is never without ssl. No dns problems. I don't know how can i