similar to: how to interconnection asterisk(sip) with mera

Hi, We are terminating around 60 channels on one of our Asterisk boxes, which the client sends in H323 mode. Client (MERA) --> H323 --> Asterisk --> IAX --> Asterisk The problem we face is that at random intervals the H323 process (as part of Asterisk) dies and can no longer accept new calls whilst Asterisk is still running happily. We have to then kill asterisk and start it

I'm having trouble to send calls to a Mera MVTS softswitch (with SIPHIT) when the asterisk box has a dynamic IP address. If the Asterisk box has a fixed IP, everything is OK. Any ideas? I'm looking for a working sample of the sip.conf in this case... user.cfg (for MVTS) is also appreciated if any special setting should be done there also.

Hi there. I have the similar problem ("Digium fax - sending fax call file vs manager originate") sending faxes with Asterisk 1.6.2.6 and Digium res_fax. Receiving is OK. I use Local channel in Call file and context [fax-out] in dialplan. My setup: Asterix<-SIP (T.38)-> Cisco(MERA MSIP v.1.0.2)<-> LocalTelco<->fax machine Debian GNU/Linux 5.0 ; Linux 2.6.26-2-686

Hi, i'm trying to install in wine-1.0.1 on Ubuntu 9.04 a management software but I receive run-time 713 error. Any solution or tip? Here the log.txt: fixme:ole:OleLoadPictureEx (0x12c8c44,35146,1,{7bf80980-bf32-101a-8bbb-00aa00300cab},x=0,y=0,f=0,0x32fac0), partially implemented. fixme:ole:OleLoadPictureEx (0x12c8c44,774,1,{7bf80980-bf32-101a-8bbb-00aa00300cab},x=0,y=0,f=0,0x32fa90),

I' got a problem, using asterisk-rc2 :IVR functions (Background...Playback...etc) doesn't works : Executing Background("OH323/RXXXXX", "vm-extension") in new stack channel.c:1650 ast_set_write_fornat: Unable to find path from GSM to G729A---Asterisk box supplied only with network adapter.---Asterisk box registered in Mera (soft-switch with H323 protocol) and doing

Hello. I'm try to set up asterisk for making outgoing calls with oh323 channel driver version 0.7.1 with Asterisk CVS-1-01/09/05-01:41:37. Our provider uses Mera MVTS softswitch and supports only H.323. We don't use gatekeeper for connection but provider requires SOURCE PHONE NUMBER for route out calls and I don't know how I can specify this number. Call with this string exten

Hi, List My SIP provider requires no authorization in incoming calls to my asterisk 11.1.0 box. I was sure previously that "insecure=invite,port" disabled authorization request during incoming calls to asterisk. But today I tried to connect to a provider (which has MERA MVTS) but could not disable auth requests in incoming calls from this provider with this option

Estimado Carlos Gil Bellosta ¿Cómo está usted? En estos lados de América del sur comienza la primavera, desde la ventana miro la parra contando las posibles uvas, siempre aparece un ave que se arrima a la ventana o incluso llegan hasta la computadora como si supiesen usarla. Ahora en R. En ese esquema un modelo lineal tendría que ir con mlib que es aportada por sparklyr, en ese caso tendría

Estimado Carlos Ortega Comprendo que hay que tener el paquete compilado para acceder al alto rendimiento, por lo cuál si está todo preparado para trabajar en un clúster y para aprovechar múltiples hilos, no habría problemas, calculo que si una librería no tiene esa tecnología no traería inconvenientes, ¿o por el contrario si está distribuido crea varias instancias y al correr separadas hay

Estimados En el sitio de https://www.rstudio.com/ hay un aviso sobre http://spark.rstudio.com/index.html ( sparklyr ). Microsoft publico un artículo donde comparan el R Server que está dentro de SQL server (o por separado, depende un poco), o el Microsoft R, junto con algunas librerías que se pueden compilar y obtener lo mismo en Ubuntu. Supongamos que tengo el dinero como para comprar por

Hi, Has anyone on this list connected with Verizon's SIP product? We are currently undergoing interop testing with Verizon, and honestly, it seems like the most convoluted process. I'd be interested in talking with someone else who has gone through this and run a few things past you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

This is from nanog; I presume there is significant interest from readers here not also on nanog.... I've edited it to only the interesting part... -JimC -------------- next part -------------- An embedded message was scrubbed... From: Daniel Golding <dgolding@burtongroup.com> Subject: Re: New VOIP Peering/Interconnection Mailing List Announcement Date: Fri, 14 May 2004 15:09:00 -0400

Hello, My colleague installed a Asterisk home as company's SIP server and I would like to integrate the Quintum gateway (SIP) but the calls don't get through. Bellow is are the configurations on each side: Quintum ******** Primary Registrar = 202.69.190.244:5060 Primary Registrar User Name= sipquintum Primary Registrar Pwd= sipquintum Primary Proxy =

Has anyone had any luck getting a SIP trunk up and working between Callmanager and Asterisk? If so were there any steps you had to take that were not in the documentation on wiki? Blake

Good morning I am new with Astersik and I want to know how can I configure my LDK to communicate with an Asterisk server via SIP. I don't know how to procede and which configuration files should I be interested in. If anyone could help me, I would be extremely grateful thanks a lot in advance. _________________________________________________________________ MSN Hotmail : cr?ez votre

My fellows, We have Asterisk@home installed and we want to interconnect it with our existing quintum gateways.. any idea how to config that? Your time is very much appreciated.. Cheers, Jessie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050220/2518797c/attachment.htm

Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a "Failed to

hello, i have trouble with nat + sip outgoing call.when make an outgoing call to a sip gateway, i have no sound. i have 2 sip gateway, one is asterisk. asterisk is on public ip and private ip other sip gateway is on public ip phone are cisco and grandstream on private ip on the same subnet as asterisk. phone are connected by sip to asterisk (i have try with or without nat=yes) incoming call

Hola, LLevo buscando desde hace tiempo como hacer el Goodness of fit test en R. Es decir, me explico, intento hacer una cosa parecida que se hace en Minitab, por ejemplo, yo tengo un conjunto de datos, y lo que quiero es sabes que tipo de distibución es, en minitab se hace un histograma para ver si se ajusta bien o no a la campana de Gauss, luego vemos si aproximar la distribución de la muestra

Hello, I'm trying to implement quota enforcement in our mailservers, and it is all working properly except that the quota warnings are not firing when the quota levels are passed.? the server stops accepting email when the quota is reached, and you can see tyhe quota usage through the email client connected through, but as the quota passes the set levels the /usr/local/bin/quota-warning.sh