similar to: SIP default gateway

Hello. I'm new in the list and sorry for my poor english :) I have this two entrys in the sip.conf file, one for incoming calls (vtele_in) an the other for the outgoing calls (vtele_out) -- piece of sip.conf --- ; entry for incoming calls [vtele_in] type=user context=sip-in host=voztele.com disallow=all allow=g729 allow=ulaw allow=alaw ;Entry for outgoing calls [vtele_out] type=peer

I have seen a number of people in this newsgroup asking for information regarding asterisk interworking with Vocal. I was able to configure Vocal and Asterisk so that calls originating from vocal can land on an extension in Asterisk. I would like to share this info with the group The scenario that I tested was as follows. A call was originated from extn. 1001 on Vocal and the call was made to

Gracias a todos, finalmente instalando primero el libxml2-dev y luego el dplR, funciono ----- Mensaje original ----- De: r-help-es-request en r-project.org Para: r-help-es en r-project.org Enviados: Miércoles, 17 de Julio 2013 12:59:47 Asunto: Resumen de R-help-es, Vol 53, Envío 26 Envíe los mensajes para la lista R-help-es a r-help-es en r-project.org Para subscribirse o anular su

Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general

I was trying to do a little searching to see if there has even been a comparison between Asterisk and VOCAL or any of the other OSS packages? "Practical Voice Over IP using VOCAL" published by O'Reilly and Associates, attempts to make a strong case about how scalable VOCAL. Of course, considering that the book is written by the makers of VOCAL, it tends to have a one sided slant.

I want to start a community based voip network projcet and am thinkimg of using VOCAL and asterisk gateways..... my question is, has anyone bench marked asterisk vs VOCAL? is it a wise idea to use VOCAL + Asterisk or Asterisk all the way.........am expecting 1000 -> 5000 users.. your thoughts would be appreciated. _________________________________________________________________ Don't

I'm trying to get Asterisk to talk SIP to Vocal and so far have only managed to get it partially working. Calls in from Vocal are working fine but outbound calls aren't. In sip.conf I have: [ivv] secret=SECRET username=08452416761 host=sip.intervivo.net fromuser=08452416761 externip=mt104.dyndns.org nat=yes canreinvite=no reinvite=no notransfer=yes In extensions.conf I

Dear colleagues, Is there any way of saving an animated plot with R? For instance, any format that could be read by Microsoft windows media or whatever. Cheers, Jesus

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

El 17 de septiembre de 2020 9:21:55 GMT-04:00, Rowland penny via samba <samba at lists.samba.org> escribi?: >On 17/09/2020 14:11, Rommel Rodriguez Toirac via samba wrote: >> *************************************** >> Hello Rowland; >> thanks for write me back. >> >> >> ?Now verifying that everything was fine: >> >> $ getent passwd

Hi guys. My conf: # 1.2.9: /etc/dovecot/dovecot.conf # OS: Linux 2.6.32-23-server x86_64 Ubuntu 10.04.1 LTS reiserfs log_timestamp: %Y-%m-%d %H:%M:%S protocols: imap imaps pop3 pop3s ssl_cert_file: /etc/ssl/mail/dovecot.crt ssl_key_file: /etc/ssl/mail/dovecot.key disable_plaintext_auth: no login_dir: /var/run/dovecot/login login_executable(default): /usr/lib/dovecot/imap-login

An example for Radius is calling cards.. I can use * for this kind of service... With platforms that use Radius Server. -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 07:16 p.m. Para: asterisk-users@lists.digium.com Asunto:

The Wine maintenance release 1.8.7 is now available. This is the final release in the 1.8.x series. What's new in this release (see below for details): - Various bug fixes. - A few more cards added to the GPU description table. - Turkish translation updates. The source is available from the following locations: http://dl.winehq.org/wine/source/1.8/wine-1.8.7.tar.bz2

*************************************** Hello Rowland; thanks for write me back. Is that first I had to join the PC to the domain :-O Now it does work: $ sudo net ads join -U Administrator [sudo] password for administrator: ? Enter Administrator's password: Using short domain name -- ATGTM00 Joined 'P107' to dns domain 'gtm.onat.gob.cu' $ sudo systemctl start winbind.service

Hi guys: I can't login as a user: The output of mail.log: Aug 31 08:39:45 socrates dovecot: auth(default): client in: AUTH^I1^IPLAIN^Iservice=imap^Ilip=178.63.82.247^Irip=81.39.27.227^Ilport=143^Irport=33715^Iresp=AHBhYmxvLnJvZHJpZ3VlekBlc2N1ZWxhc3VwZXJpb3JkZWNvYWNoaW5nLmNvbQBwZXJpa2l0YS4= Aug 31 08:39:45 socrates dovecot: auth-worker(default): sql(pablo.rodriguez at

Hi all, i'm tring ro use sip with an external sip proxy as vocal or ser. My scenario is Vocal or SER ----> Asterisk with cnah_oh323 -----> Gatekeeper I would like that sip termial register themself to Vocal or ser and the h.323 terminal to gatekeeper. When i place a call from h323 side to sip side all work.... When a try to place a call form sip to h323 nothing happen Does

Hi, I am using * to function as the voice mail system for Vocal. Since I do not have a context in sip.conf file for each vocal client, I can't set the mailbox=xxxx in sip.conf. How do I get the MWI to a Vocal client ? Cheers Sathya

I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the

Buenas, estaba siguiendo el hilo de Informes Periódicos en R de Jesus Herranz y me surgió una duda. Actualmente estoy tratando de usar markdown para todo así cualquier cosa que haga me queda presentable para informar o presentar en algún lugar. En general estoy usando html y o pdf principalmente porque cuando trato de compilar en word la verdad que queda bastante feíto. Alguién sabe, o puede

Hola Jorge Perdona que sea tan pesado, pero no encuentro el fichero docx. Tampoco me manejo muy bien con RStudio. He insertado lo que me has dicho; pero solo tengo en la pantalla la opción “Knit HTML” y siempre me genera un HTML. He instalado “pander” pero no obtengo el docx Gracias Jesús De: Jorge I Velez [mailto:jorgeivanvelez en gmail.com] Enviado el: martes, 07 de abril de 2015