search for: tenorio

...inal peer description, I'm only able to see it during an iax debug Timestamp: 00003ms SCall: 00001 DCall: 00000 [66.98.146.34:5036] VERSION : 2 CALLED NUMBER : 911214686 CALLING NUMBER : asterisk CALLING NAME : asterisk LANGUAGE : en USERNAME : tenorio FORMAT : 2 CAPABILITY : 18 ADSICPE : 2 DATE TIME : 173807980 Tkx.

...## Apdo. Postal 453 ## Ensenada, Baja California ## Mexico. ## atrujo at uabc.mx ## And the special collaboration of the post-graduate students of the 2002:2 ## Multivariate Statistics Course: Karel Castro-Morales, ## Alejandro Espinoza-Tenorio, Andrea Guia-Ramirez, Raquel Muniz-Salazar, ## Jose Luis Sanchez-Osorio and Roberto Carmona-Pina. ## November 2002. ## ## To cite this file, this would be an appropriate format: ## Trujillo-Ortiz, A., R. Hernandez-Walls, K. Castro-Morales, ## A. Espinoza-Tenorio, A. Guia-Ramirez an...

Hello there. Is there somebody with this connection working? I can't seem to make this work at all. Could someone please share some .conf files? Cheers, Vedran. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/71480bd9/attachment.htm

I've successfully got inbound/outbound calling working with our Asterisk using the Asterisk-OH323 channel driver. We are using a parent gatekeeper and the NuFone H323 channel driver would not work with the parent gatekeeper... I'm trying to determine a way to ensure that the line used for outbound calling is always available i.e. like trunking.. >From what I can tell when I place an

Does anyone know how to get any of these VOIP phones to allow me to do menu selections through asterisk, like when accessing voicemail and such. Thanks :P --

I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have

I've looked around briefly for what options are available for encrypting the media stream using asterisk. I did not see any SRTP support, and it looks like there is some initial work on iax2 encryption, but whether it works is still open for question I guess. I'm also curious of other solutions that could be bolted onto the front end of asterisk to provide encryption, and are there

Hi, I have a serious problem to configure Cisco AS5XXX and Asterisk , I trying to use asterisk for PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B) (No Nat, no Firewall) I hear (on the PSTN(A)) clearly what the other person is saying, but the other person (on the PSTN(B) side) hears nothing from PSTN(A). I use tcpdump for debug de rtp trafic, and ouput contains

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

I'm looking for Spain Voice Numbers, anyone know a trustable company that provides them? TKX, Leandro

The configuration and commands 're the same for Enterprise, Plus or Voice -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of kurt x Sent: Thursday, October 07, 2004 12:23 PM To: Asterisk Subject: [Asterisk-Users] RE: Cisco and PRI Enterprise version is way to big. You just need IOS IP Plus or IOS IP Voice.

In the next couple of weeks we will be starting the beta phase of our Guatemala POP. If you could wait, welcome. LTenorio -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Phil Astin Sent: Sunday, January 16, 2005 6:23 PM To: asterisk-biz@lists.digium.com; asterisk-users@lists.digium.com Subject: [Asterisk-biz] Guatemala DID's? I...

Anyone has an example of how a working record for agress and rates tables should look? I'been trying all the thinkable patterns, obviously not the right ones, for the last two days. Tkx, LTenorio

Try sending 5350 config and oh323.conf, versions, etc... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Goran Dj. Sent: Wednesday, December 22, 2004 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party