search for: espinoza

I can't seem to get this device to grab an ip from dhcp. We have a working dhcp server (unfortunately it is on Windows), but I don't show any leases requested by the iaxy. Anyone have any ideas? The ethernet and phone lines are plugged in before the device is powered. Thanks, Erik

I have my DTA310 getting an IP via DHCP. I loaded the unlocked firmware version (Application Code Version: DTA version 1.0 US (8x8 001111)) onto it via TFTP, so I could access the SIP configuration. Under the SIP config, I put the IP of my * system, the 5060 port, and for Domain Name, I put default (is that right?). I checked off the Send Registration Request box. Dial Plan I left at the default,

...fornia ## 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-Ra...

Hi, My Asterisk CVS is apparently not doing much (other than keeping SIP & IAX2 registrations alive and doing some ZAP calls (without echo-cancellation), but slowly the memory is filling up, so much so that 100m virtual memory is used up within 12 hours and I have to restart the asterisk application every 48 hours to make sure I have enough memory... How can I help resolve this problem?

Hi everyone, I just got a Sipura SPA-1001 and have connected my Tivo Series 1 (yes its old). When I do a test call with Tivo, the call always fails (it seems to dial the number but never connects). I can pick up the phone line and hear the Tivo "talking". I've tried looking around for anything special I need to do but its still not working. I can connect a phone to the SPA-1001

We just had a security application vendor come in. We asked about Linux support and he said that putting a security application on top of an insecure OS was useless. When I asked what he meant by insecure he replied that Linux does not have a true Auditing capability - as opposed to HP-UX & Solaris which they do support. Can anyone explain to me what he was talking about? Thanks, Marty

Hi all, I have this project that requires me to use 8 PSTN lines and possible more. I was thinking 2 TDM cards with FXO modules. The I got to read the "Qs about FXO/FXS cards" thread and that scared me. Can anybody recommend anything that is known to work ok with no mysterious problems? I was thinking OpenSwitch12 cards. What do you guys think? Any help is appreciated. Regards, Hadi

...side (Mark Phillips) 17. Codecs and bandwidth (Tim Pushor) 18. RE: Teliax to VoIPJet (Wiley Siler) 19. Re: long pause on dialing.. (Randy Williams) 20. RE: swissvoice (Florian Overkamp) 21. Re: long pause on dialing.. (Giorgio Incantalupo) 22. Re: Memory leak in asterisk CVS (Erik Espinoza) 23. Re: SoftPhones: Bad, or just bad QoS? (Time Bandit) 24. Re: long pause on dialing.. (Randy Williams) ---------------------------------------------------------------------- Message: 1 Date: Mon, 18 Jul 2005 11:29:23 -0400 From: Kurt Pasewaldt <kurtwp@gmail.com> Subject: [Asterisk-U...

Hi, I'm currently working with Asterisk 1.4.17 under ubuntu server 8.04.2. MOH stopped working suddenly a few days ago with no apparent reason. I already checked the wiki and tried different things. I already verified the following items from the wiki: 1. Make sure your asterisk user has read access to the files/folder 2. Set your moh conf up as mentioned above 3. Go into asterisk -r and do

...tification (Rich Adamson) 12. Re: Can't Make Outgoing Call (Norman Zhang) 13. Re: Recommended IAX softphone. (Bruno Hertz) 14. Re: sip seeding vs registration (Greg - Cirelle Enterprises) 15. Asterisk 1.0.3 no RedHat zaptel script? (Jerry Geis) 16. Re: Recommended IAX softphone. (Erik Espinoza) ---------------------------------------------------------------------- Message: 1 Date: Thu, 23 Dec 2004 16:51:22 -0600 From: "Brian West" <brian@bkw.org> Subject: RE: [Asterisk-Users] rtp channels not through asterisk To: "'Asterisk Users Mailing List - Non-Commercial...

I have been using Sixtel from the beginning of the year and service was getting worse and worse. Yesterday I tried to access the website to get the CDR and I got an error saying that the domain no longer exists. I checked the whois and it says that the domain is on hold. Have they finally folded? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel:

...is Bankrupt (Jay Milk) > 16. RE: Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE! > (harry gaillac) > 17. RE: LiveVoip is Bankrupt (Terry H. Gilsenan) > 18. Can anyone guide me regarding h323.cong ??? > (Adeel -31) > 19. H323 (Ronald_Wiplinger) > 20. Re: SixTel? (Erik Espinoza) > 21. Shoutcast Music On Hold problems? (hank) > 22. Re: Eicon equipment, BRI Server or PRI? (Armin > Schindler) > 23. Re: polycom soundpoint ip 300 (Wilson Pickett) > 24. RE: RTP session between two end users (Erdem > HAK?) > 25. Re: Passing called number in SIP (An...

...289-SUSE-CODE11] Server Comment --------- ------- ALBERTO Alberto Viladegut ALEX Alex Chacon ALMACEN Almacen Sopocachi ... other 30 lan computers TERESA Teresa Espinoza WILFREDO Wilfredo Romero Workgroup Master --------- ------- BAGOLIN BAGO It shows BAGO as master browser (espected data). But when samba loses to be the master browser I do the same console command and shows de followi...

Hi, I''m setting up a site that will bill on a monthly basis. I would rather not have to worry about storing customer credit card information. Does anyone have recommendations on payment processors that offer monthly billing services? I don''t want to go the paypal route as I want the user to stay on the site. I''m hoping to find an API to integrate with. I have

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Hey, Does anyone know how to use the cisco router with an fxo wic with Asterisk? I don't have enough space on this device to support an IOS that supports sip or h323. Currently the only one signaling in there says Cisco. I assume this is the skinny protocol. Does anyone know how to configure this 2600 with Asterisk? Thanks, Erik

Greetings, I recently dumped my packet8 line in favor of sixtel + asterisk/sipura spa-1000. It's working great, but I want to enable call waiting and 3 way calling. When i am on a call and get a second call, currently things go straight to voicemail. It looks like sixtel is passing on the extra calls, but my asterisk box doesn't know to route it all the way. Is there a way to set it so

Greetings, I have a Sipura SPA-1001. When I make outgoing calls, I have very jittery sound. Incoming calls work fine. This wasn't the case a few months ago, I am running head as of yesterday. Any suggestions? Thanks, Erik

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