search for: chavez

...d the second was almost a month, between the second and third a few days. We use CentOS 5.5 on a Dell 905 server with 8gb of ram. The first time it happened we were using Asterisk 1.6.2.14 so we upgraded to 1.6.2.15 to try and avoid the problem but no luck. Any recommendations? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001

Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001

On 2/22/18 3:46 PM, Antony Stone wrote: > On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote: > >> On 2/22/18 1:07 PM, Antony Stone wrote: >>> On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote: >>>> Usually phone companies set the outgoing CallerID for you but >>>> >>>> recently we got control over that and are...

...t match checksum puppetlabs-deps Puppet Labs Dependencies El 5 - x86_64 45 puppetlabs-products Puppet Labs Products El 5 - x86_64 0 Same on RHEL6. Insidious problem. Only noticed it because I was kicking some systems today. Should I report this as a bug? -- \*..+.- --Greg Chavez +//..;}; -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To post to this group, send email to puppet-users@googlegroups.com. To unsubscribe from this group, send email to puppet-users+unsubscribe@googlegroups.com. For more options, visi...

...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: +52-55-91169161 Ext 2001

I want to buy a new server to run Asterisk and after looking at prices for the Athlon XP 3000+ it costs the same as an Athlon 64 at the same speed rating. I was wondering if Zaptel/Asterisk will compile/work on an Athlon 64? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V.

..."http_caching=none" in yum.conf but that *really* sucks. If I''m being stupid and there''s an obvious solution to my problem, I beg of you to tell me what it us and humiliate me in front of the entire Puppet users community. I''d be so happy. -- \*..+.- --Greg Chavez +//..;}; -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To unsubscribe from this group and stop receiving emails from it, send an email to puppet-users+unsubscribe@googlegroups.com. To post to this group, send email to puppet-users@goo...

On 11/14/17 4:27 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: >> On 11/14/17 3:55 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: >>>> I followed the blog post and I can get video from the conference if >>>> I configure the bridge as follow_talker so I know everyt...

...ith DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only happens when calling from a digital phone on the Nortel. If I connect an analog phone to the PBX and dial from there the call can go through. Anyone has experience with this? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come over the other lines. -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director

GREAT NEWS ARE Hitting ON FR IDAY FOR CRSVF!!! DON'T miss THIS INVESTMENT MOMENT, PLACE {CRSVF} ON THE RADA`R!!! Tra de Al er t: FRIDAY, October 06, 2006 "Symb0l": CRSVF.OB Current Pri ce : $0.15 Pr evClose : $0.18 Rec ommendation: ST RON G-B UY WATCH THIS STO CK GO HIGHER AND HIGHER DON'T MI SS THIS INVE STM ENT MOMENT, PLACE CRSVF ON THE PORT FO LIO!!! About

Hi Carlos/All, Thanks for your reply. I can remove dchan=16 from zaptel.conf But according to the documentation of Digium and sangoma they mentioning to use dchan=16. Are there any specific reason you have experiance regarding this and I am confusing that what this is included to the documentations. Regards, Vidura. On Wed, 2007-09-05 at 20:26 +0530, Vidura Senadeera wrote: > Dear All,

>On Mon, 2007-11-12 at 15:46 +0100, Stefan Guenther wrote: >> > Hello, >> > >> > I have a strange situation: >> > >> > I can talk to other SIP phones and via ISDN to the outside, but I >don't hear >> > playbacks or the voicemail messages. >> > Asterisk show in the cli, that the corresponding files are played,

Can anyone recommend some White Papers or Success Cases that we can use to ease the mind of a customer that has not heard much about Asterisk? All they know is Avaya at this point. -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001

GREAT NEWS ARE Hitting ON FR IDAY FOR CRSVF!!! DON'T miss THIS INVESTMENT MOMENT, PLACE {CRSVF} ON THE RADA`R!!! Tra de Al er t: FRIDAY, October 06, 2006 "Symb0l": CRSVF.OB Current Pri ce : $0.15 Pr evClose : $0.18 Rec ommendation: ST RON G-B UY WATCH THIS STO CK GO HIGHER AND HIGHER DON'T MI SS THIS INVE STM ENT MOMENT, PLACE CRSVF ON THE PORT FO LIO!!! About

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...rres at gmail.com>> > IM: mhterres at jabber.mundoopensource.com.br > <mailto:mhterres at jabber.mundoopensource.com.br> > https://www.mundoopensource.com.br > https://twitter.com/mhterres > https://linkedin.com/in/marceloterres > > On 19 July 2017 at 17:03, Carlos Chavez <cursor at telecomab.mx > <mailto:cursor at telecomab.mx>> wrote: > > On 7/19/17 2:37 AM, Marcelo Terres wrote: > >> This is the pjsip library. >> >> Is it possible to you to update pjsip for the latest version to >> test if it solves...

I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card and an OpenVox A1200P card. Up to today everything was working perfectly. The OpenVox card has 8 FXS and 2 FXO ports. The two faxo ports are used for a GSM adapter and for an ATA connected to Vonage. The problem we started noticing today was that the Vonage line will receive a call and then cannot connect to any of the SIP

...butils 014-1 .- nut&nut-common&nut-server&nut-upsc&nut-driver-usbhid-ups 2.7.4-27 .- usbhid-ups 0.41 (2.7.4) USB communication driver 0.33 ________________________________ De: Jim Klimov <jimklimov+nut at gmail.com> Enviado: lunes, 8 de mayo de 2023 7:27 a. m. Para: Feliciano Chavez <chavezfeliciano at hotmail.com> Cc: Arnaud Quette via Nut-upsuser <nut-upsuser at alioth-lists.debian.net> Asunto: Re: [Nut-upsuser] NUT support for the new Tripp Lite AVR700U (USB 3024) under OpenWrt? I'm mostly not online for a few more days, so can't check directly. The ge...

On 2/22/18 1:07 PM, Antony Stone wrote: > On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote: > >> Usually phone companies set the outgoing CallerID for you but >> recently we got control over that and are now setting the outgoing >> Calleir ID ourselves. My problem now is that the CDR will put the >> outgoing CID in the CDR instead of the extension...