search for: telecomunicacion

Displaying 20 results from an estimated 401 matches for "telecomunicacion".

Did you mean: telecomunicación
2005 Mar 08
4
Nortel ATA not passing dtmf tones to fxo
...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
2008 May 20
7
Busy out a zap channel?
...sy 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 de Tecnolog?a +52-55-91169161 ext 2001
2008 Feb 06
3
R2 with Alestra in Mexico...
I am trying to set up Astunicall 1.4.16 with a link from Alestra in Mexico City. I have done everything I usually do for other links in Mexico but this one simply will not send or receive calls. I just get Protocol error. Anyone has any experience with R2 and Alestra? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url :...
2006 May 23
13
Now that Nufone is dead...
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
2006 Dec 20
5
Sangoma A101 with Unicall
...= 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO Basically I only get channels 1 to 15 active in Asterisk. I am using wanpipe-2.3.4-3 with Zaptel 1.2.12 for this configuration and Asterisk 1.2.14. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url :...
2011 Mar 25
2
White papers or success cases to convince a customer?
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
2005 Aug 02
5
Has Sixtel gone under?
...g 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
2008 Apr 30
1
One way audio...
...apters behave differently? I cannot think of any setting on the PAP2T that is different from the SPA. If I connect the PAP2T on the same network as the Asterisk server then audio is fine, the moment it crosses through the firewall we only get one way audio. Anyone had a similar experience? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url :...
2017 Oct 19
3
speech-recog.agi
...E "Unable to get recognition data." 3 I made sure all the dependencies are met and that my API key for Google Cloud Speech is correct (cut and paste). Any pointers to get this to work or any other quick waysto start using Google for speech recognition in Asterisk? Thanks. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)8116-9161
2012 Jun 05
3
Another IP address to block
Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL:...
2019 Jan 14
2
Various extensions ring once and go to voicemail
...      0          Running          extended res_timing_timerfd.so          Timerfd Timing Interface                 1          Running              core 3 modules loaded     This will show you what module Asterisk is using for timing. You can try doing a noload on the two you do not need. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190114/f608670e/attachment.html>
2007 May 29
7
Problem on incoming call from Zap channel to SIP phones...
..."Hookstate (FXS only): Offhook" even when the line is not in use. The other port connected to the GSM adapter says Onhook. Also, when Asterisk answers the call from Vonage I hear a loud click on the phone and after it has tried both extensions I can hear the Voicemail message play. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url :...
2007 Sep 04
6
Overhead paging over IP...
...t an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we cannot run a cable from one building to the other just for audio. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url :...
2016 Mar 24
2
PRI error "ROSE REJECT"
...e on their side, obviously. The problems started a few days ago when a user reported that incoming calls get dropped when you try to dial a particular extension from the main IVR. We are using Asterisk 1.8.15-cert2 on a CentOS 6.7 server, DAHDI 2.6.1 and libpri 1.4. Any recommendations? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez dCAP #1349 +52 (55)9116-91161
2007 Jun 06
4
meetme realtime
Hi iam using 1.2.17 does any one have information meetme in realtime and store in mysql i dont see any document could some one help me is this possible ? ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070606/36d236c2/attachment.htm
2006 Jan 11
4
Echo on phones...
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2017 Jul 18
2
Asterisk 13.16.0 segfault
...5eae8285c0 error 4 in libasteriskpj.so.2[7f5f29a49000+180000] Since this is a Freepbx distro does could the problem be related to their flavor of Asterisk? I have several other plain Asterisk servers running on this version without any problems. Any recommendations on how to debug this? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez dCAP #1349 +52 (55)8116-9161
2016 Apr 05
3
Best timing source?
...oice to sound cracked and with small interruptions. I am looking at the timing source for Asterisk and it is currently using timerfd even though we have an E1 card installed. Is timerfd better than dahdi? Any recommendations to test if timing may be a problem for voice quality and DTMF? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161
2010 May 20
3
Softphones on thin clients...
...stomer that wants to use Eyebeam (about 10 users) on a thin client platform. Each user has a little box on their desk that has a USB port, mic and headphone jacks and monitor. I am worried about conflicts when running 10 softphones on the same server since they will all try to use por 5060. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part Url :...
2018 Feb 22
2
Set external CID but retain internal extension in CDR...
...hen; what do you get in the "channel" field? > > > ??? Channels contains PJSIP/XXXX-(id) ??? Like I mentioned, the problem really lies in that the software used for call reports is coded to the "src" field.? Than is why I need src to hace the extension number. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)8116-9161