search for: telecomunicacions

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

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

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

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

I am having a problem trying to get a Sangoma A101 to work with Unicall. I have installed the sangoma drivers and everything seems to be well but when I try to run ztcfg I get the following error: CAS signalling on span 1 conflicts with HDLC with FCS check on channel 16. Here is my /etc/zaptel.conf # MFC/R2 normalmente no usa CRC4 span=1,0,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101

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

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:

I have a big headache. I have an Asterisk server connected to an Avaya PBX. Everything is working between those two. The problem is that I have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the Internet to the Asterisk server through a Fortinet firewall. When calling from a PAP2T I get one way audio, the remote site can hear me but I cannot hear them. If I do an "rtp

I want to try using google for speech recognition in Asterisk and I found a ready made AGI: http://zaf.github.io/asterisk-speech-recog/ I have followed all the steps listed in the web site but I keep getting this error: <PJSIP/2001-0000006e>AGI Tx >> 200 result=99981 (timeout) endpos=22720 <PJSIP/2001-0000006e>AGI Rx << VERBOSE "Unable to get recognition

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

On 1/14/19 4:02 PM, Duncan Turnbull wrote: > > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> wrote: > >> Duncan: >> >> You may have it right—I took one phone and set the ring time to 60 >> seconds. I now get about 4 rings on that one. >> >> I wonder how I

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

I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought 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

We've been having some problems with an E1 PRI line for a few days. We get the following errors: [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT: [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 INVOKE ID: 316 [Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 PROBLEM: Invoke: Unrecognized Operation The telephone company says that

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

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I am getting frequent segfaults on a new Asterisk installation. So far the only message I see is: Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip 00007fb2d535723f sp 00007fb25a11b5c0 error 4 in libasteriskpj.so.2[7fb2d52e5000+180000] Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188 ip 00007f4afea0c23f sp 00007f4a7f7e35c0 error 4 in

I am currently having a voice quality problem with one of our Asterisk servers. We have checked the network and we have found no problems that could cause the voice 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

Does anyone know if you can use softphones on thin clients? I have a new customer 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

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