similar to: Dialing an external number and then passing it to an extension...

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

I have a customer that has a Trendnet TEW-435BRM router which has the bad habit of rewriting all external connections so the Asterisk server only sees the IP address of the router itself. Up to today this has not been a problem since all extensions are on the local network but now they want to have a couple external IP phones (SIP). I opened up the ports on the router and my phone can register.

Lately I have been getting many complains that Eyebeam crashes when you dial a number that does not exist. This happens in both R2 and ISDN PRI lines. The softphone stops working and has to be restarted. The response I got from tech support was: the actual issue is that asterisk should not be sending a 503 service unavailable when a particular softphone is not online. The soft phone stops

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

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

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:

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

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 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

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

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I am having a problem with an Asterisk 1.6.2.15 server that runs a small call center with Queuemetrics. In the past month we've had this problem 3 times. The problem is that Asterisk simply stops responding. No calls in or out and you cannot even get to the CLI. The process seems to be running but there is simple no activity. All I see in the log files is: [Jan 14 16:30:46]

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

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

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

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

??? 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 that dialed and that causes problems for reports.? What is the proper way to set outgoing CID and keeping the original extension number in the

I am having a strange problem with an asterisk servier using R2 Unicall in Mexico. Most calls go through fine but some of them give me an error like this: -- Executing Dial("SIP/86-db41", "Unicall/g2/014448343600") in new stack -- Called g2/014448343600 Feb 9 21:44:39 WARNING[23069]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Dialing Feb 9 21:44:45

I have an Asterisk server that was running fine until Sunday. Monday there was a power outage and the server was off most of the day. This server has a TE110P, two TDM04B and an Astribank 32. Today I noticed that the TE110P started having IRQ missess. Before today it only had about two or three per month. Today, this is the miss count after ten minutes on: IRQ Misses: 1443

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