similar to: Zap channels not hanging up...

I have an asterisk servicer (1.0.5) with 3 X100P cards. Everything is working fine but two days ago I implemented call forwarding using the example from voip-info wiki. Now when I enable call forwarding on my phone and a call comes in it gets redirected to my cell and everything is apparently working. The problem is that when we hang up both Zap interfaces (the one where the original

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

Anybody knows if it's possible to disable zap echo cancellor from dialplan only for certain outbound calls ?? I share the same phone lines for voice calls and faxes. Iaxmodem works fine for me only turning off the echo cancellor, but I need it for voice calls. Any ideas ? maxx

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

Trace with 3 clients. We can hear each other but no video. https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz On 11/14/17 5:06 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 07:03 PM, Carlos Chavez wrote: >> 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

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

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 everything

On 11/15/17 11:10 AM, Joshua Colp wrote: > On Wed, Nov 15, 2017, at 01:05 PM, Carlos Chavez wrote: >> On 11/14/17 5:23 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: >>>> Trace with 3 clients. We can hear each other but no video. >>>> >>>>

Does anyone know if it is possible to upload a common directory to all Aastra phones (480i, 9133)? Is there someting equivalent to the way Polycom phones do it? -- 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

I have an Asterisk server running todays CVS (updated it just in case that was the problem). It has 3 X100P cards. The first two lines I use as my normal work lines and the third is my fax line which I use with SpanDSP. I run Fedora Core 4. I have a problem that the third X100P does not answer the call. From the console I can see that there is an incoming call with the following: --

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

We have just updated our office server to Asterisk 11.4.0 from 1.8.15 and internally everything is working fine. The problem we are having is that we cannot use any external phone connected through the Internet. This used to work fine with 1.8 but since the upgrade whenever you register any phone from an outside network the phone tries to register using the servers internal IP. I endo up

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 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]

On 11/14/17 5:23 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: >> Trace with 3 clients. We can hear each other but no video. >> >> https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz > Do you see anything in the Javascript console of the browser? We are > adding the needed media streams by sending a reinvite to

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

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 everything is working >> on the pjsip side. The only problem is that video_mode = sfu is >> apparently not valid in either confbridge.conf or

I am in need of some guidance regarding the following problem: I need to dial an external number from a list(PSTN) I need to check if the number is busy, no answer or fail If any of the above are met then I try another number from a list If none of the above happen then I first need to determine if the line answering is a fax machine or an answering machine If fax or answering machine then hangup

On Mon, Sep 12, 2016 at 3:01 PM, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 9/12/16 3:39 PM, George Joseph wrote: > > > > On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjoseph at digium.com> wrote: > >> >> >> On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cursor at telecomabmex.com> >> wrote: >> >>> Has