similar to: CLI Dialplan options...

I just installed Hylafax with Iaxmodem and I am not getting good results when receiving faxes. I can see that the modem is reporting the following: Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 2400 bit/s Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 4800 bit/s Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 7200 bit/s Mar 31 16:19:08 pbxoficina FaxGetty[5377]:

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

I am having a problem trying to use originate from the CLI on Asterisk 1.8.2.3. The SIP peer is defined correctly and it works if I dial using my IP phone. When I try to dial from the CLI I get this message: pbxoficina*CLI> originate SIP/protel-out/0445540881644 application playback tt-monkeys [Jan 18 12:00:09] WARNING[3336]: chan_sip.c:19048 handle_response_invite: Received response:

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. >>>> >>>>

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am trying to set the CDR(userfield) to a certain vaule using the application map of features.conf but I am not able to do it. When I receive a call I would like to tag it with a client code (3 digit numeric) so I can referenci it later from the CDR. I have edited features.conf with something like: code => #111,self,SET(CDR(userfield(111)) or

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

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

I am having a bit of a problem implementing the pickup command in my dial plan. I have setup this rule: exten => _*8XXX,1,Pickup(${EXTEN:2}) This works as expected when someone dials an extensions number and I can get the call. The problem I have is that when a call enters my welcome menu and does not press anything there is a timeout that sends them to the recepcionist. The rule is:

I have a SIP line that is working fine when I make calls from IP phones. I can send and receive calls. The problem is that if I try to dial from the CLI using the originate command or use an AMI connection to originate a call I get the following error: originate SIP/protel-out/0445540881644 application playback tt-monkeys WARNING[12950]: chan_sip.c:20437 handle_response_invite: Received

I am using Asterisk 1.4.4 on a CentOS 5 machine for a small call center with 6 agents. I am using realtime for queues and sip and I am also trying to use realtime static to load agents.conf. The only problem I am having is that no agents are loaded when I start Asterisk. I have to manually do a "module reload chan_agent.so" so the agents get loaded from the database. Obviously

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

I have searched for some time but I have not found an asnwer on how to fix the CDR when a call is transferred. The problem is that if someone dials a cell phone and then transfers the call to another extensi?n the CDR for the cell call stops and there is no way to track that the call was transferred so we can bill correctly. Many people have asked this question but there is no answer, only a

I have a customer that needs an Asterisk server to sell minutes for cell phones in Mexico. I do not see a problem with that since he will get the calls by SIP and then use GSM adapters to get the calls into the GSM network. My problem is that his customers only want to be identified by IP and not by a username and password. Is there a way to authenticate just by using an IP address? --

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

We have a setup running Asterisk interconnected to a Panasonic TDA200. The Asterisk server has a two port E1 card, one connected to the phone company and the other to the Panasonic. Everything is running fine and we can send and receive calls from the Panasonic and phone company. We are using MFC/R2 for both links on Asterisk 1.4.4 and Zaptel 1.4.3. The only detail we have is that we cannot

I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64 licenses for this application. Could anyone please remind me? -- Telecomunicaciones Abiertas de M?xico

I have a customer running Asterisk 1.2.13, Zaptel 1.2.11 with a TE110P, a TDM04B and an Astribank-32. They have been complaining that there is echo on calls even when they are IP to IP on the same network. There are 18 Aastra 9133i phones and 30 analog phones connected to the Astribank. I can understand there being a bit of echo on the analog phones, but I do not understand why there would be

Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- 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: