similar to: RES: Working E1 MFC/R2 M?xico !!!

Steve, Do you know if there are many differences between the Mexican and Brazilian variant ? Kind regards, Miguel Antonio Date: Tue, 05 Oct 2004 22:51:53 +0800 From: Steve Underwood <steveu@coppice.org> Subject: Re: [Asterisk-Users] RES: Working E1 MFC/R2 M?xico !!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Miguel, How many simultaneous calls (incoming and outgoing) you did using this implementation ? Kind regards, Miguel Date: Sun, 3 Oct 2004 11:21:04 -0500 From: Miguel Cavazos <miguel@cavazos.com.mx> Subject: [Asterisk-Users] Working E1 MFC/R2 M?xico !!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Hello all, R2 in mexico (telmex) is working with steve underwood libmfcr2, we have done here some test 5 incoming calls and 5 outgoing with no problems. Thank you steve for all the support to this proyect, I hope this week we can start testing other E1's like unefon, alestra, axtel and others. -- Saludos, Miguel Cavazos

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

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can

>any feedback would be awsome, the idea is to fill in the 30 channels of >the E1 all at the same time and see how stable it can be > > ----- Original Message ----- > From: "Miguel Cavazos" <miguel@cavazos.com.mx> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Sent: Wednesday,

Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of

Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details

Hi all, I'm trying to get the llvm's IR from the source code of Polybench (OMP) https://github.com/cavazos-lab/PolyBench-ACC/tree/master/OpenMP. I noticed a considerable difference between the IR generated using clang -emit-llvm -fopenmp and clang -emit-llvm: * using the -fopenmp flag I get a simplified IR in which I read a single basic block where I can highlight a llvm.memcpy

I've been trying to configure an E1 in Mexico using unicall, i went into vozdigital, googled this list, and finally followed this instructions: http://voip-info.org/tiki-index.php?page=Asterisk+MFC+R2 I have 10 PSTN numbers and 10 "lines" assigned, so i only have 10 "channels" assigned from my telco. However when i try to simulate a call using this call file: --------call

??? I am having a really bad day trying to get incoming calls to work on Asterisk 13 with PJSIP.? We just migrated from Asterisk 1.8 where everything was working but there seems that something got lost in translation.? No matter what I try I always get a 401 Unauthorized message when receiving a call from the PSTN provider.? I can make calls and the registration is working.? I have tried to

For the past month I've been having trouble dialing certain numbers. We have Asterisk 1.4.14, Zaptel 1.4.7, Libpri 1.4.2 on a CentOS 5 server. We are using PRI on a TE110P card /etcwith a provider called Alestra in Monterrey, Mexico. There are some numbers that whenever we dial them we always get a busy tone. These numbers do not all belong to the same provider, but they all do not

Hello guys, i have been on this mailing list for some weeks now, and i was wondering if someone here has installed linux on the XBOX and use it as a dedicated server. Its a 200 USD computer and could make it perfect to asterisk, its little and doesnt really take much space. My question is could this make it for a stable server??? here are some links i found for linux on XBOX

I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico. This is really the first server I have used with PRI in Mexico as we normally use MFC/R2. Everything seems to be working except that some numbers always seem to be busy when you dial them. All these numbers belong to different phone companies. I know that with R2 this problem is present if you have a "#define

I followed the how-to for R2 that coppice wrote on the webpage http://www.opencall.org/installing-mfcr2.html, now asterisk has channel unicall and when i test an incoming call I get the following errors. astbox:~/mediatel/libmfcr2-0.0.1# asterisk -r Asterisk 1.0.0, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com>

Hello Everyone, I am looking into using Asterisk as our company PBX and voicemail system. I am very familiar with Linux, but the VOIP stuff is new for me. We are a non-proffit organization, so keeping things as cheap as possible is very important. I am looking on some advice for best implementing Asterisk. Here is a rundown of our current system: We have 14 phone lines coming into the building.

I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is having calls dropped. Sometimes you can stay on the phone for a long time and sometimes the call is dropped within a minute. There are 9 lines connected to 3 TDM04B cards. The Panasonic Pbx we replaced did not have this problem. There are 8 SIP phones and 16 analog phones connected to two Astribank-8 units and everyone

FYI to anyone else who may be experiencing random hangups; I removed the busydetect=yes lines from the conf files on my asterisk servers, and haven't had a hangup since. I had done that once before and it didn't seem to have much of an effect, so I'm not breaking out the champagne yet. But so far over dozens of calls both made and received since I took that line out, I

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Has anyone on the list been able to locate and try out the 1.1.0 firmware? It was released in November, but I have yet to get my hands on it. The Polycom site has way more docs online, but the link to the firmware only brings up the release notes. -sb