similar to: Wildcard TE110P in Mexico

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'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 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

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

Miguel, How many simultaneous calls (incoming and outgoing) you did ? 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:

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:

??? 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

After I have finished to key in the area codes for Mexico I would like to propose: The guy how created the numbers should be stoned to death with the dice he created the numbers !!!

Hi, I've been trying to dial out to a cellphone, but all my calls get redirected to 066 (the emergency number at my city, like 911) does anyone know how to fix this, any ideas,? does anyone from mexico has done this? Any comment will be highly appreciated, Regards, Claudio -------------- next part -------------- An HTML attachment was scrubbed... URL:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a astunicall-1.4 setup with a te110p to a nortel pbx in Mexico. (Hate R2!). This is what I get when trying to call to * box using testcall: ./testcall Chan 31, class 'mfcr2', variant 'mx,20,4', end 0, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 31: Call control(9) MFC/R2

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, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18

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

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

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

Helo, I have problem with Digium TE110P connected to CISCO 3640 (port on NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I tried with real PBX problem was exactly same. I have this messages in Asterisk conole and log sometimes: NOTICE[1115] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Usually 2-5 such messages in series, can be repeated after 10

>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, I am considering A101/102/104 cards for my asterisk installations. Has any of you used these or any Sangoma cards in such environment? Any thoughts? How do they stack up against Digium cards? Any input would be greatly appreciated. robert

Hi Folks, I have a T100P ordered from Digium in the past. Its working perfectly fine but I now need to more the server to India (out-sourcing) and we will be getting an E1 line there. Does anyone know if T100P was capable (upgradeable) or swappable so that I could use it with the E1 lines instead? Any pointers to driver/firmware update would be highly appreciated. Thanks. R -------------- next

Hello, We are looking for someone who can act as our 'remote Asterisk hands' in Mexico. One of our customers has opened an new office in the 'Col. San Rafael, Delg. Azcapotzalco' region. We need someone (or organisation) to perform the onsite installation and connections for an Asterisk server (and maybe other servers too). If you're able and interested to help us with the