search for: ameal

Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Share your knowledge, use free software.

...uetooth, but I've a Motorola V300 and it doesn't have that feature, so I wish to connect it via USB cable, is it pissible con use my cellphone with asterisk like that? I 've not been able to find information on how to do this, I'l appreciate any help. Thanks in advance! -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 FWD: 741664 MSN: asado<at>lamorcilla<dot>com<dot>ar ICQ: 74005793 Open your mind, use open source.

...free (perhaps for a limited time) reliable hosting with the benefits of being able to install everything we want (like mediawiki, drupal, tiki-wiki or whatever) with complete access to mysql databases. Please, anyone who is interested in this send me a private e-mail. Best regards! -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Share your knowledge, use free software.

...read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works ok, then I-m planning to install it in other places. thank you in advance. regards, -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Open your mind, use open source.

I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Open your mind, use open source.

...;) in new stack localhost*CLI> Ouch ... error while writing audio data: : Broken pipe Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0). [1]+ Segmentation fault asterisk -vvvv I 'll really appreciate any help. Thanks in advance! -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 FWD: 741664 MSN: asado<at>lamorcilla<dot>com<dot>ar ICQ: 74005793 Open your mind, use open source.

...es echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel => 32-46 channel => 48-60 Thanks in advance! Greets! -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Share your knowledge, use free software.

Hello everyone! In my humble try of creating a Blog, I've made this: http://fameal.blogdns.org. By now, it's hosted in my own server but shortly it'll be moved to a serious hosting. All post are written in spanish, so it's only for spanish talking people, I will try to make it grow and have english articles. If someone is interested in writing in english (obiously I...

28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the

...al_sip:5060 stargate2 105 Registered local_sip:5060 stargate1 105 Registered from sip phone I can any other phone (cisco with sccp or iax protocol) but I can't call any other sip phone, or receive phone calls. Facundo Ameal <fameal@gmail.com> wrote: are you sure your sip phone is registering ok? 2006/2/1, abc def : > Thanks Facundo for instruction but it didn't work. there is nothing new in > your suggestion compare to my conf files nevertheless I tried it but it > didn't work. I can make cal...

I thought I read somewhere that you no longer have to do a special make command for the 2.6 kernel. Is this true, or should I still make linux26? I'm having problems getting anything zaptel to load properly. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sorry, I haven't received a message in a few hours, just testing to see if it is alive.

Hello, I've tried to transfer a IAX call to a number configured on a traditional PBX, but it doesn't work. I have a traditional PBX connected with a zap channel to Asterisk in the following way: IAX/SIP client --> Asterisk (FXO) --> (FXS) traditional PBX ---> OFFICE Phones Asterisk is connected to the PBX with an internal number configured inside it. In other words i keep an

Hi, I've been trying to bring our Asterisk server to the latest version. I've grabbed the latest CVS and upon trying to compile zaptel, I get the following errors: gcc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c gcc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB

...o find the occasional needle of useful content. Please consider storing long machine input/output data at a URL, then emailing just a link to it. On Mon, 2006-11-20 at 07:55 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Mon, 20 Nov 2006 10:20:19 -0300 > From: "Facundo Ameal" <fameal@gmail.com> > Subject: Re: [Asterisk-Users] Unicall MFC problems in 0.0.3+asterisk > 1.2 > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <a563d36f06112...

Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of

A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish?