similar to: transfer problem

Hello everybody! I've seen that you can connect your cellphone via bluetooth, 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.

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.

Dear Asterisk Fans, I'm an Asterisk consultant in Argentina and want to make an spanish wiki (something like voip-info.org). I have the idea and some concepts about this project. It won't be a comercial project, it would be free and it's target would be spanish talking asterisk enthusiasts. I'm wrinting these for the sake of finding contributors, people who want to help me

Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I 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

Hello everybody, I'm installing an Asterisk 1.2.7.1 with GNUDialer 0.98-puff18. It also has zaptel from CVS. My FXO is an X100P Clone. The agents from GNUDialer log ok, and everything is fine until the GNUDialer makes a call, as soon as it engages (the phone starts to ring) asterisk crashes with these messages: > Channel Zap/1-1 was answered. -- Executing

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

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

If Someone did that, How I connect extensions.conf with this type of Hybrid system to work with asterisk inside this schema: PSTN--->PANASONIC KX <------> Asterisk | |--------->send internal call Thanks.

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.

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 would help) I

Hello all, iam setting up an asterisk box behind NAT to get SIP calls from outside or internet. In that eschema i can setup SIP calls but, while from the outside nat people can hear me, Im unable to listen anything behind NAT. Out of firewalls settings( I checked this to port fowarding) what can i do to get this working fine?. Thanks G.

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like:

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

Hi Everyone... I am running Asterisk 1.2.22 on Debian "Etch". I installed it from sources. I have also installed tiff-v3.6.0, spandsp-0.0.3. and downloaded http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_rxfax.c http://soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/app_txfax.c and a Digium TDM card with 4 FXO ports When my dialplan

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

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

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:

Hello! I?ve an issue whit CDR using asterisk 1.4.23.1. I?ve configured mysql to store cdr information, but, while I put into cdr_mysql.conf the field ?userfield=1? and doing a query I found that this field is empty in the cdr table. On the other hand I can?t find records in the cdr table that show me calls generated through AMI using Originate Action, that?s calls are not stored in the CDR, but I

Hello all! Does anyone have a good howto to setup Asterisk and Skype. Thanks Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080526/831b3824/attachment.htm