similar to: Asterisk with USB

Displaying 20 results from an estimated 1000 matches similar to: "Asterisk with USB"

2006 Apr 18
1
Asterisk & GNUDialer issue
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
2007 Jan 26
3
International Carriers
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.
2007 Feb 22
3
Argentine Asterisk Wiki
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
2006 Jan 23
2
Home Test!
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
2007 Feb 28
5
about bluetooth channel
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
2006 Feb 06
8
change languages from an IVR
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?
2007 Jan 17
1
Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
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
2006 Jan 23
1
Video Conferencing.
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.
2007 Jan 29
1
Asterisk, VoIP and Linux Blog.
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
2006 Jan 23
1
Testing List (JUST A TEST)
Sorry, I haven't received a message in a few hours, just testing to see if it is alive.
2007 Jan 17
1
transfer problem
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
2006 Jan 23
4
make linux26
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:
2004 Jul 18
3
zaptel issues
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
2006 Jan 23
5
dial out and message playback
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
2006 Feb 02
1
Re: Contents of Asterisk-Users digest...
-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To
2006 Jan 27
1
chan_bluetooth: successful compile and outbound cell calls: Still tweaking inbound setup. WAS: Cannot compile chan_bluetooth on Asterisk 1.2.1
Editing subject line to reflect current status. On 1/26/06, Nilesh Londhe <lvnilesh@gmail.com> wrote: > Since T616 is not answering (and incoming calls are going to Cingular > voicemail after 30 sec,) I suspect the problem focus area is... > > > -- Executing Answer("BLT/T616", "") in new stack > > Is
2006 Feb 06
3
SV: callback script?
Thanks. I'm able to getting the asterisk calling back to my cellphone. But when I get to the authentication I get this message when I start to dial in my password: NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received Is this a DTMF failure of some sort? Thanks again. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com
2011 Apr 21
7
repeater
Hi all there. I am trying to set up an icecast2 server (wich is already running) to send some stream mount point to another server. I mean, my server is streaming at /radio1.mp3 mount point. We can listen it at http://myserver:8000/radio1.mp3; so, i want to know if is there any way to "clone" that strem and send it to another server, such as http://giss.tv:8000/radio1.mp3 Thanks. --
2011 Aug 10
2
Opposite of paste function
Dear All, I have vn variable > vn [1] "V300" "V376" What I want to get is 300 376 without V and "" from vn variable. Could you help me about this issue? Thank you, Soyeon [[alternative HTML version deleted]]
2006 Nov 20
2
email etiquette (was: Re: Unicall MFC problems in 0.0.3+asterisk 1.2)
It's bad enough that people insert long config/log files/dumps into messages to this list, though it's convenient. But when people include the entire redundant content in a quote in reply, it's really a waste. The digest messages are hard enough to navigate, even with an intro index, before making 95% of their content a redundant quoted dump. I know we've all got lots of bandwidth