similar to: best format for audio via asterisk...

Displaying 20 results from an estimated 3000 matches similar to: "best format for audio via asterisk..."

2007 Jun 07
3
getting at ${CALLERIDNUM}
Hi all -- I'm having awesome fun with Asterisk & voicepulse connect together. So cool. I'm trying to have the caller id read back to me. Do I need to do something to have this sent across in the sip.conf? Or is there something I need to do somewhere to enable the reading of this data? Thank you! Matt Here is my extensions.conf exten => _XX.,1,Answer() exten
2007 Jun 04
2
answer a voip call, play info.
Hi all - Not really sure where to post this question as I am just starting to research this issue. We want to allow users to dial into our did voip number. Our service will: 1. get their phone number via caller ID. look up data with the caller id. 2. generate a wave file based on the data returned & play it to the user over the established voip link. How might this be done using
2007 Jun 07
4
agi with java?
Hi all - Searching for java agi in the mailing list archives turns up ancient posts. Anyone else using java for their AGI? How well is it working & what are you using? My script is pretty simple, and I could write it with perl easy enough, but I just would feel better if I can keep most programming code for our system in java. Thank you- Matt
2010 Jun 16
1
R and LINGO?
Good Evening Does anyone in the R-help list have experience writing an R wrapper that interfaces with the commercial packages LINGO and/or LINDO.api from R? I have a set of nonlinear/mixed integer problems that solve nicely with LINGO but I would like to use R to set the problems up and analyze/plot the solutions dynamaically. I have searched the archives and have not found any R packages that
2005 Feb 17
1
Voicepulse Open Access & Asterisk Problems
I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is
2005 Mar 21
5
VoicePulse Issues
I recently switched from BroadVoice to VoicePulse Connect on my Asterisk box. Too many Meetme quality complaints (whether real or perceived). I had to make a choice to use IAX2 or SIP with VoicePulse. I first tried to go with SIP because I already had it working and all of our devices are SIP. Problem is that every time I turn my back, the Asterisk registration with the VoicePulse SIP server
2004 May 25
3
Voice Pulse
Hello: I am new to the list. I am trying to set up asterisk with voicepulse. I have a voicepulse username + password, and SIP DID. When I login to voicepulse, I have this under my devices tab: Devices *Login:* Sysxxxxxxx *Password:* xxxxxxxxxx *Context:* VPWS *Connects to:* gw5.voicepulse.com My question is: Do I need a 2.4.x kernel? Currently I am running Debian/stable stock 2.2.x ? Has
2003 Dec 09
1
dialling peer problems
I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows
2004 Sep 12
2
(no subject)
Hey guys, Im about to sign up for VoicePulse Connect. Of course, I plan on using my asterisk server to "register =>" with the service. I would rather sign up with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is behind another Linux box serving as my nat/firewall. Does anybody think I will have a problem ? Should I stick to IAX and VoicePulse Connect or can I use
2003 Dec 08
2
Problems with voicepulse.com
Greetings, I have been experimenting with Asterisk for a few weeks and finally decided to take the plunge and purchase a few DIDs for inbound calling. Our attempts at IAX/IAX2 connectivity with VoicePulse have been less than successful. We get "Registration Refused" errors from Asterisk whenever we launch the server. The front-line support folks at VoicePulse suggested that we are
2009 Apr 26
1
Error, Clue to what?
[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer '3516533812' is now UNREACHABLE! Last qualify: 86 [Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke: Peer '3516533812' is now Reachable. (98ms / 2000ms) [Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
2009 Jan 28
2
How to retrieve a phone number from call forwarding?
Hi, I'm very new to Asterisk and I have the following scenario. 1. Let's say I have a number of 1-222-222-2222 from my SIP service provider (VoicePulse). 2. I point my phone, Verizon wireless cellphone (1-111-111-1111), voicemail to the number provided by SIP service provider (1-222-222-2222). 3. I use another phone (1-333-333-333) to call 1-111-111-1111 and leave a voicemail message.
2004 Aug 23
2
VoicePluse DID problem
Hey guys, Cal someone help me. I'm register voiceplus DID i try to config fllow example but not work. When i test call to number and debug iax2 in my asterisk not found packet. My iax.conf -------- register => in-xxx:yyy@gw5.voicepulse.com [voicepulse] context = voicepulse-incoming secret=yyy auth=md5 type=friend host=gw5.voicepulse.com ------ extention.conf ---- [voicepulse-incoming]
2005 Jun 09
4
Lingo(.com) and Asterisk
Hello, A long Google search didn't turn any clear answer. Does somebody use Asterisk in combination with Lingo? Thank you, Bas
2004 Jun 14
4
Number Portability and VoicePulse
I have two questions regarding number portability... 1) If I port a DID over to Voicepulse, can I then move that DID elsewhere somewhere down the road. Or does voicepulse now OWN that DID? 2) Can I take a DID assigned by Voicepulse and transfer it to someone else? If not, why? -jwb
2004 Jan 29
1
re: help with voicepulse connect IAX2
hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am
2006 Nov 01
5
DTMF over IAX
Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no
2004 Aug 30
2
VoicePulse Connect DTMF with IAX2
Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK
2003 Oct 16
1
Weird IAX2 problem
I have an inbound and outbound account with Voicepulse (I am very happy with the service, btw). But I have a weird IAX2 problem. When I get a inbound call on my Voicepulse DID, the call hits my asterisk server correctly with the correct callerid (the DID phone number 617902xxxx). when the call gets passed on to a softphone (X-lite), the caller id that shows up on the X-lite softphone as Lee ,
2005 Jan 16
1
Guatemala DID's?
I'm looking for a company that offers Guatemala DID's. I saw that Lingo does, but Lingo isn't easily compatible w/ Asterisk, so they're a last resort. Thanks in advanced, Phil Astin.