similar to: IP-PSTN / PSTN-IP Gateway Service Providers

Displaying 20 results from an estimated 2000 matches similar to: "IP-PSTN / PSTN-IP Gateway Service Providers"

2005 Mar 19
2
Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
Hello, We are getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network (around 600 ms). There is no problem when the same user is trying to make a call with low latency network (around 300 ms). I have included the debug and log messages for Asterisk. This call is done with SJphone, the same problem exists with ATA;
2004 May 08
5
1800 Provider
Hi list, I'm interested in receiving incoming call to my Asterisk PBX thru an 1800 number. Anybody knows a provider with best minute rate? I heard that that Nufone can provide this service for around 3cents/min for calls made within 48 continental states. Any provider that can give better rate, even with additional limitation such as much few states that a call can originate? How do
2005 Feb 24
2
[Asterisk-Dev] How to monitor Agen Voice channal?
Hello, How can we monitor agents voice channels for training or quality control purpose. While agent is talking to a customer we need to be able to monitor voice channel (the actual voice conversation). If possible we would like to do that without putting agents in conference rooms. Is there any possible way to do that? Has someone done this? In addition when we tried to put the agent in
2003 Jun 16
8
SIP REGISTER
Hi! I have a new problem with my SIP device.I have done some changes and now I receive continuosly a SIP message: "501" "Not impelmented" back from the SIP Gateway. I can see that it doesn't register to Asterisk. I have in the SIP device: Registrar 1: UnRegistered to: 2222 registrar: 188.208.12.237 5060 expires: 2000 name: gateway passwd: 123 My
2004 Jan 18
5
Latest version of asterisk
Hi, Can anyone please let me know what is the latest stable version of asterisk. Thanks, shailesh --------------------------------- Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040118/fa0aba47/attachment.htm
2004 Oct 26
2
application printing text, output very small
Hello, I have gotten an application to run well in Wine. However, when this application tries to print a text document, the size of the text is so small that it looks like dots on the page. It is not readable. The printing of graphics is better. Any advice on how to get the output text to be larger would be greatly appreciated. Aram J. Agajanian
2003 Jan 12
3
Using ssh as a wrapper
Has anyone tried to wrap your SMB session inside another TCP protocol like SSH? I'm trying to sync files across two linux machines but my broadband provider is blocking SMB and port 139 traffic. If anyone has any information it would be greatly appreciated. Aram -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:
2009 Aug 20
12
IPKall and FWD
We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/4206395a/attachment.htm
2004 Nov 22
1
Using IPKall and SIP with insecure=very
Hi all, I've got one of those cool free incoming IPKall phone numbers from www.ipkall.com. These numbers just connect to the SIP proxy of your choice, they default to Frreworld Dialup. You can use them with your own sip proxy on asterisk. My config for this is below. The trouble I'm having is the incoming calls do not seem to hit the section in sip.conf for the call. With sip
2006 May 29
6
Numerical error in R (win32) (PR#8909)
Hi I had observed the following problem in R (also C, Matlab, and Python). sprintf('%1.2g\n', 3.15) give 3.1 instead of 3.2 whereas an input of 3.75 gives 3.8. Java's System.out.printf is ok though. > round(3.75,1) [1] 3.8 > round(3.15,1) [1] 3.1 Similar outcome with sprintf in R. However, the right answer should be 3.2 Regards Teckpor [[alternative HTML version
2011 Apr 26
6
vif-common.sh and iptables
Hey everyone, I have a question about vif-common.sh. I run multiple bridges attached on dummy interfaces, which allow me to put guests in seperate subnets (routed through the dom0). As you might expect I already have quite extensive iptables scripts to accomidate this kind of routing. I was just hoping someone on this list can confirm, that I understand what the iptables lines in vif-common.sh
2009 Apr 06
2
IPkall
Does IPKALL still exist? I am after a free SIP trunk - who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -------------- next part --------------
2004 Jan 20
1
PSTN Gateway
Hello, I am looking for information on setting up digium FXO card for use as a PSTN Gateway (H323-PSTN) to work with GNUGk. I am basically looking for the setup and it would be great if anyone can share his experiences with the same. Also, if there are any limitations in going for such a setup and problems that may arise/things that I should keep in consideration. Thanks & Regards, Deepak
2005 May 20
2
SecureTelephony
Ok, now thats a gadget i want to have :-) http://www.global-teck.com/english/newproduct.php http://www.global-teck.com/english/telecomproducts.php Anyone knows something similar that would work with asterisk, or any chances getting this to work? Regards, Andreas _________________________________________________________________ Need more speed? Get Xtra Broadband @
2010 Oct 24
5
Integrating Asterisk 1.8 with Google Talk and Google Voice
Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks
2011 Oct 05
11
compiling kernel in Centos Domu
Hi, Im trying to compile kernel 3.0.4 inside Centos 5 DomU The steps I do are Make bzImage Make modules Make modules_install Depmod -a mkinitrd /boot/initrd.img-3.0.4 3.0.4 but them im getting this -bash-3.2# mkinitrd /boot/initrd.img-3.0.4 3.0.4 ls: /etc/modprobe.d/*.conf: No such file or directory No module ehci-hcd found for kernel 3.0.4, aborting. If I exclude the
2009 Jan 13
9
FWD and Asterisk
I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895
2004 Feb 03
2
IPKall->FWD->Asterisk
Hi Folks, I recently setup an asterisk system in order to provide a telephone phone system for my web hosting business at a very low expense. My problem is that DTMF tones are not being recognized when calling the IPKall phone number. Calling my server via FWD and IAXTel works out fine however. Has anybody experienced this with the IPKall service? are they not passing the DTMF tones or am I doing
2006 Feb 22
2
context being ignored by inbound sip call
hello- i was messing around with a did from ipkall.com, and asterisk seems to be ignoring the context specified in the sip config. in sip.conf, i've added: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = "ipkall incoming" <7508> nat = no in extensions,conf, i have: [remote] exten => 7508,1,DISA(1111|internal) [internal] exten =>
2006 Mar 22
6
Can this box handle 8 T1s (PSTN) with Asterisk?
Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will