search for: voipinfo

...them with each of the Toshiba PBX s. This is to give IP Phones/soft phones to the users and to route these VOIP calls through the PBX to POTS. What are the Digium cards I should use in each of these cases and How should I integrate Asterisk with above systems. I read the article in http://www.voipinfo.org/wiki/index.php?page=Asterisk-ToshibaStrata and not sure whether that scenario fits mine. Also it is bit confusing to identify what Digium cards should I need for my cases. Any help is highly appreciated. Thanks, Indika.

...one. >> >> I wonder how I can change the timing source. >> > > In one version (and I can’t recall which) asterisk moved to an > internal timing system, to avoid the hardware need. > > There should be quite a lot of discussion of it in the archives or > perhaps voipinfo > > I don’t know if you can slow the VM processor speed? I am guessing it > is counting something much faster than it used to > > Cheers Duncan > > *CLI> module show like timing Module Description                              Use Count  Status Support Level res_timing_dahd...

Markus Weiler <markus_weiler at mailworks.org> schrieb: Hi > from voipinfo... > > If an Asterisk command specifies a sound file in a*subdirectory*, > Asterisk looks in that subdirectory for the language subdirectory. For > example, theSayDigits > <http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigits>command may > play the sound file &quot...

Hi All I am trying to setup : PSTN E1 ---> Asterisk------>Legacy PBX------->Legacy Analog extensions. I've followed steps using : http://www.voipinfo.org/wiki/view/Asterisk-Panasonic i get the green light (sync) on both the 2nd span of digium TE420P (that is cnnected to the legacy pbx pri card) and the pri card of the legacy pbx. but when i try to make a call to asterisk so that it can send the call to the legacy pbx using Dial command - it exi...

I want to build my own calling card system on Asterisk. I looked at this page - http://www.voipinfo.org/wiki/view/CallingCard+Applications and it has listed some applications that I thought could help speed up the development process though the link down the bottom doesn't work. Does anyone know of any AGI etc applications to build a Calling Card system on Asterisk? Michael

I have an * installation which will refuse incoming callers once a max (5 callers) is reached. Caller 6 and up should be notified of congestion...without network load on my trunk. How would I do this? The voipinfo wiki shows playing a congestion tone to the caller, but that seems stupid since I'm consuming bandwidth to send a tone. I also tried just responding with the congestion command, but the 6th+ call just hears a hangup when calling in. Can someone explain how this should be done? Thanks, MD...

Hi Everyone, I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is specific to 1.6. I am wondering if the developers cared to write about the new events that are spit out in Asterisk 1.8 somewhere on the web? I checked the tar ball for asterisk 1.8 and documentation doesn't include this event: *Event: Unlink* Privilege: call,all Channel...

I just want to pdf and email faxes coming in over pstn on a TDM400P. Outgoing faxes would just go out over pstn, not through asterisk. All the voipinfo , etc, howto's are quite complicated. And most use third party apps like Hylafax. I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm now using 1.4.22, but I'd go to 1.6 if it made this easier. But I've found no docs or sample configs for either 1.4 or 1.6. In...

Hi, 1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ? I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies. 2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read : "Also, try using: t38_udptl=yes t38pt_rtp=no t38pt_tcp=no ... in the general section of the sip.conf and under the VoIP provider account as well as the fax account. " But above, you can read "[general] t38pt_udptl = yes " Has t...

I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling are all our customers, etc. so I don't need to do any do-not-call checking. Just

Hello, A client has asked for Two B channel Transfer capability (known as TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG Path Replacement) in a new Asterisk system and so I researched the capability and came up with quite a few gaps in documentation.

Hi again I'd like to configured my Asterisk to use german sounds for the "Say"-commands... I installed the sounds-files and I tried them with "Playback(de/demo-echodone)" and it works. Now I tried to add an extension to say the current time: exten => 24,1,Verbose(2,Time asked by ${CALLERID(num)}) Exten => 24,n,Set(CHANNEL(language)=de) Exten =>

Hi, from voipinfo... If an Asterisk command specifies a sound file in a*subdirectory*, Asterisk looks in that subdirectory for the language subdirectory. For example, theSayDigits <http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigits>command may play the sound file "digits/6". Asterisk will...

...sterisk server behind a different NAT. I have a SER server (with rtpproxy installed) on a public IP adress. I've opened ports with static NAT to * and the Cisco. Without using SER, I can register the phone to *, I can complete calls, I just can't move audio. Reading the archives and voipinfo seems to indicate that I can use SER and rtpproxy as a "middleman" for the two things I want to talk to each other. The problem is that no one ever seems to mention the simplest part of that config: HOW? How do I make the Cisco talk to the * server through the SER server? Do I regis...

HI ALL; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/3408bfd5/attachment.htm

...%20channel%20transfer As for actually using this feature, you apparently need to add the following lines to the zapata.conf section that you want to be able to use 2BCT: facilityenable = yes transfer=yes To execute the transfer, you need to use the "Transfer" cmd within Asterisk: http://voipinfo.org/wiki/view/Asterisk+cmd+Transfer And according to this post, you can only do 2BCT transfers if the first call is inbound: http://www.mail-archive.com/asterisk-dev at lists.digium.com/msg25131.html Does 2BCT work with DMS100 and 5ESS right now? Are there people using this in production right...

Howdy, Excuse the neophyte questions... I was wondering: (1) what's involved in setting up a call with encrypted media (I'm on a cable network and don't want my calls snooped); (2) is there a cheat-sheet for configuring Sipura handsets/hardphones like the SPA-942, and in particular for message-waiting indicator and shared-line appearances? (3) my PSTN service provider that I

Hello All: I saw there are app_fax and app_chanspy modules in 1.6.2.6, but there is NO sample configure file for them. Is anybody know how to use them, or where is the documentation for them? Thanks -- Refer to: http://www.microsuncn.com Best Regards Alan Zheng -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Everyone, I am wondering why documentation of some of the vital parts of Asterisk is hosted on voipinfo.org (unreliable is some parts) and not on asterisk.org? For example the list of AMI events are not well documented and one has to guess which version supports which event. The documentation file for AMI for Asterisk 1.4 is really only a startup guide and it doesn't even provide a full list avai...

Duncan: You may have it right-I took one phone and set the ring time to 60 seconds. I now get about 4 rings on that one. I wonder how I can change the timing source. Thomas M. Peters | Sr. Systems Administrator | tpeters at mcts.org<mailto:tpeters at mcts.org> Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org<mailto:helpdesk at mcts.org> Milwaukee County Transit System