similar to: Mutipoint Conferencing?

What is the recommend Digium Card for a PRI in NA ? I want to interface a Asterisk Server to a Samsung iDCS System, and have available T1 w/DNIS, or a PRI w/DID, the asterisk server would need to appear as a "Telco" provided Circuit. Slot Availability. Four PCI-Express Slots x8 (1 full-length/1 half-length/2 low-profile).

I'm having trouble getting an Asterisk server to make outbound calls on my iDCS Trunks. I have idcs station to asterisk station working I have asterisk station to idcs station working However, I am unable to get Asterisk to utilize any outbound trunks on my iDCS.... Anybody have any ideas? ________________________________________________________________ Sent via the WebMail system at

Any Recommendations on a "Good" Wireless Voip Keyset that works well with Asterisk? I would prefer one that is IAX2 as it works better behind a Nat'd Firewall.. But I am reaching out to you guys as you all would know what would work the best :-) ________________________________________________________________ Sent via the WebMail system at kotbh.net

How well does the Grandstream GXW-4104 or (8) work w/Asterisk? I would use a Cisco Switch w/FXO Ports but that would be a little "Pricy" I Can't use a Digium FXO Card, as the asterisk Server is offsite. Thanks, William Stillwell KI4SWY ________________________________________________________________ Sent via the WebMail system at kotbh.net

What are people's thoughts on asterisk 1.2.26? Any show stopping bugs? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080117/57d1002d/attachment.htm

Does anybody know of a Similar list for SwitchVoX? And would like to post to proper list if one is available. I had posted on digium forum, but have not received any responses yet. http://forums.digium.com/viewtopic.php?f=38 <http://forums.digium.com/viewtopic.php?f=38&t=77031&sid=4adb81c464701e0039d e21a300aa273f> &t=77031&sid=4adb81c464701e0039de21a300aa273f

I have several remote phones that experience a slight "call" delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? Any idea what could be causing this? Thanks, Bill. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Here is sip debug from pressing "transfer->blind->dial dest->Dial Key" (note both

I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes

For those who don't know, (as I just figured out by reading the sourcecode), that all settings for a particular "channels" must be placed before the channel => entry. Ie, Immediate=no Channel=>1-24 Immediate=yes Channel=>25-48 Immediate=no Channel=>49-72 1-24 will have immediate set to no, 25-48 yes, 49-72 no Maybe someday the config will be

Anybody know a reliable fax solution for 1.4.30 branch? I am using PikaFax on another server and works very well (about 3000 faxes a week), but it appears they no longer offer their product to open source asterisk, only for there "WARP" appliance. NOT really looking to migrate from 1.4.x to 1.6.x -------------- next part -------------- An HTML attachment was

Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

Hi there, bristuff comes with these two applications - and too little info to understand what they are for. Anyone has a clue and is willing to share it? Thanks, Philipp -= Info about application 'Autoanswer' =- [Synopsis]: Autoanswer a call [Description]: Autoanswer(exten):Used to autoanswer a call for an extension. -= Info about application 'AutoanswerLogin' =-

Im sure this has been thrown around this list 1,000 times, and Im sure its been around the net too.. But I have done everything, and cannot seem to get inward calls to be processed on my asterisk box.. First, Let me tell you what works: 1) Softphones (ZoIPer using IAX2 Protocol) Can make calls behind a Natted Firewall to the FXS Port, and it rings, and calls work full duplex. 2) Soyo IP Phone

Anybody notice log delays in this list, and very small amount of traffic? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/576a9b0e/attachment.htm>

I have asked a GSM operator in my country if he can route a number or a short code to my asterisk server via SIP (since they dont give DIDs in my country) the operator said they do not support SIP, they have no way of converting GSM calls to SIP to then send them to me. I would like to know what is needed from the operator side to do this, what kind of material is needed, or what can be done from

Hi, Having problems with getting either RxFax or FaxReceive to compile. Running Asterisk 1.4 on CentOS 5. Does anyone have the free/open source executables that you could send me? Thanks for your help! P. S.: TxFax and FaxSend would also be appreciated.

My teeth are on edge after this one. A couple of perfectly good hours of my life, and I still don't know what's going on. . . . The extensions.conf.sample that comes with the current SVN trunk has this line, in an example that shows how to use ChanIsAvail: exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail) I couldn't get this to work unless I surrounded the

My colleague left our company, then I have to manage all our phones lines and asterisk: please, apologize me because I'm 'absolute beginner' about voip/asterisk!! Well... all seems work fine; we have some queues and some agents; the "music on hold" works fine when the agent press the hold button on the phone (thomson); the agents have the 'autoanwser' flag