similar to: Full SIP dial string

Hi, Here is my scenario. I have a SIP call between two SIP endpoints. A calls B. During the ringing, B disconnects (network cable is unplugged). But A continue ringing forever (until the dial timeout) even if asterisk detects that B is disconnected with the qualify. Is there any setup or asterisk configuration I need to enable to have A close its call ? Note: when A is already talking with B,

Hi all, I need to build an application that will be an SIP server program that will run on Linux and Windows. The sip server need only some features such as be able to : - Register sip endpoints - Answer a call and play a local file - Make a dial from one channel to another. I know asterisk can be stripped to exactly fit my needs. I would like to know if there

Hi, I setup an MeetMe conference. So, the admin user calls and enter the conference in talk/listen mode. (Options : dAaxs) Then other users call the same conference and enters in muted mode (options: dlmx) How can the admin user decide, when he is ready to let everybody speaks ? I didn't find such option in the admin menu. Thanks -------------- next part -------------- An HTML

Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that reflects the call from which the order originated. The typical/sample CDR table doesn't have a primary key. I can add an auto-generated PK, but the CDR is not written until the

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

Hi, I noticed that when dial terminates it does not return to the dialplan, and therefore can not execute any entry after Dial(). Is there any trick to overcome this limitation ? How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if I can not execute anything after Dial()? I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls end

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

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

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:

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.

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

Hi All I'm having some serious kernel panic while using digium cards. It may be related to IRQ shared. Can this cause a lot of drop call and bad voice quality ? Do you guys know if there is a way I can assign one IRQ for each digium card ? Thanks a lot. Here is the output of /var/log/syslog kernel: [ 3821.982893] Uhhuh. NMI received for unknown reason 20. kernel: [

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

Hello, Q/ Is it possible to create a DBMS connection automatically on startup of R? (Making sure of course that the db server has been started...) I am running MySQL on Mac OS X 10.4.2 with R2.4.1. I have tried to write a function using the RMySQL commands (below) and place them in .First of .RProfile: drv <- dbDriver("MySQL") dbcon <- dbConnect(drv, {other parameters present in

Hi, I know you can access various sip variables via 'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of the sip user - but what about variables? I have a user that has setvar=123456 in their users.conf (sip.conf if you prefer). I can read it with a 'sip show peer 201' - but that gives everything and parsing that isn't really an option. Anyone know how

Hello, We succeed to send faxes using FFA, when the files are converted to tif from PDF using gs, but it doesn't work with tif files we copy/upload directly from our PCs. We saw in the manual that the size is important, since we got the error "FAX handle 0: failed to queue document 'filename.tif'", so we set it to 1680x2285, but it's still rejected. Is there a way

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