similar to: Very simple callback application needed

I am trying to turn up DID trunks with our local phone company but do not know the correct format of extensions.conf to do this.

I've just started consulting for a SME client based in Reston Virginia. They don't know it yet but they are going to need a hosted asterisk service and some DID's. Email me if you are able to provide 10 DID's in Reston (must be able to be ported away!!) and hosted Asterisk with end user configurable IVR etc. Probably only 5-8 users at the moment BUT... they'll be

Hi, I'm trying to replace function call with call to wrapper(function_name, num_args, ...), where varargs hold args of original call. Function* launch = Function::Create( TypeBuilder<int(const char*, int, ...), false>::get(context), GlobalValue::ExternalLinkage, "kernelgen_launch_", m2); { CallInst* call = dyn_cast<CallInst>(cast<Value>(I)); if

Dear useRs, Some time ago I queried the list as to an efficient way of building a function which acts as ls() but with a different default for all.names: http://tolstoy.newcastle.edu.au/R/e6/help/09/03/7588.html I have struck upon a solution which so far has performed admirably. In particular, it uses ls() and not its explicit source code, so only has a dependency on its name and the name of

Hi, I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones. When calls come in on our POTS lines, the caller id shows up like "555-555-1234 at 192.168.1.10" where 555-555-1234 is the correct phone number and 192.168.1.10 is my pbx server IP. This format does not work for redialing on outbound calls. While there may be an outbound dialing change that could be made, it

Hi Dimitry, This makes sense if you think about it from the perspective that the string you want passing must be passed at runtime, and so can't use a const char * from compile time. You need to make the string visible in the compiled image, and use that as the argument. A string is an array of 8-bit integers, so you need to create a ConstantArray. Value *v = ConstantArray::get(Context,

Greetings folks, new to this, trying to get the syntax correct for a day of week routing. exten => 345,1,Answer() exten => 345,n,GotoIfTime(10:00-17:00|tue&thu&sat|*|*?open,345,1) exten => 345,n,GotoIfTime(10:00-19:00|wed&fri|*|*?open,345,1) exten => 345,n,Playback(afterhours) exten => 345,n,Hangup() I'll get an error stating incorrect day of week

Hello, I'm experiencing a weird problem when using the VoiceMailMain application. If I use the application after dialing a Local channel, there's strange beep just after asterisk answers the call and before the first locution. The extensions.conf I'm using is: Ruido extra?o al llamar a la aplicaci?n VoiceMailMain [default] exten => _X.,1,Dial(Local/${EXTEN}@test) [test] exten

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

Hi, I've looked high and low for any changes that streaming MoH needs on Asterisk 1.4 (.21), followed NerdVittle's article about it (http://nerdvittles.com/index.php?p=92) yet nothing worked. After creating dir stream/ and touch stream.mp3, here's my musiconhold.conf [stream] mode=mp3 directory=/var/lib/asterisk/mohmp3/stream stream =>

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

Hi All, While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use it in my dialplan. I had 2 sip extensions 555 and 666 and I called from 555 to 666, but unfortunately no value for ${RTPAUDIOQOS} appeared on Asterisk CLI. Would you please let me know what is wrong with my dialplan and/or what else should be done to get the value of ${RTPAUDIOQOS}? Following is my dialplan context

Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get "Push message cannot be displayed" back from the Polycom phone, and all I am sending is the Polycom example : <PolycomIPPhone> <Data priority=?critical?> <h1> Fire Drill at 2pm </h1>

Using Digium fax I've tried a simple dialplan: '8447' => 1. Answer() [pbx_config] 2. Set(CALLERID(num)=xxxyyyzzzz) [pbx_config] 3. Dial(DAHDI/g0/1bbbcccdddd,,G(send)) [pbx_config] [send] 4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config] 5. HangUp() But I doesn't work. It executes

Hi List, Does anyone have experiences to relate on the various Asterisk-based PBX appliances out there? Like the Aastra 160, Digium S844i, etc. Do the Epygi Quadro and Grandstream GXE also use Asterisk? Thanks, Femi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11

Hi I have a strange behaviour; perhaps someone who had a similar issue can help. I have an Asterisk-1.4.21.2 connected via sip trunk to a Cisco Call-Manager 6.1 cluster. Two phones/users from the Cisco environment call extensions on the Asterisk. Phone 1 / Call 1 is parked on the asterisk using: exten => xyz,1,Answer() exten => xyz,n,Set(PARKEXTENSION=555) exten => xyz,n,Park() Phone

Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090802/abb21767/attachment.htm