similar to: calling peer from server

Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations <sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it

Hi all, I think i understand the problem, actually I have two asterisk server. In the extension.conf file on one server I have added exten => 3923903,1,GOTO(s,1,3923903.conf) which reads the corresponding conf file when ever the extension no. through PSTN is called and learns the location of inbound.php which contains the IVR script to be executed. Now what i want is that through this

Thanks a lot Danny. I have done the part of playing a file by creating a context in my dialplan. Now I am puzzled as i wish to store the DTMF inputs done by the users who is listening to the playback. I found there are ways, but some specific way by which it is not stored in file but conveyed directly to the asterisk server. When the call landed up on the softphone, i pressed keys the

HI list-users, Greetings!! I have been using call file, i playback my file using * application:playback* and once the playback is over the call is disconnected. Is there any way it can wait and also record the dtmf inputs once the playback is over. Thanks in advace Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad contact at 9793905858 email:

Actually my problem is not related to sip.conf and extensions.conf. I have used only standard files from martin pdf which are given as example. I am able to call some system connected over LAN, when each has a softphone and are registered on a asterisk server. But now what i want is instead of using the softphone I write a function in my file which will be executed when the call is placed. In that

Thanx for the reply. The reason i wrote echo is, i was running the script on the command line, and i wanted to see if the particular function is running. Just like i do debugging in c++. I didn't know that it sends messages to asterisk. But again i was not able to see any message on asterisk server. One thing, it must be possible to run the php script file on command line, since it was the

Hello everyone, I have written simple LLVM passes, but I cannot able to write a MachineFunctionPass pass. I am following the steps form the following link but it is not working: http://www.gabriel.urdhr.fr/2014/09/26/adding-a-llvm-pass/ Please share the sample MachineFunctionPass code or steps to follow to write MachineFunctionPass. Thanks, Bala -- Thanks, Bala IIITA Allahabad --------------

Hi All, I wanted to collect the register usage stats for the program assigned by compiler(in other words "how many time each register will be accessed by the program when it will execute"). How to write LLVM pass which will analyse the code after LLVM global register allocation pass and output the stats. Regards, -Bala -- Thanks, Bala IIITA Allahabad -------------- next part

First you should learn how to write a pass and then a MachineFunctionPass. >From LLVM website, you can refer to http://llvm.org/docs/WritingAnLLVMPass.html And for outside pass, you can refer to http://adriansampson.net/blog/llvm.html It's a great article. And the author put the source code on Github <https://github.com/sampsyo/llvm-pass-skeleton>. 2016-03-01 14:36 GMT+08:00

Hello All, I am building llvm-clang cross tool chain for arm target, after successful build for Linux host now I am now building toolchain for *arm target*. However, I am getting below error message which seems to be confusing and not getting any idea how to resolve this issue. *Command :* #CC='clang' CXX='clang++' cmake -G Ninja /home/iiita/crossbuild/llvm

Hello All, I am building llvm-clang tool chain for *arm target* by following the steps from http://clang.llvm.org/docs/CrossCompilation.html *However when I tried following cmake command for build tool chain it gives the error message :* #CC='clang' CXX='clang++' cmake -G Ninja /home/iiita/llvm-exp/llvm -DCMAKE_CROSSCOMPILING=True

Hi, i'm using an elesign voip gateway esc1700 to call to one iax sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when I make the call using the esc1700 the communication is dropped, this is the log portion: Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Call accepted by 200.93.220.21 (format ulaw) Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Format for call is ulaw Feb

Hi Is it possible to receive video calls using Asterisk and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of

Hi Does anyone have experience with Synway cards like SHD-240D-CT/PCI with asterisk and SynAst driver ? Are they any good ? Do they really run on Asterisk ? Thanks. Anita Hall, Simmortel Voice www.simmortel.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100913/c97492c4/attachment.htm

Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Asterisk or Freeswitch ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does

Hi Does Sangoma 8-port card A108 support PCIe version 2.0 ? The card is here http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm Is this a good idea ? Do you have any experience

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! Making some changes in extensions.conf to test the incoming calls so that these are derived to a SIP extension, I found something that draws attention to me: if I test calling to my PSTN line from a mobile phone, when take the call from the SIP extension (softphone), if the mobile phone releases the call, sofphone do it too without problems,

I do part time consulting work. I need to setup an asterisk system to allow me to record both inbound and outbound calls to clients. I have one client that is just a PITA. The client has changed their mind three times so far and we are at step one. I have a spare slackware box and a seperate phone line for the consulting work. I have MCI Neighorhood as my carrier. What I need to know is: 1.

Hello everyone. I recently boght a Nokia BH900 headset and USB bluetooth dongle and I'd like to use them to make calls from a sofphone. I managed to this with boxe XTen-Lite and the Zoiper - but they both see the device as a simple sound card through the BlueSoleil drivers. While this is allmost usable, the headphone seems to be kept in "transmission" mode all the time and I get

Anyone know?a?good?IAX2?softphone?for?Windows that has?g729?and?it is?free? att Eduardo -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110418/024f95c7/attachment.htm>