similar to: US T1 Hangup Detection

While I myself am not a MagicJack user, I'm curious as to whether anyone here has managed to heist their MagicJack account's sip credentials, and use them to terminate calls using asterisk. Apparently it's as simple as sniffing the SIP credentials. If so, said person would enjoy unlimited termination for $20 year while retaining the flexibility of setting their CallerID to a

Job Description: Asterisk MySQL Support Engineer Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD

This is a test, please disregard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050111/3b3612cb/attachment.htm

List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n

Hi, What does retransmitting NAT means? I have a client that uses SPA 942, and his phone sometimes cannot be called. i did a sip sebug and i keep on seeing retransmitting NAT. on the realtime it shows that it is registered, so when i try to call it , asterisk thinks it is still online so it tries to reach it instead of saying it's unavailable, [Oct 9 11:10:33] -- Called 103100 it

Dear all, I've implemented an Asterisk 1.4 with SIP service for voip and video. So I can establish a voip + video connection *one-to-one* only....it works OK. But I'd like to implement a videoconference *one-to-many* in order to intercommunicate many clients, is it possible with Asterisk 1.4 ??? (multicast is better than brodcast in this situation of course) Thanks a lot, Alejandro

Hello all Asterisk Users, This is my first post here. We are in a process of moving Dialogic 240/JCT-T1 from old voicemail server to Asterisk box. Which card drivers do we need? Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Only source proves that this card work with * http://lists.digium.com/pipermail/asterisk-dev/2003-April/000244.html

How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080722/f9612f97/attachment.htm

Hi, I am using asterisk 1.4.18. I am using it for inbound only call center. The SIP phones are X-Lite. Right now when a call is proxied by Asterisk to X-Lite the agent only sees asterisk written on its CLI screen. I want the agents to be able to view the callees number. Is there any way to make this happen. Regards Syed Nasruddin -------------- next part -------------- An HTML

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 all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

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

Howdy, I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely

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 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