similar to: streaming meetme conference

Hi, Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm

Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, Anahi Ludue?a _________________________________________________________________ Descubre

Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk

Hello everybody, I have a very strange issue with a Linksys SPA-3000 (1 fxo + 1 fxs) used as PSTN gateway to asterisk in a small office. Everything works just fine, except that sometimes, and it seems that only for long incoming calls, the IVR menu appears on the middle of the call(like a three way call, call goes on with prompts playing over the parties). Dialing an extension at the prompt at

I have read more about asterisk and have succeeded in using it's app_ices function and a sample conference . I would like to learn more about lowering the latency between the Speaker on the SIPphone->Meetme Conference->ICES->Listener stream. Thank you Flash > > Let me know if there's more info you need and I'll ask my friends some > specific questions. > >

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30

I am using asterisk. I also have an access to VOIPSwitch ver 2 where I can see live calls. Many times I have seen that my calls are getting strucked and then it gets disconneected after 59 mins ( as settings are done accordingly in VOIPSwitch) What could be the reason ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'd like to know if there's an "easy way" of doing the following: SIP phone dials a custom feature code in Asterisk, call gets answered within a custom context (Answer()), anything that the caller says should be redirected/piped to an external application. Something like "monitor" except audio should be sent live. More like "app_ices" (or

Hello, I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I

Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound. What can cause that problem? Thanks in

I installed icecast-2.2.0.tar.gz and ices-2.0.1.tar.gz and referenced http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Ices. But I could not succeed to start ices-2.0.1 as follows; -- Attempting call on Local/33102@stream for 33100@stream:1 (Retry 1) -- Executing Answer("Local/33102@stream-3fa5,2", "") in new stack > Channel

Hey Guys, I am trying stream live music via icecast streaming server into a conference room, this will allow persons joining the conference to hear the music. I have been googling and i have come across a few tutorials, that give instructions as to how to get it done. But they all mention the use of a ices application module. It appears that asterisk 1.4 is not shipped with app_ices.0 by

Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090910/af1f9656/attachment.htm

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

hello I dont want to disgrace any company but i want to know from your(user)experience which one is good in case of g.729 (performace etc) is it Howler(http://www.howlertech.com/products/howlets) OR its Digium ( http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC&main_category_id=5 ) plz note i dont want to degrade any company... But to know what experience you

I have a user but I need to give that user only kill and disable all connection cut calls what is the command in the CLIC -- Bayardo S?nchez Garc?a Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server E-mail: bayardo.sanchez at gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger:

Asterisk to Asterisk voicemail not working (accessing voicemail from another asterisk). PSTN to Asterisk is working, but not between two asterisk :-( I've tried setting my asterisk dtmf to rfc2833, inband it is not working. The other Asterisk Linksys is set dtmf = auto -- Joseph

Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source & destination? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/48076c6b/attachment.htm

Hi, As you may know by now, yesterday on the Astricon the City of Amsterdam presented their large scale asterisk deployment of 20000 phones. Because they do not allow brand names to be used within the city, they call it 'IP Business Manager', but the software they use is in fact the Astium PBX, by NeoNova. Since we are very proud of this project, we have made the Astium available for

Hi all, I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs very well. On that machine I have a SIP phone. I have configured a netgear wgt634u with asterisk and a SIP phone and linked the two systems together via IAX. Audio from Ubuntu to netgear is not bad, audio from netgear to ubuntu is unintelligible. Any clues as to whether this will work? Configuration suggestions?