search for: albafotonica

Displaying 20 results from an estimated 23 matches for "albafotonica".

2009 Oct 18
4
Customising Firmware
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
2009 Dec 09
1
Problem with Asterisk and SPA-3000
...g to the new extension as normally). At the CLI there is nothing but a new incoming call from the SPA, exactly as the original call. It seems to happen with both asterisk 1.2 and 1.4, I am quite lost, Does anyone know what could be causing this problem? -- Iv?n Stepaniuk Alba Fot?nica S.L. www.albafotonica.com
2009 Oct 05
5
Networking Concept
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
2009 Oct 11
5
Call Recording and Posting
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
2009 Sep 09
2
Call getting stucked !!
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:
2009 Sep 15
3
Which is best provider for G.729
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
2009 Sep 29
2
kill sip user
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:
2009 Sep 30
1
How to finish a Meetme
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
2009 Oct 10
1
Asterisk to Asterisk access voicemail - not working
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
2009 Oct 14
2
ACD & ASR
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
2009 Oct 16
1
The City of Amsterdam has been deploying asterisk throughout the city!
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
2009 Oct 16
2
SIP to IAX to SIP
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?
2009 Nov 10
2
Gradstream Budge Tone-201
Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal
2009 Nov 10
1
Silent Dialing
Is there a way to disable ringing while dialing? Example, external users come into our IVR, and if they dial certain IVR options, these are sent off to a remote server for call handling ( Dial(SIP/extension at remoteserver) for example). It rings once, then the remote system picks up. I would like it to be more transparent to the users.
2009 Oct 08
4
No sound on voicemail from analog line
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
2009 Sep 09
2
streaming meetme conference
Hello, Our 500+ company is slowly moving away from our hosted conferencing solution to one I built a few weeks ago with Asterisk and MeetMe. When our Q3 conference call comes around, we will have the need to have approximately 300-400 users in this call. Obviously, all would be 'listen only' mode and only 1 or 2 two would be speaking as marked/admin users. Our conference hosting
2009 Oct 15
4
PSTN to SIP line ratio
Hi, I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure about PSTN incoming/outgoing line ratio for SIP users. I mean if you recall dial up internet the common line ratio is 1:10 (one line for 10 users on access server or an E1 for 300 users). Can somebody tell me what is the good ratio for incoming and outgoing analogue/ digital PSTN lines. Regards Smir
2009 Nov 03
5
Asterisk and Software Data Modem
Hello everybody I am trying to connect my asterisk to a payment equipment trough PSTN. I have a TDM400P card with an fxs module an the equipment use modem to send data! I was thinking to implement a software data modem in asterisk, but I found out that there is just faxmodem for asterisk, Is anyone here know something about software data modem working with asterisk to help out? Thanks,
2009 Sep 10
2
ASR & ACD
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
2009 Oct 08
4
Dialplan problem
Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten => 2001,1,Answer exten => 2001,n,Dial(local/3005) exten => 2001,n,Hangup exten => 3005,1,Set(__RINGTIMER=10) exten => 3005,n,Macro(exten-vm,novm,3005) exten => 3005,n,Hangup When I execute the Originate (AMI) with the argument Channel=local/2001, It rings