search for: stepaniuk

Displaying 20 results from an estimated 29 matches for "stepaniuk".

2009 Oct 18
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:
2009 Sep 07
Echo and Playtones not working on SIP after upgrade
..., moh, and everything else we are using works just fine. We are using Asterisk (on debian stable), SIP channels with both grandstream and soft phones. Everything on the same network segment. Codec does not seem to affect this behavior (tried them all) Any clues? Thanks! -- Iv?n Stepaniuk Alba Fot?nica S.L.
2009 Oct 05
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 Dec 09
Problem with Asterisk and SPA-3000
...ension (and transfers the PSTN leg 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.
2009 Sep 10
Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Oct 08
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 Oct 11
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
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 14
G.729 for Asterisk
Hello I have a confusion relating to G.729 codec. I know how to install where to get license but i really don't know why we need it? Why people use G.729 codec with asterisk? look all functionality can be done with out it ie calling from sip to iax protocol and sip/ iax to E1, then why we need this?? regards Adam -------------- next part -------------- An HTML attachment was
2009 Sep 15
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( OR its Digium ( ) plz note i dont want to degrade any company... But
2009 Sep 29
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 Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger:
2009 Sep 30
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 01
DTMF problems during a message play
I'm using the latest asterisk- and no zaptel trunks used, all SIP. I have one user that is having problems once he connects to asterisk. He's dialing from his home phone (pstn) to a Vitelity DID (SIP Trunk) which goes to my asterisk IVR. If he presses a dtmf during any message, the press is ignored unless the press was a #, 0 or *. Otherwise, he needs to wait for the
2009 Oct 02
Problem with inbound calls - asterisk
Hi all, I have a new installation with asterisk but I'm unable to receive calls from a SIP trunk: [Oct 2 14:30:09] NOTICE[21554]: chan_sip.c:18523 handle_request_invite: Call from 'user001' to extension 'user001' rejected because extension not found. Are there any changes from 1.6.0 to 1.6.1 (or there is a bug)? Below my simple configuration: sip.conf
2009 Oct 10
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
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:
2009 Oct 16
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
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
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
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.