[Asterisk-Users] iax bracking up
Good day all We have a 64kbit 128kbit burst able line between 2 city's iax.conf is configured so that the SIP users in one office can just dial the extension of the other office. the codec is ilbc and I added trunk=yes. When you call someone on the other side its clear to you but the other side brakes up? Does someone know any better tricks or tips for this scene Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] soxmix?
Good day all
I cant get asterisk to join 2 recorded files with Monitor() and sox
I have a asterisk version 0.9.1 and a zaptel card
I installed sox 12.17.6 and copyed soxmix to /bin
I added these in my extensions.conf
exten => _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten => _0.,2,Monitor(wav,${CALLFILENAME},m)
exten => _0.,3,Dial(Zap/g2/${EXTEN:1})
exten => _0.,4,Congestion
Thanks Seth
but after a call ther in extension-time|m-in.wav and
extension-time|m-out.wav
Please help
Altus
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[Asterisk-Users] record all calls
Good day all I want to record all call on my zap&vpb&internal channels. I had a look on the net and and found astGUIclient,I want something easy and simple that will save it in date/user files. Please advice Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] eyebeam video
got it working with my webcam just as is use the wizard to detect and add video for sip [EMAIL PROTECTED] wrote: Does anybody have the miracle setting required to get the video portion of eyebeam from Xten to actually work. All I get is blank screen. Or is it the product itself that is not quite ready for prime time yet… MarcG. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call another server
Good day all We have a few of asterisk server running sip and zaptel/voicetronix cards. All the server works 100 for calling in and out of the pstn and internally Some of the server are owned by the same company so I configured IAX so they can call different extensions at different branches Now,how do I configure asterisk so that if someone calls there public ip asterisk will answer as if its coming in from the pstn I.o.w I want to phone a IP instead of a phone number Can this be done? Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip+iax+firewall
Good day all We have a 2 servers setup with asterisk, both servers got 2 network cards, public and private ips I have setup SIP on both side to listen on the private ip and iax to listen on the public ip and made the so that people can phone across. No I want o install a firewall doing MASQUERADING and only allow port 5036(iax) connection with a source of each others ip. Will this work?Or will asterisk have a problem connecting to the other users now that its being masqueraded? Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] still riniging problem
Good day all I have a problem with the new asterisk version.My setup,extensions.conf is like this: If someone call in from the outside to the PSTN,asterisk wait 8s and then forwards the call to the sip user,the operator and she then transfer calls My problem is,for the first 8s the ringing sound is normal but as soon as the call goes to the sipuser(operator) the ringing gets very fast and and some people thinks its a busy signal Previous versions worked Please help Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] record
Good day all How do I record a call on a vpb channel? Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming ringsound
Good day all We have a voicetronix openline4 card If someone calls from the outside and asterisk answers the phone and diverts it to the users the ringing sound is very fast? Is there a way you can change intervals? Please Help Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] caller Id
Good day all MY caller Id does not.I have asked for caller Id on the line and in my vpb.conf I have callerid=on This is what I have in extensions.conf exten => s,1,Wait,2 ; Wait a second, just for fun exten => s,2,Answer ; Answer the line exten => s,3,DigitTimeout,3 ; Set Digit Timeout to 5 seconds exten => s,4,ResponseTimeout,4 ; Set Response Timeout to 10 seconds exten => s,5,BackGround(bi) Am I missing something,this is a voicetronix openline4 card Please Help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] divert if not here
What I want the users to do is something like pressing *333# and this will enable divert [EMAIL PROTECTED] wrote: On Tue, 12 Oct 2004, Altus Syman wrote: Good day all We have a pbx system running sip and sipphone(Bughtone) My question is.If a user is not at their desk,how do I tell it if a call comes in it should direct it to someone else Do I need a different phone for this?The only other way is that they have to switch it off and in my dialplan on stem 2 I will have to say go to that user? Please give advice on this Thaks Altus Hi Altus, Normally you would have your Dial() call just ring the phone for a limited time. For example Dial(SIP/12345,20) would ring the phone for 20 seconds. Asterisk will then carry on with the next step and you can do something else. So: exten => 6001,1,Dial(SIP/6001,20) exten => 6001,2,Voicemail(u6001) Will ring the phone for 20 seconds, then send the call to Voicemail, playing the "unavailable" message. There are lots of other examples on http://www.voip-info.org/ I am sure. Regards, Steve Davies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *8 on voicetronix OS12
Sure got it working yesterday In sip.conf add these 2 lines to each user callgroup=1 pickupgroup=1 and in extensions.conf add exten => *8,1,PickUp(1) In short and as I understand You put each user in pickup group 1 and Picjup(1) tells to pickup group (1) Remember restart Prof. Marcelo Kruk wrote: Can somebody help me on setup the Asterisk feauture *8 on voicetronix hardware? Thanks Prof. Marcelo Kruk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] remote pickup
Good day all We have a voicetronix openline4card in a new system On our old system we had a zaptel card and if a user want to pickup a remote call he just go *8 How do I do this with a voictronix card? Please Help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] billing???
So what do you suggest Kanuri, Seshu (Company IT) wrote: my bad, none of those applications really work for us and they are too rudimentary of any value. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Altus Syman Sent: Tuesday, October 12, 2004 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] billing??? my bad,wiil have a look Flynn wrote: On 10/12/2004, "Altus Syman" <[EMAIL PROTECTED]> wrote: Good day all This is most likely a new topic but I'm searching for some billing software for asterisk ,free, if can I looked at http://www.voip-info.org/wiki-Asterisk+billing and the whole of the cdr thing in my /var/log/asterisk If you look at that very same page (http://www.voip-info.org/wiki-Asterisk+billing), the section "See also" does outline several useful links: * Open Source Billing Systems * CDR mediation * Asterisk addon rate-engine: An alternative billing solution from Trollphone? * Asterisk CDR Areski GUI - PHP GUI for MySQL/PgSQL CDRs * Asterisk CDR Jon - Perl GUI for CSV CDRs * Simple Perl script for a basic calling card application ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] billing???
my bad,wiil have a look Flynn wrote: On 10/12/2004, "Altus Syman" <[EMAIL PROTECTED]> wrote: Good day all This is most likely a new topic but I'm searching for some billing software for asterisk ,free, if can I looked at http://www.voip-info.org/wiki-Asterisk+billing and the whole of the cdr thing in my /var/log/asterisk If you look at that very same page (http://www.voip-info.org/wiki-Asterisk+billing), the section "See also" does outline several useful links: * Open Source Billing Systems * CDR mediation * Asterisk addon rate-engine: An alternative billing solution from Trollphone? * Asterisk CDR Areski GUI - PHP GUI for MySQL/PgSQL CDRs * Asterisk CDR Jon - Perl GUI for CSV CDRs * Simple Perl script for a basic calling card application ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] billing???
Good day all This is most likely a new topic but I'm searching for some billing software for asterisk ,free, if can I looked at http://www.voip-info.org/wiki-Asterisk+billing and the whole of the cdr thing in my /var/log/asterisk Its all all comma separated and I'm sure It cant be that hard to write your own billing software? I dont have the time or the skill for this Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * box hangs after a couple of days...
Got the same problem on a linux redhat 7.3 box But everything hangs Its also runs mail You can ping it but cant connect on any port,not eve the gui Michael George wrote: What is the configuration of the TDM400? Port 1 - Port 2 - Port 3 - Port 4 - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] divert if not here
Good day all We have a pbx system running sip and sipphone(Bughtone) My question is.If a user is not at their desk,how do I tell it if a call comes in it should direct it to someone else Do I need a different phone for this?The only other way is that they have to switch it off and in my dialplan on stem 2 I will have to say go to that user? Please give advice on this Thaks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] eyebeam
Good day all I have got a copy of eyebeam but the quality is very very bad.If I talk it sounds to fast and as if I had a nice sniff of helium Anyone else have this ptoblem Yhaks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] caller id?
Good day all I'm totally lost with this caller id,so can someone please help me We are using a openline 4 card so in my vpb.conf I added callerid = on And we are using sip as protocol so in sip.conf. No each time a call comes in from the outside I dont se the number where its coming from on my phone? I think I'm missing something?Please help Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
