search for: shabanip

I see my own number(or remote called num) instead of caller id on incoming calls on my BT-102. but on Xlite everyything is OK. I'm using * latest CVS. - shabanip

Max TE410P card on an Asterisk Hello, Does anybody know the max number of TE410P/TE405P card we can put in an asterisk box? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/f220d3ae/attachment.htm

I get this message at CLI. what does it mean? - shabanip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040925/dbcd8d80/attachment.htm

...o talk to leave a message or talk to another person instead. now in asterisk, it seems to me that after you transfer a call to an extension, there's no way to have the caller transfered back to yourself if the called extension doesn't answer or if it's busy. is this correct? thanks, - shabanip

Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. Any Ideas? Does this work? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

...o talk to leave a message or talk to another person instead. now in asterisk, it seems to me that after you transfer a call to an extension, there's no way to have the caller transfered back to yourself if the called extension doesn't answer or if it's busy. is this correct? thanks, - shabanip

Hi, We're evaluating asterisk, somebody has measured maximum asterisk load (simultaneus calls, calls per seconds...)? Are there any stimation? Thx. Best regards. .G

I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based solution to establish a video connection with * and sip protocol? ----- Original Message ----- > Hi, > > > -----Original Message----- &...

...ows Messenger with asterisk? All documents around (google - wiki - bugs.digium.com) say that asterisk supports windows messenger with video but i have no succes yet! I can establish connection with audio but no video yet. I've used a range of windows messengers from version 4.7 to 5.0.0482. - shabanip

Am I doing something wrong? I can't get this dial command to timeout.... Dial(Zap/g1/xxxxxxx,20) -- Gary White admin@netpathway.com Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355

hello, yesterday the uniden uip200 phone was recommended to someone. i am looking for an alternative to grandstream bt-100 because i can not do a supervised tranfer with it. here my questions: 1) does the uip200 support supervised transfers? 2) can i buy the phones in europe, especially in germany? thanks in advance, jan goericke

We have set up an IP telephoney system hosted by Asterisk and its working pretty well. We primarily use SIP and hardware IP phones. We have the ability to transfer calls to another SIP phone using either the "Transfer" button on the phone (these phones are Grandstream BudgeTone 100s) or using the "#" key (the T/t flag must be set in the Dial command in asterisk for

Can anyone provide me with a step by step on how to set up Windows Messenger on a Windows XP Pro box as a SIP client with asterisk? I'm interested in doing various tests of my asterisk server from the Windows perspective of the world. In the alternative if someone could provide information on another Windows based fully functional easy to configure iax or SIP client that would suffice as

I've already installed * 0.9.0. All calls from my GS phone has no ringing sound but the other end rings! I also checked this with my CVS archive. The problem exists in CVSs from (Mar 6) up to now. and I have no problem with my Xten softphone when calling with it. anybody can help? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm using version 1.9.1 build 3908 - next problem is that the text messages won't reach by another firefly client