search for: xisco

Hi all, Now I'm working with a E400P, and I don't now if it's possible to do the following. I want that and AGI script (Perl) recieve a call, and the user introduce the date, the time and the destination phone number (where the temporized AGI must call). Before an AGI script will call to that number in the date and time introduced by user. That's possible, and it's how can I

...using SJphone on a win2k client an * as proxy SIP and GW to PSTN. I have doing some test, but I have the following question. It's possibles to make calls to external PSTN numbers without define an extension to make the call???? I will try to explain-me better. I have done some calls like sip:xisco@A.B.C.D, where in extensions.conf there are an extension like this: exten=>xisco,1,Answer exten=>xisco,2,Dial(Zap/g1/<definened number for me>) I want to make a call like sip:<my phonenumber or other number>@213.229.160.218 without define the phone number anywhere. It is...

Hi, I don't know if it's possible to stop a STREAM FILE pushing a key. Anyone know it? Best. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030507/04d18302/attachment.htm

Hi everybody, Firstly I'm going to describe the scenario where I'm working. I use a E400P with Asterisk CVS-05/22/03-11:14:50, and I'm working with asterisk trow AGI scripts (Perl). The configuration of extension.conf is: exten =>_s,1,Answer exten =>_s,2,AGI,script.agi Inside the AGI script is call Dial application as follows: print "EXEC Dial

Hi everybody, I'm trying to SIP register between two asterisk, each one have a Public IP. Asterisk told me that Unathorizae In * one sip.conf register =>usuario1:pass1@<public_ip_2> In * two sip.conf [usuario1] type=friend username=usuario1 secret=pass1 host=<public_ip_1> dtmfmode=inband Logs in * are the followings In * one logs: Sip

...ng. NOTE: Some time ago, my ISP had another network structure, and sometimes, I got different gw in each cablemodem, and load balance worked fine. I think the problem comes in the routing cache, or expiration times, really I don''t know. I would appreciate any help. Thanks in advance. Xisco Fernandez. _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

...ng. NOTE: Some time ago, my ISP had another network structure, and sometimes, I got different gw in each cablemodem, and load balance worked fine. I think the problem comes in the routing cache, or expiration times, really I don''t know. I would appreciate any help. Thanks in advance. Xisco Fernandez. _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

Hi everybody, I'm using meetme like follows (in AGI), I'm working in Spain. print "EXEC MeetMe 10|p\n"; $res = checkresult(); I select the |p option in order the users can go out of the conference, when the users press #. All work quite fine, except when the user call from a mobile and press #, then all users are removed from the conference. Somebody have or had the same

Hi everybody, I'm trying to configure * for make SIP calls. Now I'm doing several test but I have some errors. Firstly I will describe my scenario. Client Software (Private IP 192.168.0.181, SJ Phone over Windows 2000) ---- Router Adsl (Public ip A.B.C.D, and NAPT on port 5060 to 192.168.0.181) ----- FW+Router ----- Asterisk (Public IP E.F.G.H + e400p)------ Spain ISDN I

Hi everybody, We are thinking in connect out PBX (with a new PRI card) to * (with card TE110P) thought an E1. We will have to configure the framing, coding, channels, etc...our doubt is: How must we select the signalling in * 'pri_cpe' or 'pri_net'? It's depend if our PBX card emulate to be the network side or the customer side? Thanks in advance. Regards.

Hello list, We're migrating our whole network to Linux. The process involves a period of time in which there will be Linux clients and windows clients mounting remote shares from a NT server. The access is working fine, But file locking isn't working properly. I'll expose situations tested: Cases 1) Client A: Windows using Excel Client B: Windows using Starcalc (staroffice) This

Hello list, I would like to know if there is any relation between smbclient and the server config file smb.conf. I'm working with some issues with file locking and staroffice, I'm working with a default smb.conf the one shipped with RedHat Enterprise Linux 4. And I'm wondering if I have to add any directives at smb.conf to make smbmount read those parameters and be able to

Hi evereybody, I'm trying to use musiconhold during dial tones. But I only can call earing dial tones instead of music. Now will see my configuration files. AGI File(using AGI script to EXEC DIAL) print "EXEC Dial Zap/g2/numberc||m\"; $res=checkresult(); Extension.conf exten =>_numberb,1,Answer exten =>_numberb,2,SetMusicOnHold,default exten =>_numberb,3,AGI,dial.agi