search for: dtpspain

Displaying 8 results from an estimated 8 matches for "dtpspain".

2003 Sep 19
1
SIP registration between *'s
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
2003 May 07
2
Question about STREAM FILE.
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
2003 Jun 18
3
Temporized AGI Scripts.
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
2003 Jul 10
0
Problem with meetme.
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
2003 Sep 16
0
No correct IP in RTP media stream
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
2003 Sep 17
0
Aleatori PSTN number with SIP.
Hi everybody, Now I'm 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:
2003 Jun 25
1
Problems with music during tones of dial.
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
2003 Jul 02
2
Problems with musiconhold
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