[Asterisk-Users] Problem with disable call transfer
Hi, recently I have tested my asterisk and I discovered that asterisk transfer call even I not use option t,T in cmd dial. So it is problem with right charge for call transfer. Situation: 1) Somebody A calling to asterisk user B 2) Asterisk user transfer call to somebody C 3) A and B talking but I haven`t got cdr that B calling C. Anyone know how I can disable call transfer or receive right cdr for B. I have asterisk 1.0.7 on debian. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with configuration of Quintum AX with Asterisk
Hi. I'm trying to configurate Quintum AX to work with Asterisk SIP egister/Proxy server and my problem is that only the first user account get logged in and only that user is able to make call correctly. It seems to be a problem with authorization - I have noticed no Proxy-Authorization information in SIP INVITE, ACK, CANCEL messages. I have also noticed that when I remove secret from Asterisk sip.conf file (no authorization required) other users (accounts) can make a call but no media are sent. Do you know reasons of this problem and can you help me resolving it. Michał Misiak -- Have a nice day! phone: (+48 22) 4330419 mobile: (+48) 888 395 336 e-mail: [EMAIL PROTECTED] homepage: www.michalmisiak.prv.pl ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with configuration of Quintum AX with Asterisk
Hi. I'm trying to configurate Quintum AX to work with Asterisk SIP egister/Proxy server and my problem is that only the first user account get logged in and only that user is able to make call correctly. It seems to be a problem with authorization - I have noticed no Proxy-Authorization information in SIP INVITE, ACK, CANCEL messages. I have also noticed that when I remove secret from Asterisk sip.conf file (no authorization required) other users (accounts) can make a call but no media are sent. Do you know reasons of this problem and can you help me resolving it. Michał Misiak -- Have a nice day! phone: (+48 22) 4330419 mobile: (+48) 888 395 336 e-mail: [EMAIL PROTECTED] homepage: www.michalmisiak.prv.pl ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with monitor meetme
Hi,
I tried to use Monitor(wav,filename) function in dialplan to
record Meetme conference. When I monitored on IAX2 or SIP channels in that
conference It recorded all audio (in and out) but when I monitored on ZAP
channels I could hear only IN audio and piece of OUT audio (announcement get
pin and than nothing).
Anyone knows why this so happens??? I have asterisk 1.0.7
(debian package) and DIGIUM TE410P card.
I attached my configuration file.
---
[konf]
exten = 800,1,Monitor(wav,conf-201-${TIMESTAMP})
exten = 800,2,MeetMe(200|p)
exten = 800,3,StopMonitor()
---
Michał
Misiak
--
Have a nice day!
phone: (+48 22) 4330419
mobile: (+48) 888 395 336
e-mail: [EMAIL PROTECTED]
homepage: www.michalmisiak.prv.pl
___
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[Asterisk-Users] Problem with monitor application meetme
Hi,
I tried to use Monitor(wav,filename) function in dialplan to
record Meetme conference. When I monitored on IAX2 or SIP channels in that
conference It recorded all audio (in and out) but when I monitored on ZAP
channels I could hear only IN audio and piece of OUT audio (announcement get
pin and than nothing).
Anyone knows why this so happens??? I have asterisk 1.0.7
(debian package) and DIGIUM TE410P card.
I attached my configuration file.
---
[konf]
exten = 800,1,Monitor(wav,conf-201-${TIMESTAMP})
exten = 800,2,MeetMe(200|p)
exten = 800,3,StopMonitor()
---
Michał
Misiak
--
Have a nice day!
phone: (+48 22) 4330419
mobile: (+48) 888 395 336
e-mail: [EMAIL PROTECTED]
homepage: www.michalmisiak.prv.pl
___
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Problem with meetme monitor (recording)
Hi,
I tried to use Monitor(wav,filename) function in dialplan to record Meetme
conference. When I monitored on IAX2 or SIP channels in that conference It
recorded all audio (in and out) but when I monitored on ZAP channels I could
hear only IN audio and piece of OUT audio (announcement get pin and than
nothing).
Anyone knows why this so happens??? I have asterisk 1.0.7 (debian package)
and DIGIUM TE410P card.
I attached my configuration file.
---
[konf]
exten = 800,1,Monitor(wav,conf-201-${TIMESTAMP})
exten = 800,2,MeetMe(200|p)
exten = 800,3,StopMonitor()
---
Michał Misiak
--
Have a nice day!
phone: (+48 22) 4330419
mobile: (+48) 888 395 336
e-mail: [EMAIL PROTECTED]
homepage: www.michalmisiak.prv.pl
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
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