Asterisk PBX Integration Zimlet
Asterisk PBX Integration Zimlet
[quote user="sky_09"]Need HELP...
i have installed zcs 6, zimlet for asterisk and asterisk server,,
anything goes right,, but when client A call to client B it can't happen, in GUI no error,, but
when i see in /opt/zimbra/log/mailbox.log i get error like
PWC6032: Unable to load tag handler class "com.zimbra.cs.taglib.ZimletConfig" for tag "z:zimletconfig"
i hope u will response my question, i'm sorry for my english...[/QUOTE]
i've solve my problem,, just update asterisk to ver 1.6.x.x and zcs 6.0.12,,
but i 've get new problem,
in my asterisk cli,, he said,
--Got SIP response 482 "Loop Detected" back from 192.168.3.2
>Channel SIP/101-09c402e8 was never answer
any idea,, how to solve it??
i have one question,, client use zimbra from his browser (mozilla/http) and when he call another client his use browser too.. and my question,, what voip can works over http??? because i see in my asterisk 1.6 cli, he use RTP (real time protocol)..??
what i need addons???
i hope someone can solved my problem... pleaseeeeeeeeeeeeeee.......
thanks...
i have installed zcs 6, zimlet for asterisk and asterisk server,,
anything goes right,, but when client A call to client B it can't happen, in GUI no error,, but
when i see in /opt/zimbra/log/mailbox.log i get error like
PWC6032: Unable to load tag handler class "com.zimbra.cs.taglib.ZimletConfig" for tag "z:zimletconfig"
i hope u will response my question, i'm sorry for my english...[/QUOTE]
i've solve my problem,, just update asterisk to ver 1.6.x.x and zcs 6.0.12,,
but i 've get new problem,
in my asterisk cli,, he said,
--Got SIP response 482 "Loop Detected" back from 192.168.3.2
>Channel SIP/101-09c402e8 was never answer
any idea,, how to solve it??
i have one question,, client use zimbra from his browser (mozilla/http) and when he call another client his use browser too.. and my question,, what voip can works over http??? because i see in my asterisk 1.6 cli, he use RTP (real time protocol)..??
what i need addons???
i hope someone can solved my problem... pleaseeeeeeeeeeeeeee.......
thanks...
Asterisk PBX Integration Zimlet
Sorry SIP uses RTP as part of its transport. You cannot use HTTP/S.
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- Joined: Sat Sep 13, 2014 1:59 am
Asterisk PBX Integration Zimlet
[img=http://s1.postimage.org/co5p0pxg/error.jpg] />
okey that was a problem...any idea to resolve it
okey that was a problem...any idea to resolve it
Asterisk PBX Integration Zimlet
Check the AMI username and password and that the allow statement will let your Asterisk server connect.
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- Posts: 6
- Joined: Sat Sep 13, 2014 1:59 am
Asterisk PBX Integration Zimlet
[QUOTE]Check the AMI username and password and that the allow statement will let your Asterisk server connect.[/QUOTE]
where can i check the AMI..
is it in trixbox, isn`t it...??
where can i check the AMI..
is it in trixbox, isn`t it...??
Asterisk PBX Integration Zimlet
Yes it will be in Tribox. AMI = Asterisk Manager Interface.
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- Joined: Sat Sep 13, 2014 1:59 am
Asterisk PBX Integration Zimlet
thanks a lot ubod...
i`ll try....
i`ll try....
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- Posts: 6
- Joined: Sat Sep 13, 2014 1:59 am
Asterisk PBX Integration Zimlet
hai ubod...
i`m still confused when i configure in trixbox.
can you show me, the contain of files that i have to configure.
sip.conf
extensions.conf
voicemail.conf
manager.conf
thank you.
i`m still confused when i configure in trixbox.
can you show me, the contain of files that i have to configure.
sip.conf
extensions.conf
voicemail.conf
manager.conf
thank you.
Asterisk PBX Integration Zimlet
Hello, fuadprasetyo77. I haven't configured this in a long time so I am only guessing but are you sure your permit statement in manager.conf is correct? I'm guessing that should be the address of the Zimbra server yet it is the same as your bindaddr statement which I assume is the address of your Asterisk server.
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- Posts: 6
- Joined: Sat Sep 13, 2014 1:59 am
Asterisk PBX Integration Zimlet
thanks for your help gracedman,, it helps me