Asterisk Wiki

Interested in talking about Mash-up's? This is the place.
marcmac
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Postby marcmac » Thu Aug 31, 2006 9:47 am

[quote user="henrik_b"]still doesn't work. It still uses the 192.168.. IP instead of the IP I edited in conf.xml. Ok, I know how to handle this. Simply adjust the IP in the .zip -file before deploying it.


Doesn't work either, but its another problem with my Debian-install.

OK. doesn't help either. Asterisk calls the other phone, but my desk-phone doesn't ring. Looking on the line with ethereal it seems like Zimbra is trying to transfer the call using a REFER-request but asterisk doesn't recognize that properly.

How are the extensions defined in your asterisk? Perhaps we have to search the problem there? Could you post the relevant part of your extensions.conf? Any special settings in sip.conf?
thanks Henrik[/QUOTE]
Ok - I've provided a debug option, and told you where it outputs the debug info. So, please post some of the debug info.


marcmac
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Postby marcmac » Thu Aug 31, 2006 9:49 am

[quote user="sakilaine"]Hello,
is necessary it to insert something of specific in the dialplan??
The URL following in file "com_zimbra_asterisk.xml" correspond to what??



http://192.168.1.254">

http://www.zimbra.comhttp://http://www.zimbra.com>


And the http://www.zimbra.com ???
The URL full it's an application in to ZIMBRA ??
Thank's

Sakilaine


I
have no idea why I included the referrer in there, I don't think it's used by the asterisk management portal. Having the target hardcoded is lame - I'll try to pull that into the config.xml in my next rev.
BTW, folks - I'm assuming you're on some version of 4.0 for this - I don't think it will go anywhere on 3.1.x
marcmac
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Postby marcmac » Thu Aug 31, 2006 9:51 am

[quote user="1739miguel"]It's full workin' here! :)
Thank you very much marcmac![/QUOTE]
Cool! Any changes required on your server, or any tips for the crowd? I've only got one asterisk to play with, and it's in production, so I can't do too much fiddling with it; somewhat limits my troubleshooting options.
One thing I did notice, tho - if you're I don't think you can connect a call to the same extension/username you're using to authenticate.
marcmac
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Postby marcmac » Thu Aug 31, 2006 9:53 am

Couple of things I'd like to change:
- code cleanup (it's a mess) - not sure how much I'll be able to do, since I don't really do much java programming

- support NAT - I think I'll need two config items, the public and private IP of the server, but I haven't played with it much yet.

- Figure out the MWI - I think you can get that via SIP

- Address book functionality - add a config for your directory on the server.
1739miguel
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Postby 1739miguel » Fri Sep 01, 2006 10:48 am

Well, I didn't take any special change neither asterisk or zimbra sides.. Just rebuilt the com_zimbra_asterisk.zip you post here, including the 2 jar files and changed the config_template.xml:
my.asteriskbox.ip

my.zimbrabox.ip
and com_zimbra_asterisk.xml:


restart">http://my.asteriskbox.ip">
restart
tomcat, got it working, drag & drop contacts working, everything.. BTW, I can connect a call if I use the same extension/username used to authenticate...
One cool thing would be the possibilty to call a contact this way: Imagine.. you're reading an email, when you right click on the "Sent by:" name it would appear a new option "Call with asterisk" or something.. This would be very useful in my opinion..
Thanks for the your good work!

[quote user="marcmac"]Cool! Any changes required on your server, or any tips for the crowd? I've only got one asterisk to play with, and it's in production, so I can't do too much fiddling with it; somewhat limits my troubleshooting options.
One thing I did notice, tho - if you're I don't think you can connect a call to the same extension/username you're using to authenticate.[/QUOTE]
marcmac
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Postby marcmac » Fri Sep 01, 2006 12:47 pm

Thanks for the details.
You're right, adding it to the right click menu would be sweet - not sure if the zimlet framework supports that.
8131crash
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Postby 8131crash » Fri Sep 01, 2006 5:41 pm

I've made the changes listed above and still have the same issue. Attached is the sip debug info.

sip_debug.txt

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Postby marcmac » Fri Sep 01, 2006 7:10 pm

[quote user="8131crash"]I've made the changes listed above and still have the same issue. Attached is the sip debug info.[/QUOTE]

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP MY_ZIMBRA_IP:44732;branch=z9hG4bK839a023416a08000110f141709d2de5e;received=MY_ZIMBRA_IP

From: ;tag=Zimbra36641

To: ;tag=as1c6d7d7d

Call-ID: 25574aeb47d076f5cfdb239c91cb6af1@MY_ZIMBRA_IP

CSeq: 2 INVITE

User-Agent: Asterisk PBX

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY

Contact:

Content-Type: application/sdp

Content-Length: 229
21.1.5 183 Session Progress
The 183 (Session Progress) response is used to convey information

about the progress of the call that is not otherwise classified. The

Reason-Phrase, header fields, or message body MAY be used to convey

more details about the call progress.
Not familiar with the 183 response, but that's what's breaking it. There are a few of these 18x responses that I don't deal with (which is dumb, since I don't really have to DO anything).
New asterisk.jsp attached - let me know if it works.

asterisk.jsp.zip

8131crash
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Postby 8131crash » Fri Sep 01, 2006 7:21 pm

That fixed it. :D
alohatone
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Postby alohatone » Fri Sep 01, 2006 9:32 pm

That works much better!!! Thanks so much!
I do have a question re: the dial flow. Why does it dial the destination before the source?
I am having to play a recording so that the recipeint of the call WAITS while the system then dials my extension or mobile phone.

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