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14319KevinH
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Postby 14319KevinH » Sat Aug 12, 2006 12:18 pm

Can you connect external to your mobile? So a true external to external call? IF not maybe your bridge ext is not setup the way the Zimlet expects.


gmsmith
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Postby gmsmith » Sat Aug 12, 2006 8:12 pm

[quote user="14319KevinH"]Can you connect external to your mobile? So a true external to external call? IF not maybe your bridge ext is not setup the way the Zimlet expects.[/QUOTE]
No, I can not. I can get my cell phone to ring, if I try to ring it from my "desk" phone, the cell phone will ring, but I will have dead air when I pick it up. My desk extension will not ring at all, though the logs say the call was made from my extension to my cell phone.
14319KevinH
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Postby 14319KevinH » Mon Aug 14, 2006 7:35 am

Hmm what about a mobile to mobile call. This is our favorite one to demo. Just enter two mobile numbers and connect them. First one should ring and after you answer the 2nd mobile should ring.
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Postby gmsmith » Mon Aug 14, 2006 8:30 am

[quote user="14319KevinH"]Hmm what about a mobile to mobile call. This is our favorite one to demo. Just enter two mobile numbers and connect them. First one should ring and after you answer the 2nd mobile should ring.[/QUOTE]
Yeah, so that didn't work. The "to" phone rings, the from phone never rings. Both numbers are off system.
Greg
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Postby seanh » Mon Aug 14, 2006 12:55 pm

[quote user="14319KevinH"]It will use the Asterisk bridge number(ext) to make two calls one to the first number and one to the second. The 2nd should ring after the first anwsers.[/QUOTE]
I'm not seeing this fuctionality. My asterisk box calls the end user but does not dial my desk extension.
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Postby marcmac » Mon Aug 14, 2006 6:26 pm

Run sip debug on your asterisk box, and you should see something like "attempt native bridge of..."
Note - you'll need 2 phone lines for this, obviously.
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Postby 8131crash » Wed Aug 16, 2006 10:49 am

[quote user="marcmac"]Run sip debug on your asterisk box, and you should see something like "attempt native bridge of..."
Note - you'll need 2 phone lines for this, obviously.[/QUOTE]


Is this working for others? If so there must be some piece of config I'm missing. I followed the wiki instructions and am having the same issue of only one side of the call ringing. I am not seeing any attempt to bridge the call on asterisk. Is there a step missing in the wiki instructions?
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Postby 1739miguel » Thu Aug 17, 2006 5:49 am

Not working here too, desk phone doesn't ring.. and I can't find no "attempt native bridge of.." on sip debug.
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Postby marcmac » Fri Aug 18, 2006 7:20 pm

moebis
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Postby moebis » Wed Aug 23, 2006 10:15 am

[quote user="marcmac"]
Marcmac
, this is a crock. Everyone here is having the same problem with the call not "bridging" or "originating" from the extension end. Another user fixed the problem with the configuration because of a bug, which you refused to answer or fix. "It just works" you say. Well we all only started to get this Zimlet to half work when we changed the configuration first, then rezipped the file and then Deployed the zimlet because your instructions do not work on a new install. I went through 3 reinstalls to see if I was doing something wrong, alas it was a bug, and if you had taken the time to try your instructions on a fresh install (instead of assuming everyones Asterisk server is 192.168.1.254 you would have seen that).
Now onto the new issue..... it does not work, as a matter of fact it does not for everyone in this forum thusfar, except you, the developer. Why are you using a Bridging system anyways? A similar plugin for SugarCRM simply forwards the call to the extension, and guess what? It does work, as a matter of fact it works for everyone who has ever tried to integrate Asterisk with SugarCRM.
I'm sorry if this post sounds harsh, but it does accurately convey my level of frustration about the time I/we have wasted. Reflect on an old adage Marcmac, if everyone is telling you something is wrong, and you're the only one who thinks its ok, then usually everyone else is right.

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