it work perfectly avec passing the astNoExtenCheck to false. Thanks a lot !!!
it work perfectly avec passing the astNoExtenCheck to false. Thanks a lot !!!
Hi,
just installed and configured the zimlet and already did some debugging...
our SIP channels are like: phone-manufacturer-#
e.g: phone-linksys-2 is the channel name!
everytime i dial the log is:
does the SOAP "kill" the "-" ?zimlet - ch_bnc_asterisk caller: phonelinksys2 callee 203
so phone-linksys-2 -> phonelinksys2
any ideas? (changing the SIP name isn't a solution)
Thanks Martin
There is no good day since you make it one!
Sorry for not updating in so long. But I have good news!
Simply adding the hint lines to my phones context solved it!
After adding that and reloading, I clicked and my desk phone rang right away. I picked it up and sure enough it went right out. Two way audio is working as well.
So thank you so much!
Hi Martin
No SOAP does not kill the '-'. Actually I do some number checking in asterisk.jsp. In fact this is not necessary for the caller (which is taken from the zimlet user propery srcPhone). So you can try to remove line 496 in asterisk.jsp that contains this code:
Then i would expect it to work. I think i will remove that that check in future versions, too ...Code:caller=checkNumber(caller);
Let me now if it worked
Hi,
thanks that helped! works great now!
good work... keep going
Martin
There is no good day since you make it one!
Wow, that was intense. After 6 hours of head banging, I got it working. Problems turned out to be Asterisk 1.6 related.
1) If you run Asterisk 1.6.x, you'll need to add 'originate' privs (new in 1.6) to your manager user:
... otherwise the 'originate' command will get rejected with "Permission denied", although you won't see this in the mailmanager log, you'll just see a generic originate failure. I'm not sure if it needs originate on both read and write, I just put it on both to be sure.Code:read = system,call,log,verbose,command,agent,user,originate write = system,call,log,verbose,command,agent,user,originate
2) If you run 1.6.0.3, you'll need to patch your Asterisk as per this bug:
0014208: Channel not specified - Asterisk-1.6.0.3 - Digium Issue Tracker
... (file you need is main/manager.c) then make, make install and restart Asterisk. Without this, the 'originate' command always gets rejected with "Channel not specified", even though it is specified.
If you run FreePBX in vanilla form, you'll probably need to config with:
Whoo HOOOO! It works.Code:<property name="astDialContext">from-internal</property> <property name="astDialChannelType">Local</property>
Thanks chlauber, this is awesome.
-- hugh
Can we pretty please have a dial prefix config option? Similar to srcPhonePrefix, but for prefixing the callee with?
-- hugh
Probably a simple question but how do you adjust so it can dial different number formats?
For example, right now it's just dialing 5555555 for me but I also need 555.555.5555 and 555-555-5555.
Thanks!
You need to fiddle with the matchingRegEx in your zimlet config. Which can be a little scary if you aren't familiar with regex pattern matching.
Someone mentioned building a better, multi-purpose regex earlier in this thread, but I haven't seen any followup on that yet. I'm about to start rebuilding my matchingRegEx to recognize more formats, I'll post when I have something working.
-- hugh
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