I am having problems... i setup a sip trunk and every time i try to dial out i get the "all circuits are busy now" female voice! How can I go about fixing this problem?
all circuits are busy now!! driving me crazy!!
Submitted by ericosuave on Wed, 02/10/2010 - 11:30am.
well, i got it figured out
Wed, 02/10/2010 - 1:59pm
well, i got it figured out to hit the VOIP server, but now when i try to call out its giving me another english accent lady's voice saying "VOIP message 100 the peer you are calling from is not configured correctly"?
Also when I call the PBX number from a cell phone it answers correctly and goes through the menus that i set up, but when i dial an extension it takes like 15-20 seconds to ring and connect me to that extension? Thats kinda crappy. any ideas?
about that Wiki...
Fri, 02/12/2010 - 6:46am
While I am often the one asking simple questions I know I am taking this post a bit off topic, but since the Wiki came up I want to draw attention to a few links within:
http://trixbox.org/wiki/trixbox-ce-release-test-plan
The MOST useful information so far on the Wiki...
http://trixbox.org/wiki/trixbox-ce-faq
NO useful info here/\
http://www.trixbox.org/wiki/how-big
almost addresses scalability...not quite and certainly not directly...
http://trixbox.org/wiki/trixbox-quick-install-guide
Extremely BASIC guide - real useful tip here is the system-config-network command
http://trixbox.org/wiki/Files_systems%2C_hard_drives%2C_mounting_....
Thank god this is in the Wiki...
http://trixbox.org/wiki/trixbox-ce-manager-docs
Amen to this! Def. not the place I would look for info on managing a system...
http://trixbox.org/wiki/trixbox-ce-end-user-docs
Good thing there are no end users looking here for information (or maybe there are, and this empty page (since 1/21/2008!!!) is why they continue to come to the forum for simple questions)...
http://trixbox.org/wiki/trixbox-ce-certified-hardware
aaaaaand....the incomplete list of endpoints - maybe we need to add the new line of Aastra that was relabeled to keep more in line with the EU naming of the same phones.
Perhaps there needs to be an update to it - I am working on a guide for SCCP and Cisco phones to set them up since there is a lot of information out there, but none on what SCCP version is best to use and most stable. If everyone who had specific knowledge in an area contributes to the wiki, we would really have a good guide put together (including FAQ or frequent issues one might run into)
Sorry - back on topic now!
-Jon
start your live log from the
Fri, 02/12/2010 - 7:29am
start your live log from the CLI ( asterisk -cvvvvvr )
Watch the log and make a test call.
You'll see something like
-- Requested transfer capability: 0x00 - SPEECH
-- Called SIP/5551014
-- SIP/7-1 is proceeding passing it to SIP/1092-08979870
-- SIP/7-1 is ringing
Basically, paste the log of your call into the forum, and mark in there at the point where the log 'stalls' while waiting for your call to continue. We can put a little analysis into it for you.
Member Since:
2010-02-10