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Forum profile page for Help on http://www.trixbox.org.
This report page is the aggregated overview from a single forum: Help, located on the Message Board at http://www.trixbox.org.
This forum profile page summarizes the general forum statistics such as: Users Activity, Forum Activity, and Top Authors, which are reported in either a table or graph below for a given reporting time period.
Additional forum profile information for "Help" on the Message Board at http://www.trixbox.org is also shown in the following ways:
1) Latest Active Threads
2) Hot Threads for Last Week
Warning: These statistics are generated using 'best efforts' and can experience delays and reporting errors at times. Please note that such statistics do not constitute a forum's popularity and/or exact posting volumes at any given reporting period.
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Posting activity on Help:
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Week
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3 Months
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Threads:
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103
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416
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1,279
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Post:
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180
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771
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2,514
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Help Posting activity graph:
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Top authors during last week:
user's latest post:
Full Log File States...
Published (2009-12-04 23:43:00)
Quote: What are some of those issues? If you are referring to dial plan looping it is either the result of a misconfiguration IVR, Ring Group and Follow Me or some combination of these. This is a very easy problem to spot and solve. Lost frames is almost always due to network issues, layer 2 issues creep up all the time. Let me give you an example. I am amazed at the folks who buy expensive managed switches and then cascade them together...
user's latest post:
Inbound round picking up too...
Published (2009-12-04 16:41:00)
rofz, I going to give you a good advice, just buy a Digium or Openbox FXO card, the X100P is a very cheap card but with many issue one of then noisy problem.
user's latest post:
Iristel SIP Trunk
Published (2009-12-01 14:33:00)
It's embarrassing that I've been posting 6 times in a row... Anyways, here's an update. My provider wants to try static registrations. Since I have a static, the registration expiry wouldn't be used... Just curious if anyone has any insights on trixbox conf for a SIP trunk that doesn't require registration refreshes? Thnx guys!
user's latest post:
Call Pickup not posible
Published (2009-12-01 16:27:00)
Here are some instructions posted by christopherb it worked for me but I guess you may have to add a line with the name of the context that is specified in your ISDN config file if it is different to the ones included below. I'd like to be able to pick up a call from a phone that is ringing because it is in a group, its call would be for ext 600 so the following doesn't work under that condition. 1)Disable the callpickup feature on...
user's latest post:
flite and trixbox 2.8.0.1
Published (2009-11-28 19:44:00)
I was able to get festival running. yum -y install festival and then some tweaking from the following page to customize it. I used method 1. http://www.voip-info.org/wiki/index.php?comment_page=2&page_id=45... and then start the festival server /usr/bin/festival_server & This will have to be performed at each reboot to start the festival server. (unless I automate the startup). Voice sounds pretty robotic, not really what...
user's latest post:
VM problem
Published (2009-12-02 20:24:00)
Well I have tried disabled VM on one of stension. Rebooted Trix and enabled again. No luck. The same think - "You have " and hung up. MST
user's latest post:
Full Log File States...
Published (2009-12-03 20:17:00)
Same problem, same situation as above, as of yet no resolution other than to reload the system... Anybody out there have an answer? If not how does one escalate an issue like this to get an answer? Thanks in advance...
user's latest post:
Best way to failover
Published (2009-12-04 12:24:00)
This will require custom programming, cronjobs, and a pretty good understanding of linux and networking. There isn't an easy answer :(
user's latest post:
TRUNKS
Published (2009-12-03 08:23:00)
any help???
user's latest post:
Site to Site VPN and Phone...
Published (2009-11-30 13:24:00)
Dont Know if this wiil help, but i have to different trixboxs at two different locations and they are working fine, i have connected them through a iax trunk using port 4569 udp, both trixboxs can can ring out and transfer to Extension. i spent hours configuring my systems to get them to work but hay thats life. . Regards Seanshonag
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Latest active threads on Help::
Started 7 months, 1 week ago (2009-04-27 10:27:00)
by aaronhun22
I found out the at the problem was with MISC Applications. When I put a feature code on one of my IVR's and used it I started getting this problem. Since then I haven't been using MISC Applications and I'm now at 15 weeks for uptime with no full error logs. Log into asterisk and see if it is doing anything. I knew I had a problem when it was repeating code on the interface....
Started 2 months, 2 weeks ago (2009-09-18 02:00:00)
by Domhampton
A bodge could be to route the call through a ring group and tell it to fail over to the ivr when nobody picks up form a dummy extention
Started 3 days, 7 hours ago (2009-12-03 23:23:00)
by ddwyer
this is easy to to , in your incomming routes set a prefix on your DID.
ie maybe one line is sales for set prefix to "sales-" and you will see the number displayed on the phone as "sales-99988877766"
this is a simple 30 second task.
Started 10 months, 2 weeks ago (2009-01-22 13:37:00)
by dickson
I think (after a quick look and some assumptions) they might be load balancing based on departments or groups of phones.
Ok, if you are talking LOAD BALANCING and not high availablilty, dialplan can easily do this.
For example, you could have a box that strictly handles all your inbound PRI lines. It could maybe do all your IVR. Then depending on call selection, lets say French...
Started 2 days, 18 hours ago (2009-12-04 12:24:00)
by b14ck
This will require custom programming, cronjobs, and a pretty good understanding of linux and networking. There isn't an easy answer :(
Started 2 days, 21 hours ago (2009-12-04 09:09:00)
by SkykingOH
FIrst, by not editing sip.conf as it clearly states in the top of the file.
canreinvite=yes needs to be set on the peers.
Did you restart asterisk after your change?
Started 4 days, 16 hours ago (2009-12-02 14:26:00)
by stechnique
In pfSense:
Under Firewall -> NAT
On the Outbound tab.
Select Manual Outbound NAT rule generation (Advanced Outbound NAT (AON)).
Once that's done and saved, edit the default rule that popped below and click to select YES under Static Port.
Save and reload. This disables outbound port randomization.
Started 2 months, 1 week ago (2009-09-25 00:18:00)
by yiannos
Started 1 month, 3 weeks ago (2009-10-16 12:07:00)
by fitzrik
It sometimes feels that asterisk is ringing for 15 seconds and somehow transfers from one agent to another on seconds 13 or 14.
Also is there a way of getting the queue to keep ringing even when announcements are being made?
Thanks
R
Started 3 days, 6 hours ago (2009-12-03 23:56:00)
by SkykingOH
Welcome to the forum. Please do not post multiple messages.
Before asking for help please read the getting started guides on our wiki.
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Hot threads for last week on Help::
Started 1 week ago (2009-11-29 18:23:00)
by jdwebcc
Are you sure that your provider allows port 80? I typically change the web server to another port for security purposes.. please do this and retest.
JD
Started 1 week, 1 day ago (2009-11-28 22:42:00)
by SkykingOH
Did you change your firewall or network at all.
It would be useful to know the Asterisk version of the old and new systems.
You also need to post the trunk config.
Started 6 days, 20 hours ago (2009-11-30 09:58:00)
by dbaddorf
The ASA has a command to run a "fixup" on the SIP packets:
policy-map global_policy
class inspection_default
inspect sip
service-policy global_policy global
Is this code in place on the ASA?
My concern is that the Netgear may not be handling the SIP packets in the same way as the ASA (changing ports or something). I'm assuming that you have full access from one ...
Started 3 weeks, 2 days ago (2009-11-13 06:46:00)
by IcelandDreams
We would need more information about your layout.
If you are trying to reach multiple providers you don't need to "listen" on multiple ports, you are making the connection. You can run multiple trunks on the same port to multiple providers. If you need to route over specific interfaces (at the router or nics at the server?) then that is a routing function and easily done with ...
Started 2 weeks ago (2009-11-22 19:34:00)
by SkykingOH
Wow - a year and 6 months later the thread picks up.
#included files are parsed when asterisk reloads the dialplan or corresponding module.
Started 6 days, 19 hours ago (2009-11-30 11:06:00)
by b14ck
I'm using the same version of firefox and not having this issue. Can you reproduce it using another browser?
Started 4 days, 13 hours ago (2009-12-02 16:31:00)
by krayt3ch
onne i remember mine i shall post link :)
EDIT: i can only seem to locate the firmware downloads for SPA9xx
EDITEDIT: Apparently i dont have a valid Technical Support Services Agreement :(
Started 4 days, 16 hours ago (2009-12-02 14:26:00)
by stechnique
In pfSense:
Under Firewall -> NAT
On the Outbound tab.
Select Manual Outbound NAT rule generation (Advanced Outbound NAT (AON)).
Once that's done and saved, edit the default rule that popped below and click to select YES under Static Port.
Save and reload. This disables outbound port randomization.
Started 7 months, 1 week ago (2009-04-27 10:27:00)
by aaronhun22
I found out the at the problem was with MISC Applications. When I put a feature code on one of my IVR's and used it I started getting this problem. Since then I haven't been using MISC Applications and I'm now at 15 weeks for uptime with no full error logs. Log into asterisk and see if it is doing anything. I knew I had a problem when it was repeating code on the interface....
Started 6 days, 10 hours ago (2009-11-30 20:12:00)
by SkykingOH
I have never had any luck building against the modified Asterisk distributed with CE.
I build DAHDI, Asterisk, LIBPRI, Asterisk-Addons and channel_skype in one big package and it works great.
On a P4 machine it takes about an hour to do.
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