would you like to try this?
bridge_hangup_cause = session:getVariable("bridge_hangup_cause") or
session:getVariable("originate_disposition");
if (bridge_hangup_cause == "NORMAL_TEMPORARY_FAILURE" or
bridge_hangup_cause == "NO_ROUTE_DESTINATION" or bridge_hangup_cause
== "CALL_REJEC
grr...continue_on_fail...ignore my ignorance ;)
but it would still be nice getting a response back from the session:execute
bridge
--matt
On Thu, May 21, 2009 at 11:09 PM, Matthew Fong wrote:
> hrm...it's also seems to be that if my lua script looks like
> session:execute("bridge", "sofia/gatew
hrm...it's also seems to be that if my lua script looks like
session:execute("bridge", "sofia/gateway/XXX/0X")
session:execute("bridge", "sofia/gateway//XXX")
if the first bridge fails, the session is immediately hungup, even if
hangup_after_bridge is set to false...is this the intended be
Hey Brian,
Will have a look at ZRTP :)
Not sure I understand your comments regarding its all over once
receiving the 415 from the B party. Is'nt that what parm
continue_on_fail does? The fact that it sends the invite back out
sorta proves this.
The other point of interest here is that if you s
I'm using a lua script to control an IVR, and would like to know how I can
tell if a
session:execute("bridge","sofia/gateway/blahblah");
was successful or not
it seems the response from session:execute is nil regardless if the bridge
was successful or not
whats the best way? Thanks
--matt
_
Hi all,
I want to use FreeSWITCH as a SIP transparent proxy in session border
controller application. Please let me know the changes in configuration files
required to achieve this behaviour
Thanks very much for the help.
Regards,
Sridhar
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If I had a few bucks for every time the telco has said this to me I
could just about retire! You using 100% SIP?
/b
On May 21, 2009, at 10:09 PM, Dale Trub wrote:
And, the telco says many other people are in the same set-up as us
and don't have any issues, so they're insisting it's on our
Jim,
You seem to be making the whole ordeal overly complex for no reason.
expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"
break="never">
You can not accept the call and send it out and get a 415 back and
expect to do the proc
Hey guys,
I'm about to start my own ITSP with FreeSWITCH, and I'm looking some
cool names for my VoIP company, if you know some please tell me :)
Diego
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Hi All,
Have been trying to workout how to solve a call scenario involving
SRTP and need some help.
The scenario is:
Eyebeam >FS->Eyebeam with make and accept only encrypted calls set.
What I am hoping to acheive is, if the A leg does not have SRTP set
and no SRTP Descriptors are sent in
FYI,
We just want to let everyone know that we have made a few updates that
will require a rebootstrap. One of the key updates was a security fix
for libsndfile. In this particular case it won't be possible simply to
"make current" like you normally do. Here is a common set of commands
fo
Dale Trub wrote:
> Freeswitch svn revision: 12148
It's time for an upgrade.
The developers have fixed a lot of bugs since then.
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Thanks Brian! To answer your questions:
Freeswitch svn revision: 12148
Centos rev: 2.6.18-92.el5
And apologies, actually I guess we're using g711 not 729.
Jason: I agree it would seem to be on the switch/telco side. And, the
telco says many other people are in the same set-up as us and don't h
Dale Trub wrote:
> Does anyone have any recommendations about how to troubleshoot this?
sofia profile external siptrace on
and watch the SIP traces to see what happens.
>
> Any known issues/patches in FS that could be biting us?
You didn't say which version you were running. Does the problem s
On May 21, 2009, at 9:15 PM, Dale Trub wrote:
We're running FreeSwitch as part of a teleconferencing service,
inside a telcom (so no
internet latency/NAT issues) and using g.729
So you're using g729 with conferences?
We are receiving some complaints of dropped calls,
including from landli
I have experienced the same a while ago, I originated calls from my
freeswitch server to some landlines and calls would simply drop after
X minutes.
I tried to debug the thing but found nothing relevant, maybe I had the
same issue as you.
Let me know if you figure it out what it was.
Diego
On T
We're running FreeSwitch as part of a teleconferencing service, inside a
telcom (so no
internet latency/NAT issues) and using g.729
We are receiving some complaints of dropped calls,
including from landlines. This means they join the conference, and x
minutes in they simply drop.
I know that ce
Try pressing * during the greeting and make sure you have the vmain
extension so you can login.
/b
On May 21, 2009, at 6:56 PM, Lars Zeb wrote:
I want to setup a dialplan for a single DID. I would like it to go
to a specific extension, and if not picked up in 15 seconds, go to
voicemail.
I want to setup a dialplan for a single DID. I would like it to go to a
specific extension, and if not picked up in 15 seconds, go to voicemail.
I have set this scenario up and it works. But I would also like this person
to be able to call this DID from outside FS via a phone and be able to
ret
We just tested this on our 64bit system and it works fine. Can you
open a jira with a backtrace attached? Also how did you update?
/b
On May 21, 2009, at 3:29 PM, Lon Baker wrote:
I updates to the latest trunk this morning and still get the same
segmentation issue on 32 bit Ubuntu server
Also open a jira please.
/b
On May 21, 2009, at 3:29 PM, Lon Baker wrote:
I updates to the latest trunk this morning and still get the same
segmentation issue on 32 bit Ubuntu server 9.
The exact command I am issuing is fsctl shutdown asap.
Lon
Brian West
br...@freeswitch.org
-- Meet us
prefix it with bgapi fsctl shutdown You can't do a shut down and wait
for it to stop and give you a response because XML RPC is blocking.
/b
On May 21, 2009, at 3:29 PM, Lon Baker wrote:
I updates to the latest trunk this morning and still get the same
segmentation issue on 32 bit Ubuntu s
I updates to the latest trunk this morning and still get the same
segmentation issue on 32 bit Ubuntu server 9.
The exact command I am issuing is fsctl shutdown asap.
Lon
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No, this doesn't exist yet in FreeSWITCH but I do like the idea.
/b
On May 21, 2009, at 3:17 PM, Yossi Neiman wrote:
Now that I've said that I don't believe it currently has these
features,
I'm waiting for Brian West to come in and correct me like he usually
does. :-)
Brian West
br...@fre
I don't *believe* there is anything of this nature built in at this
time. I would imagine it can be implemented. If you yourself don't
know C, but would like this functionality, three ways to get something
like this added would be:
A) Put up a bounty on the bounty page of the Wiki and hope s
Thanks for catching those little glitches!!!
-MC
On Thu, May 21, 2009 at 9:44 AM, Giovanni Maruzzelli wrote:
> fixed
>
>
>
> On Thu, May 21, 2009 at 6:31 PM, Brian West wrote:
> > Its an error on the wiki you should have $${domain} in there
> > /b
> > On May 21, 2009, at 11:22 AM, Larry Marshall
fixed
On Thu, May 21, 2009 at 6:31 PM, Brian West wrote:
> Its an error on the wiki you should have $${domain} in there
> /b
> On May 21, 2009, at 11:22 AM, Larry Marshall wrote:
>
> On the page http://wiki.freeswitch.org/wiki/Configuring_SIP under
> Configuration, it speaks about the vars.xml
Its an error on the wiki you should have $${domain} in there
/b
On May 21, 2009, at 11:22 AM, Larry Marshall wrote:
On the page http://wiki.freeswitch.org/wiki/Configuring_SIP under
Configuration, it speaks about the vars.xml file. Specifically it
states, “In this file, there is only one pa
On the page http://wiki.freeswitch.org/wiki/Configuring_SIP under
Configuration, it speaks about the vars.xml file. Specifically it states,
"In this file, there is only one parameter that you need to specify. That
parameter is $${sip_profile}."
I can't find the variable, nor can I grep for its
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