Hi:
I try to dial sip url from my softphone but seems like the sip address is
being processed by sofia before it pass to the dialplan. The example here is
:
*X-lite(softphone) dials - 1...@4.2.2.2 (it's fake sip address, the purpose
was just to test what's being passed to dialplan)
sofia
So there seems to be some sort of error when bridging directly like
originate
{ignore_early_media=true}sofia/gateway/.com/91415992http://epik.com/914159927717
bridge(sofia/gateway/.com/91415465 http://epik.com/914154650027
)
BUT
if I get FS to send media to leg A, and then
Hi
I checked and there is no looping in cdr.
Also, only a very small percentage of the channels become zombie.
What could cause fs to not releasing the channels?
Also, it seems to happen on under high traffic. Could the fact that FS does
not receive BYE or BYE timing out on the uac side may
No, you don't get the full sip uri in the dialplan like that. You do
have a whole bunch of variables of the parsed sip header you can use.
Use the info application to see all the vars so you can see what you
have to route the call on.
Mike
On Aug 22, 2009, at 2:40 AM, Henry Huang
It that case, the example of dialing sip_uri in the dialplan/default.xml
should be removed to prevent confusion. Because according to what you said,
one can never be able to hit this extension:
!-- dial via SIP uri --
extension name=sip_uri
condition field=destination_number
Henry Huang red.rain.se...@gmail.com wrote:
It that case, the example of dialing sip_uri in the dialplan/default.xml
should be removed to prevent confusion. Because according to what you said,
one can never be able to hit this extension:
It is entirely possible to reach this extension, but
Jason:
I fully understand how the regex works in the dialplan. If you look closely
in my original email and check out the pastebin. You will see that sofia
does not pass the sip: to dialplan. I can do any combination of letters
that dials from my softphone, and it will pass them to the dialplan.
a call coming from sofia would never hit that in the dialplan. That
extension is useful for dialing a sip url from mod_portaudio.
Mike
On Aug 22, 2009, at 10:09 AM, Henry Huang wrote:
Jason:
I fully understand how the regex works in the dialplan. If you look
closely in my original email
Remember the dialplan is agnostic... it has no clue about SIP, IAX,
Jingle, H323... it routes... you have various other variables you can
condition on also... route on destination_number and you'll be fine.
/b
On Aug 22, 2009, at 9:09 AM, Henry Huang wrote:
I fully understand how the regex
Brian:
but why can't I pass sip: to dialplan? seems like it's being truncated by
sofia..
Can you confirm that?
On Sat, Aug 22, 2009 at 10:30 PM, Brian West br...@freeswitch.org wrote:
Remember the dialplan is agnostic... it has no clue about SIP, IAX,
Jingle, H323... it routes... you have
Because the dial plan is technology agnostic... you have been told
more than once it won't pass it to the dialplan from mod_sofia...
/b
On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass sip: to dialplan? seems like it's being
truncated by sofia..
Can you
Brian:
Sorry, it's my English. I didn't understand what you meant by agnostic
back there. Now I know.
Thank you.
On Sat, Aug 22, 2009 at 10:59 PM, Brian West br...@freeswitch.org wrote:
Because the dial plan is technology agnostic... you have been told
more than once it won't pass it to the
Brian:
Oh, and again, if it's not passing it to the dialplan. I had suggested to
remove the sample sip uri extension in the default.xml dialplan. because
no one can reach the dialplan with prefix sip: because sofia is going to
remove that prefix.
!-- dial via SIP uri --
extension
You were told already this was used by mod_portaudio. So that you can
pa call sip:b...@domain.com which portaudio passes the exact string
you dial with pa call to the dialplan.
/b
On Aug 22, 2009, at 10:07 AM, Henry Huang wrote:
Brian:
Oh, and again, if it's not passing it to the
On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang red.rain.se...@gmail.comwrote:
Brian:
Oh, and again, if it's not passing it to the dialplan. I had suggested to
remove the sample sip uri extension in the default.xml dialplan. because
no one can reach the dialplan with prefix sip: because sofia
I think there's something wrong with the script at
http://wiki.freeswitch.org/wiki/Examples_cnam.js.
If you use it as is, it displays Content-type: text/html for the
effective_caller_id_name. In cnam.pl, the first two output lines are
generated by:
if (!$debug) {print Content-type:
Michael:
Thank you for making it in for dummies format. :P
These are really nice tips I can use. thanks.
On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins m...@freeswitch.orgwrote:
On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang red.rain.se...@gmail.comwrote:
Brian:
Oh, and again, if it's
Yes try enabling session timers
On Aug 22, 2009 3:10 AM, Woody Dickson woodydick...@gmail.com wrote:
Hi
I checked and there is no looping in cdr.
Also, only a very small percentage of the channels become zombie.
What could cause fs to not releasing the channels?
Also, it seems to happen on
Hi,
I have a FS gateway (SVN revision 14537) that is is receiving SIP calls from
different source gateways and sending it to one single destination gateway.
Now each source gateway can talk in one specific codec and FS itself is not
doing any transcoding. So i enabled all possible codecs that
Can you provide a little bit of log detail?
/b
On Aug 22, 2009, at 3:53 PM, Muhammad Shahzad wrote:
Can you guys suggest why it is happening and what are the possible
solutions, other then transcode of course.
___
FreeSWITCH-users mailing list
i just upgraded it to 14599 and its working fine now.
Thank you.
On Sun, Aug 23, 2009 at 2:07 AM, Brian West br...@freeswitch.org wrote:
Can you provide a little bit of log detail?
/b
On Aug 22, 2009, at 3:53 PM, Muhammad Shahzad wrote:
Can you guys suggest why it is happening and what
X-lite I believe handles the sip: by itself sometime and therefore will try
and place a call to the sip address directly from x-lite without touching
FreeSWITCH. Be aware of this while testing and watch for this behavior
because it might throw off your expectations.
Regards,
Kevin Green
On
Nice work, keep up the great work :).
On Fri, Aug 21, 2009 at 6:29 PM, Moises Silvamoises.si...@gmail.com wrote:
So, I finally took some days to put up OpenR2 working with OpenZAP, which
means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
support for. Including Mexico,
23 matches
Mail list logo
