On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote:
On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote:
On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote:
Well, if you're running multiple machines, waiting for it to drainstop
isn't that big of a deal unless you're in some sort
I've experience with a commercial SBC, these are two machines running in
cluster mode. In that case if one SBC is going down then other will take all
new calls including the call which were active on broken SBC (SIP only).
Thats quite ideal for wholesale traffic where the SBC will never be idle.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Anthony,
yes, that's true. Works well now.
Thanks a lot!
regards
Helmut
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
iD8DBQFKMk1W4tZeNddg3dwRAhzrAJ45OpGkPkpLEPRw17HUpR3CTaxVVwCcD4/0
TJpI0jZez6uOdETu3OtDbc8=
=yWhz
-END PGP
Can you comment some more on how this is configured?
Would it be something that could be added to the wiki in the SBC setup
page?
Best regards,
Even André Fiskvik
On 12. juni. 2009, at 12.16, Saeed Ahmad wrote:
I've experience with a commercial SBC, these are two machines
running in
Lars, don't get me wrong but you have been asking questions that are all
answered on the wiki:
http://wiki.freeswitch.org/wiki/Sofia#Flushing.2Frebooting_registered_endpoints
Might be a good idea to value the work of lots of ppl who have been
documenting by actually using the documentation, no?
No, its not too harsh, João, but I hope not all of my questions were
answered on the wiki.
I do try to go to the wiki first. I think that my total ignorance of the
environment makes it difficult for me to do a search on the wiki or Google.
I did try before asking this list. My query to Google
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I just want to let you know, that I started some work on Q.931 TE state
timers in current openzap ozmod_isdn stack. The current stack has some
problems with freeing ressources when far end doesn't follow Q931 state
machine cleanly. On my side
No its not possible yet. Voicemail has an option to record in 11025
but not arbitrary rates for recording otherwise.
/b
On Jun 12, 2009, at 9:37 AM, Andy wrote:
Hi,
Sorry but I just can't find this in the documentation. I'm using
recordFile to record incoming messages. I'd like the
Its one of the 3000 settings you can change on the identity ... but in
your case I am not sure it would have removed the old registration
before the new one was registered... but check on the preferences or
the identity there is a setting to unregister on reboot.
/b
PS: sofia profile xxx
Hi,
Is there any where I can find more documentation on the XML-RPC interface of
freeswitch. I am trying to initiate a conference and add in users to it from
flex. I would appreciate any insights.
Thanks
San
___
Freeswitch-users mailing list
http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC
http://wiki.freeswitch.org/wiki/Mod_commands
http://wiki.freeswitch.org/wiki/Mod_conference
Mike
On Jun 12, 2009, at 3:35 AM, Santhosh wrote:
Hi,
Is there any where I can find more documentation on the XML-RPC
interface of freeswitch. I am
please make sure you stay tuned into #openzap and coordinate with stkn and
the other guys
doing work on the stack. That way we can make sure we get the best out of
the code.
On Fri, Jun 12, 2009 at 9:32 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
-BEGIN PGP SIGNED MESSAGE-
Hash:
Looked easy enough so i added
record_sample_rate variable that should influence it for you if you set it
in advance.
On Fri, Jun 12, 2009 at 9:49 AM, Brian West br...@freeswitch.org wrote:
No its not possible yet. Voicemail has an option to record in 11025 but
not arbitrary rates for
Excellent, thanks Anthony, I'll give it a go.
_
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: 12 June 2009 17:03
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users]
Well, Nextone for instance has a database the keeps most of the state of calls,
and it's replicated between the two nodes. (I seem to recall the database was
GNU dbm, but I might be mistaken.) However, as of 4.3 anyways, the CDRs still
get truncated when there's any kind of switchover.
But
Helmut Kuper wrote:
Hello,
I just want to let you know, that I started some work on Q.931 TE state
timers in current openzap ozmod_isdn stack. The current stack has some
problems with freeing ressources when far end doesn't follow Q931 state
machine cleanly. On my side I mainly miss a
On Fri, Jun 12, 2009 at 9:26 AM, Andy a...@fabulous4.co.uk wrote:
Excellent, thanks Anthony, I'll give it a go.
Andy, can you report back on your success with this variable? Also, we would
appreciate it if you could add an entry to the wiki on the channel_variables
page. Let me know if you
Saeed Ahmed wrote:
No idea at all,
It's a commercial SBC.
I wish if we can have same functionality in FS.
You could accomplish parts of this with hearbeat and ldirectord the
in-session calls aren't going to go anywhere, but if the server crashes,
the second one can take over the ip of
Gang,
There have been crazy rumors flying around. Tony sets the record straight.
Please go here now and digg this story:
http://digg.com/software/Anthony_Minessale_of_FreeSWITCH_Discusses_Barracuda_Rumors
Thanks!
-MC
___
Freeswitch-users mailing list
Hi all,
We're running into a problem with rfc2833. Here's the situation:
A registered SIP endpoint sends DTMF as rfc2833 to our FreeSWITCH.
The FS console shows it receives the digits once.
We send the call to a tier-1 carrier
The digits are played read back, but often the first several digits
What device are you using? RTP traces, debug logs something to see
what might be taking place.?!?!
/b
On Jun 12, 2009, at 1:59 PM, Ben Jones wrote:
Hi all,
We're running into a problem with rfc2833. Here's the situation:
A registered SIP endpoint sends DTMF as rfc2833 to our FreeSWITCH.
Testing was done with SJphone for Mac, dtmfmode rfc2833 pt 101.
Hopefully this debug log can help:
http://pastebin.freeswitch.org/9374
If I need to add, change, whatever, let me know. Thanks for the help.
-benj
Brian West wrote:
What device are you using? RTP traces, debug logs something to
Greetings,
I'm new to freeswitch, just playing with it in a home/small office at
present. Overall I've been really impressed with it.
One issue we've noticed is a pronounced latency on some or perhaps all
calls over roughly 500 seconds in duration. The latency is
approximately 3 seconds or more
I need to hire an experienced FreeSwitch developer to build a small chat
bridge that supports VoIP and 1-4 PRI's of TDM. Please contact me at
h...@powercom.com or (805)845-8906.
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Can you describe your setup? Need to know what kind of OS and hardware is
running FS as well as what kind of phones. Any NAT involved?
-MC
On Fri, Jun 12, 2009 at 3:14 PM, David Burgess apt@gmail.com wrote:
Greetings,
I'm new to freeswitch, just playing with it in a home/small office at
On Fri, Jun 12, 2009 at 4:33 PM, Michael Collinsm...@freeswitch.org wrote:
Can you describe your setup? Need to know what kind of OS and hardware is
running FS as well as what kind of phones. Any NAT involved?
-MC
FS is running inside pfsense, which is a freeBSD-based firewall
(pfsense.org).
On Fri, Jun 12, 2009 at 3:53 PM, David Burgess apt@gmail.com wrote:
On Fri, Jun 12, 2009 at 4:33 PM, Michael Collinsm...@freeswitch.org
wrote:
Can you describe your setup? Need to know what kind of OS and hardware is
running FS as well as what kind of phones. Any NAT involved?
-MC
Try action application=set data=rtp_autoflush=true /
Math
On 12-Jun-09, at 7:28 PM, Michael Collins wrote:
On Fri, Jun 12, 2009 at 3:53 PM, David Burgess apt@gmail.com
wrote:
On Fri, Jun 12, 2009 at 4:33 PM, Michael Collinsm...@freeswitch.org
wrote:
Can you describe your setup?
Upgraded from Apr 3 svn to svn 13769.
Calling from openzap to (music on hold) works.
Calling from openzap to 9995 (5 sec echo test) works.
Calling from openzap to vmail works.
Calling from Grandstream to (music on hold) works.
Calling from Grandstream to 9995 (5 sec echo test) doesn't
Hi everyone,
I just found this project which seems to be a new VM for running
languages... the parrot project aims to create a virtual machine for Perl 6
and other dynamic languages. You can take a look at it here:
http://www.parrot.org/
It already supports many different languages:
BTW: in all cases show channels says PCMU 8000 is being used
for the read and well as write codec.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax:
Here mercutioviz gave me some interesting info about Parrot.
20:47 @mercutioviz diegoviola: parrot isn't tied specifically to perl 6
but it is an offshoot of the perl 6 effort
20:47 @mercutioviz they were smart to break apart the big project into two
separate projects
20:47 @mercutioviz parrot is
On Fri, Jun 12, 2009 at 5:34 PM, Mathieu Renemrene_li...@avgs.ca wrote:
Try action application=set data=rtp_autoflush=true /
Thanks, I will try that.
Hmm... Might want to ask Mark Crane (IRC: mcrane) if he's seen anything like
this with FS+pfSense.
Yeah, we've discussed it. He's experienced
Diego Viola diego.vi...@gmail.com wrote:
Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot
support also.
Are you offering to write it?
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
No, I'm not a C programmer, just offering the idea (feedback).
On Fri, Jun 12, 2009 at 9:15 PM, Jason White ja...@jasonjgw.net wrote:
Diego Viola diego.vi...@gmail.com wrote:
Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot
support also.
Are you offering to write
Yet more information ... a packet trace of a openzap to Grandstream
call shows:
SourceDestination Packet
FreeSWITCHGrandstream SIP Request: INVITE ...
Grandstream FreeSWITCH SIP Status: 100 Trying
Grandstream FreeSWITCH SIP Status: 180 Ringing
I think possibly that the configs changed, specially the auto-nat stuff
Yep ... a closer look at the packet trace showed FreeSWITCH settings
the Contact as 10.10.10.1 instead of the actual IP address of the machine.
If you have modified those two files then I recommend looking at the new
37 matches
Mail list logo
