Hello,
While trying to display the *called party *name on SNOM phones I've found
that the field sent to the phone needs to be changed slightly in order to
make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects
Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and
Currently, is there any way to set the timeout on an outbound event socket?
In case, for whatever reason, the socket application at 192.168.1.108:
is unresponsive or offline, I would like the call to not wait the
extraordinary amount of time it takes to timeout so that I can handle it in
other
Hi!
I have FS natted and am connecting with an 'external' extension that
is registered to FS. ie the extension 2000 is registered on the
internet with a public IP through my router to FS (192.168.1.120 IP
address). uPnP works and I see that the extension is registered
successfully.
The problem
hi all:
how can i get the value of the myArg1 myArg2 in test.js.
like this originate sofia/example/1...@somewhere.com 'javascript(test.js
myArg1 myArg2)'
thanks!
2009-11-09
god.nirvana
___
FreeSWITCH-users mailing list
Can you elaborate on this and provide a patch on jira?
/b
On Nov 8, 2009, at 2:46 AM, Yehavi Bourvine wrote:
Hello,
While trying to display the called party name on SNOM phones I've
found that the field sent to the phone needs to be changed slightly
in order to make SNOM work:
I'm trying to get through the noobie tutorial that c888 recommends in
IRC, but PortAudio doesn't seem to build properly on Mac OS X 10.6.
It failed due to some code that wasn't 64bit ready, apparently. The
error I got was exactly the same as this 4 month old error from
MacPorts:
The problem is the patch isn't backwards compatible and blows away any
chance of being so. We have looked at this... and that patch IS NOT
RIGHT.
/b
On Nov 8, 2009, at 3:12 AM, Bruce Fletcher wrote:
I'm trying to get through the noobie tutorial that c888 recommends in
IRC, but PortAudio
OK, I'll ignore that MacPorts patch for now and try to find a better
approach.
I'll look into this further tonight, but this morning I found a more
recent promising patch on the PortAudio site:
http://www.portaudio.com/trac/changeset/1418
It seems to push some data types to 32 bit
Hello
I am also having trouble with portaudio. I still haven't figured out what it
is that is wrong. I have had it working in the past on this same machine same
install of debian lenny. Now it just reports
[ERR] mod_portaudio.c:964 Cannot find an input device
Also seeing this on a fedor
If you can figure out a clean way for us to do this with proper ifdefs
in tree in a way that will not break others that would be the most
preferred.
Mike
On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote:
OK, I'll ignore that MacPorts patch for now and try to find a better
approach.
I'll
You don't have ext-rtp-ip set in your config.
Mike
On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote:
Hi!
I have FS natted and am connecting with an 'external' extension that
is registered to FS. ie the extension 2000 is registered on the
internet with a public IP through my router to
Hi Mike,
I should have put that in also.
I do have external_rtp_ip set in my config. I have it set to my domain name:
X-PRE-PROCESS cmd=set data=external_rtp_ip=host:mydomainname/
I should also mention that if I use flaphone.com (which registers with
an external IP address), then I get audio.
Your packet traces would disagree with the statements below. It is
sending your internal address in rtp, so its not set correctly on
whatever profile your using to call out,
MIke
On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote:
Hi Mike,
I should have put that in also.
I do have
OK.. thanks Mike.
I assume I am using the Internal profile. I have defined user 2000
in the 'directory' using a context called family: switch_ivr.c:1367
Transfer sofia/internal/1...@192.168.1.120 to xml[2...@family]
This is an extract from sofia:
sofia status profile internal
before playing with mod_sofia, did you try the sip_cid_type variable?
http://wiki.freeswitch.org/wiki/Variable_sip_cid_type
On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine yehavi.bourv...@gmail.com wrote:
Hello,
While trying to display the called party name on SNOM phones I've found
that
Hi again,
Actually, changing the param name=ext-rtp-ip value=auto-nat/ to
param name=ext-rtp-ip value=$${external_sip_ip}/ means that I
now see the IP address in the INVITE message:
v=0
o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 124.xxx.xxx.xxx
It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise,
no go
Have you changed the ext-sip-ip too?
Regards,
JM
On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi again,
Actually, changing the param name=ext-rtp-ip
I was not aware of this variable; I will take a look on it tomorrow.
However, when looking in the code I did not find something which looks like
Remote-Party-ID'.
Thanks! __Yehavi:
2009/11/9 SP spr...@gmail.com
before playing with mod_sofia, did you try the
Is there a way to determine if FS has detected nat? I am behind UPnP
and I can see on the router the mappings for Freeswitch.
2009/11/9 João Mesquita jmesqu...@freeswitch.org:
It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise,
no go
Have you changed the ext-sip-ip
I think I've fixed it, but I had to change a few things...
I had a host name set in vars.xml for external_rtp_ip and for
external_sip_ip. Having the external_rtp_ip set to a hostname, sofia
showed the
RTP-IP 192.168.1.120
Ext-RTP-IP host:myhostname
SIP-IP
Closed. As (almost) usual the reason was me.
Anthony's hint works perfectly:
api uuid_transfer uuid bridge:sofia/gateway/somegateway/somenumber
inline
Sorry for bothering!
On Sat, Nov 7, 2009 at 3:08 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
If you know the reason, why are
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