In my case the 1001 resides in -
/usr/local/freeswitch/conf/directory/default/1001.xml
And you set the Caller Name and ID by adding -
On Tue, Jun 23, 2009 at 12:26 AM, Edward Q. q.edw...@gmail.com wrote:
Edmar
Eso esta en freeswitch/conf/vars.xml en ese archivo.
If i am not mistaken
Hi.
I have been searching for an alternative PBX to asterisk (which has not
been all that stable) to run on Dragonfly BSD. I spent a fair amount of
time a couple months ago trying to compile freeswitch without success.
I have since tried Yate, but it consumes 85-95% of the cpu when idle
(not
Hello All,
I am an absolute newbee in the voip world but have a project where I believe
freeswitch will work and need very, very basic guidance
on how to setup a testbed in a thin client environment.
I am using K12LTSP 5EL (Centos5) for my terminal server/host testbed and
utilize
Sorry Edmar
I missundertood you .. I thought you wanted to change the number showing
once you were going out not the 1001.xml file.
In this case Harmeet is right. There you have those values to to make the
changes.
My bad.
Ed
On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh harm...@litatel.com
Actually the extension_caller_id=Extension 1001 and
extension_caller_number=1001 is set as Harmeet says but the same issue
FreeSwitch the caller name and the number is 000 i just want 1001 the
caller number and the id
Edmar
Edward Q. wrote:
Sorry Edmar
I missundertood you .. I thought
depending how do you make out going call.
On Jun 23, 2009, at 2:39 PM, Edmar Cruz wrote:
Actually the extension_caller_id=Extension 1001 and
extension_caller_number=1001 is set as Harmeet says but the same issue
FreeSwitch the caller name and the number is 000 i just want
1001 the
Hi!
I am trying to call from my corporate network (firewalled) using Gtalk
to Freeswitch. I am not getting any audio.
In the logs I see that mod_dingaling is using my internal corporate IP
address which is not publically addressable.
2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634
Hello,
I once found in the wiki a page explaining how to substring a channel
variable,
something like
@[intra]lanman 12345 would be 345 if you do ${var:2}
I can't find that page on the wiki anymore, any hint on were it could
be? :-)
Also do you think it could be useful to extend this
Check your dialplan where you call bridge to gateway to make outgoing
calls. Stick in the following lines before the bridge call -
action application=set
data=effective_caller_id_number=${effective_caller_id_number}/
action application=set
You are way off base in a few places, let me see if I can clarify a bit.
Here are at least 2 pointers:
1) The release tarballs do not come with bootstrap because they already are
bootstrapped.
2) FreeSWITCH does not depend on system libs so all the stuff about apr is
barking up the wrong tree.
try adding this to your jingle profile in client.xml
param name=candidate-acl value=wan/
then edit acl.conf.xml and add this list
list name=wan default=allow
node type=deny cidr=10.0.0.0/8/
node type=deny cidr=172.16.0.0/12/
node type=deny cidr=192.168.0.0/16/
/list
this tells
play with it from the cli
freeswitchglobal_setvar foo=12345
API CALL [global_setvar(foo=12345)] output:
+OK
freeswitch eval ${foo:2:1}
API CALL [eval(${foo:2:1})] output:
3
freeswitch eval ${foo:2:3}
API CALL [eval(${foo:2:3})] output:
345
freeswitch eval ${foo:3:2}
API CALL [eval(${foo:3:2})]
No it snot because of this.. you have to understand how Jingle works
and if you notice it has three candidates
2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked
an IP [146.xx.xx.xx]
Its already picked this one, maybe a packet capture would clear this up.
/b
On
Hi Michael,
Using loopback solves my problem. Thanks a lot.
There is a strange thing i observed though. I need to paste my extension in
the default.xml file. Having them in the default directory isn't enough. Is
that normal?
Max.
___
On Jun 23, 2009, at 7:04 AM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Hi Michael,
Using loopback solves my problem. Thanks a lot.
There is a strange thing i observed though. I need to paste my
extension in the default.xml file. Having them in the default
directory isn't
The file is located under /usr/local/freeswitch/conf/dialplan/default/. The
name is: mysocket.xml. The content is:
include
extension name=mysocket
condition field=destination_number expression=^242.*
break=on-true
action application=socket data=192.168.50.66:1 full
If you can supply a patch to expose this as a config option for us it
would be appreciated. Patches can be posted to http://jira.freeswitch.org
.
Mike
On Jun 17, 2009, at 3:22 PM, Muhammad Shahzad wrote:
Ok, thanks, i will take care of it in my code where necessary.
Thank you.
On Thu,
How are you configuring your polycom?
On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb larc...@yahoo.com wrote:
I’m sorry Chris, but I don’t know where the look for the “global sip.cfg
and mac/phone specific cfg” settings. I also looked for digitmap but could
find nothing.
Can you be more
On Mon, Jun 22, 2009 at 9:31 PM, murrah boswellotrc...@isp-systems.net wrote:
Hello All,
I am an absolute newbee in the voip world but have a project where I believe
freeswitch will work and need very, very basic guidance
on how to setup a testbed in a thin client environment.
I think this
Try turning up your logging level to debug to see why the call is
hanging up.
Mike
On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote:
My freeswitch has a mysql database consists of freeswitch tables,
registrations and nibblebill on mysql configured it correctly and
working...
Issue is when
if you need to use the same tags, we should be using the whole same nh
in the code. There is code to do this by call uuid but I can't recall
if thats for NOTIFY or INFO. If its the wrong one, we should add teh
same for what you need.
Mike
On Jun 21, 2009, at 6:05 AM, Christian
Can anyone suggest a good way to do the following;
1 - I want to be alerted [via the event API] to a new incoming call.
2 - I do not want that incoming call to be answered but just stay
ringing.
3 - Then via the API I want to send a redirect command to push the call
off to a new destination
On Jun 23, 2009, at 11:31 AM, Richard Lamkin wrote:
Can anyone suggest a good way to do the following;
1 - I want to be alerted [via the event API] to a new incoming call.
See below.. ie park. You should get an event via event socket you can
decide what to do.
2 - I do not want that
if you turn up the debug logs it should tell you why.
On Jun 22, 2009, at 11:38 PM, Edmar Cruz wrote:
Nope. I just want to call a mobile number with no register number.
Brian West-3 wrote:
I'm going to guess you're calling a registered user? If so replace
the @ with %
/b
On Jun 22,
Hi Guys,
Scratching my head on this one, under load FS is not playing an audio
file, OR and lua script is not getting executed. Not all the time, just
some. I've changed ulimit -n to 9 but no diff, and ideas where else
I might look?
Regards,
Basically read the polycom manual ... it is the polycom producing the
dialtone and deciding when to dial the number you are entering, using its
own dialplan and interdigit timers.
On Tue, Jun 23, 2009 at 10:38 AM, Rupa Schomaker r...@rupa.com wrote:
How are you configuring your polycom?
On
Does the log show anything? if the lua script fails to execute it should
appear in freeswitch.log
On Tue, Jun 23, 2009 at 9:45 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
Scratching my head on this one, under load FS is not playing an audio file,
OR and lua
Hmm,
Looking at console I'm seeing this, does this offer any additional clues
to anyone?
2009-06-23 17:44:29.856574 [CRIT] mod_local_stream.c:234 Leaking stream
handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1015]
2009-06-23 17:44:33.620372 [CRIT] mod_local_stream.c:234 Leaking
Via a web browser.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 23, 2009 8:39 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Polycom configuration
Ok, most of us configure the polycoms via a provisioning interface. usually
ftp or http.
Anyway, when using the web interface, you want to look at:
Goto the web interface, Click on SIP.
Scroll down to the Local Settings section and you need to modify digitmap
and digitmap timeout. the syntax
Are you making many calls share a single local_stream?
This error usually means a handle open to a local_stream is not reading from
that stream source, such as if you paused during playback of a local_stream.
They are only a real issue if you are getting them with no calls up.
On Tue, Jun 23,
hi
thank you for your reply
how can we procced?
br
On 2009-06-23 18:20, Michael Jerris wrote:
if you need to use the same tags, we should be using the whole same nh
in the code. There is code to do this by call uuid but I can't recall
if thats for NOTIFY or INFO. If its the wrong one, we
They're reading an audio file from a ram disk. Wouldn't have thought
that this would cause a problem or am I wrong. Running at around 400
concurrent calls
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
the lines you pasted indicate something stuck playing local_stream (hold
music) and not actually reading it.
playing a file from a ram disk with 400 is for sure fine. I have done many
thousand before.
if you turn up your debugging do you see anything else about the box going
wrong?
On Tue, Jun
Hi,
I've got some news on this. When i move my extension to a different
directory (/usr/local/freeswitch/conf/dialplan/sockets/) and add an include
element at the very sample place where the default is included, things work
just as expected. That is, my default.xml now include following:
I love it when users figure it out AND report back what they did to solve
the issue! Nice work.
-MC
On Tue, Jun 23, 2009 at 12:11 PM, Max Bridgewater max.bridgewa...@gmail.com
wrote:
Hi,
I've got some news on this. When i move my extension to a different
directory
I know you are all eagerly anticipating the arrival of the coolest
conference around! We want to make sure that everyone is aware of the
following information:
* The last day to get the early-bird registration is Wednesday, July 1.
Early birds get into the conference for only $499. After July 1
Also, if and when you get this working please send a message to the list.
I'd like to make sure that your setup gets documented on the wiki.
-MC
On Tue, Jun 23, 2009 at 6:44 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
try adding this to your jingle profile in client.xml
param
Curious - what kinds of SIP phones do the clients support? Have you decided
what you'd be using?
-MC
On Mon, Jun 22, 2009 at 8:31 PM, murrah boswell otrc...@isp-systems.netwrote:
Hello All,
I am an absolute newbee in the voip world but have a project where I
believe freeswitch will work and
Did you ever get resolution on this? If not, join us on IRC and we'll
discuss it.
-MC
On Sat, Jun 20, 2009 at 6:28 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
I have a problem with voicemail in that freeswitch fails to let users
leave their message. Something wrong in the
You're using native files and you have no native files in PCMU...
/b
On Jun 23, 2009, at 3:15 PM, Michael Collins wrote:
Did you ever get resolution on this? If not, join us on IRC and
we'll discuss it.
-MC
On Sat, Jun 20, 2009 at 6:28 PM, Mark Campbell-Smith mcampbellsm...@gmail.com
Thanks to Rupa and Chris for this help. I didn't know enough to understand
Chris was pointing me to the Polycom phone rather than FS. I would never
have figured this out.
Are Polycoms the only SIP phones which have this feature?
Lars
From: freeswitch-users-boun...@lists.freeswitch.org
Nope other phones have this also.
/b
On Jun 23, 2009, at 4:57 PM, Lars Zeb wrote:
Thanks to Rupa and Chris for this help. I didn’t know enough to
understand Chris was pointing me to the Polycom phone rather than
FS. I would never have figured this out.
Are Polycoms the only SIP phones
Every sip phone I've used has this feature. Even ATAs -- though they tend
to ship with more forgiving defaults.
On Tue, Jun 23, 2009 at 4:57 PM, Lars Zeb larc...@yahoo.com wrote:
Thanks to Rupa and Chris for this help. I didn’t know enough to
understand Chris was pointing me to the Polycom
Did anyone have any suggestions on this? Just to reiterate...
- 8000 is a local extension defined in the default dialplan... see
http://pastebin.freeswitch.org/9450 for definition
- didn't work: originate sofia/default/8...@192.168.10.35
txfax(storage/fax/test.tif) ... see
Brian,
Thank you for suggesting I try PARK.
I tried PARK but unfortunately it sends out a 183 with SDP which stops
the originator hearing ringing (ring back).
If you know of a way to park without sending a 183 that would solve my
problem.
Regards
Richard Lamkin
Is 8000 just a dialplan extension? I'm curious about the whole
8...@192.168.10.35 thing. I doubt that's necessary. For kicks try something
like this:
originate loopback/8000 txfax(storage/fax/test.tif)
That will drop the A leg right into extension 8000.
-MC
On Tue, Jun 23, 2009 at 3:12 PM, Tim
Curious - what kinds of SIP phones do the clients support? Have you decided
what you'd be using?
-MC
I am still experimenting! I have a zoiper 2.0 installed on one of my test
clients. zoiper seems to work fine, so now I am attempting
to get the freeswitch/zoiper interface working. I will
/freeswitch-users/attachments/20090623/6df939cc/attachment.html
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UNSUBSCRIBE:http
Where can i find this logs?
Michael Jerris wrote:
Try turning up your logging level to debug to see why the call is
hanging up.
Mike
On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote:
My freeswitch has a mysql database consists of freeswitch tables,
registrations and nibblebill on
'. Stop.
gmake[7]: *** [all-recursive] Error 1
Making all in packages
gmake[6]: *** [all-recursive] Error 1
gmake[5]: *** [all] Error 2
gmake[4]: ***
[/u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la]
Error 2
gmake[3
Thanks Anthony.
I am getting closer. I had to put in the 146 address, which is the
firewalled address I get at work. The problem now is that when the
call is bridged, I do not hear audio.
2 scenarios:
1 - the local extension is not registered. There is two way audio -
I hear the voicemail in
On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote:
Ok. I did this.
Compilation still failed but there are significant improvements since
the last time.
Here is what I did and the results:
It looks like some the games that sofia plays with errno makes Dragonfly
unhappy. I
Thanks Brian, but still no luck with the email.. I have configured exim4 so
that I can send messages from the command line using 'mail' command and
these are sent successfully.
I still get a core dump in the log when freeswitch is trying to send the
mail:
/bin/cat: write error: Broken pipe
sh:
Please see the debugging pages on the wiki
On Jun 23, 2009, at 10:10 PM, Edmar Cruz darklio...@yahoo.com wrote:
Where can i find this logs?
Michael Jerris wrote:
Try turning up your logging level to debug to see why the call is
hanging up.
Mike
On Jun 19, 2009, at 7:13 AM, Edmar Cruz
/ports/freeswitch-20090623/work/
freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-
ua.la] Error 2
gmake[3]: *** [mod_sofia-all] Error 1
gmake[2]: *** [all-recursive] Error 1
Making all in build
+ FreeSWITCH Build Complete ---+
+ FreeSWITCH has been
Dear All,
Look like nibblebill does't work with multiple gatreway.
I try
action application=set
data=nibble_account=0838833133/
action application=bridge
data={absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734...@203.xxx.xxx.xxx
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