That's great. And how do I pass the parameters when calling the script
from my dialplan?
e.g.
Anthony Minessale wrote:
currently it would entail binding everything to 1 script and detecting
which kind of section it
was in the script. Making it support many would require some coding
which w
Hi,
I tried to use the Phrase API for playAndGetDigits, but it only works if the
phrase does not take any input. When an input is specified, it gives error:
2008-11-21 22:42:55 [ERR] switch_ivr_play_say.c:201
switch_ivr_phrase_macro() Can't find macro ivr_prompt,welcome.
2008-11-21 22:42:55 [WAR
After transferring a call, caller id is not changed with
effective_caller_id_number.
effective_caller_id_number affects the caller id of first INVITE message.
But after transferring a call with REFER message, new caller id is not sent
to IP phone again.
How do I do to send the new caller id
On Fri, Nov 21, 2008 at 1:29 AM, Sangwoo Jin <[EMAIL PROTECTED]> wrote:
> I have checked WIKI already.
> But I can't find variable to change caller id of IP phone after
> transferring
> a call.
> effective_caller_id_number change the caller id of initial call, not
> transferred call.
>
The wiki s
I have checked WIKI already.
But I can't find variable to change caller id of IP phone after transferring
a call.
effective_caller_id_number change the caller id of initial call, not
transferred call.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:freeswitch-
> [EMAIL PROTECTED] O
On Thu, Nov 20, 2008 at 1:50 AM, Birgit Arkesteijn
<[EMAIL PROTECTED]> wrote:
> Hi Michael,
>
> Sorry to nag you, but did you manage to experiment with the Javascript
> session in this capacity?
> I hope you reached your office. :-)
>
> Cheers, Birgit
Birgit,
After reviewing my dialplan I realize
Check out the variables Wiki page. It tells you the various variables
that affect callerid.
/b
On Nov 20, 2008, at 7:01 PM, Sangwoo Jin wrote:
> You may misunderstand my email.
> My posting in JIRA is the problem of "show calls" command of
> freeswitch.
> But, this question is that how to ch
You may misunderstand my email.
My posting in JIRA is the problem of "show calls" command of freeswitch.
But, this question is that how to change the caller id display of IP-Phone.
Do you think these are same problem?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:freeswitch-
> [E
You have a jira on this already and I'm looking at this.. and can you
please NOT hijack threads.
Click NEW message input the address. DO NOT click reply.. delete the
body and change the subject aka hijack.
/b
On Nov 20, 2008, at 6:35 PM, Sangwoo Jin wrote:
> I have tested to transfer a cal
I have tested to transfer a call with the last Freeswitch and 3 IP phones.
My test scenario is the following.
1. 202 makes a call to 201.
2. After 201 holds a call from 202, 202 can hear MOH and 201 makes a call to
200.
3. After 201 transfers a call from 202 to 200, the both displays of 202 and
I wonder if sofia is not matching the notify to the dialog so we are not
associating it with the channel.
I know for a fact sofia tears it down for us when it gets the notify.
do you have x-lite/eyebeam?
When i was testing I called into park ext with fs and did
show channels
uuid_deflect
sip:[EM
okay, this confirms my suspicion and that the fix is correct. let us
know.
Mike
On Nov 20, 2008, at 6:37 PM, kokoska.rokoska wrote:
> Michael Jerris napsal(a):
>> I added some code to protect against this segfault in svn revision
>> 10483, but it seems like you are getting an invalid presence
Michael Jerris napsal(a):
> I added some code to protect against this segfault in svn revision
> 10483, but it seems like you are getting an invalid presence packet.
> Can you update and test and make sure it doesn't segfault anymore, and
> paste a sip trace of the message that causes this i
how about the uuid_deflect with latest?
On Thu, Nov 20, 2008 at 2:33 PM, Andy Spitzer <[EMAIL PROTECTED]> wrote:
> Woof!
>
> On Thu, 20 Nov 2008 15:14:52 -0500, Anthony Minessale <
> [EMAIL PROTECTED]> wrote:
>
> > how were you calling the deflect the way that had no change?
> > every time i tri
I added some code to protect against this segfault in svn revision
10483, but it seems like you are getting an invalid presence packet.
Can you update and test and make sure it doesn't segfault anymore, and
paste a sip trace of the message that causes this issue?
Mike
On Nov 20, 2008, at 5
Brian West napsal(a):
> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace
>
Thank you very much, Brian, for the link above!
I'm not sure if I caught backtrace correctly, but I did what I can :-)
The pastebin links:
http://pastebin.freeswitch.org/6243
http://pastebin.frees
http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Getting_a_Backtrace
/b
On Nov 20, 2008, at 3:32 PM, kokoska rokoska wrote:
>
> Brian West napsal(a):
>> Including a backtrace of the segfault would be most helpful... do you
>> have it?
>>
>
> I'm sorry, but I have no idea how to get backtrace
Brian West napsal(a):
> Including a backtrace of the segfault would be most helpful... do you
> have it?
>
I'm sorry, but I have no idea how to get backtrace :-)
If you point me to some "howto" I catch it very quickly because the
segfault is reproducable.
Thanks, regards,
kokoska.rokoska
Including a backtrace of the segfault would be most helpful... do you
have it?
/b
On Nov 20, 2008, at 2:21 PM, kokoska rokoska wrote:
>
> Hi all,
>
> yesterday I start to experiment with FreeSWITCH and IPV6 and first
> what
> I saw is segfault :-)
>
> 1. I'm using FreeSWITCH 1.0.trunk (10479
Woof!
On Thu, 20 Nov 2008 15:14:52 -0500, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> how were you calling the deflect the way that had no change?
> every time i tried it sofia has taken down the channel once it completed.
Hmm...
dialplan:
Call from "207" to "[EMAIL P
Hi all,
yesterday I start to experiment with FreeSWITCH and IPV6 and first what
I saw is segfault :-)
1. I'm using FreeSWITCH 1.0.trunk (10479) runnig one of Sofia profiles
on IPv6 (sip and rtp) on Centos 5.2, 32 bit.
2. As client phone I try Sip Communicator on WinXP, 32 bit. (BTW: Do you
know
the seg was a typo (resolved)
how were you calling the deflect the way that had no change?
every time i tried it sofia has taken down the channel once it completed.
On Thu, Nov 20, 2008 at 1:16 PM, Andy Spitzer <[EMAIL PROTECTED]> wrote:
> Woof!
>
> Thanks for the changes!
>
> On Wed, 19 Nov 2
Woof!
Thanks for the changes!
On Wed, 19 Nov 2008 21:09:19 -0500, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> try latest code and see how that works.
FreeSWITCH Version 1.0.trunk (10481)
No difference with just "deflect"--the call does not clear when the REFER is
completed, nor are there a
How are your endpoints pointing to the 2 boxes? Using dns srv
records. I think the cleanest way is to change the srv records, do a
"pause" on freeswitch 1, wait for all calls and registrations to fall
off of box 1, all new registrations and calls will then be to box 2,
once it is clear yo
2008/11/20 Anthony Minessale <[EMAIL PROTECTED]>:
> Again for good measure, we do not do SIP specific forked dialing/proxy
> fantasy that alice and bob and the white rabbit are having with the
> cattipilar and his hooka. I am glad you are here to provide a check and
> balance be be sure to respec
Hi,
On Thu, Nov 20, 2008 at 10:45:42AM -0500, Michael Jerris wrote:
> What devices support srtp and video?
CounterPath eyeBeam - http://www.counterpath.com/
CounterPath Bria
Kapanga - http://www.kapanga.net
Regards, Artem
___
Freeswitch-users mailing
Thanks, David,
here are my coments:
>(a) can you not do something where you deregister them one at a time,
or in batches, on FS1
>while registering them on FS2?
A batch is a good method, and reduces the downtime of course
>(b) use method 1, but set a short period for re-registration initially,
a
We can offer to you help on build and testing this modules onto Sun
servers under Solaris 5.10 and 5.11.
Unfortunately we cannot help you with code writing bacause:
1. it needs very much time for understanding already available code;
2. now we working on a module for SS7.
We already made bui
Hi Peter,
A quick brainstorm:-
(a) can you not do something where you deregister them one at a time, or
in batches, on FS1
while registering them on FS2?
(b) use method 1, but set a short period for re-registration initially,
and then increase it once FS1's
taken down?
(c) use method 1, but t
2008/11/20 Michael Jerris <[EMAIL PROTECTED]>:
> You have a catch-22 in this situation, which media path would you want
> to pass along as you may have multiple?
Yes, this is also an undocumented "feature" of SIP and each
implementation chooses what to do. Normally the devices choose the
first ear
I'm not able to compile FS where I am, but the "-ERR no reply" response is
due to the fact that the api returns no output string, this can be solved by
adding a line: "stream->write_function(stream,"+OK\n");" just after
inserting the DTMF before the "goto done;"
I haven't realized this because
Updated with "make current" to rev 10479, but sill not good :(
Failure log sent to pastebin:
http://pastebin.freeswitch.org/6224
Cheers,
Viktor
Hirdetés (x)
Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a
kötelező biztosítások kiindulópontja!
__
currently it would entail binding everything to 1 script and detecting which
kind of section it
was in the script. Making it support many would require some coding which
would need to be done in every language module to keep them uniform and we
don't have the time for it right now. you can provid
I have the following scenario for a 2-server FS system with failover
functionality:
* I have a number of Gateways registered at FS1
* I have another number of Gateways registered at FS2
* In case I want to do maintenance on FS1 I would like all external
gateways to be registered
Thanks Michael for your quick reply!
1. Yes, we are interested in working on that very much.
2. We need to do this as quick as it possible. That's why we need
to know when you'll get this updates?
Thanks, Evgeniy.
В Чтв, 20/11/2008 в 10:28 -0500, Michael Jerris пишет:
> We still need to
post_load_switch.conf
post_load_modules.conf
are both processed after the initial load.
so you can put just xml_curl in the real one
then when it asks for these post_load you can specify what you want.
On Thu, Nov 20, 2008 at 9:21 AM, Michael Jerris <[EMAIL PROTECTED]> wrote:
> Those settings
I use "make current" for updates, I made just one a few minutes ago.
Cheers,
Viktor
Hirdetés (x)
Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a
kötelező biztosítások kiindulópontja!
___
Freeswitch-user
It said eyeBeam in the user-agent ;)
/b
On Nov 20, 2008, at 9:45 AM, Michael Jerris wrote:
> What devices support srtp and video?
___
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http://lists.freeswitch.org/mailman/listinfo/fre
I've refreshed to latest freeswitch (revision 10478), and tried
again.
Unfortunately, its still not working. I put the log from failure into pastebin.
http://pastebin.freeswitch.org/6223
Cheers,
Viktor
Hirdetés (x)
Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.b
What devices support srtp and video?
On Nov 20, 2008, at 10:31 AM, Brian West wrote:
>
> On Nov 20, 2008, at 3:35 AM, Artem Makhutov wrote:
>
>> Can somebody please check it?
>
> http://wiki.freeswitch.org/wiki/Bounty
>
> If you really need that functionality I would post a bounty.
>
> /b
>
__
We are not a proxy we are a b2bua so we are not going to send bob or carols
sdp to alice
alice and FS have a private sdp between them.
We are not doing SIP forked dialing here, we are doing FS forked dialing
that is designed to be
protocol agnostic. Keep that in mind because it's important to not
You have something much more powerful than NoOp(), you have access to
the logging API of FS (If I understand correctly).
So you can log in red in your console with CRIT level, you cal also
use the other levels like DEBUG. Search the wiki,
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_log
On Thu, Nov 20, 2008 at 7:22 AM, Brian West <[EMAIL PROTECTED]> wrote:
> "sofia_contact [EMAIL PROTECTED]" will also work at the CLI :P
>
> /b
I swear I tried that and got an unknown command! Of course, when I
tried it again it worked perfectly. My punishment will be to put this
in the wiki somew
On Nov 20, 2008, at 3:35 AM, Artem Makhutov wrote:
> Can somebody please check it?
http://wiki.freeswitch.org/wiki/Bounty
If you really need that functionality I would post a bounty.
/b
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HINT: even though it takes a few minutes longer, it is advisable to
use "make current" because it will make 100% sure that you are on the
latest SVN, no build skew, etc.
-MC
On Thu, Nov 20, 2008 at 7:18 AM, Anthony Minessale
<[EMAIL PROTECTED]> wrote:
> did you actually update again.
> I am almost
We still need to merge the latest cross platform code from sangoma. I
should have updates from them very soon, are you interested in working
on that?
Mike
On Nov 20, 2008, at 10:12 AM, Evgeniy Zolotov wrote:
> 1.
>
> We have successfully configured A-104 + WANPIPE-3.3.14 +
> FreeSWITCH( o
You have a catch-22 in this situation, which media path would you want
to pass along as you may have multiple? I think in this situation we
pass the media of the first media connection that we setup unless you
configure for fs to generate the ringback.
Mike
On Nov 20, 2008, at 10:18 AM, I
2008/11/20 Anthony Minessale <[EMAIL PROTECTED]>:
> if you want to wait for the first one to answer instead of indicate progress
> you add
> {ignore_early_media=true} to the beginning of the dial string
> data="{ignore_early_media=true}sofa/profile/[EMAIL
> PROTECTED],sofia/profile/[EMAIL PROTECT
On Thu, Nov 20, 2008 at 6:58 AM, Brian West <[EMAIL PROTECTED]> wrote:
> ${sofia_contact([EMAIL PROTECTED])} will return "error/user_not_registered"
>
> /b
>
Side note: if you wanted to check from the CLI then is this correct?
sofia status profile internal reg [EMAIL PROTECTED]
Or is there bette
Hello,
I have tested video with FreeSWITCH and found out that video with SRTP
is broken. Without SRTP the video works fine.
I have attached a capture of the SIP SDP negotiation.
I am pretty new to FreeSWITCH so I don't know where exacly the problem is.
It looks like FreeSWITCH does not understa
"sofia_contact [EMAIL PROTECTED]" will also work at the CLI :P
/b
On Nov 20, 2008, at 9:18 AM, Michael Collins wrote:
> Side note: if you wanted to check from the CLI then is this correct?
>
> sofia status profile internal reg [EMAIL PROTECTED]
>
> Or is there better way? (The above worked for
Those settings may be pulled from switch.conf before we load the
modules (including mod_xml_curl) so it would not get those from the
web-server where other parameters may pull later, which would explain
why you still see it pulled from the server. Could you open a bug on
jira.freeswitch.or
Doesn't matter.. the api call is the same via event socket or cli.
(as in they call the exact same code with NO differences)
/b
On Nov 20, 2008, at 9:07 AM, Gopala krishnan wrote:
> And also forgot to say one thing, I am using event socket.
>
> --
> Thank you with regards,
> Gopal,
>
>
It worked by default on mine... I'm on the Mac version of eyeBeam.
/b
On Nov 20, 2008, at 9:03 AM, Gopala krishnan wrote:
> Is there any dtmf setting that needs to be changed in the eyebeam
> phone?
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Freeswitch-user
did you actually update again.
I am almost positive this is fixed in the current trunk.
When I say latest you need to take me literally and update again every time.
2008/11/20 x y <[EMAIL PROTECTED]>
>
> I've done as you said.
>
> http://pastebin.freeswitch.org/6220
>
> I'll try to stay on IRC m
1.
We have successfully configured A-104 + WANPIPE-3.3.14 +
FreeSWITCH( over mod_openzap) under CentOS 5.2.
But we cann't unite SVwanpipe-i386-5.10.pkg + SVzaptel-i386-5.10.pkg
+ FreeSWITCH( over mod_openzap ) under Solaris 5.10
Does it possible?
Do we need any other packages for S
And also forgot to say one thing, I am using event socket.
--
Thank you with regards,
Gopal,
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.
>
> Is there any dtmf setting that needs to be changed in the eyebeam phone?
>
--
Thank you with regards,
Gopal,
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UNSUB
I dont understand, can you please brief me?
--
Thank you with regards,
Gopal,
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${sofia_contact([EMAIL PROTECTED])} will return "error/user_not_registered"
/b
On Nov 20, 2008, at 8:41 AM, Peter P GMX wrote:
> Is there a way to check if a user is registered locally?
>
> I have the following Scanrio: If a call comes in from an external
> gateway then Freeswitch1 shall check i
Your phone must not be rendering them. I just tested this and its
working fine. X-Lite/eyeBeam
/b
On Nov 20, 2008, at 8:55 AM, Gopala krishnan wrote:
> Hi,
>
> I am using the event socket in freeswitch with audiocodes, and the
> client as a softphone.
__
Hi,
I am using the event socket in freeswitch with audiocodes, and the client
as a softphone.
--
Thank you with regards,
Gopal,
___
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you have to remember that just because you send DTMF to a phone via
RTP or SIP INFO the phone doesn't have to render them. The best way
to test this is with an ATA since it will render the tones most
likely. Many ip phones do NOT render the tones to the speaker.
/b
On Nov 20, 2008, at 8:3
I've done as you said.
http://pastebin.freeswitch.org/6220
I'll try to stay on IRC more.
Thx,
Viktor
Hirdetés (x)
Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a
kötelező biztosítások kiindulópontja!
___
Is there a way to check if a user is registered locally?
I have the following Scanrio: If a call comes in from an external
gateway then Freeswitch1 shall check if the destination UA is registered
locally. If not then redirect (302) the call to Freeswitch2.
But how to determine that the UA who shal
What if I want to use one binding for "directory", one for
"configuration" and one for "dialplan"?
While we are at it, how can I pass parameters so that I can "fill up" my
%XML_REQUEST when
the perl script is called from the xml dialplan? e.g. :
On Thu, Nov 20, 2008 at 1:50 AM, Birgit Arkesteijn
<[EMAIL PROTECTED]> wrote:
> Hi Michael,
>
> Sorry to nag you, but did you manage to experiment with the Javascript
> session in this capacity?
> I hope you reached your office. :-)
>
> Cheers, Birgit
>
>
Sorry, day job was rough yesterday and I di
Hi,
I was trying this dtmf stuff for me also its not working. whenever i used
to send the dtmf you know i get a beep. whats wrong?
--
Thank you with regards,
Gopal,
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if you want to wait for the first one to answer instead of indicate progress
you add
{ignore_early_media=true} to the beginning of the dial string
On Thu, Nov 20, 2008 at 3:13 AM, Iñaki Baz Castillo <[EMAIL PROTECTED]> wrote:
> Hi, I read in:
> http://wiki.freeswitch.org/wiki/Dialplan_Recipes
>
no the languages only have one binding.
Do you really need more than one binding?
On Thu, Nov 20, 2008 at 6:20 AM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hi,
>
>Is there a way to declare more than one script with its binding in
> perl.conf.xml?
> Because from what I understood by read
Disable 100rel on the sofia profile or update to latest SVN trunk. We
have an issue on jira about this already and are working with the
sofia team on this issue. The solution is to disable 100REL and it'll
quit doing that.
/b
On Nov 20, 2008, at 3:13 AM, Iñaki Baz Castillo wrote:
Hi, I
You would need to setup an ACL to let that in without auth.
http://wiki.freeswitch.org/wiki/ACL
/b
On Nov 20, 2008, at 7:11 AM, paulo leonardo wrote:
ERROR
2008-11-20 10:58:29 [WARNING] sofia_reg.c:1247
sofia_reg_parse_auth() can't find user [EMAIL PROTECTED]
You must define a domain called
Good morning list,
i have a litle problem again ;-) !
1 - I have a freeswitch (192.168.170.101) running on port (5060, 5061); -
that's ok
2 - My freeswitch is registred to Proxy SIP (SER) (XXX.XXX.XXX.XXX); -
that's ok
3 - I have a user local (1000) registred in my freeswitch; - that's ok
4 - Wh
Hi,
Is there a way to declare more than one script with its binding in
perl.conf.xml?
Because from what I understood by reading the documentation, is that
there are
no different sections to define different perl scripts with bindings
like for example in the
xml_curl.conf.xml :
Figured that out by myself. One has to raise the console debug output to
"debug".
[EMAIL PROTECTED] wrote:
Tried that, but the output of a simple data="hello" /> does not appear in my console.
I verified that the context that the eval is in gets executed.
I have the loglevel set to "debug" in
Tried that, but the output of a simple data="hello" /> does not appear in my console.
I verified that the context that the eval is in gets executed.
I have the loglevel set to "debug" in my switch.conf.xml by the way.
Any help?
Michael Collins wrote:
Try eval
http://wiki.freeswitch.org/wiki/
Hi Michael,
Sorry to nag you, but did you manage to experiment with the Javascript
session in this capacity?
I hope you reached your office. :-)
Cheers, Birgit
On 18/11/08 18:10, Michael S Collins wrote:
> Birgit,
>
> I'm almost to my office. I will give you more info soon. I have not
> use
Hi, I read in:
http://wiki.freeswitch.org/wiki/Dialplan_Recipes
the following:
---
= Forked dial example =
Forked dial is when you want to attempt to ring 2 destinations at the
same time. Freeswitch will attempt to call both bridge options
simultaneously. The first bridge leg
Hi,
I tried using curl to configure switch.conf to set the min rtp port and max
rtp port. The request did arrive in webserver and the response is correctly
returned.
However, Freeswitch still did not use the port in that specified range. So,
I manually modified the switch.conf under autload_conf
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