Sure, I'll try that. I'm just building freeswitch-snapshot that I
downloaded from files.freeswitch.org
I also experience, when bridging a
call from an FXS to FXO the call is cut after a random time (this does not
appear when bridging SIP to FXO). Might this upgrade fix this problem also
?
Fr
Dear All,
I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card). I
use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) and
freeswitch 1.0.5 pre 7
My Server Intel(R) Core(TM)2 Quad CPUQ6600 @ 2.40GHz and 2
GB memmory.
when i start freeswitch if e
Hi,
I found that with bypass_media=true, freeswitch would change c= to FS's own
IP.
I think this is a misconfiguration. Does anyone know what config could have
caused that?
thanks,
jb
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch
In the following trace,102 is FS IP, 104 is calling party and 13 is
called party.
with bypass_media, FS still changesc=IN IP4 192.168.1.102
Any idea why?
freeswi...@localhost.localdomain> recv 951 bytes from
udp/[192.168.1.104]:5060 at 22:56:33.782715:
---
On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote:
> Dear All,
>I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card). I
> use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) and
> freeswitch 1.0.5 pre 7
>
>My Server Intel(R) Core(TM)2 Quad CPU
What is the jira bug number on this voicemail email issue? I don't
recall seeing it.
Mike
On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine
wrote:
> > Are you on SVN trunk? As far as I recall the callee_id_number/name
> stuff isnt in 1.0.4.
>
> No, because the SVN has problems with Emailing th
The only way this would happen would be if this is set to proxy media
not bypass. Are you setting both?
Mike
On Dec 1, 2009, at 10:08 AM, Juan Backson wrote:
In the following trace,102 is FS IP, 104 is calling party and 13
is called party.
with bypass_media, FS still changesc=IN
2009/12/1 Moises Silva
> On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote:
>
>> Dear All,
>>I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card).
>> I use zaptel driver from http://e400p.phoniceq.com/driver/ (1.4.12.1) and
>> freeswitch 1.0.5 pre 7
>>
>>My Serv
It is MODAPP-373.
Thanks, __yehavi:
2009/12/1 Michael Jerris
> What is the jira bug number on this voicemail email issue? I don't
> recall seeing it.
>
> Mike
>
> On Dec 1, 2009, at 2:04 AM, Yehavi Bourvine
> wrote:
>
> > > Are you on SVN trunk? As far as I recall the callee
On Sun, Nov 29, 2009 at 10:06 AM, Puskás Zsolt wrote:
> Hi Guys!
>
> I'm using the latest svn (15711) with the default xml config. Only modified
> cdr_csv.conf.xml the line to name="legs"
> value="ab"/>
>
> Here is what i do:
>
> 1. 1000 calls 1001 (1001 answers the call)
> 2. 1001 do blind tra
On Wed, Nov 25, 2009 at 9:42 PM, Joseph L. Casale wrote:
> I need to make faxing easy for some very computer illiterate folk. I am
> using an email
> service and going to use procmail to print anything incoming automatically
> but they cant
> get the hang of scanning to an email app, so I am goin
Dear All:
I try to register from a feature code of an analog phone like Elastix.
It is useful for DID.
There is an idea that I use dynamic dialplan to implement it and it's not
really register to FS.
And I need to run script to insert or delete dialplan to the database when
dialed.(Input logon's E
I'm not entirely sure that I understand your question, so I am going to ask
a few questions to clarify.
Are you looking to have analog telephones receive incoming calls, like in a
call center? Is that why the user of the analog phone would need to log in
and log out?
I would recommend checkout out
I attempted to do a make current with revision 15739, but some of the Sofia
source files will not compile with revision 15739. Those source files were not
changed between revisions 15738 and 15739. I am using GCC 4.1.2 to compile
FreeSWITCH. I used the following to get revision 15738, which was
On Tue, Dec 1, 2009 at 12:02 PM, Dome Charoenyost wrote:
>
>
> 2009/12/1 Moises Silva
>
> On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote:
>>
>>> Dear All,
>>>I got problem about openzap. I plan to use 4E1 R2 (Phone EQ card).
>>> I use zaptel driver from http://e400p.phoniceq.com
the updating of the display code is significantly improved in trunk.
Please figure out your email problem and use that. Most likely you need an
alternate configuration.
What mailer client are you using in switch.conf.xml ?
On Tue, Dec 1, 2009 at 11:11 AM, Yehavi Bourvine
wrote:
> It is MODAPP-37
yes he did you can see it in his trace.
you can not use both of them together..
On Tue, Dec 1, 2009 at 10:46 AM, Michael Jerris wrote:
> The only way this would happen would be if this is set to proxy media not
> bypass. Are you setting both?
>
> Mike
>
> On Dec 1, 2009, at 10:08 AM, Juan
I think I just fixed this a few minutes ago, it is running test builds on the
build servers now to verify.
On Dec 1, 2009, at 2:19 PM, John Platts wrote:
>
> I attempted to do a make current with revision 15739, but some of the Sofia
> source files will not compile with revision 15739. Those
Dear Michael:
Yes, I want to distribute a real phone number to each analog phone (direct
inward dialing).
One FXS one analog phone.
I guess the user maybe want to add a number mapping this FXS port.
Thank you, Michael. mod_xml_curl is really a powerful module. :D
-- Forward --
Fr
I have tried to do a blind transfer from a phone that is registered with
FreeSWITCH, and it will fail, even when proxying and media bypass are enabled.
Details about this issue can be found here:
http://jira.freeswitch.org/browse/MODENDP-272
__
2009/12/2 Moises Silva
>
>
> On Tue, Dec 1, 2009 at 12:02 PM, Dome Charoenyost wrote:
>
>>
>>
>> 2009/12/1 Moises Silva
>>
>> On Tue, Dec 1, 2009 at 8:43 AM, Dome Charoenyost wrote:
>>>
Dear All,
I got problem about openzap. I plan to use 4E1 R2 (Phone EQ
card). I use zap
Esben Stien writes:
> trying to record and play back video
So nobody is using video with freeswitch?.
--
Esben Stien is b...@e s a
http://www. s tn m
irc://irc. b - i . e/%23contact
sip:b0ef@ e e
jid:b0ef@
I'd like to set up an extension that would just ring forever.
When a person calls this extension, it would ring until the end of
times.
I've tried several ways to do this, without luck, and I don't find any
information on this on the wiki.
Any pointers how to do this?.
--
Esben Stien is b..
Hi,
I'm trying to migrate from Asterisk to FreeSWITCH (really like the way how
it can be programmed), but ran into one issue with sound quality that I just
cannot workaround by myself. I would describe the sound problem as being
"choppy". From time to time small portions of the other party's voic
I should also add, after browsing through some topics here, that my SIP
provider sends 172-byte RTP frames, which is in accordance with ptime:20
that it gives to FreeSWITCH.
eaf wrote:
>
> Hi,
>
> I'm trying to migrate from Asterisk to FreeSWITCH (really like the way how
> it can be programmed
do you want to generate ringback forever or no?
The calling party will probably abort at some point.
put both of these in your context
then use one of these 2 sets of actions in your main ext
On Tue, Dec 1, 2009 at 4:30 PM, Esben Stien wrote:
> I'd like t
>In this case, the $1 will only contain whatever is in the parens in your
>expression, i.e.
>
>
>What do you have for your expression?
>-MC
Well, untested of course as I am busy with school:) But what I wrote up to try
at
Christmas (with your addition) was:
On Tue, Dec 1, 2009 at 2:09 PM, Joseph L. Casale
wrote:
> >In this case, the $1 will only contain whatever is in the parens in your
> expression, i.e.
> >
> >
> >What do you have for your expression?
> >-MC
>
> Well, untested of course as I am busy with school:) But what I wrote up to
> try at
> C
linksys has had a bug for eons that can be fixed by setting the ptime (or
rtp packet size in their terms)
in it's firmware to .20 instead of .30
Asterisk does not use async RTP like we do so it's never a problem
you can disable the timer by setting the channel var rtp_timer_name=none or
sofia para
Just remove the terminating '/' at the end of the second condition tag
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, December 01, 2009 5:23 PM
To: freeswitch-users@lists.freeswit
>Just remove the terminating '/' at the end of the second condition tag
>
>
I tried to see based on examples if it was obvious to me why that should not be
there
but it didn't jump out:) Cuold you explain that please?
The gateway will arrive at the end of the week, but I probably won't get t
"Joseph L. Casale" said:
> >Just remove the terminating '/' at the end of the second condition tag
> >
> >
>
> I tried to see based on examples if it was obvious to me why that
should not be there
> but it didn't jump out:) Cuold you explain that please?
It is a multi-line condition. If a c
Anthony Minessale writes:
> do you want to generate ringback forever or no?
Yes, I want the dialing party to get a ring forever. This is because I
cannot transfer the party to my SIP phone, because my SIP phone is
broken for incoming calls.
I'll solve it by letting the party hear a ring and th
are you on older REV?
try answering first to compare.
On Tue, Dec 1, 2009 at 6:29 PM, Esben Stien wrote:
> Anthony Minessale writes:
>
> > do you want to generate ringback forever or no?
>
> Yes, I want the dialing party to get a ring forever. This is because I
> cannot transfer the party to
Anthony Minessale writes:
> are you on older REV?
I think I'm on 15334
> try answering first to compare.
?, that's what I do:
--
Esben Stien is b...@e s a
http://www. s tn m
irc://irc. b - i . e/%23contact
Hi folks,
I'm doing a little survey to get an idea of what everyone prefers to use for
their operating environment, like 32 vs. 64 bit, Linux vs. Windows, etc.
Please log in to the main page and check out this node:
http://www.freeswitch.org/node/206
Select the environment that you use the most
upgrading always helps *something* not sure. but that is where we have to
start because we have changed that code alot.
On Tue, Dec 1, 2009 at 2:37 AM, François Legal wrote:
> Sure, I'll try that. I'm just building freeswitch-snapshot that I
> downloaded from files.freeswitch.org
>
> I also ex
Thanks. I tried that... Just forcing SPA to 20ms didn't change anything. Just
installing SVN trunk didn't fix it either, but setting that option afterwards
surely did the trick.
One thing I've noticed while staring at the console is that it *looks like*
that w/o the new setting the stuttering happ
To expand on what Russell said: XML always has a start and an end tag,
possibly with other stuff in between.
... content ...
If there is no content, you get:
Or, on one line, . You're allowed to abbreviate that to just
.
So in your case:
<--- these are themselves abbreviations of !
>To expand on what Russell said: XML always has a start and an end tag,
>possibly with other stuff in between.
>
> ... content ...
>
/snip
Ahh, so must all the actions be contained within at least one condition tag as
content, or could have I kept the
last "/" on the last condition and dropped
Wow... Thinking about this timer setting and about how it converted
send()/recv() from non-blocking to blocking, I straced freeswitch when it was
supposed to be idle. It never pauses! It keeps going in and out of select()
every millisecond! Why??
-- Original Message --
Received: Tue, 01 De
Joseph L. Casale wrote:
> Ahh, so must all the actions be contained within at least one condition
> tag as content,
Yes.
> or could have I kept the
> last "/" on the last condition and dropped the line?
No. Think of the tags as a begin/end pair that surround the content. If there
is no content
Indeed, all actions must be contained with a condition tag.
FS just processes the conditions inside the extension top-to-bottom, so:
if the top condition (without actions in it) doesn't match, it stops
processing that extension.
-Original Message-
From: freeswitch-users-boun...
Hi all,
Did anything ever progress with this? Is there an option for OSP in
FreeSWITCH?
Thanks,
Max
On 10/31/08 7:58 AM, Anthony Minessale wrote:
> We're here all the time if you want to collaborate on it.
> We have 100+ users and developers in our irc channel and on this list so
> it should no
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