On 1/9/09, Kristian Kielhofner wrote:
> On 1/8/09, Miles Keaton wrote:
> > I'm new to FreeSWITCH and hope I could ask the community's suggestion
> > or experience with how the location of your FreeSWITCH server affects
> > the quality of the VoIP call.
> >
> > I'm using JunctionNetworks (
On 1/8/09, Miles Keaton wrote:
> I'm new to FreeSWITCH and hope I could ask the community's suggestion
> or experience with how the location of your FreeSWITCH server affects
> the quality of the VoIP call.
>
> I'm using JunctionNetworks (http://pstn.junctionnetworks.com/) for SIP
> trunking.
Jeng,
Your condition expressions are not right. Could you describe what you
hope to accomplish with those two expressions? Once you get the
regular expressions figured out then it should all work.
-MC
Sent from my iPhone
On Jan 8, 2009, at 6:44 PM, Jian Yuan Peng wrote:
Hi,
Can you he
I'm new to FreeSWITCH and hope I could ask the community's suggestion
or experience with how the location of your FreeSWITCH server affects
the quality of the VoIP call.
I'm using JunctionNetworks (http://pstn.junctionnetworks.com/) for SIP
trunking. Their servers are in NYC.
I'm trying to decid
Hi, Can you help me to look at why I got this error. See this link for
detailhttp://pastebin.com/m3be7932cAlso, one more question, what is most used
java sip library connected with fs for make phone calls. I am looking at
jain-sip now. I open for any.Thnaks,-Jian Yuan Peng
__
one of hidden feature of freeswitch ;)
http://wiki.freeswitch.org/wiki/Mod_vmd
On Thu, Jan 8, 2009 at 10:30 PM, Kristian Kielhofner
wrote:
> On 1/8/09, Adam Wilt wrote:
>> Hi, I have two issues I'd appreciate some help with.
>>
>> A) I'm testing VMD and I'm getting a success rate of well under
On 1/8/09, Adam Wilt wrote:
> Hi, I have two issues I'd appreciate some help with.
>
> A) I'm testing VMD and I'm getting a success rate of well under 50%. I know
> part of the reason is that some of the voicemail beeps it's encountering are
> very short in length (I've noticed this for T-Mobile a
Hi, I have two issues I'd appreciate some help with.
A) I'm testing VMD and I'm getting a success rate of well under 50%. I know
part of the reason is that some of the voicemail beeps it's encountering are
very short in length (I've noticed this for T-Mobile and Sprint voicemails,
and there may be
I'm trying to program an extension in the dialplan to do an intercom
announce. I read through the wiki and wrote the following based on what
I thought I understood from it, but it's not working the way I expect:
Yes, I think so. It seems that it just doesn't work while doing an api call
such as session.execute("playback", "/path/to/file.wav")
Is this the correct behaviour? Are calls made with execute "blocking"?
Thanks!
Erik
On Thu, Jan 8, 2009 at 4:01 PM, Michael Collins wrote:
> Sorry for the late
mod_dptools/event doesn't have a wiki page...
Anyone care to give me some syntax so I can make one?
Thanks!
--
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
___
Freeswitch-user
yes it will be there in ext 500. any vars in {} are set when the channel is
created and are present the rest of
the life of the channel.
you can do regex on it in the dialplan by referencing it with a ${} in the
field
...
and any apps you execute will also be able to get the value of ${myvar}
Here's an example:
bgapi originate {myvar=blah}sofia/gateway/gw/19415551212 &transfer(500
XML default) 9185551212 60
So... I'd like to be able to read ${myvar} from the the 500 extension
in the default context.
On 1/8/09, Anthony Minessale wrote:
> the variables should still be set on the chan
Sorry for the late followup. Did you ever get this working? (I'm not a
Python guy so it's a bit out of my area of interest/expertise).
-MC
On Wed, Jan 7, 2009 at 4:09 PM, Erik Wickstrom wrote:
> Hi all,
>
> I'm trying to get a setInputCallback function working with mod_python. I'm
> using a curr
So the issue is not happening right now? If it is then we would want
you to pastebin the debug output of a few test calls. If not, please
watch to see if this issue happens again and then report it back with
a debug trace.
-MC
On Thu, Jan 8, 2009 at 8:46 AM, ahgindia wrote:
>
> Hi all,
> I have a
Can you pastebin a complete call history where the first call works,
gets hung up and then the second call comes in? I would like to see
the entire d-chan trace.
-MC
On Thu, Jan 8, 2009 at 9:15 AM, Peter P GMX wrote:
> We have a strange Issue on Openzap with a Digium PRI card (Digium TE220
> inkl
the variables should still be set on the channel in your application?
you mean a remote call-leg or the channel itself once it's in the app?
On Thu, Jan 8, 2009 at 5:50 PM, Michael Collins wrote:
> On Thu, Jan 8, 2009 at 9:06 AM, Kristian Kielhofner
> wrote:
> > On 1/8/09, Anthony Minessale wr
Could you go to pastebin.freeswitch.org and paste your config changes?
-MC
On Thu, Jan 8, 2009 at 7:32 AM, bahbie wrote:
>
> I think there is some problem with the bridge and context. I get a message
> from FS saying the person you are trying to contact is not available.
> --
> View this message
On Thu, Jan 8, 2009 at 9:06 AM, Kristian Kielhofner
wrote:
> On 1/8/09, Anthony Minessale wrote:
>> put them in {} comma separated.
>>
>>
>> {foo=bar,test=true}sofia/default/u...@dom.com
>>
>> if you are doing forked dial you can set them per leg with []
>>
>> [var1=foo]sofia/default/u...@dom.com
Snom has already responded to my issue and are going to be providing
me a firmware for testing this tomorrow.. its still going to default
to the WRONG way.. but has a toggle to turn it to the right way.
/b
On Jan 8, 2009, at 3:23 PM, Kristian Kielhofner wrote:
>
> Yes, it should use the deci
Thanks for posting the solution. I was following the issue with much
curiosity!
On Thu, Jan 8, 2009 at 1:47 PM, Matthew Kaufman wrote:
> Anthony Minessale wrote:
> > This is a very unique problem as many people get this basic situation
> > working daily so
> > it must be a network issue of some
On 1/8/09, Helmut Kuper wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
>
> Hi Brian,
>
> sorry, got lost ... It's a bug in Snom and AVM's phones?
>
> Maybe we misunderstood. RFC 2833 emtioned in chapter 3.10 following DTMF
> Events:
>
> 3.10 DTMF Events
>
>Table 1 summarizes
If i use the internal profile with the following two lines
param name="ext-rtp-ip" value="$${external_rtp_ip}"
param name="ext-sip-ip" value="$${external_sip_ip}"
I can use the trunk however I get the following two errors on the cli.
Don't know if they are serious errors
2009-01-08 20:49:59 [
On the examples to configure offsite phones I read the following:
Below are steps to get remote Exts 20xx able to call one another and call
on-site telephones.
How to change those examples for one of the offsite phones to *also* use a
trunk to call other pstn numbers ?
http://wiki.freeswitch.or
Anthony Minessale wrote:
> This is a very unique problem as many people get this basic situation
> working daily so
> it must be a network issue of some sort.
As I said yesterday, a network problem makes the most sense, but the
behavior was still very strange.
I have now tracked down the problem
I think so. I would do it the following:
pass your variables for your outgoing number in front of your originate
string:
originate {var1=xxx, var2=xxx}dingaling/gmail.com/atomic.devter...@gmail.com
Then bridge it to a destination in your app or dialplan and set some
vars there.
Is that what solv
We have a strange Issue on Openzap with a Digium PRI card (Digium TE220
inkl. VPMOCT064 Octasic DSP-based echo cancellation module)
Every second incoming call is successfoll, every second incoming one
fails. For me it seems as if FS tries to use a channel which may be
still occupied?
Anybody has
On 1/8/09, Anthony Minessale wrote:
> put them in {} comma separated.
>
>
> {foo=bar,test=true}sofia/default/u...@dom.com
>
> if you are doing forked dial you can set them per leg with []
>
> [var1=foo]sofia/default/u...@dom.com,[var1=bar]sofia/default/us...@dom.com
>
Tony,
Thanks for getting
On 1/8/2009 10:02 AM, kriko wrote:
> I wrote a java socket client that originates a call, for e.g.:
> originate dingaling/gmail.com/atomic.devter...@gmail.com
> &bridge(loopback/1003/java_gmail_bridge)
>
> This works fine, however both ends doesn't really see each other numbers,
> instead they
Hi all,
I have a freeswitch setup for bridging calls between two gateways.
i.e. Originator -> Gateway A -> freeswitch -> Gateeway B -> terminator user
I have it running very well for last 43 days.
But from last one week, I noticed that freeswitch was not able to pass more
than two calls at once. A
I have set up an external phone per this example
http://wiki.freeswitch.org/wiki/Example_Offsite_phones
http://wiki.freeswitch.org/wiki/Example_Offsite_phones
I have set up a sip gateway that works with internal phones but does not
work when dialed from external phones.
What am I missing. I w
I think there is some problem with the bridge and context. I get a message
from FS saying the person you are trying to contact is not available.
--
View this message in context:
http://www.nabble.com/external-users--%3E-gateway-*help-please*-tp21351221p21352577.html
Sent from the Freeswitch-u
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Brian,
sorry, got lost ... It's a bug in Snom and AVM's phones?
Maybe we misunderstood. RFC 2833 emtioned in chapter 3.10 following DTMF
Events:
3.10 DTMF Events
Table 1 summarizes the DTMF-related named events within the
telephone-event
And if its stereo it will be a bit bigger. Also I don't recommend
recording in mp3 at all if you want to scale far.
/b
On Jan 8, 2009, at 10:15 AM, Andy Ayers wrote:
Thanks Brian, am I correct in saying therefore that all mp3 streams
generated by the recordFile command(with mod_shout insta
Thanks Brian, am I correct in saying therefore that all mp3 streams
generated by the recordFile command(with mod_shout installed) will be
64Kbps?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
You have to use SVN of both opal and ptlib. Its bleeding edge.
/b
On Jan 8, 2009, at 10:06 AM, Apostolos Pantsiopoulos wrote:
> Hi,
>
> I have installed ptlib 2.4.3 and opal 3.4.3 and still cannot get
> freeswitch to
> compile mod_opal. The error I am getting is this :
>
> mod_opal.h:58: error:
Hi,
I have installed ptlib 2.4.3 and opal 3.4.3 and still cannot get
freeswitch to
compile mod_opal. The error I am getting is this :
mod_opal.h:58: error: expected class-name before β{β token
I am using the 11094 revision of FS on CentOS 5.2.
Has anyone faced anything similar?
--
--
I wrote a java socket client that originates a call, for e.g.:
originate dingaling/gmail.com/atomic.devter...@gmail.com
&bridge(loopback/1003/java_gmail_bridge)
This works fine, however both ends doesn't really see each other numbers,
instead they see freeswitch number and id.
Is it possible to
I just opened a bug with Snom on this one.. it should NEVER send a *
or # in the signal line. But we do work around this but its wrong.
/b
On Jan 8, 2009, at 9:32 AM, Helmut Kuper wrote:
INFO sip:mod_so...@85.16.246.6:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP
85.16.245.220:1024;bran
I am using fs 1.0.2 running on CentOS 5.2.
You're right, the gxw is not alerting fs at all, i did some testing and
there is absolutely nothing on the debug console (after pressing F8) between
the moment i put the user on hold and the moment i pick up the call again.
"You might need to use
tone_de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello Mike,
I guess the workaround refers to INFO messages with "dtmf" mime type
instead of "dtmf-relay"?
With the actual trunk (11090) it works.
DTMF event F is still converted to zero (0). Im not sure if this event
will ever transmitted via INFO.
bitrate nor sample rate are configurable. The format depends on the
extension of the filename. The sample rate is recorded at the
channels native rate.
/b
On Jan 8, 2009, at 8:37 AM, Andy Ayers wrote:
Hi,
Is the bitrate, sample rate or format of the audio stream created by
session.re
Hi,
Is the bitrate, sample rate or format of the audio stream created by
session.recordFile configurable at all? Apologies if I've missed something
in the docs.
cheers
Andy
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http:/
Nope.
Currently only gtalk → sip ringback works, sip → gtalk doesn't.
If soemone needs, I'm pasting my extensions used.
sip → gtalk (ringback not working):
gtalk → sip:
Thank
put them in {} comma separated.
{foo=bar,test=true}sofia/default/u...@dom.com
if you are doing forked dial you can set them per leg with []
[var1=foo]sofia/default/u...@dom.com,[var1=bar]sofia/default/us...@dom.com
On Thu, Jan 8, 2009 at 12:50 AM, Kristian Kielhofner <
kristian.kielhof...@g
I see, looking at users inside directory I found why.
Thanks
On Thu, 08 Jan 2009 15:07:04 +0100, Anthony Minessale
wrote:
> authenticated users can override the context setting with variables in
> their
> account or domain.
>
>
> On Thu, Jan 8, 2009 at 6:32 AM, kriko wrote:
>
>> Hello!
>>
>
you may want to try
jingle has no concept of telephony early media waiting for answer and all
that so it's not an exact fit into sip.
On Thu, Jan 8, 2009 at 7:32 AM, kriko wrote:
> Now I've made a small dialplan to call from sip phone directly to gtalk:
>
>
>
> expression="^g
This workaround was added to address phones that specifically send
info dtmf incorrectly. Do you have a specific device that is not
working with 1.0.2? If so, can you please show the exact packet it is
sending.
Mike
On Jan 8, 2009, at 6:46 AM, Helmut Kuper wrote:
> -BEGIN PGP SIGNED
authenticated users can override the context setting with variables in their
account or domain.
On Thu, Jan 8, 2009 at 6:32 AM, kriko wrote:
> Hello!
>
> I'm a bit confused how sofia profiles works.
> If I did understood correctly it is something like this:
> the param inside a external sofia
Did you ever find out if the rtp was making it to your phone?
Did you get around to testing the echo exten? That is the most basic call
you can do
it is 1 leg call just playing your own audio back. Also 9998 plays the
tetris song with the tone generator.
We for sure can see rtp packets in the
Hi all,
I'm trying to get a setInputCallback function working with mod_python. I'm
using a current svn checkout for my build and the hello world via call
example from the wiki (
http://wiki.freeswitch.org/wiki/Mod_python#Hello_World_via_call ).
I've tried repeatedly, but I can't get the callback
Now I've made a small dialplan to call from sip phone directly to gtalk:
Simple, calling works. However still can't get ringback to work. In this case
the first leg is not yet aswered.
If I apply same stuff onto SIP to SIP call
Hello Antony,
I remember the fund raiser, but I didn't know that a mod_opal was
available for testing some time know. Perhaps I missed the announcement
I guess, this is why I thought that no work was done on it.
I'll proceed to testing it asap.
Best regards,
Vlasis Hatzistavrou.
Anthony Mines
Hello!
I'm a bit confused how sofia profiles works.
If I did understood correctly it is something like this:
the param inside a external sofia
profile will force
to process calls via extern context (e.g. dialplan/extern.xml).
Then if I use inside my jingle profile,
will force calls
from the s
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
today we found a DTMF problem in an older FS build. We weren't able to
pass "*" and "#" to PSTN. FS sents those chars as "0". I fixed that on
my own. After that I searched in build 11055M to if it was already fixed
in trunk.
I found it was fix
Thanks for all suggestions. Ufortunately I cannot get it working.
Seems like packets are not coming to phone behind nat (freeswitch is on public
ip).
When registering I can see multiple notify retries like this:
send 802 bytes to udp/[10.99.10.6]:5060 at 10:49:31.762605:
Yes, I am doing it via event socket.
On a system with pure SIP it works, but on a system where on both logs
openzap is used, it doesn't work. So maybe it's an openzap problem?
I do the following via event socket:
Leg B is doing the bridge via "uuid_bridge " while leg A is playing a soundfile
Thanks for your help! Its working! I insert the user and the password in
odbc.ini.
_
Von: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Im Auftrag von Chris
Danielson
Gesendet: Mittwoch, 7. Januar 2009 19:32
An: freeswitch-users@l
Hello!
I have question about
http://wiki.freeswitch.org/wiki/Queue_which_calls_extensions .
The javascript connectqueue.js access on tables in the database. Do I have
to create them myself?
How do the agents register to the queue?
Thanks! fidibus
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