Just a question,
do you use Freeswitch in bypass-media-mode in this scenario? Then media
negociation should be handled outside Freeswitch.
Best regards
Peter
Jerry Richards schrieb:
After establishing an audio call between two Bria softphones, and then
starting video at the caller phone, FS
Hello,
in a bigger installation with some thousand endpoints in the field we
see, that the endpoint equipment is always using INFO messages (standard
setting is auto, so the endpoint decides which method to use). I have 2
questions to that scenario:
1. Is there a way that Freeswitch
were
you thinking) either way... set the transfer_ringback variable.
/b
On Dec 18, 2009, at 4:00 AM, Peter P GMX wrote:
Should I open a JIRA for this?
Best regards
Peter
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we do this based XML-Curl.
Jerry Richards schrieb:
Is it possible to allow/deny REGISTER requests based on the User-Agent
header? I need to know/manage what devices are registering.
Best Regards,
Jerry
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Should I open a JIRA for this?
Best regards
Peter
Peter P GMX schrieb:
Hello,
we have the following scenario:
A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For
the called FS user, call forwarding has been enabled to another PSTN
extension (B) .
Result: The calling
I try to attach Bravis video conference clients to Freeswitch. Those
video conference clients are really working good (Multilingual clients
for testing ca be downloaded here: http://www.bravis.eu/). Some big
companies here in Germany use them in large installations. They are
based on SIP, but do
Hello Oliver,
I have the same on Ubuntu wth newest trunk.
Best regards
Peter
Oliver Schönbeck schrieb:
Hello,
we are running freeswitch 1.0.trunk and are currently trying to get
the mod_voicemail to send the received messages to the user by using
exim4 on a debian machine.
So
Hello,
we have the following scenario:
A PSTN call (A) is coming in to Freeswitch through a Patton gateway. For
the called FS user, call forwarding has been enabled to another PSTN
extension (B) .
Result: The calling party does not hear any ringing tone. Here an
Abstract of the SIP protocol we
We currently use Patton gateways SN4116 for attaching fax and modem
equipment to our Freeswitch system. Freeswitch is in bypass-media-mode,
so media flow goes the following way:
Modem/Fax = Patton_SN4116 = Patton_SN46XX =PSTN/ISDN
However modem connections are not very reliable. We exchanged the
Hello,
I try to invite a user into a conference by
loopback/255 8000 Conference
255 is the user, I invite the user via loopback as that way I can also
invite external numbers.
It processes the user's local dialplan correctly (as if the user was
normally dialled), however it only offers L16
schrieb:
this action can be accomplished using Group Dialing (Sequential). this
may not answer your problem but have you considered it?
-nandy
On Tue, Dec 8, 2009 at 2:31 AM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
I have a Problem with continue_on_fail
Hello,
in our dialplan we have enabled multiple-registrations, so 2 phones can
register on a single directory entry.
param name=multiple-registrations value=true/
Both phones are registered, both phones can be called and each phone can
call the other phone.
However in an attended_transfer
Hello,
is there a way to manually force a presence status update?
In our scenario we have a Freeswitch cluster. As phones sometimes
register on one and one time on another machine via the load balancer,
we cannot dial via user/exten. Instead we dial each phone by it's
register string via
I have a Problem with continue_on_fail.
I have setup a hunt group
action application=set data=continue_on_fail=NO_ANSWER,USER_BUSY/
action application=bridge
data=sofia/external/2...@10.11.12.243,sofia/external/2...@10.11.12.234,sofia/external/2...@10.11.12.188,sofia/external/1...@10.11.12.245/
Hello,
i now changed the $${domain} name of the server to the domain name the
phones register with.
Now messaging (MWI, notify) works.
Best regards
Peter
Peter P GMX schrieb:
Hello Anthony,
I did some checks today
Here is how the phones are registered:
mysql select sip_host
Hello,
i now changed the $${domain} of the server to the domain name the
phones register with.
Now messaging (MWI, notify) works. Thanks to all for your support.
Best regards
Peter
Peter P GMX schrieb:
Hello Anthony,
I did some checks today
Here is how the phones are registered:
mysql
Concerning,
Which I'm kinda confused about, I don't have any 192.168 net here??
I think, this is a default entry in the acl.conf.xml. Please check the
entries there. But normally this shouldn't stop freeswitch from working
and handling requests.
Can you set the console_log_level to debug in
Hello,
I would like to offer a dictation service to a secretary.
Means:
* the boss is dictating some text on a certain phone number
* the secretary picks up the recording on the phone and types the
text into the computer
As the secretary is not able to type in as fastly as heir
on the featureset required.
On Dec 6, 2009 10:30 AM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
Hello,
I would like to offer a dictation service to a secretary.
Means:
* the boss is dictating some text on a certain phone number
* the secretary picks up the recording
.
On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX
prometheus...@gmx.net mailto:prometheus...@gmx.net wrote:
I tried now with phones directly attached to the freeswitch
(without an
OpenSIPS in between). I also added the alias. But the
behaviour is as
before
Hello,
is there a chance to have the voicemail system to play announcment #1
only and not play announcement and then record the voicemail?
Means: Can I switch off the recording part?
Best regards
Peter
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passing the call to an IVR application that plays the message instead
of passing it to the voicemail application.
-AF
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Peter P
GMX
Sent: Friday
inexpensively maybe $100
On Fri, Dec 4, 2009 at 11:26 AM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
I would like to manage this in the voicemail menu.
Press 6 to enable recording
Press 7 to only play announcement
or so. So hte user can manage it's
I tried now with phones directly attached to the freeswitch (without an
OpenSIPS in between). I also added the alias. But the behaviour is as
before:
No notify message from freeswitch, neither after register nor after a
voicemail is recorded.
Best regards
Peter
Brian West schrieb:
Yes an alias
Hello,
I have a similar problem with Freeswitch behind OpenSIPS as a load balancer:
When registering, Freeeswitch does not send a MWI NOTIFY message for a
Phone which has voicemails. Even after recording a new voicemail there
is no NOTIFY message sent. And there are no error messages on the
/profile
cannot be resolved to the correct
sofia profile to send the notify
The event starts out as a freeswitch event and is translated into the
notify by mod_sofia but only if it can
match the event to a real sip user
On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX prometheus...@gmx.net
I sorted it out.
Something went wrong with the odbc database. I deleted the voicemail
database tables, restarted FS and let FS create the tables again. Now it
works.
I can even share the voicemails across 2 Freeswitch boxes.
Best regards
Peter
Peter P GMX schrieb:
I now created a file
/msg_4d484a7e-d76c-11de-84d4-a1cd7fa320b3.wav.5..B_NORMAL.
Both filenames can be read.
Best regards
Peter
Peter P GMX schrieb:
I installed all sounds from SVN, but
usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA
isn't there. I checked another, older installation and couldn't this
file
to venture to guess maybe the file was recorded in a
different codec and NOT pcma?
/b
On Nov 20, 2009, at 6:56 PM, Peter P GMX wrote:
2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening /
usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA
Hello,
i have a couple of problems with voicemail. Voicemails are recorded but
not played in any way.
1) when I call my voicemail, I can hear the number of new messages, but
I canot not hear the recorded files itself. I hear the following
* You have 1 urgent new message in forder inbox
on pastebin i can tell you what's happening.
On Thu, Nov 12, 2009 at 2:38 AM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
Thanks Anthony,
however this rather deteriorated the situation.
Now it works the following
- A calls B
- B enters *4 gets
loopback_bowout=true so it's on
the loopback leg
On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX
prometheus...@gmx.net mailto:prometheus...@gmx.net wrote:
Hello,
I have some problems with attended transfer and loopback
Scenario how id does work
Hello,
I have some problems with attended transfer and loopback
Scenario how id does work
- A calls B
- B enters *4 gets an announcement and enter digits for C (A get MOH)
- C is called
- As soon as C picks up the call, A and C are connected and B is dropped
How it should work until here:
- A
Hello,
in a freeswitch cluster (FS1 and FS2) behind an OpenSIPS I want
Freeswitch to register to external gateways through the OpenSIPS load
balancer, in order to later receive incoming calls through the load
balancer.
Is there a way to tell Freeswitch in it's Gateway definition to define
an
mode, right?
Best regards
Peter
Michael Jerris schrieb:
There was just a bunch of work on UPDATE, can you confirm this is the
same behavior with trunk?
On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote:
Hello,
we have the following problem.
2 Fax machines are communicating via
Hello,
we have the following problem.
2 Fax machines are communicating via Freeswitch. One is externally
attached via a Telco who is able to handle T.38. The other one is
attached locally.
When 2 Fax machines start syncing each other, the Telco sends a SIP
UPDATE message with T.38 SDP, as it
Has anybody managed to get this to work already?
How do you play the announcements dependent on the variable in the dialplan?
Best regards
Peter
Michael Collins schrieb:
On Thu, Sep 10, 2009 at 12:32 PM, Diego Viola diego.vi...@gmail.com
mailto:diego.vi...@gmail.com wrote:
Lets make
Hello,
I read in the wiki that binding blocks are processed in sequential order
in a failover matter.
So I created the following bindings for XML-Curl:
However grepping the network traffic I can see that Freewitch always
fetches both servers fo one binding. So there is no real failover.
How can
Anybody tried siptapi with freeswitch?
http://sourceforge.net/projects/siptapi/
This may enable Click2Dial e.g. from Outlook to Freeswitch.
So anybody has experience with that solution?
Best regards
Peter
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Content-Length: 0
On Fri, Sep 18, 2009 at 2:53 AM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
Hello ,
I try to hangup aa call with a certain cause code.
If the user dials a number which we currently do not serve we send
action
Hello,
I finally solved it by using
action application=hangup data=${originate_disposition}/
Best regards
Peter
Peter P GMX schrieb:
Hello Anthony,
I did further testing on a second machine and found out the following:
After
action application=set data
Hello ,
I try to hangup aa call with a certain cause code.
If the user dials a number which we currently do not serve we send
action application=set data=sip_ignore_remote_cause=true/
action application=hangup data=NO_ANSWER/
which gives a
SIP/2.0 480 Temporarily Unavailable.
Thanks Anthony,
that did the trick.
Best regards
Peter
Anthony Minessale schrieb:
you can edit mod_xml_curl.c line 64
and increase XML_CURL_MAX_BYTES
On Thu, Sep 3, 2009 at 12:31 PM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
Hello,
in a B2BUA
Hello,
in a B2BUA scenario we have 2000 defined gateways (defined but not
registered yet).
When reloading mod_sofia Freeswitch complains about the XML-Curl File
size 1MB and deactivates all gateways:
mod_xml_curl.c:121 Oversized file detected [1056100 bytes]
Is there any way to overcome
I had the same problem. Must have been changed something in lua since
this morning.
Please install swig.
E.g. on Debian
sudo apt-get install swig
That did it for me.
Best regards
Peter
Lars Zeb schrieb:
I just updated using “svn up” which brought the source to 14741. After
running
, 2009, at 5:25 PM, Peter P GMX wrote:
Hello,
is there a chance to have sofia_reg_external in odbc/mysql instead of
sqlite?
In a B2BUA environment we have thousand of external registrations
during
a migration phase, and it would be good to have easy external control
over the registered
Sure this works,
you can set rtp or srtp independently to every call leg (if FS is in
media path) and even mix them in a conference.
Best regards
Peter
NOx-WHV schrieb:
Hi,
i have a problem using SRTP Encrytion. All intern calls are SRTP encrypted.
Some of my Gateway don´t support SRTP
If you do not explicitely set bypass_media to true, then FS is in the
media path.
Best regards
Peter
NOx-WHV schrieb:
How can I see if the FS is in media path?
Or how can i set the FS in media path?
Peter P GMX wrote:
Sure this works,
you can set rtp or srtp independently
Hello,
is there a chance to have sofia_reg_external in odbc/mysql instead of
sqlite?
In a B2BUA environment we have thousand of external registrations during
a migration phase, and it would be good to have easy external control
over the registered gateways (like in fs_internal.
I got it,
gateways have to be preloaded (rescanned) before they can be registered.
Best regards
peter
Peter P GMX schrieb:
Hello,
I am using XML-Curl to handle the configuration of freeswitch
When I try to register a gateway via event-socket with
sofia profile external register gw-name
I have found a strange thing in my FS installation,
FS is registered via a Gateway to an external provider (QSC) in the
external context.
But when a call is coming in, FS does not seem to go to any context, but
tries to lookup the user, as I receive the following message
2009-08-27
And yes,
external profile is on Port 5080 and all request go to 5080.
Best regards
Peter
Peter P GMX schrieb:
I have found a strange thing in my FS installation,
FS is registered via a Gateway to an external provider (QSC) in the
external context.
But when a call is coming in, FS does
Hello,
I am using XML-Curl to handle the configuration of freeswitch
When I try to register a gateway via event-socket with
sofia profile external register gw-name
I receive back invalid gateway.
After reload mod_sofia the gateway is there. Question: Does this command
work with xml-curl or
Hello,
is there any chance to limit the listening ips of the xml-rpc server
(which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)?
Best regards
Peter
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, Aug 21, 2009 at 1:38 PM, Brian West br...@freeswitch.org
mailto:br...@freeswitch.org wrote:
You can ship me one whois bkw.org http://bkw.org, I can add it
to my lab.
/b
On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
BTW: We can ship you a FritzBox if you need one
Hello Michael,
I made some tests with Freeswitch and Fritzbox and found by Wireshark that:
within one call
* Freeswitch starts sending 20msec packets, then after ~0,2 second
sends 30msec packets
* FritzBox always sends 30msec packets.
The average jitter is below 2 msec in both
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 21-Aug-09, at 11:38 AM, Peter P GMX wrote:
Hello Michael,
I made some tests with Freeswitch and Fritzbox and found by
Wireshark that:
within one call
* Freeswitch starts
Hello,
when calling from Fritzbox to a Snom Phone , sound is fine. But when
calling an internal Freeswitch number (conference, mailbox) i hear a
very choppy voice coming from the fritzbox side. I think it may have to
do with the ptime 20msec/30msec.
Example: When calling from the fritzbox to a
sure you have
the latest firmware? Try setting the ptime on the fritz to 20ms?
I really can't trust a product that has fritz in its name... it might
go on the fritz :P pun intended.
/b
On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:
Any more hints
Hello,
anybody has a clue what this message means?
[WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from
PROGRESS_MEDIA to PROGRESS
What does VETO mean here?
Best regards
Peter
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Hello,
I setup libpri and a sangoma card A108DE, but I cannot dial out.
At startup I receive on the D channel
TDM API: CMD: 18 : Operation not supported
When dialling Libpri debug shows that the numbering plan is fine and
that it accepts the screened number, but then it finally hangs up
clears the error message.
I was asked by telco to specifically use telco, will using Q931 cause
any issues for me?
Thanks again,
Niall.
2009/7/31 Peter P GMX prometheus...@gmx.net:
Firstly I would try get get rid of the sangoma errors. There are 2 errors:
Short Circuit: ON
Loss
Hello Luis,
are you using encrypted TLS instead on SIP on this phone? I experienced
a similar behaviour with 31 seocnds on TLS.
Best regards
Peter
Luis F Urrea schrieb:
Hi all,
I am experiencing a behavior that I cannot clearly understand.
Basically I autocall a few phones into a conference
Thanks,
I have found the sources in
contrib/jmesquita/fsgui
Any recommendatioins how to compile it under Linux?
Best regards
Peter
João Mesquita schrieb:
Dear folks,
Even tho it might be premature, I would like to already spread the word.
Check out FsGUI and feel free give feedback if this
Well,
that's very useful for us in order to have this info in our FS Operator
panel.
Best regards
Peter
freeswitch-users@lists.freeswitch.org schrieb:
Hi,
I usually find it very useful when I can retrieve a list of the
currents calls along with durations. I noticed that the 'show
channels'
sofia_reg_handle_sip_i_register() NO CONTACT!
Please give any suggestions to rectify this error..
Thanks in Advance,
Regards,
K.Velusamy.
On Jul 11, 2009, at 8:46 AM, Peter P GMX wrote:
I have several Twinkles running against freeswitch on a locally
installed machine (FS acts
I have several Twinkles running against freeswitch on a locally
installed machine (FS acts as a SIP/TLS proxy).
So in general Twinkle works (on various Ubuntu machines from 7 upto 9
with various Twinkle versions). It must be some kind of setting in
Twinkle. E.g.
* set the local Twinkle SIP
Hello Helmut,
I looked at these dic files. Their content (look at all the qq's) is
quite different from the dic files supplied with freeswitch pocketsphinx.
As I remember the CMU dict file format has changed in April 2008.
Best regards
Peter
Helmut Kuper schrieb:
Hi,
I try to change
Hello Helmut,
I looked at these dic files. Their content (look at all the qq's) is
quite different from the dic files supplied with freeswitch pocketsphinx.
As I remember the CMU dict file format has changed in April 2008.
Maybe there is a converter somewhere?
I was thinking of just enhancing
Hello,
I installed a Sangoma A108 with openzap and libpri. Signalling (E1)
works sometimes (inbound and outbound calls are connected) but not always.
Sound is just distortion but connection is stable.
2 questions:
1) What is the best way to go with Sangoma? OpenZAP with libpri or
without libpri?
Hello,
I have the following problem: Every call stops after 30 seconds when TLS
is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP.
The phones are behind NAT. So I expect, that every 30 seconds an Options
request is sent.
Wiresharking the traffic I can see
* that there are ongoing UDP
: NORMAL_UNSPECIFIED
Best regards
Peter
Peter P GMX schrieb:
Hello,
I have the following problem: Every call stops after 30 seconds when TLS
is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP.
The phones are behind NAT. So I expect, that every 30 seconds an Options
request is sent
... doesn't make much sense.
/b
On Jul 2, 2009, at 6:11 AM, Peter P GMX wrote:
Hello,
I have the following problem: Every call stops after 30 seconds when
TLS
is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP.
The phones are behind NAT. So I expect, that every 30 seconds
or simply
sofia status
for all gateways
Jason White schrieb:
Brad Tuan brad.t...@gmail.com wrote:
As title ,Does FS keep the status of gateways??
sofia status gateway gateway-name
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.
Best regards
Peter
Brian West schrieb:
click on the AA button? :)
/b
On Jun 15, 2009, at 7:19 PM, Peter P GMX wrote:
What is the best way to have this done? Move the call to park and then
retransfer again with intercom, or is there a better solution
is gone away. So I would
like the web app to drive answering the call. It gives a better
visibility about what he is doing to the callcenter agent.
Best regards
Peter
Raymond Chandler schrieb:
Peter P GMX wrote:
Hello Brian,
this is too easy :-).
This is for a small callcenter app and I
is hit in the web ui.
Mike
On Jun 16, 2009, at 4:24 AM, Peter P GMX prometheus...@gmx.net wrote:
Hello Brian,
this is too easy :-).
This is for a small callcenter app and I only want the user to pickup
the call once (to accept the call in X-Lite (or a Snom phone) and to
start
May this help also: I just tried current Zoiper with TLS. Outbound is
working, inbound not.
Zoiper registeres with the following contact info:
7233213
sip:7233...@217.xx.xx.xxx:51989;rinstance=85925cbf0ecc0ab4;transport=TLS
When a call comes in, Zoiper rings once and then hangs up. It shows
issue?
Best regards
Peter
Michael Jerris schrieb:
The transfer should work but it sounds like offhook agents is what
your really trying to accomplish so I would go with brian's suggestion.
On Jun 16, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net wrote:
Hello Michael,
I want
with parking the call and then
forward it.
Best regards
Peter
Michael Jerris schrieb:
uuid_setvar unique_id sip_invite_params intercom=true should be
unnecessary.
Mike
On Jun 16, 2009, at 1:49 PM, Peter P GMX wrote:
It mainly works now by uuid_transfer the following way via event
I have managed to have a realtme status of a phone on a web page with
event_socket and a push service to the web bowser.
What I am now trying to do is roughly the following:
* when a call comes in, a flashing banner appears on the web page
with an underlying link (this works so far)
I saw that xmpp is supported in Fresswitch. See wiki:
http://wiki.freeswitch.org/wiki/Mod_xmpp_event
Has anybody already set this up? I have found no mod_xmpp neither in my
mod directory nor in the source?
There was also a question:
Q: Is it possible to send commands to fs via xmpp?
Answer: Yes.
I have a problem where FS gives a core file when an voicemail email
shall be sent via exim.
I am on 13438.
No entry in debug log in FS.
No entry in exim log.
Best regards
Peter
Jason White schrieb:
Luis M. Zuccolo luismzucc...@yahoo.com.ar wrote:
1.0.4pre8
It works for me with
JIRA opened:
*FSCORE-375 http://jira.freeswitch.org/browse/FSCORE-375*
Brian West schrieb:
Please Open a JIRA ASAP. We are working to get 1.0.4 out and these
are the types of issues that should have been reported weeks ago if
they were happening.
/b
On May 30, 2009, at 6:27 AM, Peter P
And mine with the same behaviour on Linux.
Best regards
Peter
Diego Toro schrieb:
Hi, my job with FS has been on Windows.
Diego
--- On *Thu, 5/28/09, Brian West /br...@freeswitch.org/* wrote:
From: Brian West br...@freeswitch.org
Subject: Re: [Freeswitch-users] The calls are
I want to do the following:
Originate a call via event_socket, I get back a job_uuid. Then I want to
control the call when it's established (2 call legs).
Scanning the variables of the 2 call legs I currentyl cannot see any
relation between the job_uuid and the uuid of the resulting call legs.
I
the core to give you a new uuid with the create_uuid FSAPI call.
On Wed, May 27, 2009 at 4:46 AM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
I want to do the following:
Originate a call via event_socket, I get back a job_uuid. Then I
want to
control
I had a similar behaviour with dropped calls. After I changed the
firewall on the FS machine it worked. In my case some ports on the FS
machine were not open for outbound traffic (inbound were ok). Check SIP,
TLS, RTP, STUN, DNS ports.
Best regards
Peter
FERNANDO VILLARROEL schrieb:
Hi Diego,
Hello,
today I tried uuid_chat via event socket.
A simple chat application works: bgapi chat
sip|age...@fqdn|age...@fqdn|Message.
uuid_chat uuid however returned +OK, but nothing happens. Neither
is there a debug line on the console, nor a SIP (in my case TLS) message
is sent to the UA.
Has
We have used FS on ec2 for testing purposes only. It was ok. We havn't
done any performance test though.
Best regards
Peter
Ing. Edwin Villarreal schrieb:
Hello my friends.
Has anyone used the EC2 for production? Tests?
I’m wondering if it would be “better” to have a FS system in
Just a side notice about how to name a company.
If you use a descriptive name e.g. GlobalSIP as sugested before, it
may be difficult to register this name later on as a brand name when
your company becomes successful.
At least here in Europe it is not possible to register a brand name when
the
This is also interesting for me, as I love freeswitch, and maintaining a
single platform is easier, than handling various different ones.
In the past years I did a couple of projects with OpenSER /openSIPS.
These projects comprised:
* registrar for the SIP user agents
* handle invite
Hello Jean-Yves,
did you ever try a call-trough? (a person dials in (1234567, see below)
types the target number as DTMF and gets connected to this number?
A basic dialplan can be like this:
extension name=Dialthru
condition field=destination_number expression=^(1234567)$
action
chance that you would speak but we
won't know exactly which day or time.
Please let me know what you think.
-MC
Sent from my iPhone
On May 16, 2009, at 6:14 AM, Peter P GMX prometheus...@gmx.net wrote:
Hello Michael,
I see that there are still some time slots available on 6th of Aug
Hello Michael,
I see that there are still some time slots available on 6th of Aug. I am
thinking of doing a presentation on an application server and Web GUI
for Fresswitch we have developed.
Is it still possible to register for a full speaker slot?
Best regards
Peter
Michael Collins schrieb:
Peter
Michael Collins schrieb:
On Thu, Mar 5, 2009 at 12:20 PM, Peter P GMX prometheus...@gmx.net wrote:
Hello Brian,
concerning
Well you should use ESL then ;)
I simply do not understand what you mean by this. Is it sarcastic? Am I
asking stupid questions?
ESL
I record them to file.wav and they play perfectly. I think it's recorded
in a raw-format here. See:
http://www.nabble.com/Recording-ULAW-files-td21587474.html
Best regards
Peter
Peter Olsson schrieb:
Hello again,
I also have a problem when I try to record messages. I record to
Hello,
I habe the following problem when re-registering to an external SIP
provider during a call which results in immediate call-hangups.
- FS re-registers with nonce
- 2ms later FS re-registers without nonce
- external SIP provider asks for credentials
- FS re-registers with nonce
- External
Hello Helmut,
I also have problems with my Snom300s and Snom320s and G711 and SRTP.
They may be related to this problem, but I am not sure.
The phones disconnect the media stream after a while (2..10 minutes)
because the Snom media port is blocked all of a sudden.
I have opened a bug report at
Hello,
I tried conferencing for FS und tried to lock/unlock conferences.
While conf-is-locked.wav was played, conf-is-unlocked.wav was
missing in the file system.
Any idea where I can download this?
Best regards
Peter
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