Hello,
did anybody get Twinkle with ZRTPworking?
I tried this with 2 Twinkle clients with both zrtp enabled.
Twinkle sends a=zrtp in the sdp message but this dows not arrive at the
other Twinkle instance after the message passed Freeswitch.
Any help is welcome.
Just to get it: I just run a zrtp daemon and then FS can handle it?
Where can I get zrtp daemon?
best regards
Peter
Brian West schrieb:
Or you can run the zrtp daemon on the linux box and it works also
right to FS.
/b
On Jul 17, 2008, at 4:23 PM, Michael Jerris wrote:
it should in
Hello,
I have the standard Ubuntu installation for Hardy and freeswitch mostly
works fine so far.
Now I test the xmlrpc functionality in order to try the new Telegraph
project which is based on freeswitch now (
http://code.google.com/p/telegraph
Thanks. Now it works!
Brian West schrieb:
Edit conf/autoload_config/modules.conf.xml and uncomment the line so
the module will load.
/b
On Jul 17, 2008, at 5:29 PM, Peter P GMX wrote:
What must I do in order to start xmlrpc for freeswitch?
Brian West
sip:[EMAIL PROTECTED
Hello,
has anybody managed to setup TLS? When I change tls to true in
internal.xml, then freeswitch doens't listen on any ports (5060 5061).
I use freeswitch 1.0.0-0ubuntu1~ppa4
Is there any tutorial available (could not find it while googling)? I
would like to set it up with Snom phones
-an but there are no processes listening
on port 5061.
Also gentls_cert with real values didn't work.
Without TLS enabled it works well on port 5060.
Any hints how to continue?
Brian West schrieb:
This should get you started.
http://wiki.freeswitch.org/wiki/Tls
/b
On Jul 23, 2008, at 1:29 PM, Peter P
()
Creating agent for internal
2008-08-03 18:57:32 [*ERR] sofia.c:552 sofia_profile_thread_run() Error
Creating SIP UA for profile: internal*
Best regards
Peter
Brian West schrieb:
Did you turn tls on the profile on?
/b
Sent from my iPhone
On Aug 3, 2008, at 6:44 AM, Peter P GMX [EMAIL
the call is setup correctly and then it hangs up.
Did I miss something?
Best regards
Peter
Brian West schrieb:
And you have everything in conf/ssl right?
/b
On Aug 3, 2008, at 12:01 PM, Peter P GMX wrote:
Hello Brian,
Yes it's turned on:
!-- TLS: disabled by default, set to true
and AES_CM_128_HMAC_SHA1_80. I highly recommend you only enable one
cypher suite...
/b
On Aug 3, 2008, at 2:14 PM, Peter P GMX wrote:
I got TLS working right now. It turned out that the modified start/
stop
script for freeswitch which I had from the Ubuntu package caused that
problem
Has anybody successfulkly tested a Softphone that works with TLS/SRTP
and Freeswitch?
- I tried Minisip (I think it works with MIKEY) - no success
- I tried Zoiper Biz - it did not like to connect via TLS (any hints?)
- more ??
Best regards
Peter
___
Hello Darren,
I've tried the GUI and it looks fine. However I would like to use TLS in
Freeswitch. Is there a way to use TLS with this GUI?
Best regards
Peter
Darren Schreiber schrieb:
Hi folks,
Nice to see that interest in this project found it's way here...
The FreeSwitch GUI
Hello,
did anyone manage to get a TLS and SRTP connection working between 2
Freeswitch servers?
For my understanding Freeswitch should just behave like a normal UA. So
TLS and SRTP should also be possible, when routing calls between 2 FS
servers, hein?
Maybe someone may also post a sample
all the nat settings.
/b
On Aug 6, 2008, at 7:47 AM, Peter P GMX wrote:
Hello,
I have successfully set up Snom phones for TLS and SRTP so the whole
communication should be encrypted.
However I see OPTIONS SIP requests in clear text coming from
Freeswitch.
Is there a chance
Brian West schrieb:
Open a Jira on this. SO we can track it at jira.freeswitch.org i'm
not sure what is required but it should work like it is. The
transport=tls won't work on gateways because you set the transport in
the gateway config.
/b
On Aug 29, 2008, at 4:38 PM, Peter P GMX wrote
Concerning TLS and SRTP on S60 see
http://mosh.nokia.com/common/download/4452B13D5F854A8DE040050A45306C1B/original/Developing_3rd_party_VoIP_clients_on_S60_platform_v1_0_en.pdf
But I a not sure whether they use SDES or Mikey for key exchange.
Brian West schrieb:
Eric,
I wasn't aware that
Hello,
as explained everything works fine, except that I do not get any
sofia.conf requests.
For section=configuration I receive requests for
key_value=post_load_modules.conf
key_value=event_socket.conf
key_value=acl.conf
key_value=post_load_switch.conf
key_value=switch.conf
at
Hello Michael,
yes, mod_sofia is loaded. I would not be able to route calls to external
gateways defined under sofia.conf in the xml file if it was not loaded
Best regards Peter
Michael Jerris schrieb:
On Sep 16, 2008, at 4:02 PM, Peter P GMX wrote:
Hello,
as explained everything
fs_curl[3450]: [key_value] = 'limit.conf'
Sep 16 15:56:20 lmdt fs_curl[3448]: [key_value] = 'dingaling.conf'
Sep 16 15:56:20 lmdt fs_curl[3648]: [key_value] = 'sofia.conf'
Sep 16 15:56:22 lmdt fs_curl[3450]: [key_value] = 'iax.conf'
-Ray
Peter P GMX wrote:
Hello
:20 lmdt fs_curl[3648]: [key_value] = 'sofia.conf'
Sep 16 15:56:22 lmdt fs_curl[3450]: [key_value] = 'iax.conf'
-Ray
Peter P GMX wrote:
Hello,
as explained everything works fine, except that I do not get any
sofia.conf requests.
For section=configuration I receive
Hello,
I have done it the following way:
xml_curl.conf.xml:
configuration name=xml_curl.conf description=cURL XML Gateway
bindings
binding name=example
param name=gateway-url
value=http://192.168.0.35:3000/xml_curls/directory;
bindings=configuration|dialplan|directory/
/binding
/bindings
According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to
set up gateways dynamically, but I do net get it to work:
I always get : Invalid profile
My assumptions for a right xml answer back to FS are as follows, but I
think at least one of it is false:
1. ) I start with
document
Nobody has an idea anbout this?
Peter P GMX schrieb:
According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to
set up gateways dynamically, but I do net get it to work:
I always get : Invalid profile
My assumptions for a right xml answer back to FS are as follows, but I
think
Hello Raymond,
I had a look at your code. You are submitting a complete profile for
sofia.conf, right?
Best regards
Peter
Raymond Chandler schrieb:
Peter P GMX wrote:
According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to
set up gateways dynamically, but I do net get
schrieb:
Peter P GMX wrote:
According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to
set up gateways dynamically, but I do net get it to work:
I always get : Invalid profile
My assumptions for a right xml answer back to FS are as follows, but I
think at least one
Hello,
I have setup Freeswitch with xml_curl and provide configs almost
identical to the local xml files. I can call external gateways, however
when I call a local phone, I cannot connect and it drops me to the VM of
the phone.
2008-09-24 23:30:55 [DEBUG] switch_core_state_machine.c:140
parameter in this case:
user/[EMAIL PROTECTED])
Where does user come from? Is it an implicit internal context?
May I use another context in that case when the phone is registered in
the internal context?
Best regards
Peter
Peter P GMX schrieb:
Hello,
I have setup Freeswitch with xml_curl and provide
Hello Michael,
thanks for the hint, but how shall a dial-string param look like? I
looked up the internet but could not find an example.
Can you provide an example?
Best regards
Peter
Michael Jerris schrieb:
On Sep 25, 2008, at 7:19 AM, Peter P GMX wrote:
I figured out (via ngrep
Thanks for the hint. With the following dial-string in the directory it
works:
param name=dial-string value=[EMAIL
PROTECTED],transfer_fallback_extension=${dialed_user}}${sofia_contact([EMAIL
PROTECTED])}/
Best regards
Peter
Michael Jerris schrieb:
On Sep 25, 2008, at 4:04 PM, Peter P
I have TLS working with Snom phones. I could not get TLS working with
ZoiperBiz, as the current 2.06 Linux version of 31-Jul simply didn't
communicate to Port 5061.
Atractel has - as a reply to the submitted problems - announced a new
Zoiper release 2.17 for Windows, but It still isn't available
Today I had a strange behaviour:
I am routing calls to an Asterisk PBX. It has a very sophisticated
dynamic least cost router built in - so I use it to terminate mobile and
international calls.
For a first test I created a dialplan which checks for ^231$ in the
dialled number and then routes the
:
^(234)$
$1 will = '231'
-MC
-Original Message-
From: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[mailto:freeswitch- mailto:freeswitch-
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]] On Behalf Of Peter P GMX
Sent: Friday, October 03, 2008 4:14
Hello,
I want Freeswitch do do the following thing:
* after a call is setup between UA1 and UA2, then if UA2 presses
5 the call is diverted from UA1 to UA3
Is this possible? If yes, how can I do this?
Best regards
Peter
___
My Freeswitch on an 64bit Ubuntu server crashed with a core dump and
there are no infos in the log. Sometimes it crashes while loading the
codecs, sometimes it finishes loading and then crashes.
After all prerequisites were finished it compiled and installed well.
But program execution crashes:
to the dir
where you build FS and do the following:
# cd libs/libedit
# make clean
# sh configure.gnu
# make
once that is done
# cd ../..
# make clean
# make install
On Wed, Oct 8, 2008 at 9:40 AM, Peter P GMX [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
My Freeswitch
, 2008 at 6:25 AM, Peter P GMX [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hello,
is there any way to determine whether the outbound leg is ringing?
For example when (in SIP terms) the called UA is sending back a
Ringing
message via SIP, can I somehow recognize
I have another 3 Questions. I know I had already 2 before within the
last 15 minutes, but I need to qualify whether we can build this special
app with freeswitch or not.
The questions are:
1.)
When I do a uuid_playback I want to be able to stop this playback #1
immediately. My Idea is that I may
Hello,
is it possible to play special ring tones or a wav file before answering
the call? I do not really believe so, but maybe there is a chance?
Best regards
Peter
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
I've seen in the XCML files that passwords and credentials e.g. for
directory entries are always stored in clear text. Is there a way to use
encrypted passwords?
Bet regaerds
Peter
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
, Peter P GMX wrote:
Thanks,
I got it for the directory password (a1-hash).
But what about the voicemail-password and the passwords stored for
external gateways?
Best regards
Peter
___
Freeswitch-users mailing list
Freeswitch-users
Thanks for your support for the vm-passwords.
The most important part for us however is having hashed passwords for
external gateway definitions (we have a lot) and securing pins for
conferences.
Do we have a chance to add this also?
In our environment DTMF is of course transported via SRTP so
OK, I got it. The replay vulnerability only happens when key exchange is
done via unencrypted SIP. I understand that with TLS the Invite message
cannot be replayed as it cannot be seen in clear text.
Brian West schrieb:
Its called TLS...
/b
On Oct 21, 2008, at 4:30 PM, Peter P GMX wrote
Hello,
I receive the following message during CS_INIT
*Failed to load library libceplang_de.so due to:
/opt/swift/lib/libceplang_de.so: undefined symbol: cst_rx_int*
Later however, FS at least tries to speak:
2008-10-28 23:40:05 [NOTICE] mod_dptools.c:605 answer_function() Channel
Hello Michael,
No, I startet with a 5.1 installation.
Cepstral works on the command line
opt/swift/bin/swift -o hello.wav 'Hallo Peter'
And the voice is registered:
[EMAIL PROTECTED]:/opt/swift/bin# ./swift --voices
Swift command-line synthesis program
Version 5.1.0 of July 2008
Copyright (c)
PROTECTED] On Behalf Of Peter P GMX
Sent: Tuesday, October 28, 2008 4:16 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cepstral 5.1 no sound
Hello Michael,
No, I startet with a 5.1 installation.
Cepstral works on the command line
opt/swift/bin/swift -o
... sorry I couldn't be of further
assistance.
-MC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:freeswitch-
[EMAIL PROTECTED] On Behalf Of Peter P GMX
Sent: Tuesday, October 28, 2008 4:36 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users
I reverted to this old libs/apr
and after compiling the complete freeswitch (compiling only apr did't
work) it finally worked.
Thanks for your support
Peter
Wasim Baig schrieb:
On Wed, Oct 29, 2008 at 7:20 AM, Peter P GMX [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I did a svn log
I tried to read dtmfs via event socket and came across this thread.
What do you mean by:
If you're using event socket you have really no reason to use the read
application.
Is there another chance to do this?
Currently I do the following and this isn't successful:
I send the following message
It's rather simple
- Setup a sip user on asterisk with username/password
- Setup a gateway in freeswitch with the asterisk user credentials (ip,
username, password of asterisk)
- Define a route in the dialplan (e.g. default.xml) to route certain
numbers to the asterisk gateway
e.g.
extension
I receive a FACILITY_NOT_SUBSCRIBED message when I call from 1000 to 1001.
I've read in the wiki:
50 FACILITY_NOT_SUBSCRIBED requested facility not subscribed
[Q.850 This cause indicates that the user has requested a
supplementary service, which is available, but the user is not
} ${dialed_extension})
[EMAIL PROTECTED] schrieb:
Send a full debug from the FS console when a call is placed. That
should give more of a clue as to where the issue is.
Peter P GMX wrote:
I receive a FACILITY_NOT_SUBSCRIBED message when I call from 1000 to 1001.
I've read
Addendum:
I fixed the sip_secure_media=[undef] problem. However no change.
Best regards
peter
Peter P GMX schrieb:
Hello,
here is the complete Log of the relevant part:
[EMAIL PROTECTED] 2008-11-10 18:12:56 [DEBUG] sofia.c:3467
sofia_handle_sip_i_invite() IP xxx.xx.xx.186 Rejected
But what does this error message mean? I started with the latest SVN
version with standard setup, tried to do an internal call (all phones
are behind NAT on different public IPs) and it failed.
I cannot reach either of the phone, although sofia status profile
internal shows me the correct
Nobody has a clue why I cannot connect the 2 phones internally? I
googled around, I can see some other people asking for this problem at
various places, but no reply so far.
Best regards
Peter
Peter P GMX schrieb:
Addendum:
I fixed the sip_secure_media=[undef] problem. However no change
regards
Peter
Anthony Minessale schrieb:
that can't be latest trunk?
did you update with make current and see it actually update?
and revert to the default config?
do you possibly not have mod_dptools loaded?
On Tue, Nov 11, 2008 at 11:22 AM, Peter P GMX [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED
On Nov 11, 2008, at 4:14 PM, Peter P GMX wrote:
http://pastebin.freeswitch.org http://pastebin.freeswitch.orgasked for
login credentials. Any idea where to get them from? I googled around, no
solution fund. Wiki credentials don't work.
Best regards
Peter
The second part is in:
http://pastebin.freeswitch.org/6099
Peter P GMX schrieb:
Aaargh, being able to read can be a real advantage sometimes.
I have now put the log to
http://pastebin.freeswitch.org/6098
Best regards
Peter
Brian West schrieb:
If you look very close in the dialog box
:
It appears to have been cutoff. The last line that I see is:
2008-11-11 23:38:27 [NOTICE] switch_loadable_module.c:281
switch_loadable_module_process() Adding F
Peter P GMX wrote:
Aaargh, being able to read can be a real advantage sometimes.
I have now put the log
not name your profile the same name as your domain or at
least have an alias for the profile that is equal to your domain name
you will not be able to use the user channel.
You can always opt for the alternate method of sofia/profile
name/user%domain
On Tue, Nov 11, 2008 at 6:53 PM, Peter P GMX
Please have a look at the wiki
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples
In which country do you need a provider?
So far I got every provider I tried (5) to work with freeswitch; even
with FS behind NAT.
Andy Ayers schrieb:
Hi,
Can anyone recommend any good VOIP providers that
Dear all,
when I get an incoming call from my SIP provider I do receive an invite
on the trunk number, e.g. in Germany 0x9. However I have an
extension block of 0x90 to 0x999.
In the dialplan I checked the condition field=destination_number but
this compares to the number of the trunk
. variable_sip_to_user
is empty.
What can I do?
Best regards
Peter
Anthony Minessale schrieb:
route the call to the info app and look for the one that has it.
On Wed, Nov 12, 2008 at 1:36 PM, Peter P GMX [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Dear all,
when I get an incoming
Thanks,
this works
Best regards
Peter
Anthony Minessale schrieb:
here's another hint.
use field=${sip_to_user}
On Thu, Nov 13, 2008 at 3:43 AM, Peter P GMX [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I checked your hint. The variable sip_to_user has the right value
If you tell FS to use Port 5090 then FS will be contacted through port
5090. But ports are not always symmetrical on both sides. The UA (e.g.
Zoiper) will use an available port on his side, that can be 5060 but
often is 2051 or even a higher port. Imagine, you may have several local
phones with
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
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, 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,
and
On Nov 20, 2008, at 12:23 PM, Peter P GMX wrote:
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
I have set up a FS behind a NAT.
Calling internal numbers, VM etc. works from a different network
(Caller's phone is also natted).
However if I call an external Gateway the call is terminated as soon as
the remote party lifts the handset. And Freeswitch is delivering
SIP/2.0 500 Cannot Get IP
Nobody has an idea?
Best regards
Peter
Peter P GMX schrieb:
I have set up a FS behind a NAT.
Calling internal numbers, VM etc. works from a different network
(Caller's phone is also natted).
However if I call an external Gateway the call is terminated as soon as
the remote party lifts
Brian West schrieb:
Ok this is why.. DO NOT put hostnames in the external RTP ip or the
sip ip's those are ONLY IP addresses.
/b
On Nov 21, 2008, at 2:00 PM, Peter P GMX wrote:
o=FreeSWITCH 1227284490 1227284492 IN IP4 my.domain.com
s=FreeSWITCH
c=IN IP4 my.domain.com
I do and I got it finally working under Ubuntu 8.041:
Here is how it worked under Ubuntu 8.04:
First, disable mod_flite as they are incompatibel
Then set environment var export SWIFT_HOME=/opt/swift
Check if /opt/swift /lib/libswift.so.5 link exists
create a file /etc/ld.so.conf.d/swift.conf
I have installed OpenZAP with a TE220 card and EuroISDN.
When I bridge an outgoing call I get a Received unhandled message 125
Any idea what that means? As far as I know there is no external result
code 125 defined in the ISDN protocol.
Best regards
Peter
See the following logs.
2008-11-27
has anybody already used playAndGetDigits via event socket and has a
working example?
Best regards
Peter
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Hello Maxim,
can you reach another internal device except the GSM one in order to see
whether it's GSM codec specific?
However I can see that you're using local IPs (10.x.x.x) so I expect
that they are natted. This often causes one way audio when the external
rtp-ip is not set. Please try to set
Did you add
action application=export data=sip_secure_media=true/
into youy dialplan before bridging that call. How is your internal.conf,
is TLS enabled there?
Best regards
Peter
matrim schrieb:
Hi,
I'm having problems using TLS to receive calls.
I'm using a Nokia N95 to test TLS against
I am building an IVR application where an incoming call is initiating an
outgoing call. When I pass a variable_other_uuid (the uuid of the
incoming channel) at originate time, I am able to reference to the
incomig call, once the outgoing call is set up. So the outgoing call can
see the uuid of the
be used for my issue.
Best regards
Peter
Michael S Collins schrieb:
What is your originate string?
-MC
On Dec 5, 2008, at 3:54 AM, Peter P GMX [EMAIL PROTECTED] wrote:
I am building an IVR application where an incoming call is
initiating an
outgoing call. When I pass a variable_other_uuid
I try to play mp3 I generated through Cepstral TTs and which I encoded
via lame.
However they won't play, so my question is: Which mp3 formats are supported?
I generate the wav files by the following
/opt/swift/bin/swift -n Katrin -p
I use Telegraph with Ruby on Rails to listen on the event socket
interface. With Telegraph you can register on any FS event and get all
channel variables in a hash for further processing. Interactions then
can be done via event_socket intreface.
Telegraph is not finished yet, but for me it was a
according to the wiki wiich should be
iunstalled, and they are there: Configure does not show any warnings.
Nobody has a clue what may be the problem here?
Best regards
Peter
Peter P GMX schrieb:
I try to play mp3 I generated through Cepstral TTs and which I encoded
via lame.
However
When I send a stop_dtmf command via event-socket, I get a
channel_execute and a channel_execute_complete message back. However FS
still accepts DTMFs and sends them via event-socket. In addition the
other party will hear the DTMF. So I expect the stop_dtmf command is not
really executed by FS.
for the inband dtmf listener, I would guess you are
getting dtmf via rfc2833 or some other method. If you want to
understand why we generate all those packets have a read of rfc 2833.
Mike
On Dec 29, 2008, at 5:46 PM, Peter P GMX wrote:
When I send a stop_dtmf command via event-socket
As I see on the Wiki page uuid_playback seems to be implemented, however
it doesn't work on the console or via event_socket.
Also in the code I could not find it (svn 10438).
So for now I use uuid_brodcast to play announcements to one or both parties.
Question: What is the status of
Here's the link
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_playback
Brian West schrieb:
Wiki link please.
/b
On Dec 31, 2008, at 8:28 AM, Peter P GMX wrote:
As I see on the Wiki page uuid_playback seems to be implemented,
however
it doesn't work on the console or via
Generally speaking you will need to open an UPD port range for the RTP
stream. This can be configured on FS. Eg. we use 12000-13000 on our system.
Then If you do not hear any sound you may put
param name=ext-rtp-ip value=stun:stun.freeswitch.org/
in your external and internal profile, if FS
at 5:00 AM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
Hello Anthony,
I updated to SNV=11084 and still have the problem. The behaviour is
slightly changed now.
Step 4+5 (a s below in my mail)
4) When I bridge A and B, A and B canNOT hear each
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
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
zap_channel_done() channel
done 1:1
Michael Collins schrieb:
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 prometheus...@gmx.net wrote
Hello Michael,
sorry for the inconvenience. It turned out that our Telco had to reset
the second PRI line. Now it works.
Best regards
Peter
Peter P GMX schrieb:
Hello Michael,
here is a log of 2 calls. The first is one successfull, the second not.
Bestr regards
Peter
2009-01-08 17:57:29
of the call into park and then
bridge via uuid_bridge.
Still I have no idea why this is different from a former intstallation.
Best regards Peter
Peter P GMX schrieb:
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
to you send to the socket and exact apps with args
you execute in the dialplan.
are you on irc, or im so i can ask you more questions live?
On Fri, Jan 9, 2009 at 2:03 PM, Peter P GMX prometheus...@gmx.net
mailto:prometheus...@gmx.net wrote:
I investigated a bit more in this.
If I
From my experience it shows red when:
- the cables are not connected
- the pri lines are not synchronized yet
- sometimes the telco provider has to reset the line
Best regards
Peter
Michael S Collins schrieb:
This looks like a zaptel issue. Do we have any zaptel users familiar
with this
what kind of ec2 machine is it? Linux/Distribution? Windows?
best regards
Peter
Rehan Allah Wala schrieb:
i am looking for a consulant to send me a quote to run this for me on amazon
ec2
Rehan
Ciao FreeSWITCHers,
mod_skypiax is now usable, for Skype calls and finding bugs :-).
After a time I receive the following error when a call comes in on our
OpenZap span 2:
parse error [-3012] [Q931E_INVALID_CRV]
Here's the log
2009-01-14 13:14:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got
an event! Type:[4d] Size:[103] CRV: 23 (0x17, CTX: Originator)
2009-01-14 13:14:11
]
Peter P GMX schrieb:
After a time I receive the following error when a call comes in on our
OpenZap span 2:
parse error [-3012] [Q931E_INVALID_CRV]
Here's the log
2009-01-14 13:14:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got
an event! Type:[4d] Size:[103] CRV: 23 (0x17, CTX
to pony up serious $$ to support
OpenZAP development which means it is progressing at the speed of
developers' free time.
-MC
On Wed, Jan 14, 2009 at 9:44 AM, Peter P GMX prometheus...@gmx.net wrote:
After a time I receive the following error when a call comes in on our
OpenZap span 2
Is there a way to use the hardware timers e.g. of a PRI card in
fresswitch? Or other question: Is it recommended to use those if they
are available?
I have installed a dual PRI card, and show timer shows one soft timer.
Best regards
Peter
___
15.01.2009 09:20, schrieb Peter P GMX:
Hello Michael,
how much $$ are we talking about? I need this issue to be solved quickly
and it's worth to spend some money.
I've read the following post:
http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg05792.html
and have
Maybe it's a good idea to implement a wireshark export for those
messages in FS. This will make debugging easy and cheap.
Hope it helps a bit.
regards
helmut
Am 15.01.2009 12:06, schrieb Peter P GMX:
Helmut,
can you give me a hint, how you worked around this?
Best regards
Peter
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