Hi all,
I've created a wiki page, which contains the example configuration for
making Digium TE220 to work.
I request you people to check this, and give feedbacks.
http://wiki.freeswitch.org/wiki/Configuration_OpenZap-DigiumTE220-Example
___
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Hello Anthony,
On 19.10.2009 22:07, Anthony Minessale wrote:
please update and test trunk
1) I changed the core to remove the excess data by default in your scenario
2) I added variables you can use to control it
origination_callee_id_name
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Hello,
I wonder if there is real stereo support planned for FreeSWITCH in terms
of streaming music/video-audio to desktop? I ask, because I heared that
celt codec is supporting stereo.
regards
helmut
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Hi Mark,
I confirm that it works fine. I use it for two years now and there were
no problems compiling or running FS on ubuntu 8.04 caused by the OS.
This is my system:
lsb_release -a
No LSB modules are available.
Distributor ID: Ubuntu
TC
TCcall flow is SIP_user = FS = H323_endpoint is failing ..
coredumped
TChttp://pastebin.freeswitch.org/10703
i fix some bugs now,
ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this
is
updated version, try it, if you experience no audio try enable rtp proxy in
Hello,
I am using the same set of extensions for testing the system during
development, they include XLite, Cisco sip phone and several extensions that
just play some audio file.
Sometimes, very rarely, this message Can not record session. Media not
enabled on channel. appears on FS console.
Yeah. A call to B and C eavesdrops call. I send dtmf for C to talk with B,
but B can't hear C. Here, what I had done in details:
[r...@centos4-4-vm ~]# telnet localhost 8021 Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is '^]'.
Content-Type:
Hello,
I us a call forward on freeswitch to forward calls to my mobile phone. If now
Freeswitch forwards a call, the number information presented to my mobile phone
is that of the original caller, because that is what there is appearing in its
display. What I want
to see is that
Hi,
Consider below mob_nibblebill configuration,
configuration name=nibblebill.conf description=Nibble Billing
settings
param name=db_username value=root/
param name=db_password value=password/
param name=db_dsn value=MySQL-freeswitch/
param name=db_table value=accounts/
Dear Sir,
I'm using mod_odbc_query and mod_nibble_billing for my calling card
solutoin. i found mod_odbc_query cant work with high load (200 calls
concurrent)
i got error
[STATE: 53200 CODE 7 ERROR: [unixODBC]ERROR: out of shared memory;
Error while executing the query
]
in FS console
How to fix
On 20/10/09 07:53 +0200, ineya ineya wrote:
IP should be OK:
[r...@franta /opt/freeswitch]# ifconfig eth1
eth1 Link encap:Ethernet HWaddr 00:4f:4e:62:ad:83
inet addr:10.80.62.40 Bcast:10.80.62.255 Mask:255.255.255.0
inet6 addr: 2000:2::1/32 Scope:Global
This address
Hi Dome,
Have you tried increasing global_heartbeat to reduce the frequency of
odbc calls? What is it currently set to?
Rob
On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote:
Dear Sir,
I'm using mod_odbc_query and mod_nibble_billing for my calling card
solutoin. i found mod_odbc_query
2009/10/20 Rob Forman rob4manh...@gmail.com:
Hi Dome,
Have you tried increasing global_heartbeat to reduce the frequency of
odbc calls? What is it currently set to?
Now 1 min
Rob
On Oct 20, 2009, at 8:31 AM, Dome Charoenyost wrote:
Dear Sir,
I'm using mod_odbc_query and
Have you tried setting the effective_caller_id_number before
bridging? Such as:
action application=set data=effective_caller_id_number=9185551212/
Cheers,
Rob
On Oct 20, 2009, at 6:55 AM, Durk de Beer wrote:
Hello,
I us a call forward on freeswitch to forward calls to my mobile
phone. If
Hi,
I haven't run into that problem yet, but did you try increasing the
maximum shared memory in /proc/sys/kernel/shmmax (sysctl
kernel.shmmax) ?
regards,
Leon
On Oct 20, 2009, at 3:31 PM, Dome Charoenyost wrote:
Dear Sir,
I'm using mod_odbc_query and mod_nibble_billing for my calling
Try 300 seconds (5 minutes) and see if it improves.
On Oct 20, 2009, at 9:00 AM, Dome Charoenyost wrote:
2009/10/20 Rob Forman rob4manh...@gmail.com:
Hi Dome,
Have you tried increasing global_heartbeat to reduce the frequency of
odbc calls? What is it currently set to?
Now 1 min
Rob
No, it's the same thing with /64 on both ends.
I tried to built SVN version and modified just IP addresses in
sip_profiles, but still can't call from one phone to another. It goes
straght to voicemail.
Is there a softphone, which you know works with freeswitch on IPv6?
Maybe the error is in
Well how are you trying to dial users?
/b
On Oct 20, 2009, at 9:16 AM, ineya ineya wrote:
No, it's the same thing with /64 on both ends.
I tried to built SVN version and modified just IP addresses in
sip_profiles, but still can't call from one phone to another. It goes
straght to voicemail.
Gabriel Gunderson wrote:
On Mon, Oct 19, 2009 at 11:35 AM, Mark Sobkow
m.sob...@marketelsystems.com wrote:
Everyone I've emailed with on the dev list is running the current
release of Ubuntu, not 8.04/Hardy.
Well, what issues?
Gabe
2009/10/20 Rob Forman rob4manh...@gmail.com:
Try 300 seconds (5 minutes) and see if it improves.
Ok. i'll try
On Oct 20, 2009, at 9:00 AM, Dome Charoenyost wrote:
2009/10/20 Rob Forman rob4manh...@gmail.com:
Hi Dome,
Have you tried increasing global_heartbeat to reduce the frequency of
2009/10/20 Leon de Rooij l...@scarlet-internet.nl:
Hi,
I haven't run into that problem yet, but did you try increasing the
maximum shared memory in /proc/sys/kernel/shmmax (sysctl
kernel.shmmax) ?
Recommend me please i have 4GB RAM.
is posible to increase to 50% RAM ?
Dome C.
regards,
Now I am building a PHP SOAP Web Service to access the database of FS. Anyone
has idea about how to access sqlite database of FS through PHP ? I have read
about socket event in FS, but I don't know whether it can response with the
query of database or not.
Thanks for your help.
--
View this
On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC TC
TC TCcall flow is SIP_user = FS = H323_endpoint is failing ..
TC coredumped
TC TChttp://pastebin.freeswitch.org/10703
TC
TC i fix some bugs now,
TC
On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC TC
TC TCcall flow is SIP_user = FS = H323_endpoint is failing ..
TC coredumped
TC TChttp://pastebin.freeswitch.org/10703
TC
TC i fix some bugs now,
TC
You could, but I would try just doubling whatever it is to see if
thats improves the issue first. The default is 32MB. You could
double it to 64MB and test again.
What is it currently set to (run: sysctl kernel.shmmax)? You can
change it on the fly with sysctl. Once you're done testing
2009/10/20 Rob Forman rob4manh...@gmail.com:
You could, but I would try just doubling whatever it is to see if
thats improves the issue first. The default is 32MB. You could
double it to 64MB and test again.
Ok' let's me try 64MB first
i found some information
On 2009-10-20 10:17 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC TC
TC TCcall flow is SIP_user = FS = H323_endpoint is failing ..
TC coredumped
TC TChttp://pastebin.freeswitch.org/10703
TC
TC i fix some bugs now,
TC
By numbers, I have 2 numbers registered, so I dial 1006 or 1007.
This worked for IPv4, so I haven't thought about doing it differently for IPv6.
SIP messages looked OK to me - you just gave me an idea to compare SIP
messages for IPv4 and IPv6. I'll try that tommorow.
On Tue, Oct 20, 2009 at 4:21
Those are recommended values for Oracle. Freeswitch != Oracle. They
behave and use a server's resources very differently.
I wouldn't change more than you need for now. Tweak shmmax then go
from there.
Cheers,
Rob
On Oct 20, 2009, at 10:14 AM, Dome Charoenyost wrote:
2009/10/20 Rob
maybe next time test and/or search the mailing list before asking. I
was a little worried when I read that it do not works on Hardy. Good
to be reassured, it works. :-)
On 10/20/09, Mark Sobkow m.sob...@marketelsystems.com wrote:
Gabriel Gunderson wrote:
On Mon, Oct 19, 2009 at 11:35 AM, Mark
OMG such a stupid user error!
I have to write
action application=bind_meta_app data=7 ab s
/usr/local/freeswitch/conf/dialplan/execute_extension::roar XML features/
with double colon instead of a single one.
Thanks everyone for help!
Uncle Johny wrote:
Hi guys,
I hope you can help me
There is not enough spec/devices using spec on stereo to try to implement it
at this time.
Maybe some day.
On Tue, Oct 20, 2009 at 2:35 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
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Hello,
I wonder if there is real stereo support planned for
Now i'm setting to 64MB
I found channel problem also. many channel not disconnect
freeswi...@internal show calls count
87 total.
freeswi...@internal show channels count
466 total.
freeswi...@internal status
UP 0 years, 0 days, 1 hour, 53 minutes, 23 seconds, 872 milliseconds,
0 microseconds
The defaults will NOT work with ipv6 out of the box because the
sofia_contact on the directory only looks at the internal profile NOT
the internal-ipv6 profile... open up the directory default and change
the sofia_contact to prepend the internal-ipv6/u...@domain
/b
On Oct 20, 2009, at
Dear All
What's CS_REPORTING state ?
I found many channels not hang up ans state is CS_REPORTING
e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20
On Tue, Oct 20, 2009 at 12:25 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
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Hello Anthony,
On 19.10.2009 22:07, Anthony Minessale wrote:
please update and test trunk
1) I changed the core to remove the excess data by default in your
Or just set the var to what you want it to say?
/b
On Oct 20, 2009, at 11:19 AM, Michael Collins wrote:
Under what conditions did you see unknown? I'm wondering if the
user can just pick a default other than unknown if he wants
something else to be displayed.
Thoughts?
-MC
Hmm, I didn't about this.
so in directory/default.xml I have:
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}/
and the modified version for IPv6 would be ?...
param name=dial-string
FYI,
The conference call agenda for this week is online:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_23
Also, we will be discussing whether or not Friday is the best time to be
doing this call. Just remember that the FS devs have lots of work to do and
they can't always accommodate
Hello all,
As you may know we have a weekly conference call each Friday. This week's
agenda has several wiki cleanup sub-projects. I wanted to crowdsource this
because, as the saying goes, many hands makes the work light. In other
words, if everyone can help a little bit then we won't be dumping
William,
Where is mod_skypiax.so? Isn't freeswitch-skypiax package expected to
contain it?
- Dmitry Bely
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If you really wanted: http://php.net/manual/en/book.sqlite.php
But I would recommend you make use of ODBC to use a client/server RDBMS.
Here's some good reading:
http://www.sqlite.org/cvstrac/wiki?p=WhenToUseSqlite
On October 20, 2009 10:53:01 am homqua wrote:
Now I am building a PHP SOAP Web
Thanks. I added minor changes, plus a category and see also. Thanks for
adding this content!
-MC
On Tue, Oct 20, 2009 at 12:18 AM, lakshmanan ganapathy lakindi...@gmail.com
wrote:
Hi all,
I've created a wiki page, which contains the example configuration for
making Digium TE220 to work.
Hello everyone,
I'm trying to use proxy media across two profiles. The codec
settings are identical, they both have late negotiation enabled, and
they both have inbound-proxy-media set to true (I also tried setting
proxy_media from the dialplan).
FreeSWITCH ends up clearing the call with
Fixed in 15181 =D 1 revision higher doh =D
On Tue, Oct 20, 2009 at 2:56 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Hello everyone,
I'm trying to use proxy media across two profiles. The codec
settings are identical, they both have late negotiation enabled, and
they
you are making FS to play wav file when sending a call in G711 or GSM or
some other codec.
you might use mod_native_filehttp://wiki.freeswitch.org/wiki/Mod_native_fileto
avoid transcoding.
T.
On Tue, Oct 20, 2009 at 9:56 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Hello
Updated and now working.
Thanks!
On Tue, Oct 20, 2009 at 4:11 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
Fixed in 15181 =D 1 revision higher doh =D
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
Hello everyone,
Now that proxy media mode is working again I'm trying to figure out
why T.38 with re-INVITE doesn't...
Everything goes well until my end tries to re-INVITE to T.38:
U 10.16.5.129:5060 - 65.196.170.191:5060
INVITE sip:mod_so...@65.196.170.191:5060 SIP/2.0.
Via: SIP/2.0/UDP
Fix this or set it to silence.
/b
On Oct 20, 2009, at 3:57 PM, Kristian Kielhofner wrote:
EXECUTE sofia/s2s/+19412848...@65.196.170.129 playback
(local_stream://moh)
2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file
format [local_stream] for [moh]!
1. Can I email the voicemail message to multiple email addresses?
I revisited this again after not requiring it for a while. A comma
separated list in the extension.xml file does work. The problem was
the template file. Once I removed and in the To: field, I can
send to multple emails no
What's in the dialplan for this channel? Is bypass-media or proxy-media set
to true? Do a debug trace and post it in pastebin.
-MC
On Tue, Oct 20, 2009 at 3:30 AM, Maciej Aniserowicz
maciej.aniserow...@gmail.com wrote:
Hello,
I am using the same set of extensions for testing the system
REPORTING is the state that it writes to CDR. If you have calls stuck
in this state, take one and try to use uuid_kill on it and see if it
goes away, then get a core off of it and pastebin the thread apply all
bt (with no other calls up). What modules are you using for cdr and
with what
If you really want to access this information outside I would strongly
recommend using odbc instead of the internal sqlite db, it does not
handle locking contention well. If you need access to things in the
core db (like show calls and show channels information) you will need
to write a
Brian,
It's already set to silence :).
My guess here is that Sofia is mis-interpreting the re-INVITE as not
having any media (a la RFC 2543 hold).
I don't want to place the call on hold. I want to negotiate T.38 ;).
On Tue, Oct 20, 2009 at 5:35 PM, Brian West br...@freeswitch.org wrote:
Anthony,
I will the next chance I get.
Thanks!
On Tue, Oct 20, 2009 at 5:34 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
issue:
console loglevel debug
sofia profile internal siptrace on
and put it on pastebin
http://pastebin.freeswitch.org
--
Kristian Kielhofner
Done.
On Tue, Oct 20, 2009 at 5:34 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
issue:
console loglevel debug
sofia profile internal siptrace on
and put it on pastebin
http://pastebin.freeswitch.org
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
ineya ineya ine...@gmail.com wrote:
Hmm, I didn't about this.
so in directory/default.xml I have:
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}/
and the modified version for IPv6 would be ?...
param
Hi to all of you guys (and ladies too).
I would like to bother all of you and take just a little of your time.
I am a total NOOB on freeswitch and would like a little help.
I need to know from all of you guys which would be the hardware recommended
from all of you guys to be able to deal with a
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