hi,
do you have idea if is possible detect if a call to device(1) is from
queue? (i.e. if app_queue set some variable)
exten => 800,1,queue(sales) ; queue pick exten 20
exten => 20,1,noop("detect variables")
exten => 20,n,Dial(SIP/20)
(1) its through a local interface i.e Local/20@phones
hi,
what's your experience with asterisk compiled with libsrtp 2.x and
WebRTC(pjsip)?
issues/crashes/speed/cpu usage?
Marek
official status https://wiki.asterisk.org/wiki/display/AST/libsrtp
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hi,
i have webrtc client chrome69/jssip which is connecting to asterisk
13.23.1/pjsip
i have strange problem where pjsip aor stays in status "created"
sip trace on asterisk looks ok.
do you think if this can be bug?
test*CLI> pjsip show aors
Aor:
Contact:
hello,
i met with this interesting situation
[Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport '8' since no request was received in 32 seconds
[Jan 24 13:48:37] NOTICE[1049] res_pjsip_transport_management.c:
Shutting down transport '8' since no request
Dne 26/09/2017 v 22:33 Joshua Colp napsal(a):
On Tue, Sep 26, 2017, at 05:29 PM, marek cervenka wrote:
hi,
i want use asterisk+pjsip as voip client with multiple registrations
(perf testing)
i'm using this example configuration for one account
[308]
type=registration
outbound_auth=308
hi,
i want use asterisk+pjsip as voip client with multiple registrations
(perf testing)
i'm using this example configuration for one account
[308]
type=registration
outbound_auth=308
server_uri=sip:3...@example.com:5060
client_uri=sip:3...@example.com:5060
[308](auth-userpass)
username=308
my perftest suite
call generator
sipp, but creating sipp scenario is not easy. i'm using more user
friendly(but its for win) - http://startrinity.com (REST API available)
device emulation
using asterisk as SIP client in docker - 30 SIP endpoints per instance
reports
pbx cpu/load/.. -
hi,
i know about architecture limits of app_queue
https://issues.asterisk.org/jira/browse/ASTERISK-25806
what CPUs are you actually using for asterisk + app_queue ? (my actual
scenario 90simult calls, 50agents, call recording to SSD (mixmonitor),
no transcoding, CDR/CEL via odbc to
hi,
is there somebody who is using say.conf mode=new in Asterisk 13?
i'm searching for tips what to try in
https://issues.asterisk.org/jira/browse/ASTERISK-15421
Marek
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point:ENDPOINT,
app:'originate-example',
appArgs:'dialed',
callerId:'7' });
can i specify it in endpoint somehow?
Dne 30/06/2017 v 10:45 marek cervenka napsal(a):
my use case is for performace testing
scenario
asterisk14 - sip - tested asterisk - sip - clients (asterisk 1
call e.g. for this example
https://github.com/asterisk/node-ari-client/blob/master/examples/promises/originate.js
Dne 29/06/2017 v 13:38 marek cervenka napsal(a):
hi,
do you have someone example of
http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/
in node.js
hi,
do you have someone example of
http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/
in node.js asterisk-ari ?
thanks
Marek
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hi,
i have mix of realtime and static configuration of pjsip
https://pastebin.com/YVFwVsMD
pjsip.conf
[global]
endpoint_identifier_order=username,ip,anonymous
user_agent=ipbx
...
transport definition
extconfig.conf
[settings]
ps_endpoints => odbc,configDb
ps_auths => odbc,configDb
can you someone confirm
https://issues.asterisk.org/jira/browse/ASTERISK-27065
its easy to replicate
Marek
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hi,
i'm using hangup handlers on Asterisk13
with standard answered calls i have 1 CDR per call
with scenario call from voip->mobile, call rejected on mobile i have 2 CDRs
i dont want the second CDR
without hangup handlers i have 1 CDR
do you think its bug or its feature of hangup handlers?
.c:1235
__clframe = {__cancel_routine = , __cancel_arg =
0x7f19a9c25700, __do_it = 1, __cancel_type = }
ret =
a = {start_routine = 0x5e65e0 , data =
0x7f1a1800ae00, name = }
#17 0x7f1a1e89ddc5 in start_thread () from /lib64/libpthread.so.0
No symbol table info available.
#18 0x7f1a1db7d7
when run from console without systemd i found its segfaulting
turned core dump on because it was off
Dne 09/05/2017 v 13:52 marek cervenka napsal(a):
hi,
i have strange problem with asterisk 13.15.0+pjsip bundled/centos
7/systemd start script
we are using chan_pjsip only for webrtc
hi,
i have strange problem with asterisk 13.15.0+pjsip bundled/centos
7/systemd start script
we are using chan_pjsip only for webrtc endpoints . switched from sipml5
to jssip with upgrade to 13.15.0(from 13.9.0) few days ago
today i have problems with stopping/crashing asterisk
hi,
what kernel version are you using for asterisk?
are you satisfied with distro kernel (centos 6 2.6.32, centos 7 3.10, ...) ?
are you using newer kernels from elrepo.org?
which kernel features are most critical for Asterisk performance pattern?
thanks
Marek
--
hi,
i have similar problem to
https://issues.asterisk.org/jira/browse/ASTERISK-25806
do you know about some workarounds/patches for better scalability?
thanks
marek
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thanks for confirmation
dns name in /etc/hosts & dnsmgr enabled solved my problem
Dne 14/12/2016 v 13:50 Joshua Colp napsal(a):
On Wed, Dec 14, 2016, at 08:47 AM, marek cervenka wrote:
i found this issue https://issues.asterisk.org/jira/browse/ASTERISK-26280
but its not c
hi,
i'm trying get report about missed calls per agent. i'm using queue_log
and RINGNOANSWER event
but i found problem described here
---
https://www.thirdlane.com/forum/queue-log-problem
RINGNOANSWER only happens if the call TIMES OUT ringing the agent and it
returns to the queue. If your
i found this issue https://issues.asterisk.org/jira/browse/ASTERISK-26280
but its not clear if this problem can be in chan_sip/udp created
channels & pjsip module is active only for wss transport
Dne 14/12/2016 v 12:14 marek cervenka napsal(a):
hi,
i have strange problem with no
hi,
i have strange problem with no rtp packets from asterisk after dns
query. see pcap below
centos6/asterisk 13.9 + chan_sip
172.23.0.3 - asterisk
172.23.5.1/2 - voip phones
any ideas/hints?
1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711
PCMA, SSRC=0x334508F6,
is around 10 dialplan commands (execif,set) + 1x fastAGI
do you think it's bug or timing "limit" of Asterisk?
Dne 30/11/2016 v 22:17 marek cervenka napsal(a):
hmm. i think customer will not agree this is correct behavior
from pcap it looks like there is missing CANCEL to the second dev
and the external call channel
connected.
The second device simply off hook but his channel have no external
channel to connect.
It's looks like a simple telephony glare.
Sam
בתאריך 30 בנוב' 2016 7:00 PM, "marek cervenka" <cerva...@gmail.com
<mailto:cerva...@gmail.com>> כ
hi,
our customer reports problem when 2 agents answer the call in the same time
faster operator (device) answer the call, but the second is showed up
(on device) and call is without sound
asterisk 13.9/app_queue with strategy ringall/operators via Local
channel with sip device (chan_sip)
hi,
can you recommend open source helpdesk solution with working Asterisk
integration?
marek
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Dne 20/10/2016 v 16:32 Matt Fredrickson napsal(a):
On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka <cerva...@gmail.com> wrote:
i tested this
# cat /etc/asterisk/extconfig.conf
[settings]
queue_log => sqlite3,cdrDb
# cat /etc/asterisk/res_config_sqlite3.conf
[cdrDb]
dbfile = /var/lib
28804||QUEUESTART|NONE|NONE|NONE
2016-10-20 11:40:36.690313||CONFIGRELOAD|NONE|NONE|NONE
column types needs modification to something more appropriate
can someone with confluence access ad info to
https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration ?
is there somebody usi
hi,
is it possible log cel/queue_log to sqlite?
via odbc?
any experience?
marek
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hi,
i'm testing CONNECTEDLINE function
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
example dialplan
same => n,set(CONNECTEDLINE(name,i)=aastra)
same => n,set(CONNECTEDLINE(name-pres,i)=allowed)
same => n,Set(CONNECTEDLINE(num,i)=5551212)
same =>
ansible.com
Dne 18/10/2016 v 11:46 Duncan napsal(a):
Hi All
We have about 15 different asterisk boxes around the place and on my
list has been automate deployment updates and keep a revision history.
They are mostly not publicly accessible, and external SIP access is
closely firewalled ,
hi,
i'm trying configure $subj
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger
but there is a ton of "informational" messages
[Sep 30 14:40:16] SECURITY[18311] res_security_log.c:
using in production
last asterisk 13 + pjsip bundled + pjsip patch for RTP/SAVPF (search
pjsip conf) + sipml5 version from roginvs
https://github.com/DoubangoTelecom/sipml5/pull/238
Dne 08/09/2016 v 23:36 Annus Fictus napsal(a):
Hello list,
before to lost my time, I'd like know if someone
hi,
i want switch my application server(dynamic routing) in node.js from
fastAGI to CURL because of
- easier development of REST API server
- testing and debuging
- AGI is not known in the web dev world
what do you think about curl from performance view? (10cps, 500 simult
calls per
Le 2015-08-10 13:54, Marek Cervenka a écrit :
Dne 6.8.2015 v 21:00 Sylvain Boily napsal(a):
Hello,
Le 2015-08-06 09:24, Marek Cervenka a écrit :
hi,
there is updated skills based routing patch for asterisk queue
please test if you have time
https://issues.asterisk.org/jira/browse/ASTERISK
flowing only in one direction)
call from person2(chrome) to person 3(chrome) - no audio on both side
(RTP flowing only in one direction)
BUT
call from person2(chrome) to person 3(Jitsi sip client) - works!
any tips howto find the problem?
--
---
Marek Cervenka
Dne 6.8.2015 v 21:00 Sylvain Boily napsal(a):
Hello,
Le 2015-08-06 09:24, Marek Cervenka a écrit :
hi,
there is updated skills based routing patch for asterisk queue
please test if you have time
https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22
You can
hi,
there is updated skills based routing patch for asterisk queue
please test if you have time
https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22
--
---
Marek Cervenka
.
Thanks in advance.
Best regards,
Ruban.S
--
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Dne 3.6.2015 v 17:57 Marek Cervenka napsal(a):
hello,
my webrtc calls ends after ~60seconds with res_rtp_asterisk.c: DTLS
failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3
alert unexpected message', terminating. any ideas where can be
problem? or howto debug this problem
,firefox)
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dtlsenable=yes was missing
thank you joshua
Dne 21.5.2015 v 22:53 Marek Cervenka napsal(a):
hi,
is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?
or is chan_pjsip better supported?
or the recommended way for asterisk is use respoke.io?
my problem with asterisk13+chan_sip
request:
cf2990ba-3f12-3d9e-adb6-52889c414ed3
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Marek Cervenka
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?
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Marek Cervenka
===
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.
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bounty offer prolonged to 31.4.2015 (end of april)
Dne 3.3.2015 v 16:22 Marek Cervenka napsal(a):
hi,
i'm offering bounty[1] $500 (five hundred) US dollars for resolving
https://issues.asterisk.org/jira/browse/ASTERISK-22708
fix must be available for asterisk 11.x and asterisk 13.x
hello,
i'm searching second BOUNTY donor ($250)
for
https://issues.asterisk.org/jira/browse/ASTERISK-22708
if you want participate, please contact me privately
--
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Marek Cervenka
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hi,
is it possible connect call to queue to specified agent?
like
Mr. Neo called helpdesk queue, call picked by agent Smith
Mr. Neo is calling again and i want connect him with agent Smith
--
---
Marek Cervenka
hi,
i upgraded few asterisk systems from last asterisk 1.8 to 11 and i see
in graph that cpu usage is ~50% higher
any ideas? configuration, modules, .. is the same
--
---
Marek Cervenka
hi.
i have dialplan with 2 simultaneous calls - dial(sip/phone1sip/phone2).
when i cancel call on phone1 (push reject button), the call is still
ringing on phone2
can i cancel call on both phones from one place(one phone)?
thanks
--
---
Marek Cervenka
hi,
i want convert mixmonitor recorded speech audio from wav to mp3 or aac
can you recommend your settings for speech audio? filters, noise
elimination, compression ratio, ...
i will probably use lame
thank you
--
---
Marek Cervenka
://bugzilla.redhat.com/show_bug.cgi?id=1140324
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New
lines).
1 out of 4 hunks FAILED -- saving rejects to file res/res_rtp_asterisk.c.rej
thanks
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192.168.10.1
are there some ways for this scenario?
1) chan_pjsip?
2) kamailio in front of asterisk on the same server?
3) iptables magic?
4) ...
thanks
--
---
Marek Cervenka
transport=transport-udp-net2
can you someone confirm this solution?
Dne 29.8.2014 v 11:26 Marek Cervenka napsal(a):
hi,
i need migrate customers from severeal to one asterisk server with
multiple ip aliases
like
eth0 192.168.10.1
eth0:1 192.168.10.20
eth0:2 192.168.10.30
i must preserve
hello,
can you recommend good asterisk-SugarCrm integration plugin?
i googled a lot, but i want something what is used on daily basis
thank you
--
---
Marek Cervenka
it's old. sugarcrm v7 is not supported
Dne 28.8.2014 v 14:54 Scott Griepentrog napsal(a):
I've used this before, and it appears to still be an active project.
https://github.com/blak3r/yaai
On Thu, Aug 28, 2014 at 7:29 AM, Marek Cervenka cerv...@fpf.slu.cz
mailto:cerv...@fpf.slu.cz wrote
on this.
Lorenzo
--cite--
Dne 24.1.2014 10:42, Gareth Blades napsal(a):
On 23/01/14 23:37, Marek Cervenka wrote:
can someone confirm that mp3 is unsupported? is patch available?
what about patch for Opus?
uncle google doesnt know
MP3 is only supported for reading not writing. Its a patent
i'm talking about native mp3,opus support in mixmonitor application.
read the first answer from Gareth Blades
Dne 24.1.2014 1:39, Patrick Lists napsal(a):
On 24-01-14 00:37, Marek Cervenka wrote:
can someone confirm that mp3 is unsupported? is patch available?
Iirc mp3 was in asterisk
hi,
which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor
can i record to Opus?
--
---
Marek Cervenka
can someone confirm that mp3 is unsupported? is patch available?
what about patch for Opus?
uncle google doesnt know
Dne 23.1.2014 16:31, Gareth Blades napsal(a):
On 23/01/14 15:21, Marek Cervenka wrote:
hi,
which file extensios are supported in mixmonitor application?
https
hi,
i have strange problem with call-limit/groupcount limiting. i set up
limit of 2 calls.
i'm using both methods but a for few times i have problem with
successfull fraud with more calls than 2
asterisk is 1.8.22
someone with the same problem?
any ideas how to solve or debug this problem?
Dne 14.8.2013 13:35, Marek Cervenka napsal(a):
hi,
i have strange problem with call-limit/groupcount limiting. i set up
limit of 2 calls.
i'm using both methods but a for few times i have problem with
successfull fraud with more calls than 2
asterisk is 1.8.22
someone with the same problem
it is possible? any recommendations?
--
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Marek Cervenka
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hello,
any news about WebM/VP8 support in asterisk?
some bounty where can i contribute?
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hello,
do you have someone connector to salesforce?
http://wiki.developerforce.com/page/Open_CTI
i need it for Mac OS X (the web/javascript way, not the old CTI Adapter way)
i'm using Asterisk 1.8
thanks
--
---
Marek Cervenka
10.0.0.213 - 10.0.0.193 SIP Status: 200 OK
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New to Asterisk
) quality.
check asterisk testsuite
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation
thereis scenarios for console sip client pjsua(from pjproject) which can
perform speech quality measurement
marek cervenka
https://wiki.asterisk.org/wiki/display/AST/CEL+Function
on this wiki is THIS IS NO LONGER TRUE REWRITE
is there some way to write userfield,accountcode to the cel?
Dne 5.6.2012 13:21, Marek Cervenka napsal(a):
hello,
is there someone who successfully get info about attended transfer
from CEL
Dne 20.6.2012 18:40, Marek Cervenka napsal(a):
https://wiki.asterisk.org/wiki/display/AST/CEL+Function
on this wiki is THIS IS NO LONGER TRUE REWRITE
is there some way to write userfield,accountcode to the cel?
solved. it's set(CHANNEL(userfield)=something)
another question
i'm using
with D (consultation)
time A with D
time A with everyone (full time - from start to the end of call)
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Dne 29.5.2012 18:23, Kevin P. Fleming napsal(a):
On 05/29/2012 07:57 AM, Marek Cervenka wrote:
is it possible with simple CDR fully describe axfer? (axfer is asterisk
native, not phone function)
No, it is not. CDRs (Asterisk or otherwise) are only capable of
directly (simply) describing
ring time?)
is it possible? if yes, can you post some example?
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Dne 30.4.2012 11:09, Bharat Lalcheta napsal(a):
Hiii all,
I am using asterisk 1.8.9.2 and compile all modules related to calendar.
neon version is 0.29.6. OS is ubuntu 11.10.
I configured ical for zimbra, caldav for google mail and ews for
exchange 2010 calendar.
ical and caldav setup
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On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote:
Am 05.10.2011 20:42, schrieb Marek Cervenka:
hello,
is there some way to notify people in the same pickup group about call
from caller to callee?
i.e. i have call from 111 to 222
there are 222,333,444 in the same pickup group
333,444 see
this on their sip openstage phones. how they do this?
thanks
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New
]: Leaving directory `/root/rpmbuild/BUILD/asterisk-1.8.5.0/main'
make: *** [main] Error 2
error: Bad exit status from /var/tmp/rpm-tmp.FeglRm (%build)
--
---
Marek Cervenka
hi,
is there some way to balance accross sip trunks by the number of calls?
example
3 trunks
alfa(max 30 channels, priority 1),delta(60,priority 2),omega(90,priority
3)
alfa have 25 calls now
i want next call terminate to delta. how to find in asterisk the current
calls number on sip trunk
the information is available here:
http://fail2ban.aleph-com.net/fail2ban_sharing If you're interested in
the server code just drop me an email.
i'm interested in the server code. thanks
--
---
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(unassigned) number
X-Asterisk-HangupCauseCode: 1
how can i resend HangupCauseCode from AsteriskB to SOMEPBX?
i'm tried this on AsteriskB
exten = _X.,1,Dial(SIP/AsteriskA,${EXTEN})
exten = _X.,n,Hangup(${SIP_HEADER(X-Asterisk-HangupCauseCode)})
thanks
--
---
Marek
---
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instead of my public IP address on the
firewall.
try asterisk 1.6.2.9
---
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On 06/22/2010 04:38 PM, marek cervenka wrote:
On 06/22/2010 02:51 PM, Johann Steinwendtner wrote:
On 2010-06-22 12:36, Remco Bressers wrote:
Dear list,
I've got the following setup :
[FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]
On the PBX's we run Asterisk 1.4.33
darilion and daniel ferenci(asterisk t.38
developers) and i can arrange fixing bugs
my jabber is cerv...@njs.netlab.cz
look forward for better t.38 days
---
Marek Cervenka
jabber - cerv...@njs.netlab.cz
://activa.sourceforge.net/readme.html
many thanks to Activa Team
---
Marek Cervenka
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asterisk-users mailing
problem:
it is only for caller. i dont know how to log call leg B
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenka
Sent: Monday, November 16, 2009 8:50 AM
To: asterisk-users@lists.digium.com
hi,
i want add info about remote party ip address to the asterisk cdr table
can you recommend me the system way?
thanks
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astDB SIP/Registry
- set some variable
really doesnt exist some cleaner way?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenka
Sent: Monday, November 16, 2009 8:50 AM
To: asterisk-users
testers needed
-- Forwarded message --
Date: Wed, 11 Nov 2009 17:48:04 -0600
Subject: [Asterisk 0013405]: [patch] T38 gateway
A NOTE has been added to this issue.
==
https://issues.asterisk.org/view.php?id=13405
card with Tylersburg(intel 5520/5500)
chipset?
thanks
p.s. sorry for offtopic :(
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Marek Cervenka
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,linux,mac)
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Marek Cervenka
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(with ffmpeg), supported free audio codecs
(http://en.wikipedia.org/wiki/Flash_Video#Format_details)
* uncompressed PCM
* ADPCM
* AAC
can you someone recommend solution/combination which works?
tnx
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Marek Cervenka
can anybody help me to give Opensource SIP client information which can be
modified as per our requirment
http://www.qutecom.org
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Marek Cervenka
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...@njs.netlab.cz
thanks!
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Marek Cervenka
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