[Asterisk-Users] mangle + to 00

hi, i want change prefix from +XXX. to 00XXX. but this doesnt work [incoming] exten = _+.,1,SetCIDnum(00${CALLERID:1}) exten = _+.,2,goto(incoming,${EXTEN},1) exten = _X.,1,Noop(CALLERID: ${CALLERID}) exten = _X.,2,goto(route,${EXTEN},1) can you help? --- Marek

Re: [Asterisk-Users] chan_cornet

HI4K to asterisk? any example h323 conf for asterisk? --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz

Re: [Asterisk-Users] :: Success Case = Motorola 62802-51 as FXO device ::

, audio experience (echo, delay, ...) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz

[Asterisk-Users] 1x fxs + 1x fxo transfer

)? --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list

[Asterisk-Users] zaptel vanilla kernel

hi, to digium maybe some individuals: do you plan add zaptel drivers to vanilla kernel? for users is this very good thing --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA

Re: [Asterisk-Users] Asterisk 1.0.4 and more ... + rpm spec

rpmbuild -ba asterisk.spec if this file will be contained directly in the tarball (like openvpn or other good software), then simply run rpmbuild -ta asterisk-1.0.5.tar.gz --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF

Re: [Asterisk-Users] Asterisk 1.0.4 and more ... + rpm spec

rpmbuild -ba asterisk.spec if this file will be contained directly in the tarball (like openvpn or other good software), then simply run rpmbuild -ta asterisk-1.0.5.tar.gz sorry, file is in attachment now --- Marek Cervenka Centrum Vypocetni Techniky CVT

[Asterisk-Users] ogg vorbis

hi, what are the reasons why ogg player is not included in asterisk?(for onhold music) technical, political, no coders? --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA

Re: [Asterisk-Users] SIP jitter?

with it to your heart's content--like the rest of us? It would work just like it had really been put into CVS-HEAD. less testers less bug reports for production use is stable version (asterisk doesnt have good roadmap and versioning :( ) --- Marek Cervenka Centrum

Re: [Asterisk-Users] click to dial extension number functionality ?

.= fread($socket, 8192); } fclose($socket); echo pre; echo ASTERISKMANAGEREND $wrets ASTERISKMANAGEREND; echo /pre; } ? --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA

Re: [Asterisk-users] Asterisk 1.0.6

about Asterisk. there is unofficial fast mirror in europe (md5 will be useful) ftp://ftp.ipex.cz/pub/mirrors/ftp.asterisk.org/pub/telephony/asterisk/ --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http

[Asterisk-Users] re: PA1688 Chipset IP Phones ATA's

192.168.1.100: PHONE 192.168.1.100: V1.37.008 192.168.1.100: Updating ... Please Wait 192.168.1.100: upgrade binary mismatch any help? --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA

Re: [Asterisk-Users] re: PA1688 Chipset IP Phones ATA's

binary mismatch any help? i'm found the problem in the settings debug - no check --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz

[asterisk-users] asterisk1.2 to 1.4 g711a fax

hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 -

Re: [asterisk-users] asterisk1.2 to 1.4 g711a fax

(like a2billing etc) then fax FAIL any ideas? Marek Cervenka ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

Re: [asterisk-users] asterisk1.2 to 1.4 g711a fax

(); $dialstr = SIP/asterisk1/1|300|HgL(61:61000); $myres = $agi-exec(DIAL $dialstr); $agi-hangup(); ? thanks! Marek Cervenka ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list

Re: [asterisk-users] Faxing through a PAP2

too --- Marek Cervenka === ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Redundancy / Failover

is master? (heartbeat) - now i have custom script for this question2: it's possible read registration data from astdb from python/php (or it is possible write sip registrations to mysql/sqlite? i do not want realtime because of NAT issues) --- Marek Cervenka

[Asterisk-Users] fax pass-through

result is '(null)' any ideas? --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit

Re: [Asterisk-Users] fax pass-through

DEBUG[28047] pbx.c: Function result is '1139871035.6' Feb 13 23:52:00 DEBUG[28047] pbx.c: Function result is '(null)' any ideas? --- Marek Cervenka === ___ --Bandwidth and Colocation

Re: [asterisk-users] chan_ss7 0.10

developed by www.dicea.dk. http://www.dicea.dk/company/downloads it's used on production servers. it is very stable solution --- Marek Cervenka === ___ -- Bandwidth and Colocation

[asterisk-users] incoming call popup

hi, can you recommend cleansimplestable solution for incoming call popup (in browser)? i'm using flash operator panel now but i want something without flash (maybe something in AJAX?) thanks --- Marek Cervenka

[asterisk-users] (announce) asterisk T.38 gateway

--- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users

Re: [asterisk-users] (announce) asterisk T.38 gateway

marek cervenka wrote: hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs

[asterisk-users] asterisk stops sending qualify

--- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users

Re: [asterisk-users] G722 capable soft phone?

Does anyone know where I might purchase a G.722 capable SIP soft phone? Counterpath say that they offer one, but only in the OEM versions do they support G.722. I need only a couple of licenses. www.qutecom.org --- Marek Cervenka

[asterisk-users] Re: [asterisk-dev] SRTP implementation

= 610,n,hangup p.s. sorry for cross post --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz

[asterisk-users] 1.2.17 - 1.2.18 asterisk crash

hi, i am updated to latest asterisk stable (because of security problems), but now asterisk crashes within a hour log is clear do you someone have this problem too? --- Marek Cervenka

Re: [asterisk-users] Re: Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card

interrupt line scenarios. --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] SRTP testers needed

please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compilerun clients with srtp (linksys,gxp-2000, minisip, twikle, ...) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http

Re: [asterisk-users] SRTP testers needed

On Fri, 2007-03-23 at 16:12 +0100, marek cervenka wrote: please look at http://www.voip-info.org/wiki/view/Asterisk+SRTP and try compilerun clients with srtp (linksys,gxp-2000, minisip, twikle, ...) Does this work on 1.2 or 1.4 too or is it trunk only? trunk only ... now no testers

Re: [asterisk-users] Any free video (or audio) softphone VOIP client under Linux with touchscreen friendly interface ?

there are some skins for existing clients that are more touchscreen friendly ? http://www.qutecom.org it is successor to openwengo --- Marek Cervenka === ___ -- Bandwidth

[asterisk-users] SRTP testers needed

...@njs.netlab.cz thanks! --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-users] Open source SIP client

can anybody help me to give Opensource SIP client information which can be modified as per our requirment http://www.qutecom.org --- Marek Cervenka === ___ -- Bandwidth

[asterisk-users] playing media(moh,prompts) from flash player

(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 --- Marek Cervenka

Re: [asterisk-users] Ekiga, Twinkle and from where to start with open source

,linux,mac) --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

[asterisk-users] looking for GXV-3000 users

hi, i want try Grandstream GXV-3000 video part. i'm looking for GXV users. i have asterisk-trunk available. please contact me privately (or at jabber:[EMAIL PROTECTED]) --- Marek Cervenka

asterisk t.38 (was RE: [asterisk-users] trixbox t38 pass through)

T38 passthrough doesn't seem to work in trunk at the moment. that's true http://bugs.digium.com/view.php?id=7679 http://bugs.digium.com/view.php?id=7844 t.38 in asterisk 1.4 http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 --- Marek Cervenka

[asterisk-users] digium compatibility notes

hi, what is mean by partially incompatible in http://www.digium.com/en/docs/misc/compatibility_notes.php i have server with E7221+te110p mobo and i think i dont have any problems thanks --- Marek Cervenka

[Asterisk-Users] OT: pocket pc + ilbc/g729

hi, can you recommend some pocket pc sip client with iLBC or G729? i'm googling over a hour and found nothing --- Marek Cervenka === ___ Asterisk-Users mailing list Asterisk

[Asterisk-Users] codecs order

--- Marek Cervenka === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] ftp.digium.com HTTP mirror, Digium's FTP server

The ftp server has been broken for months. If you keep trying you will eventually get a listing or a file. i'm using ftp://ftp.ipex.cz/pub/mirrors/ftp.asterisk.org/ --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU

Re: [Asterisk-Users] Grandstream GXP-2000 headset

--- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users

[Asterisk-Users] gxp-2000 tftp cfg

hi, can you someone post tftp template for gxp-2000? like http://www.grandstream.com/DOWNLOAD/Configuration_Tool/Windows/Grandstream_Configuration_File_Template_1.0.6.x.txt thanks --- Marek Cervenka

[Asterisk-Users] sip nat bug

--- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth

[Asterisk-Users] grandstream GXV-3000

hi, do you someone test this http://www.grandstream.com/y-gxv3000.htm? video works? (it's have H264 video codec) i want this topology gxv-3000 - nat -{Internet}- Asterisk -{Internet}- nat - gxv-3000 --- Marek Cervenka LCNA- http://lcna.slu.cz

[Asterisk-Users] Astricon - materials

hi, will be somewhere materials (videos, presentations) from astricon? thanks --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz

Re: [Asterisk-Users] Astricon - materials

marek cervenka wrote: hi, will be somewhere materials (videos, presentations) from astricon? Registered attendees will get information about the material soon. No videos where recorded this year. any chance for not registered? astricon was too far for me (europe) my english is terrible

[Asterisk-Users] sqlite + stable asterisk

[1]: Leaving directory `/usr/src/asterisk/cdr' make: *** [subdirs] Error 1 --- Marek Cervenka === ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing

[Asterisk-Users] link quality monitor

;) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation sponsored by Easynews.com

[asterisk-users] show channels

? thanks --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

[asterisk-users] sip realtime

you point me to some examples? thanks --- Marek Cervenka === ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] t.38 asterisk-trunk

and in this bug are questions without answers http://bugs.digium.com/view.php?id=5090 like can i have an ATA behind NAT? what mean chan_sip.c:4586 process_sdp: Unsupported SDP media type in offer: image 10912 udptl t38? --- Marek Cervenka FPF SLU OPAVA - http

[asterisk-users] t.38 bounty

hi, bounty for t.38 is $11,750. that looks good! http://www.voip-info.org/wiki/view/Asterisk+T.38+Bounty how high must be bounty for Digium to hire programmer for this? thanks --- Marek Cervenka

[asterisk-users] chan_ss7 0.10

/remote hangup info in NOTICE (cervajs at freevoice.cz) please test and report thanks --- Marek Cervenka === ___ --Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] chan_ss7 0.10.1

/chan_ss7-0.10.1.tar.gz md5sum a3ca3031f8f4ef96d505be3b297b47cc --- Marek Cervenka === ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing

[asterisk-users] asterisk chan_sip tuning

reg. timeout: 20 secs Outbound reg. attempts: 0 asterisk 1.4 thanks --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

[asterisk-users] asterisk optimalization

is about 100% which optimalizations do you recommend for ~1500 peers scenario? (behind nat, reregistrations) --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] asterisk optimalization

marek cervenka [EMAIL PROTECTED] writes: hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores

Re: [asterisk-users] asterisk optimalization

marek cervenka [EMAIL PROTECTED] writes: hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores

[asterisk-users] supermicro hardware + sangoma

card with Tylersburg(intel 5520/5500) chipset? thanks p.s. sorry for offtopic :( --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] [Asterisk 0013405]: [patch] T38 gateway (fwd)

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

[asterisk-users] asterisk cdr - remote ip address

hi, i want add info about remote party ip address to the asterisk cdr table can you recommend me the system way? thanks --- Marek Cervenka === ___ -- Bandwidth and Colocation

Re: [asterisk-users] asterisk cdr - remote ip address

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

Re: [asterisk-users] asterisk cdr - remote ip address - SOLVED

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

[asterisk-users] ANNOUNCE: New version of Activa TAPI driver

://activa.sourceforge.net/readme.html many thanks to Activa Team --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

[asterisk-users] Asterisk T.38 Gateway code testing

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

Re: [asterisk-users] Asterisk T.38 Gateway code testing

--- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] UDPTL T38 via NAT

instead of my public IP address on the firewall. try asterisk 1.6.2.9 --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] UDPTL T38 via NAT

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

[asterisk-users] resending cause codes

(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

Re: [asterisk-users] Sharing Fail2ban data

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 -- --- Marek Cervenka

[asterisk-users] sip trunk balancing

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

[asterisk-users] asterisk rpm build problem

]: 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

[asterisk-users] call pickup

this on their sip openstage phones. how they do this? thanks -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] call pickup

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

[asterisk-users] cdr documentation - new fields

-- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Calendar Integration Problem

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

[asterisk-users] axfer with simple CDR

ring time?) is it possible? if yes, can you post some example? -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] axfer with simple CDR

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

[asterisk-users] attended transfer with CEL

with D (consultation) time A with D time A with everyone (full time - from start to the end of call) -- --- Marek Cervenka === -- _ -- Bandwidth

Re: [asterisk-users] attended transfer with CEL

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

Re: [asterisk-users] attended transfer with 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

Re: [asterisk-users] Proactive problem monitoring on SIP on 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

[asterisk-users] AGI not generating sip 180/183 status

10.0.0.213 - 10.0.0.193 SIP Status: 200 OK -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

[asterisk-users] salesforce opencti

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

[asterisk-users] WebM / VP8 support

hello, any news about WebM/VP8 support in asterisk? some bounty where can i contribute? -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation

[asterisk-users] sip video endpoint with asterisk

it is possible? any recommendations? -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

[asterisk-users] groupcount fraud problem

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?

Re: [asterisk-users] groupcount fraud 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

[asterisk-users] mixmonitor extension

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

Re: [asterisk-users] mixmonitor extension

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

Re: [asterisk-users] mixmonitor extension

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

Re: [asterisk-users] mixmonitor extension

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

[asterisk-users] asterisk SugarCrm integration

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

Re: [asterisk-users] asterisk SugarCrm integration

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

[asterisk-users] asterisk multiple ip

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

Re: [asterisk-users] asterisk multiple ip

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

[asterisk-users] opus 11.12.0

lines). 1 out of 4 hunks FAILED -- saving rejects to file res/res_rtp_asterisk.c.rej thanks -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Tutorial: compiling and installing Asterisk 13

://bugzilla.redhat.com/show_bug.cgi?id=1140324 -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

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