[asterisk-users] announcement: JS library for asterisk pattern matching and number manipulation

0999888-77', myPatterns) // last pattern (_+X) from variable myPatterns is valid, // (9)[+420]999888[-77](123) => result 0999888123 parseNumber('+420999888-77', myPatterns) Marek Cervenka -- _ -- Bandwidth and Colocation

[asterisk-users] analysis of attended transfer in CEL

DEDTRANSFER.bridge1_id  =   BRIDGE_ENTER.bridge_id then i can find right HANGUP   by uniqueId what do you think about it? is there more simple method? Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digi

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

ch from *10.2.152.36 to **2.2.2.2* it looks like**its somewhere in the learning phase* * Dne 12/12/2019 v 11:51 Joshua C. Colp napsal(a): On Thu, Dec 12, 2019 at 6:39 AM marek <mailto:cervaj...@gmail.com>> wrote: hi, i have following topology PSTN - Asterisk internet

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is not helping ... going back to read res_rtp_asterisk.c & decrypting pcaps with wireshark Dne 12/12/2019 v 13:02 Joshua C. Colp napsal(a): On Thu, Dec 12, 2019 at 7:57 AM marek <mailto

[asterisk-users] asterisk pjsip webrtc rtp to private IP

? is it possible debug Asterisk STUN request/response ? or is it hidden in pjsip internals? Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

btw we are using tools like sipcapture.org,voipmonitor.org, callstats.io, elasticsearch+filebeat, ... but without informations whats happening inside asterisk is harder to solve problems Dne 12/12/2019 v 16:00 Joshua C. Colp napsal(a): On Thu, Dec 12, 2019 at 10:57 AM marek <mailto:c

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

rovider we dont know yet what technology is the problem but "sometimes" respond ip of some core router ( ISP - isp core/edge router ip - customers router ip - customers private ip ) to stun request pjsproject debug config pjproject.conf [startup] log_level=4 type=startup btw some examp

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

https://issues.asterisk.org/jira/browse/ASTERISK-28656 Dne 12/12/2019 v 16:49 Joshua C. Colp napsal(a): On Thu, Dec 12, 2019 at 11:40 AM marek <mailto:cervaj...@gmail.com>> wrote: examples of "interesting" information like ICE result and howto make &quo

[asterisk-users] astricon videos

Hello, any plans for astricon videos? Thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk

Re: [asterisk-users] astricon videos

ohh, that's bad news :( what about presentations? Marek p.s. i saw your presentation. good one . thank you Dne 21/01/2020 v 14:53 Joshua C. Colp napsal(a): On Tue, Jan 21, 2020 at 9:51 AM marek <mailto:cervaj...@gmail.com>> wrote: Hello, any plans for astricon videos?

Re: [asterisk-users] New RTP engine

patches welcome Philip Asterisk is Open Source so everyone can help Marek Dne 12/05/2020 v 07:02 Saint Michael napsal(a): Asterisk needs urgently to push the RTP engine to the Kernel, away from userland, like professional and commercial softwares do. I measured the cost

[asterisk-users] do not start MoH when caller pres HOLD on mobile

hi, its possibe to "dont start" music on hold when caller (from sip operator trunk) press HOLD (i.e. on mobile phone) Asterisk acts on SDP a=sendonly i want pass trough media from SIP trunk provi

[asterisk-users] chan_sip -> pjsip - address binding

sorcery_config_internal_load: Could not create an object of type 'transport' with id 'udp' from configuration file is there some other way? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

Re: [asterisk-users] chan_sip -> pjsip - address binding (SOLVED)

Dne 25/11/2020 v 16:29 Joshua C. Colp napsal(a): On Wed, Nov 25, 2020 at 11:24 AM marek <mailto:cervaj...@gmail.com>> wrote: tried chan_sip/udp + pjsip/tcp but the same problem sip.conf [general] udpbindport=5060 udpbindaddr=0.0.0.0 tcpenable=no

Re: [asterisk-users] chan_sip -> pjsip - address binding

'pjsip.conf' anybody know if its possible preload pjsip before chan_sip? (tried preload=  but its complicated. a lot of dependecies) its Asterisk 13 current Marek Dne 25/11/2020 v 14:55 marek napsal(a): hi, i want gradually migrate to pjsip i have 2 network interfaces with a lot of endpoints

[asterisk-users] 180 Ringing missing

n if the called party     isn't actually ringing. Allow interruption of the ringback if early media     is received on the channel. it changed to Asterisk 13 (Dial(${ARG1},300,R) -> INVITE <- 100 Trying <- 180 Ringing <- 183 Session Progress ( <- RTP -

Re: [asterisk-users] [External] 180 Ringing missing

So this is actually correct. One should not rely on any of these 1xx "Provisional" messages. They may or may not be sent, without violating SIP standards. Am 01.12.20, 12:20 schrieb "asterisk-users im Auftrag von marek" : hi, after upgrade from Asterisk 11 to Aste

Re: [asterisk-users] 180 Ringing missing

Dne 01/12/2020 v 12:58 Joshua C. Colp napsal(a): On Tue, Dec 1, 2020 at 7:20 AM marek <mailto:cervaj...@gmail.com>> wrote: hi, after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know, its old. customer is very conservative...) i have problem with mi

Re: [asterisk-users] [External] 180 Ringing missing

e is some recommendation - https://www.sipforum.org/technology/sipconnect/ Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.

[asterisk-users] monitoring pjsip status

Hi, can you recommend way to test status of PJSIP endpoint (SIP trunk to the operator)? is there something better than parsing asterisk -rx "pjsip show contact operator/sip:operator@1.1.1.1:5060" ? We are using icinga2/promethe

Re: [asterisk-users] Disable Session Progress in PJSIP

it sounds similar to my problem http://lists.digium.com/pipermail/asterisk-users/2020-December/295463.html i'm using workaround for now - Dial with R param   same => n,Dial(PJSIP/alice,,R) R: Default: Indicate ringing to the calling party, even if the called party     isn't actually ringing.

[asterisk-users] asterisk 11 -> 16 ForkCDR

  159 | 800800800 111222333 | 666777888 | 159 | 800800800 Tried "e" and "v" ForkCDR options.  no change any ideas why CDR records changed after update and what to change? Thanks Marek -- _ -- Band

[asterisk-users] BUILD_NATIVE vs Xeon E5 v2 vs Xeon E5 v3

RD_LOOPBACK, sa_data=00:00:00:00:00:00}}) = 0 ioctl(3, SIOCGIFHWADDR, {ifr_name="eth0", ifr_hwaddr={sa_family=ARPHRD_ETHER, sa_data=filtered}}) = 0 futex(0x910ba0, FUTEX_WAKE_PRIVATE, 2147483647) = 0 futex(0x910900, FUTEX_WAKE_PRIVATE, 2147483647) = 0 ---

Re: [asterisk-users] BUILD_NATIVE vs Xeon E5 v2 vs Xeon E5 v3

Uh i forgot. the build machine is a VMware VM and asterisk build is inside a docker container (centos7) as RPM Dne 20/05/2021 v 11:57 marek napsal(a): hi, i builded Asterisk 16.18 with BUILD_NATIVE on Xeon E5 v3 (Intel(R) Xeon(R) CPU E5-2670 v3 @ 2.30GHz) on machine when i want run

Re: [asterisk-users] BUILD_NATIVE vs Xeon E5 v2 vs Xeon E5 v3

/943755/gcc-optimization-flags-for-xeon/25095818 Dne 20/05/2021 v 12:31 marek napsal(a): Uh i forgot. the build machine is a VMware VM and asterisk build is inside a docker container (centos7) as RPM Dne 20/05/2021 v 11:57 marek napsal(a): hi, i builded Asterisk 16.18 with BUILD_NATIVE on Xeo

[asterisk-users] mobile phone as queue agent - best practice

hi, my customer wants call center with mobile phone agents only. What is your experience with this setup? Whats your best practice? - custom device state? - sw sip phone + redirect to mobile number in docker ? - ... tnx Marek

[asterisk-users] dialplan goto - bad priority

block #2) exten => _X.,n,Set(GWNAME=out) my problem is when call arrive with +421 so i want strip this example prefix from callerid then i expecting that call jump(goto) to the line exten => _X.,1,noop(main block) but it jumps to exten => _X.,n,Set(GWNAME=out) any idea

[asterisk-users] check if call is from chan_sip or chan_pjsip

? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https

Re: [asterisk-users] check if call is from chan_sip or chan_pjsip - SOLVED

${CHANNEL(channeltype)} Dne 17/09/2021 v 14:41 marek napsal(a): hi, i need check sip headers of incoming calls i have hybrid configuration with chan_sip and chan_pjsip enabled so i need check if incoming call is through chan_sip or chan_pjsip because i cant use i.e. ${PJSIP_HEADER(read

[asterisk-users] Opus Codec - real world adoption

) - Any experience with "better" audio on slow/problematic internet connection? - Any experience with call recordings on endpoint side(pbx user) and speech to text transcription? (compared to calls in alaw/ulaw) Thank you Have a nice weekend Mare

[asterisk-users] asterisk playback ogg files

be a file which can be played natively on Android, and in browsers (at least Chrome and Firefox). The quality of the encoding is considerably higher than MP3 while using approximately the same bitrate. Marek -- _ --

Re: [asterisk-users] asterisk playback ogg files

modified exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output.ogg) changed to exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output-ogg) file [root@dev cz]# file /var/lib/asterisk/sounds/output-ogg.ogg /var/lib/asterisk/sounds/output-ogg.ogg: Ogg data [Dec 22 14:39:26]

Re: [asterisk-users] asterisk playback ogg files (SOLVED)

asterisk doesn't support .ogg file format (digged through apps/app_playback.c, main/file.c) dev*CLI> core show file formats Format Name   Extensions --    -- slin   mp3    mp3 slin48 ogg_opus   opus so with /var/lib/asterisk/sounds/output-ogg.opus

[asterisk-users] asterisk prometheus grafana dashboard

hi, i'm testing https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_res_prometheus anybody who can share grafana dashboard json  ;) ? thanks Marek -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] Astricon 2021 Videos

Hi, will be available videos from Astricon 2021? thank you Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New

[asterisk-users] problem with asterisk multiple queues and call assigning

queue2 is assigned to agent. i need assign the longer waiting call from queue1 Asterisk 16 any hints/tips? Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

[asterisk-users] app TRANSFER PJSIP URI format

asterisk1 to asterisk2 is   SIP/2.0 404 Not found so what's the correct URI for TRANSFER? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new A

Re: [asterisk-users] app TRANSFER PJSIP URI format (SOLVED)

mistake in __TRANSFER_CONTEXT Dne 07/02/2022 v 11:17 marek napsal(a): hi, i have two asterisk boxes need transfer call from second box to first one pstn -> asterisk1 -> dial(number 555) -> asterisk2 -> TRANSFER (number 444) -> asterisk1 dialplan on asterisk1

[asterisk-users] parsing P-Asserted-Identity if Privacy: id

hi, what's best method from perfomance view to parse this header P-Asserted-Identity: P-Asserted-Identity: i need +44111222333 (if Privacy: id) PJSIP_PARSE_URI ? STRREPLACE/CUT? FastAGI? other options? thanks Marek

Re: [asterisk-users] parsing P-Asserted-Identity if Privacy: id

isk/asterisk/blob/97b3459bd249eb3ad4c5fc96732fadf447cd8e6d/res/res_pjsip_caller_id.c#L176 which pill will you take neo? Dne 12/09/2022 v 22:39 marek napsal(a): hi, what's best method from perfomance view to parse this header P-Asserted-Identity: P-Asserted-Identity: i need +44111222

[asterisk-users] pjsip endpoint reachable

hi, we are migrating from chan_sip to pjsip i want logs like this about pjsip endpoints [Oct 14 17:20:36] NOTICE[35629] chan_sip.c: Peer 'endpoint22' is now Reachable. (15ms / 2000ms) is it possible? thanks Marek

[asterisk-users] is KTLS usable with asterisk+libsrtp?

with asterisk+libsrtp (for webrtc clients)? any idea about cpu usage impact?  smaller percentage units? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum

[asterisk-users] mailing list working?

there are new versions of Asterisk but mailing list is empty http://lists.digium.com/pipermail/asterisk-users/ Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

[asterisk-users] AEAP experience

and STT from IBM? https://cloud.ibm.com/docs/speech-to-text?topic=speech-to-text-models-ng#models-ng-telephony Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community

[asterisk-users] aeap wss connection

request 1/1] what do you think? is it bug? Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk

[asterisk-users] SIP_HEADER GET_TRANSFERRER_DATA chan_pjsip

? or is there other way? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here

[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

- sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is changed in 1.4 in g711 pass-through faxing? thanks marek ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing

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] Channel banks for E1

(O), but I have Digium E1 card and I want use for one line 30 channels and not only 24. Thank you very much. Sincerely Jan Marek -- Ing. Jan Marek | Nez mi poslete prilohu .doc, .xls University of South Bohemia | nebo .ppt, prectete si, prosim, Academic Computer Centre | WWW

[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] Digium TE405 card and Matra PBX

timesource on the computer... Is there any other possibility? Thanks for your advices. Sincerely Jan Marek -- Ing. Jan Marek | Nez mi poslete prilohu .doc, .xls University of South Bohemia | nebo .ppt, prectete si, prosim, Academic Computer Centre | WWW stranku uvedenou na poslednim

[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

[Asterisk-Users] Problems setting up X100 FXO card

... ) please help me sort it out or point out what i'm doing wrong :( thanks, Marek - here is output from kernel/dmesg: Aug 24 16:11:13 arnor kernel: Zapata Telephony Interface Registered on major 196 Aug 24 16:13:32 arnor kernel: ACPI: PCI Interrupt

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] Hangup problem

() [pbx_config] 't' =1. Hangup() [pbx_config] please tell me what i'm doing wrong :) thanks in advance, Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] Huge Echo

piece of the call could be causing the trouble so i can look into it? thanks, Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk

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