[Asterisk-Users] 1.0.5 and h323 compiling problem

Hello! Looks like h323 compiling is FAQ, but I didn't found an answer... The same problem with 0.6.5 and 0.7.1: gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/var/local/files/asterisk-1.0.5/include -I../wrapper -g -c -o

[Asterisk-Users] meetme2 compilation

Hello! Do somebody knows how to compile meetme2 with 1.0.6. I readed wiki, applied patches, but no luck ;-( Me be someone can give me working meetme2.c ? :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] meetme2 compilation

: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:302: error: for each function it appears in.) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dmitry Melekhov Sent: Wednesday, March 16, 2005 4:36 AM To: asterisk-users

Re: Re: [Asterisk-Users] Meetme2 compilation problem

Looks like there is no anoynomous access for this dir :-( Could you mail me this file? Squid sent the following FTP command: PASS yourpasswordand then received this reply Login incorrect. - Original Message - From: Asterisk To: Anil Kumar K ; Giovanni Powell Cc:

[asterisk-users] wcte12xp0: Missed interrupt. when disable echocanceller

Hello! I run asterisk 1.6.1.0, dahdi 2.1.0.4 with TE122. I always (and only) have missed interrupt when dahdi disables echo canceller (ng2 or oslec- no difference). Dec 29 14:00:54 asterisk kernel: dahdi: Disabled echo canceller because of tone (rx) on channel 1 Dec 29 14:00:54 asterisk

[asterisk-users] rtcp to cdr for calls from dahdi to sip

Hello! I want to get rtcp stats to cdr. (btw, I run asterisk 1.6.2.11) There is howto here http://www.voip-info.org/wiki/view/Asterisk+RTCP But I (and my users) do bridged calls from dahdi to sip, so in h extension channel is dahdi , and it doesn't contain rtcp stats. There is info about

[asterisk-users] func SHARED, how to use?

Hello! Could somebody tell me how to use SHARED function? I want to get RTCP stats from SIP , but current channel is DAHDI. How can I get SIP channel? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] func SHARED, how to use?

21.09.2010 18:57, Philipp von Klitzing пишет: Hi! Could somebody tell me how to use SHARED function? http://www.voip-info.org/wiki/index.php?page=Asterisk+func+shared There are no examples there :-( I want to get RTCP stats from SIP, but current channel is DAHDI. How can I

Re: [asterisk-users] func SHARED, how to use?

22.09.2010 14:50, Philipp von Klitzing пишет: Hi! I see. I want to use SHARED function! Do you have example how to to export them to the local call leg/channel ? Have you considered using Google (or your favourite search engine)? Shure, I searched and find nothing. The

Re: [asterisk-users] func SHARED, how to use?

22.09.2010 15:12, Andrea Cristofanini пишет: Could you, please, give me link ? :-) Google is not difficult to use... BTW http://www.voip-info.org/wiki/view/Asterisk+func+shared There is no example here! I already wrote about this... --

Re: [asterisk-users] func SHARED, how to use?

22.09.2010 16:08, Philipp von Klitzing пишет: Hi Dmitry! Hello! And the third hit in my google result is this: http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html Since I mentioned in my previous message that you will find the answer in the archive of this list you

Re: [asterisk-users] func SHARED, how to use?

23.09.2010 16:06, Philipp von Klitzing пишет: Hi! And the third hit in my google result is this: http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html Since I mentioned in my previous message that you will find the answer in the archive of this list you could have

Re: [asterisk-users] func SHARED, how to use?

23.09.2010 17:27, Philipp von Klitzing пишет: Hi! There are 2 things I can't understand - 1. how can I know channel name? ${CHANNEL} Thank you! Really, I get SIP channel name in macro (I thought I'll still have DAHDI there...) 2. where should I call this SHARED function?

Re: [asterisk-users] func SHARED, how to use?

27.09.2010 09:26, Dmitry Melekhov пишет: Since I have not done this with 1.6 or 1.8: See if you can get the RTCP data without using CHANNEL(), and instead use the individual xxxBRDIGED RTCP channel variables as illustrated on the Wiki. Your SIP channel is the 2nd channel (= the bridged one

Re: [asterisk-users] func SHARED, how to use?

27.09.2010 16:25, Philipp von Klitzing пишет: Hi! Well, only problem I see, is to how pass channel name from macro to h extension... SHARED() or CDR(userfield) Philipp Looks like I still don't understand how SHARED works :-( Let's say, I dial my softphone:

Re: [asterisk-users] func SHARED, how to use?

28.09.2010 15:35, Philipp von Klitzing ?: Hi! Looks like I still don't understand how SHARED works :-( exten=6052,n,Dial(SIP/6052,,M(test)) exten=6052,n,Dial(SIP/6052,,M(test^${CHANNEL})) Hello! Thank you! I can pass this constant , but I need RTCP stats And this

Re: [asterisk-users] func SHARED, how to use?

28.09.2010 16:19, Dmitry Melekhov ?: btw, about bridged variables- they are really what I need. Looks like there is bug in asterisk- if call is dropped from dahdi side- there is no info in these variables. I think I have to fill bug. Thank you! I got what I want :-) Thank you again

[asterisk-users] outbound fax over t38 gateway can't pass

Hello! I have problems with outbound faxes with asterisk 10.2 t38 gateway. There is asterisk box, connected to panasonic kx-td500 over PRI link with TE122. If we try to send fax with following path: panasonic 500 extension fax machine panasonic500- asterisk- ooh323- cisco 3845- fax machine

Re: [asterisk-users] outbound fax over t38 gateway can't pass

btw, played with res_fax.conf if I set maxrate=7200 fax machines try (and fail) 9600 anyway. Why? If limited ti 7200? looks like bug... So I set maxrate=4800 and modems=v27. Faxes pass Looks like problems with V29... 29.02.2012 07:56, Dmitry Melekhov пишет: Hello! I have problems

Re: [asterisk-users] Video conferencing?

25.07.2012 22:24, Ken D'Ambrosio пишет: Hi, all. I'm 99% sure that Asterisk technically *supports* videoconferencing well, confbridge supports sort of videoconferences , but our users refused to use them because asterisk switches video in the middle of stream and this leads to broken

Re: [asterisk-users] Video conferencing?

27.07.2012 19:25, Matthew Jordan пишет: Hi Dmitry! Hello! So, our original conversation is here: http://lists.digium.com/pipermail/asterisk-video/2012-April/003621.html As I said in our previous conversation, we don't currently have plans to implement a re-transmission of a new source's

Re: [asterisk-users] Video conferencing?

30.07.2012 22:52, Matthew Jordan пишет: - Original Message - I don't normally say something this blunt, but your point of view is wrong. May be, I even will agree when something will be at least planned ;-) Refusal of Feature: Person A: Do you have plans to implement feature Y?

[asterisk-users] motif and psi - no sound

Hello! I'm trying to use psi+ to conect to asterisk using chan_motif and vise versa. Connection looks good, but no sound. As I see there is some traffic (22.229 is my desktop with psi) 08:38:37.463506 IP 192.168.22.229.8010 192.168.22.19.17012: UDP, length 82 08:38:37.481325 IP

Re: [asterisk-users] motif and psi - no sound

19.10.2012 08:40, Dmitry Melekhov пишет: Hello! I'm trying to use psi+ to conect to asterisk using chan_motif and vise versa. Connection looks good, but no sound. As I see there is some traffic (22.229 is my desktop with psi) 08:38:37.463506 IP 192.168.22.229.8010 192.168.22.19.17012: UDP

[asterisk-users] confbridge and talker

Hello! We use meetme, but, as I understand it will be soon removed from asterisk (already marked as deprecated), so I'm thinking about confbridge migration. Really, we use self-developed (really my ;-) ) web interface to control meetme. We use cli ( over manager ) command to get users list

Re: [asterisk-users] confbridge and talker

11.02.2013 18:19, Matthew Jordan пишет: On 02/11/2013 01:20 AM, Dmitry Melekhov wrote: Hello! We use meetme, but, as I understand it will be soon removed from asterisk (already marked as deprecated), so I'm thinking about confbridge migration. Really, we use self-developed (really my

Re: [asterisk-users] Echo Cancellation

25.07.2013 13:51, bilal ghayyad пишет: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? oslec, imho. -- _ --

[asterisk-users] ooh323 and tcp timeout?

Hello! I run asterisk 11.5.0 and I connect to some peer over ooh323. But there is alternate path to the same peer over sip, but this not we can usually use. So, if peer is not available over h323 I'd like to dial over sip. I tried to test this: iptables -D INPUT -s 192.168.6.0/24 -j DROP

[asterisk-users] is it possible to compile chan_h323 with 11.5.0?

Hello! Tried to compile, but : [CC] chan_h323.c - chan_h323.o chan_h323.c: In function '__oh323_update_info': chan_h323.c:349: error: dereferencing pointer to incomplete type chan_h323.c:350: error: dereferencing pointer to incomplete type chan_h323.c: In function 'oh323_rtp_read':

Re: [asterisk-users] is it possible to compile chan_h323 with 11.5.0?

ok, changed to ast_channel_writeformat ast_channel_readformat at least, got it compiled :-D 22.08.2013 13:24, Dmitry Melekhov пишет: Hello! Tried to compile, but : [CC] chan_h323.c - chan_h323.o chan_h323.c: In function '__oh323_update_info': chan_h323.c:349: error: dereferencing

[asterisk-users] CONNECTEDLINE and ooh323, do it work?

Hello! Just read http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE tried on dahdi, it works, i.e. if I call asterisk user from my pbx connected phone I see what I set in Set(CONNECTEDLINE(name)= But if I call the same user over h323 ( no matter is it asterisk with ooh323 or

Re: [asterisk-users] CONNECTEDLINE and ooh323, do it work?

30.10.2013 13:25, Dmitry Melekhov пишет: Hello! Just read http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE tried on dahdi, it works, i.e. if I call asterisk user from my pbx connected phone I see what I set in Set(CONNECTEDLINE(name)= But if I call the same user over h323

[asterisk-users] TE420, is it possible do disable span (red blinking)?

Hello! Just got new server with TE420. Not all four spans will be used immediately, but spans not configured or not connected blink red light. Is it possible to turn span off, so my colleagues will not eventally tell me that something is wrong with asterisk? :-) Thank you! --

[asterisk-users] e1 , hdlc data link?

Hello! I want to use TE121 for E1 data link to cisco. Really only for tests now. So I wrote: span = 1,1,0,ccs,hdb3,crc4 #span=1,1,0,esf,b8zs nethdlc=1-31:hdlc0 may be first line is wrong, problem is somewhere else anyway. Just because I get: # dahdi_cfg DAHDI_CHANCONFIG failed on channel 1:

[asterisk-users] overlapdialing and no digits in setup problem

Hello! I have asterisk which is connected to avaya definity. I set trunk to overlap. When I call to this trunk (so called tac in avaya) without any number I hear dial tone for some time, any digit I try to dial are ignored by asterisk- still tone, then call is rejected: -- Accepting

[asterisk-users] CONNECTEDLINE and panasonic 500

Hello! I have following connections over isdn pri: avaya definity---pri--asterisk--pri-panasonic 500 Just because panasonic 500 can't send user's names. I also want to have reverse callerid for avaya users. But if there is no answer in dial plan: exten =

Re: [asterisk-users] CONNECTEDLINE and panasonic 500

18.11.2013 20:51, Richard Mudgett пишет: On Mon, Nov 18, 2013 at 1:21 AM, Dmitry Melekhov d...@belkam.com mailto:d...@belkam.com wrote: Hello! I have following connections over isdn pri: avaya definity---pri--asterisk--pri-panasonic 500 Just because panasonic 500 can't

Re: [asterisk-users] e1 , hdlc data link?

Hello! Could somebody at least tell me is such data link still supported? Thank you! 14.11.2013 09:15, Dmitry Melekhov пишет: Hello! I want to use TE121 for E1 data link to cisco. Really only for tests now. So I wrote: span = 1,1,0,ccs,hdb3,crc4 #span=1,1,0,esf,b8zs nethdlc=1-31:hdlc0

Re: [asterisk-users] e1 , hdlc data link?

19.11.2013 21:31, Shaun Ruffell пишет: On Tue, Nov 19, 2013 at 01:54:16PM +0400, Dmitry Melekhov wrote: Hello! Could somebody at least tell me is such data link still supported? Thank you! Yes, but you will need to edit include/dahdi/dahdi_config.h and ensure that CONFIG_DAHDI_NET is defined

Re: [asterisk-users] e1 , hdlc data link?

20.11.2013 07:40, Dmitry Melekhov пишет: 19.11.2013 21:31, Shaun Ruffell пишет: On Tue, Nov 19, 2013 at 01:54:16PM +0400, Dmitry Melekhov wrote: Hello! Could somebody at least tell me is such data link still supported? Thank you! Yes, but you will need to edit include/dahdi/dahdi_config.h

[asterisk-users] dahdi-dahdi native bridging and audio level

Hello! I use asterisk with TE420 as PRI switch for two channels : ;panasonic uplink group=3 context=panasuplink ;relaxdtmf=yes ;immediate=yes rxgain=0.0 txgain=0.0 mohsuggest=default jbenable = no ;jbenable = yes ;

[asterisk-users] 1TE133F and first pci-e slot

Hello! At the end of year 2013 I had problem on my supermicro system with TE133, namely it losted connect to another system. So I opened case and finally card was replaced by TE220. Now I need to buy another 1 port card for almost the same supermicro. Could somebody tell me is this problem

Re: [asterisk-users] 1TE133F and first pci-e slot

you! This is not the cause of my problem.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dmitry Melekhov Sent: Thursday, July 17, 2014 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion

[asterisk-users] originate , callerid

Hello! I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten =

Re: [asterisk-users] originate , callerid

25.12.2014 15:46, Anthony Messina пишет: On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11

[asterisk-users] how asterisk detects silence?

Hello! As I see there is dsp_drop_silence switch in confbridge. Could you tell me how asterisk detects silence? Is it possible to change silence level, so, let's say some not loud enough background noises will be recognized as silence and only loud enough human voice will be recognized as

Re: [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.

05.03.2015 11:42, Dmitry Melekhov пишет: 05.03.2015 11:29, Dmitry Melekhov пишет: Hello! Just installed asterisk 13.2.0 and see many such messages in log, I see them in console during calls, really something like this: -- Executing [6166@kanbaikal:2] Dial(OOH323/kanbaikal-6, SIP/6166

Re: [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.

10.03.2015 16:18, Matthew Jordan пишет: On Tue, Mar 10, 2015 at 5:00 AM, Dmitry Melekhov d...@belkam.com wrote: 05.03.2015 11:42, Dmitry Melekhov пишет: 05.03.2015 11:29, Dmitry Melekhov пишет: Hello! Just installed asterisk 13.2.0 and see many such messages in log, I see them in console

Re: [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.

10.03.2015 00:37, Paul Belanger пишет: On Thu, Mar 5, 2015 at 2:29 AM, Dmitry Melekhov d...@belkam.com wrote: Hello! Just installed asterisk 13.2.0 and see many such messages in log, I see them in console during calls, really something like this: -- Executing [6166@kanbaikal:2] Dial

Re: [asterisk-users] how asterisk detects silence?

19.03.2015 09:31, Dmitry Melekhov пишет: Hello! As I see there is dsp_drop_silence switch in confbridge. Could you tell me how asterisk detects silence? Is it possible to change silence level, so, let's say some not loud enough background noises will be recognized as silence and only loud

Re: [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.

05.03.2015 11:29, Dmitry Melekhov пишет: Hello! Just installed asterisk 13.2.0 and see many such messages in log, I see them in console during calls, really something like this: -- Executing [6166@kanbaikal:2] Dial(OOH323/kanbaikal-6, SIP/6166@asterisk) in new stack == Using SIP RTP

[asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.

Hello! Just installed asterisk 13.2.0 and see many such messages in log, I see them in console during calls, really something like this: -- Executing [6166@kanbaikal:2] Dial(OOH323/kanbaikal-6, SIP/6166@asterisk) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark

Re: [asterisk-users] chan_dahdi.c: Don't know what to do with frame type '10'

Frame type '10' is a CNG (Comfort Noise Generation) frame. This is a frame that, instead of carrying audio, carries a command to for the receiver to generate comfort noise for a length of time to the local user. chan_dahdi is not currently capable of generating comfort noise in response to

Re: [asterisk-users] can ooh323 work with cisco router?

06.05.2015 10:58, s m пишет: Hello! I'm not h323 expert, may be somebody else can understand from this log what is happening, but I can't :-( Could you, please, provide log with tracelevel=6 in ooh323.conf ? Thank you! hello Dmitry thank you for your reply. Ok, you are right. i want to

Re: [asterisk-users] can ooh323 work with cisco router?

06.05.2015 10:06, s m пишет: hello every body, i have big problem to configure h323 trunk between cisco router and asterisk 11.13.1 which uses ooh323 module. any body knows if ooh323 module can work with cisco routers or not (in gateway mode, it is ok and register in cisco gatekeeper but

Re: [asterisk-users] can ooh323 work with cisco router?

07.05.2015 08:17, s m пишет: hello thanks Dmitry for your useful hints. i enable debug and solve my problem:). it was codec compatibility problem. but it is so strange; if i set codec g711alaw in cisco router and asterisk, i have the mentioned problem but if i set codec to transparent in

Re: [asterisk-users] chan_ooh323 to sip , no connected line info

by the way, just tried IAX2 between asterisks- the same problem, no connected line name 14.05.2015 15:56, Dmitry Melekhov пишет: Hello! We have asterisk connected over PRI no our phone network, so I'm avaya PBX user. Asterisk connects to another avaya system over h323. Connection can

[asterisk-users] chan_ooh323 to sip , no connected line info

Hello! We have asterisk connected over PRI no our phone network, so I'm avaya PBX user. Asterisk connects to another avaya system over h323. Connection can be shown as avaya--PRI-asterisk--h323-avaya When I do call as avaya user I see name of remote end avay user, i.e. connected line info.

[asterisk-users] chan_dahdi.c: Don't know what to do with frame type '10'

Hello! I see large enough amount of such messages on one of our asterisks. There are no complains from users, so I may be they are harmless. Could you tell me what can it be? Thank you! -- _ -- Bandwidth and Colocation

[asterisk-users] asterisk 13 n-way call problem

Hello! I need to use n-way call as it described here: http://habrahabr.ru/sandbox/52259/ It is in russian, but dial plan is quite clear. It works in asterisk 11: -- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer) priority 1 -- Executing [0@fromtransfer:1]

Re: [asterisk-users] asterisk 13 n-way call problem

I spent some time reading docs and such change is not documented, so this is bug. I'll open issue... 22.12.2015 10:53, Dmitry Melekhov пишет: Hello! I need to use n-way call as it described here: http://habrahabr.ru/sandbox/52259/ It is in russian, but dial plan is quite clear. It works

[asterisk-users] pjsip module reload problem

Hi! Installing new asterisk server and decided to use chan_pjsip. While module reload I get: y 12 15:33:04] ERROR[21137]: config_options.c:715 aco_process_var: Could not find option suitable for category '3567' named 'inband_progress' at line 867 of [May 12 15:33:04] ERROR[21137]:

Re: [asterisk-users] pjsip module reload problem

12.05.2016 15:38, Joshua Colp пишет: Dmitry Melekhov wrote: Hi! Installing new asterisk server and decided to use chan_pjsip. While module reload I get: y 12 15:33:04] ERROR[21137]: config_options.c:715 aco_process_var: Could not find option suitable for category '3567' named

Re: [asterisk-users] pjsip module reload problem

12.05.2016 16:01, Dmitry Melekhov пишет: 12.05.2016 15:38, Joshua Colp пишет: Dmitry Melekhov wrote: Hi! Installing new asterisk server and decided to use chan_pjsip. While module reload I get: y 12 15:33:04] ERROR[21137]: config_options.c:715 aco_process_var: Could not find option

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

09.09.2016 15:18, Joshua Colp пишет: Dmitry Melekhov wrote: 09.09.2016 14:08, Dmitry Melekhov пишет: And, as I already said, there was no such messages while using asterisk 13.10. I'll open bug report. I was mistaken and it is indeed a bug. I've got a fix up but even without the fix

[asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"3563" ' failed for '192.168.32.116:5060' (callid: 0_1409534529@192.168.32.116) - No matching endpoint found

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

09.09.2016 13:45, Joshua Colp пишет: Dmitry Melekhov wrote: Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"3563" <sip:3563@192.168.32.254>' failed for '192.168.32.1

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

09.09.2016 14:08, Dmitry Melekhov пишет: 09.09.2016 13:45, Joshua Colp пишет: Dmitry Melekhov wrote: Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"3563" <sip:3563@1

Re: [asterisk-users] 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed

09.09.2016 14:13, Joshua Colp пишет: Dmitry Melekhov wrote: 09.09.2016 14:08, Dmitry Melekhov пишет: 09.09.2016 13:45, Joshua Colp пишет: Dmitry Melekhov wrote: Hello! Upgraded 13.10 to 13.11.1 today and now I see messages in log: [Sep 9 12:23:16] NOTICE[6064] res_pjsip

[asterisk-users] chan_ooh323 - cisco call manager express

Hello! I need to setup h323 trunk between cisco call manager express ( I have no access to it) and asterisk ( my side ). Calls from asterisk are OK, but there is no voice if calls are from cisco to asterisk. Looks like there is signalling problem. Could you , please look at

[asterisk-users] Callee id over chan_sip trunk

Hello! I run two asterisks 13.13.1. Here is how they are connected: me---PBX--isdn pri--asterisk1--sip--asterisk2. If I call something from asterisk1 and have in dial plan: Let's say exten => 6000,n,Set(CONNECTEDLINE(name)=Conf. 6000) exten => 6000,n,Meetme(6000,TL(1080:6))

[asterisk-users] cmd AGI(), maximum script time.

Hello! It there way to limit script execution time ? I did something wrong writing my script yesterday , finally got it working, but found that there are busy ISDN channels, looks like these are with hang scripts... Thank you! --

[asterisk-users] pjsip insecure=port,invite

Hello! Looks like faq, but... Could you , please, point me on how to convert this [cisco] type=friend host=192.168.22.253 insecure=port,invite to pjsip? as you can see another side is very old cisco router, so I can't change anything there. I don't see any examples here

Re: [asterisk-users] pjsip insecure=port,invite

Thank you very much! It works. Although there is one strange thing here- there is no ringback tone on cisco's side when pjsip is use, but this is another story :-) 02.11.2017 14:17, Joshua Colp пишет: On Thu, Nov 2, 2017, at 04:50 AM, Dmitry Melekhov wrote: Hello! Looks like faq

Re: [asterisk-users] G729

20.07.2018 23:35, John Kiniston пишет: On Fri, Jul 20, 2018 at 11:41 AM Saint Michael > wrote: ​The community would benefit if a non/licensed version of G729 would be included with Asterisk​, since the license expired. The current codec source

Re: [asterisk-users] asterisk 16 manager --END COMMAND--

12.10.2018 14:10, Joshua C. Colp пишет: On Fri, Oct 12, 2018, at 3:35 AM, Dmitry Melekhov wrote: Hello! Just upgraded asterisk from 13 to 16 and found that php-agi library is not compatible. It waits for --END COMMAND-- after command is completed, but, as I see from tcpdump, now asterisk

[asterisk-users] asterisk 16 manager --END COMMAND--

Hello! Just upgraded asterisk from 13 to 16 and found that php-agi library is not compatible. It waits for --END COMMAND-- after command is completed, but, as I see from tcpdump, now asterisk does not send such string after command is completed. Could you tell me, is it possible to get

[asterisk-users] which linux for asterisk?

Hello! I use Centos for asterisk for long time. And planned to install Centos 8 on new servers. But because Centos is declared dead, what is best choice ? Oracle? Ubuntu? Thank you! -- _ -- Bandwidth and Colocation

Re: [asterisk-users] which linux for asterisk?

09.12.2020 16:52, Jeff LaCoursiere пишет: This machine I visited yesterday in our data center... it is running Ubuntu 14... I would say this is a pretty stable platform :) Ubuntu 14... It is not supported for years now. This is not our method, we are replacing Centos 6 servers now...

Re: [asterisk-users] which linux for asterisk?

Well, I already got on result: As I suspected using Debian or Ubuntu leads to running unsupported OS version, because even LTS is too short.. Anybody knows which linux Sangoma developers are using now? 09.12.2020 11:03, Dmitry Melekhov пишет: Hello! I use Centos for asterisk for long

Re: [asterisk-users] which linux for asterisk?

09.12.2020 20:13, Joshua C. Colp пишет: From an open source project perspective we would  accept issues filed when the underlying Linux distribution is one of those, as they are from RHEL. What we don't support is for example Gentoo, Arch, Slackware, that kind of thing. Thank you very

Re: [asterisk-users] which linux for asterisk?

holes, this is not our method of operation  :-) --- Michel FACERIAS 11 bis, Chemin de BADASSAC 34510 FLORENSAC Cel : +33/0 638 42 91 93 http://www.facerias.org Le 2020-12-09 14:06, Dmitry Melekhov a écrit : 09.12.2020 16:52, Jeff LaCoursiere пишет: This machine I visited yesterday in our

Re: [asterisk-users] which linux for asterisk?

09.12.2020 18:52, Joshua C. Colp пишет: On Wed, Dec 9, 2020 at 10:47 AM Dmitry Melekhov <mailto:d...@belkam.com>> wrote: Well, I already got on result: As I suspected using Debian or Ubuntu leads to running unsupported OS version, because even LTS is too short..

Re: [asterisk-users] Asterisk and CentOS 8

10.12.2020 03:25, Patrick Wakano пишет: In case anyone out there is working with CentOS, you might reconsider that decision: https://blog.centos.org/2020/12/future-is-centos-stream/ Oracle? :-) --

Re: [asterisk-users] which linux for asterisk?

09.12.2020 16:03, Julian Beach пишет: Re: [asterisk-users] which linux for asterisk? On Wednesday, December 9, 2020, 11:07:55 AM, Antony Stone wrote: *> Upgrading a Debian server to the next release is a whole lot easier than doing > a CentOS one. *It will be slightly easier for me having

Re: [asterisk-users] which linux for asterisk?

09.12.2020 13:20, Frank Vanoni пишет: On Wed, 2020-12-09 at 11:03 +0400, Dmitry Melekhov wrote: what is best choice ? Oracle? Ubuntu? I'm running Asterisk since several years on Ubuntu without any issues. Debian should be fine too. Thank you. This gives me just about 3-4 years of support

Re: [asterisk-users] which linux for asterisk?

09.12.2020 15:07, Antony Stone пишет: This gives me just about 3-4 years of support, considering 2 years between LTS, and upgrading remote server can be pain. Upgrading a Debian server to the next release is a whole lot easier than doing a CentOS one. There was no need to upgrade