similar to: Trixbox 2.6.1.13 OpenR2

I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work. -----Original Message----- From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81

Hi all, Is it Openr2 supported by asterisk 1.6.2 without pach instalation ? I am a little bit confuse about that. My asterisk 1.6.2 show me the following warning: Unknown signalling method 'mfcr2' at line 29. I had downloaded and instaled openr2-1.3.0 but the messages is still shown. Which files I must to change in order to have everything working properly. Best regards! Att,

Dear All, I'm looking for someone who has implemented OpenR2 in Thailand successfully. Any settings, advice, caveats etc. are welcome. Best regards, Peter Lindqvist www.voxion.net

Hello to the R world... I have some problems regarding a GLM - repeated measures analysis. I want to test overall differences between AgeClass and Treatment (between subject) with OpenR1+OpenR2+OpenR3 (repeated measures, within subject). The table looks kind like this: AgeClass Treatment OpenR1 OpenR2 OpenR3 1 1 0 0 12.63 1 1 12.67 3.83 45.67 1 1 38.46 65.38 75.21 1 1 14.46 0 17.96 1 2 27.83

So, I've compiled and installed libpri-1.4.11.5, dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but chan_dahdi is not getting built. If I do a "make menuselect" in asterisk I see it listed with XXX, meaning that dependencies are not met. XXX chan_dahdi Depends on: res_smdi(M), dahdi(E), tonezone(E), pri(E), ss7(E), openr2(E) res_smdi gets built fine, dahdi is

Hi, I have installed Elastix 1.5.2 (Barranquilla, Colombia (TELCO: METROTEL)) with a TE220P (2xE1) and TDM2400P (16FXS), openr2 is included in 1.5.2 version. The outcoming calls are ok, but with incoming call i have an error: ERROR*[*14972*]* chan_dahdi.c: Chan 2 - Protocol error. Reason = Multi Frequency Cycle Timeout, R2 State = Seize ACK Transmitted, MF state = Category Request Transmitted,

Hi folks! I have an asterisk, version 1.4.22.2, with dahdi 2.1.0.4 and OpenR2 2.1.1.0. Sometimes twice a day, sometimes after 3 days, all sip devices looses registry, but asterisk doesn't show nothing strange, no error log, and all calls in E1 trunk continue running, but sending to voicemail. What could be this problem? What should I research do find a solution? Thanks a lot! Carlos

I have a problem with my internal extensions, I'm using Asterisk 1.6.2.5 and freePBX 2.6. When I call betwen extensions these don't connect. There is a long silence and finally hang up. I have an E1 whit r2 and I use openr2, but I don't have problems to do calls to the PSTN......my problem it only with internal extension. Please help!!!!...it's an Asterisk bug??..... Thanks :S

Hi all, Recently I have raised up a bug (related to segmentation fault of asterisk process) due to a core dump generated by Asterisk under the following environment: CentOS release 5.7 (Final) Kernel 2.6.18-238.12.1.el5 #1 SMP Tue May 31 13:23:01 EDT 2011 i686 i686 i386 GNU/Linux Asterisk 1.8.23.0 DAHDI Version: 2.6.1 Echo Canceller: HWEC, OSLEC libpri-1.4.14-0 libpri-devel-1.4.14-0

Hi guys, I'm hoping I've made a silly mistake here, but I've been staring at the screen for the past few hours and I can't work it out. I upgraded to 1.2.16 recently, and am having problems with zaptel. The card is detected, I get a reasonable output from ztcfg -vv, and zttool shows the installed module (TDM400) with one FXS module. But when I start asterisk, I get

Hi list, I'm trying to setup an asterisk 1.8.0 with MFC/R2, but I'm having these error messages loading chan_dahdi: > module load chan_dahdi.so ERROR[9241]: chan_dahdi.c:11848 mkintf: Signalling requested on channel 1 is MFC/R2 but line is in ISDN PRI signalling ERROR[9241]: chan_dahdi.c:16180 build_channels: Unable to register channel '1-15' I got the config in

Hello, I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't install DAHDI channel, if I issue command connect*CLI> core show channeltypes I would have response like: connect*CLI> core show channeltypes Type Description Devicestate Indications Transfer ---------- -----------

Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, do I have to establish any special setting in Asterisk 1.4 ??? Or the presence status (online,

hi list, I'm currently trying to get Asterisk running with an HFC-compatible ISDN card (no-name product, but supposed to work with Asterisk according to the packaging). the ISDN-card is connected to a alcatel ISDN-system where it should act just like a normal ISDN-phone. I went with http://www.misdn.org/index.php/MISDN_with_Asterisk and installed the package from beronet.com including:

Hi Guys! I got a Diva passive ISDN card and I can't get it work with asterisk 1.4, It is supported in the kernel as an isdn4linux device but I can't find "Modem" channel type when i type in: core show channeltypes. I'm guessing it is removed in asterisk 1.4. Tried with capi interface but it does not work :( Anybody got some idea how can i make it work or got a link to a

Hello all, I have seen some people asking how to configure asterisk to work with h323 but i did not manage to do fix it yet (i am not an asterisk expert). Can someone help me configuring asterisk? It is already compiled asterisk 1.4.5 with H323 support. Everything looks fine. Then i understand i need to configure several files: -sip.conf -ooh323.conf -extensions.conf do i also need to configure

Hi, I installed a 1-port FXO on my Ubuntu 8.4. I was earlier only hearing a fast clicking sound and now I am not hearing any dial-tone. The FXO card has 2 slots: [phone | line ]. I hve connected the wall-phone-input to the "line" slot and "phone" to my home-phone. I do not hear any dial-tone on my home-phone. Asterisk seems to recognize my hardware....here are the relevant

I recently upgraded a server to Asterisk 1.4.22 with OpenR2. Previously I was using 1.4.18. It seems that 1.4.22 has a big bug using chan_alsa.so for overhead paging. After rebooting the server it would work once or twice and then I just got an error on the CLI: [Jan 7 10:35:14] ERROR[26164]: chan_alsa.c:693 alsa_read: Read error: Resource temporarily unavailable I had to switch to chan_oss

Hi Leonardo At first should be useful to post your message at asterisk-r2-request at lists.digium.com group. By the other way let me advice, to make an explained detail of your problem as; Asterisk version Openr2 version Configurations files Dialplan dahdi pattern detail Detail of the call process (inbound or outbound call failed?) In case of outbound call failed extension dial wait time of

I thought I could try to post this question here since I have at times found excellent support here on A@H issues as well as pure/straight digium/asterisk issues. I am looking to finally build a permanent asterisk server and would prefer, if at all possible, to stick with the "new" A@H called trixbox. Does anyone know if the CentOS behind trixbox is a relatively complete CentOS system? I