similar to: ISDN Random Hangup Problems

I'm using happily this cheap phones, but I still have a little problem. Configuring the phone is extremely easy on * and I've a couple of them perfectly working, except when i try to call some toll-free number (in italy 800xxxxxxx ). If the number called is an IVR system, often with GrandStream (but also with Cisco 7905.h323) it's impossible to make the menu choices via the Dialpad.

I need to configure an * box to connect to a primary * box which is attached to a PRI in order to make calls using both the same E1 connection. I know the best solution is to use IAX (altough i could connect the IP phones on the remote site directly to the main * box that is VPNed with the remote one). I've some doubt on how to config IAX to work in this situation. I think that: On the

Greetings, I need if possibile an explanation on how to manage the incoming callerid for an incoming call. Let me explain the situation: We have two different companies in this office that shares the same PBX (* box). Each company have its own number for the incoming calls. What i'd like to implement is something that, depending on the incoming line that is involved in the call, plays a

Greetings. Today I've installed a fresh new E100P on a EuroISDN PRI. It seems to work well, accepting calls, but, when I start *, I have the screen flooded with this message: PRI got event: 8 (or 6) If i look into /var/log/asterisk/messages i've this Error: File chan_zap.c, Line 5341 (zt_pri_error): PRI: Read on 35 failed: Unknown error 500 Repeated indefinitely. Also, i've a

I've to project and build a fresh new box with * on. Basically, i'll have this situation: The office is connected to the phone carrier with a PRI. I need to let users continue to use their analog phones in a office, and an IP-phone based solution on the remote office, that will call using outside using the PRI on the first one. PRI---> Asterisk ---> Analog phones |

After a long working evening yesterday, now my * box place and receive calls with H323,SIP and ISDN line. Calling from the office to an outside line, happens: - If I call a mobile number and the called answers, all goes ok - If I call a number at home/office, and it's answered , all goes ok - If I call a toll-free number with an IVR system, nothing happens: it continues to ring indefinitely

Is there any way to run asterisk without a fxo card? I am looking only run SIP and a single fxs card.

Hello, R developers, I would like to contribute to the R project: I prepared Russian .po-files (and translated a few lines). List of changes: * files *ru.po are generated and a few lines are translated; * files ./po/LINGUAS and ./po/THANKS are modified. I am not sure whether to post the .diff- and archive files here would be such a great idea, so I put them on-line: (1)

I would appreciate ideas about MS Windows install issues. I'm at our stats summer camp and have been looking at a lot of Windows R installs and there are some wrinkles about R_LIBS_USER. On a clean Win7 or Win8 system, with R-3.0.1, we see the user library for packages defaulting to $HOME/R/win-library. I think that's awesome, the way it should be. Yea! But it does not appear that way

Dear plotting experts, I need to determine the width of a string before plotting it to the current device because I want to adjust `mai' to have the necessary space available. It seems to me that this is an "hen and egg" problem: R> a <- "test" R> strwidth(a) Error in strwidth(a) : plot.new has not been called yet # ok, I need an open device R> plot(1:2,

Dear All, I wonder if there is a step by step guide somewhere that will allow me to use Fortran and C codes from packages already on CRAN? I am using Windows XP and "R-2.2.1". I could link to these packages but the problem is that they get updated and they are not backward compatible. Also, I only require a few functions from these packages so it does not seem efficient to load

On Mon, Mar 07, 2016 at 08:14:41PM +0200, noxdafox wrote: > As the API documentation says, this is the low level API which I > have provided as an example. > > I took inspiration from the guestfs_ls0 API which does a similar job > storing the content of a directory onto a host file. > > If I understood correctly (the dynamic code generation is still > confusing me a bit),

I post a patch for comment on ext2-devel for the NFS cookie bug. Did not get any feedback yet. As Ted suggested, it set the cookie to -1 on EOF, even though it is not seek able to there. I only test it with Stephen's "readdir.c". Not have chance to run it on a NFS server yet. Do you have more information about the cache trashing bug? Regards, Chris ===== dir.c 1.5 vs edited

On 07/03/16 13:29, Richard W.M. Jones wrote: > On Sun, Mar 06, 2016 at 05:42:24PM +0200, Matteo Cafasso wrote: >> As discussed in the topic: https://www.redhat.com/archives/libguestfs/2016-March/msg00018.html >> >> I'd like to add to libguestfs the disk forensics capabilities offered by The Sleuth Kit. >> http://www.sleuthkit.org/ >> >> The two APIs

I need help creating a channel for my phone device (Quicknet PhoneJack). I have installed and loaded the driver and phone devices listen in /dev (phone0 - phone15). [phone.conf] mode=dialtone format=slinear device => /dev/phone0 fxoks=2 ;Quicknet PhoneJack [extensions.conf] ... exten=>_NXXNXXXXXX,1,Dial,Phone/phone0 ... When I try to make a call, I get the following output: Executing

Hello, I'm trying to understand why when I make a call from a SIP phone to an external number who has an IVR system in which I've to choose some options using the dialpad, it does'nt recognise the key pressed and remains still waiting for my choose. I'm tryng using Grandstream 102, and i've tryed with all the 3 modes possibile: Dtmf inband, rfc2833 and INFO (obviously

Has anyone ever seen these errors generated by a cisco 7960? none of our other brand phones seem to generate these erros: Jan 27 21:54:07 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie s exceeded on call 000ded24-d7000024-5d2ca17a-29c81cf4@65.204.176.54 for seqno 1 01 (Response) Jan 27 21:54:08 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie s exceeded on call

I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as my PSTN interface. I'm experiencing random dropped calls on the various SIP devices I have tested. Network connectivity to the SIP devices looks ok, and I have tried a variety of the devices including all of the following. Grandstream 286 Grandstresm 486 Sipura SPA 1000 Mediatrix 2102 Some example lines from my logs

Hello all, I'm becoming mad in trying to solve that issue. If I make a call from any of the phone here (I have some Grandstream and a couple of Snom105 - quite one of the best phones i've ever seen, this last one), to an outside IVR system, if i try to send dtmf to choose one of the IVR options, i notice in the /log/asterisk/messages this line: WARNING[43028]: Discarding too big frame

Hello all, I was playing with the zaphfc driver and i had these issues: I've tried to configure multiple spans (in TE mode, not NT) but it always give me errore "No such Device (6)"... I've 3 HFC-S based cards in the machine, but it seems to load only the first one. If i try to load only the first card, asterisk starts correctly, but if i try to place a call it gives the error