similar to: Multitech MVP130 as FXO with asterisk

Hi All, I'm new to asterisk, can I use asterisk with a Multitech mvp 210? Thanks, Steve

We recently had a customer install that went horribly wrong. Serious echo (pots lines into a cac cb) that, although * did a good job getting rid of alot of it, could not get rid of it all. We tried everything, every canceller, gain setting, etc... combination possible to no avail. Both the vegastream and mediatrix boxes also could not get rid of all of the echo. So, on an off chance, we

Hi List, I have a Multitech H323 Gateway MVP400 box with 1 phone on port FXO 1. I have Asterisk ruuning in Fedora Core 3. Both are in the same network. But I can't figure what I have to do in Asterisk to make that box work. What files I have to configure? Can anyone help me? I will really appreaciate youu help. Luis. _________________________________________________________ Do You Yahoo!?

Everyone, I have a centos 5.10 machine set up to be a fax server using hylafax+ with three multitech MT5634ZBA modems. I am still in the testing phase of using this system for production, and during my tests I have detected a problem I have not been able to solve except for unplugging the modem from the usb port and plugging it back into the usb port. In the middle of a fax production of

What FXO interface methods are folks using successfully in the UK? I'm looking for good, known-to-work solutions for commercial use for two PSTN trunks on an Asterisk box. Here's the options I have, as I see it: i) Two Digium X100Ps. Pro - cheap (c. ?120), CE approved. Con - UK line impedance mismatch, with resulting echo problems, plus needs two PCI slots. ii) Digium TDM400P with two

Dear all, I have a problem with constructing a nested loop. I have two matrices: pnr: 800 rows 14 columns where rows are 40x20 meaning that 40 rows belong to one of twenty objects in the matrix pnr mvp: 20 rows and 14 columns I want to: calculate a distance value with the first 40 rows of pnr but 20times with the 1st row of mvp then the next 40 rows of pnr and 20times with the second row

Ok, I am back from my thanksgiving holiday, and I find there was a big snow storm here in Seattle. Apparently during the storm there where multiple brown out/black outs. I have struggled since day one to get a high quality PSTN gateway configured with my very long loop and Mac based asterisk. I originally tried the HT-488, which had multiple issues, and was unacceptable. I then purchased

--- include/cube.h | 18 +++------ plugins/cube.c | 120 +++++++++++++++++-------------------------------------- 2 files changed, 43 insertions(+), 95 deletions(-) diff --git a/include/cube.h b/include/cube.h index 0a87626..293bad1 100644 --- a/include/cube.h +++ b/include/cube.h @@ -87,16 +87,11 @@ typedef void (*CubePaintInsideProc) (CompScreen *s, CompOutput *output,

hi .. i've just converted myself back to a newbie by trying to experiment with some new stuff .. I have connected two grandstream Budgettone 100 phones to my asterisk, and trying to experiment with them .. I am trying to get into the asterisk sample basically .. when I dial 1000 asterisk receives the call, but I do not hear any sound on the phone. Dialling from phone to phone direct (via

Hello. I have two records in dialplan: exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall at sip.linphone.org) exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall at iptel.org) Calling testA works fine while testB fails with "CONGESTION". Adding debug for console shows that pjsip_resolver.c does `New queries added, performing parallel resolution again` for linphone after

It seems a new Linphone 4.2 is to be published next week ! Hopefully, ... Le ven. 5 juin 2020 à 13:34, John Hughes <john at calva.com> a écrit : > On 26/05/2020 15:33, Olivier wrote: > > Hi John, > > 1. Could you get any further, in your quest for working BLF with linphone ? > > The patches to get linphone-3.12 BLF working with Asterisk are here: > >

Signed-off-by: Alex Nelson <ajnelson at cs.ucsc.edu> --- images/mkrlenvalue_test_hive.py | 37 +++++++++++++++++++++++++++++++++++++ images/rlenvalue_test_hive | Bin 0 -> 12288 bytes 2 files changed, 37 insertions(+), 0 deletions(-) create mode 100755 images/mkrlenvalue_test_hive.py create mode 100644 images/rlenvalue_test_hive diff --git a/images/mkrlenvalue_test_hive.py

On 26/05/2020 15:33, Olivier wrote: > Hi John, > > 1. Could you get any further, in your quest for working BLF with > linphone ? The patches to get linphone-3.12 BLF working with Asterisk are here: http://perso.calvaedi.com/~john/linphone-3/ They're pretty damnned trivial: 1. add the "Accept" header to the SUBSCRIBE message so asterisk doesn't reject it. 2.

I'm experiencing one-way voice paths, followed by a hangup on one softphoine and not the other. Also, caller ID has odd outputs -- and I wonder if the problems are related. My configuration has Asterisk and a Linphone softphone running on the same box. I have a PC, and on that PC I use X-Lite or SJPhone to connect to the Linphone instance. When I call from the PC to Linphone: * I call

I'm planning a new * system which will utilize 48 stations (Polycom Soundpoint 501s mostly) and a dual span PRI card and I have some questions. The system will host MeetMe conferences of 10-15 users on a regular basis and see fairly high usage as it is going into a medical setting. 1. I haven't built a system this big before, will a processor such as the Intel Pentium D 830 3.0GHz / 2MB

Hi folks an issue I don't understand. I'm running * stable 1.0.3 on public internet, with following iax.conf / sip.conf entries: iax.conf [100] type=friend username=Foo context=default auth=md5,plaintext,rsa secret=secret host=dynamic callerid="Foo" <100> qualify=no sip.conf [10] type=friend username=Bar context=default callerid=Bar <10>

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone

Today's "frustrated programmer" award goes to Linphone, which has the following debug output: > (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that