similar to: asterisk 16.9 function HINT with option n does not return anything

Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that

Hi ! On Cisco 7960 (with or without 7914 add-on module) when I press speakerphone button (or select line with line button - which automatically put second line on speakerphone) after about 15-20 seconds of dialtone Asterisk stable dies (seg fault). Tested versions of Asterisk are 1.0.2, 1.0.3 or 1.0.5, chan_sccp is newest form CVS of chann-sccp.sourceforge.net ). Firmware of 7960 is

Hi ! Does anyone managed to send multipage faxes (in single TIFF file) with app_txfax from spandsp package (i'm using 0.0.2pre18, libtiff 3.7.1)? If so, I'm interested in format of TIFF file that has been sent sent succesfully (tiffinfo <fax-filename>). I'm having problems with app_txfax, sending first page successfuly 99 % of the time, but never managed to send second or

Thanks for the information Lee ! Still, something is still strange to me, since this two Panasonic Fax machines are receiving at least 20 multi-paged faxes a day (they are in same office as both Asterisk boxes, and me :) ). Beside that, POTS lines in those two faxes are from same PSTN switch as line in X100P in one of Asterisk boxes, and the ISDN PRI lines in other Asterisk box is from same PSTN

Anyone have any luck with connecting Asterisk to the Panasonic KX-TD500. I have Asterisk connected via crossover to the TE110P. We are able to make internal calls into the Asterisk Box but the PBX vendor (I know nothing about the KX-TD500) tells us it is not possible route DID over the trunk. I find this hard to believe. Anyone have any luck with this? Thanks! -------------- next part

Hi ! Did anyone had issues/managed to solve issues with DISA over Zap channels on * 1.0.X (STABLE) ? I have a situatuion where DTMFs that should be recognized in DISA work over SIP channels and do not work over ZAP channels (Zap channels are on TE110P) I have in default context: exten=> 299,1,DISA(no-password|default) and I have SIP extension 200 in [default] and I have Zap trunk which

OK I see the ponit (although I never said that second page is interrupted - I said that in some combinations of resolutions and TIFF options receiving fax "spits" another blank sheet of paper beside the clearly received first page). I have read someware (some faxing tutuorial) that there is some kind of control code (6 EOLs I belive) that should be in TIFFs between pages and that

Hi everyone ! I have some aditional info on problem with TXFax and sending multi-page TIFFs. I have made 6 different multi-page TIFFs (Group3 1D with fillbits EOL aligned - 3 pages one TIFF in lowres and one in hires, Group3 2D -3 pages againg in both resolutions , and Group 4 - 3pages in both resolutions), and then tried to send them to Panasonic KX-F1100, Panasonic KX-F500 and SpanDSP

Dear All, I'm having problem with spandsp/txfax, I'm not able to send a multi paged tiff file, the fax machine receives the first page of the document and complains about communication problem. The file what I'm trying to send has 2 pages and is received & generated by spandsp/rxfax. Searched in the archives, but found no solution. I'm using: Dell Optiplex PC Inoteska Quad

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

On Mon, Apr 25, 2016 at 11:11 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > 2016-04-25 18:14 GMT+02:00 George Joseph <gjoseph at digium.com>: > >> >> >> On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com> >> wrote: >> >>> >>> >>> On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07

On Mon, Apr 25, 2016 at 10:00 AM, George Joseph <gjoseph at digium.com> wrote: > > > On Mon, Apr 25, 2016 at 9:29 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I've just discovered PJSIP 's support of set_var setting in pjsip.conf. >> Is this setting also supported in pjsip_wizard.conf ? >> On a fresh 13.8.2, it

Howdy, I'm trying to re-write my voicemail check extension. I formerly used the SIPPEER function to get the mailbox for a peer with ${SIPPEER(${peer},mailbox)} Is there a way to do this with PJSIP now that I've converted over? I see a function PJSIP_ENDPOINT and it has a mailboxes subset but I'm not retrieving any data from it when I query it. -- A human being should be

Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36

Hi all, I'm new to the forum. Oh no....newbie question coming, I hear you all cry! I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've

Hi all ! I'm trying to make a automonitor generated filename to "make its way" into CRD(usrefiled), so I can keep track of recorded conversations in CDR logs. Looking how to do that, I have found cool (but almost undocumented) option of res_monitor: if you set monitor format in form of "format:<string>" (i.e. "wav:monitor"), res_monitor will prefix the

On Mon, Sep 24, 2018 at 6:30 PM Richard W.M. Jones <rjones@redhat.com> wrote: > On Mon, Sep 24, 2018 at 10:00:21AM +0200, Fabien Dupont wrote: > > Hi, > > Hi Fabien, sorry I didn't respond to this earlier as I was doing some > work. If you CC me on emails then you can usually get a quicker > response. > > > I've read the virt-v2v OpenStack output code

On Thu, Sep 19, 2019 at 01:47:45PM +0200, Nenad Peric wrote: > A theoretical way to test this against any other protocol is to have a > firewall rule which can be triggered locally which just drops the traffic > and then re-enables the traffic in a while (short or long, can be > decided/tweaked). We generally want our tests to run without root and without doing weird stuff to the

Hi, I'm using spandsp-0.0.3 [http://www.soft-switch.org/downloads/snapshots/spandsp/ spandsp-20061116.tar.gz] on a bristuffed asterisk (1.2.13) [http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0- PRE-1v.tar.gz] libtiff is at version 3.6.0 Running on: Linux router2 2.6.17-2-686 #1 SMP Wed Sep 13 16:34:10 UTC 2006 i686 GNU/Linux Debian testing distro. I've tried many

I have a virtual machine that had been running fine using a RAW disk image (IDE). I am in the process of converting this virtual machine to use a single LVM and virtio. (For LVM backup snapshots and easy resizing purposes (no LVM within the guest)) I create the LVM and copied all the files from the existing RAW image to an ext3 under the LVM. However, I cannot get it to boot. It has to do