similar to: Spandsp garbage

On the /proc/zaptel it was lost interrupts, you haven't got any so that's good! On opencall.org there's a known issues link and that mentions some fax machines that have issues, might be worth a quick check there. I can receive from our fax machine this end, if you'd like I'll send you a test fax if you send me your fax number, if you can receive from us and tpc then it's

>From couple of weeks i am working on asterisk fax but was not successful. I am able to receive only half fax documents and its ending with blurred lines. I have tried with almost all the versions of spandsp 0.0.2pre4,pre10 etc but of no use. I have digium card to which i have plugged in 2 pstn lines and i have removed echocancellation too(zapata.conf) and also enabled g7111 alaw&Ulaw

I'm trying to install the RxFax in Asterisk. After much trouble getting app_rxfax.c (and friends) compiles and installed into Asterisk (thanks to the many asterisk-users messages) it did get it installed. I have been unable to receive a legible fax. app_rxfax provides lots of debugging information but none useful to me. I was hoping that someone has an idea what might be going on.

more detailed output Sep 21 15:54:31 DEBUG[1120357296]: pbx.c:1255 pbx_extension_helper: Launching 'RxFAX' Sep 21 15:54:31 DEBUG[1120357296]: channel.c:1699 ast_set_read_format: Set channel Zap/1-1 to read format SLINR Sep 21 15:54:31 DEBUG[1120357296]: channel.c:1666 ast_set_write_format: Set channel Zap/1-1 to write format SLINR Changed from phase 0 to 1 Slow carrier up Slow carrier

Sorry to bore you more with the clock issue, but have you check /proc/zaptel/<span> to make sure it's not missing interrupts? There's also an option to record the audio for the fax, you could listen to that vs a recorded file that will receive correctly on a fax machine and see whether there is an obvious difference? (Good luck, that'll be really scraping the barrel!!) Does it

Hi, so far I had our PRI line connected via Digium TE410P and ZAP channel to asterisk which worked perfectly. I now have a second line coming in through a H.323 gateway and chan_oh323. rxfax() still works and receives faxes with G.711 alaw codec, but I always get one empty first page via H.323 - a one-page-fax becomes a two-page-fax with one empty page in front of it. It seems the empty page

Hi, I just sent this to Steve Underwood, but then found a bunch of posts on the mailing list about similar issues.. does anyone have the fix? I'm running asterisk CVS-HEAD-06/28/04-18:13:13, spandsp 0.0.1k, libtif 3.5.7 one thing i just noticed is that calls come in with format '72' which is G711A-law or LinearPCM.. it uses PCM for the call, i assume this is ok the results of RxFAX

I'm running asterisk CVS HEAD from 20040601 with spandsp 0.0.1k and libtiff 3.6.0 (no other copies are installed). I've put the audio files up at http://www.mixdown.ca/~andrew/dump/akohlsmith-faxsegfault.tgz -- the machine I am faxing from is a Canon IR3300 printer/copier/fax, but I get similar crashes from $29.95 fax machines too. :-) I'm trying to get a decent backtrace but

Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme

Hello everybody, when I upgraded from RC1 to RC2 I didn't had any audio between my ATA286 and H323 EP (my post from 13/08/04) I checked further and discover that problem is with ATA286 who is unable to call. I always get an 404 error. Coming back to RC1 everything works fine again. I tried to modify my dtmfmode from rfc2833 to info but in change nothing. Local call to asterisk are

I've been reading the RHEL 5 manual/guide linked from centos, and the idea of LVM with expandable partitions seems like a great idea (yes duh why else would it exist if not?). Although the documentation shows a screenshot of how the final setup would look like with LVM & RAID together, it doesn't actually gives the step. 1. Am I correct in thinking it should be done in this way -

Hi list! I have a 3TB usb drive with an NTFS filesystem on it, that I can't mount.: [bent at bnlaptop ~]$ sudo /sbin/mount.ntfs-3g /dev/sdb1 /mnt Failed to read vcn 0xf28: Input/output error Failed to mount '/dev/sdb1': Input/output error NTFS is either inconsistent, or there is a hardware fault, or it's a SoftRAID/FakeRAID hardware. In the first case run chkdsk /f on Windows then

Hi Am 10.01.25 um 02:49 schrieb Andy Yan: > Hi Thomas, > > At 2025-01-09 22:56:56, "Thomas Zimmermann" <tzimmermann at suse.de> wrote: >> Add drm_modes_size_dumb(), a helper to calculate the dumb-buffer >> scanline pitch and allocation size. Implementations of struct >> drm_driver.dumb_create can call the new helper for their size >> computations.

Hello, I recently cloned an old install from one server chassis to another via rsync. The old setup only had one disk and the new one is set up with two disks in a softraid. I managed to solve the problem and thought I'd present it here to benefit others. attempting to add sda3 to md2 says: mdadm: hot add failed for /dev/sda3: Invalid argument dmesg says: md: could not bd_claim

Hi, I'm having trouble with an IRQ conflict I can't figure out how to resolve. I tried KDE (which is great), but this required the NVIDIA driver. However, this driver conflicts with my modem. If the driver is loaded, then the system locks up when accessing the modem. If I remove the NVIDIA driver from /boot/loader.conf, modem works fine. I tried moving the modem to different slots, but

A BETA firmware upgrade toasted my ATA286. It now has limited operations. It will get an IP address via DHCP and register to the last configured SIP server, but the web interface is gone as is the voice config menu. Apart from registration, there doesn't appear to be any other SIP functionality. An Ethereal dump does not show the device trying to grab a new firmware via tftp on bootup, so

Dear Everyone, I am using R 2.6.2 on my Redhat AS4. I installed the RPM offered on the cran website. I would like to use R from my bash console with the following command: /path/to/R --vanilla --slave --args < /data/myscript.R this script invokes a range of functions and tools, among others to process a bunch of raster images in a loop using GDAL. The thing is that releasing this command in

Call drm_mode_size_dumb() to compute dumb-buffer scanline pitch and buffer size. The hardware requires the framebuffer width to be a multiple of 8. The scanline pitch has be large enough to support this. Therefore compute the byte size of 8 pixels in the given color mode and align the pitch accordingly. Signed-off-by: Thomas Zimmermann <tzimmermann at suse.de> Cc: Philipp Zabel <p.zabel

Hi list, would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side,

Hello everybody I am trying to connect my asterisk to a payment equipment trough PSTN. I have a TDM400P card with an fxs module an the equipment use modem to send data! I was thinking to implement a software data modem in asterisk, but I found out that there is just faxmodem for asterisk, Is anyone here know something about software data modem working with asterisk to help out? Thanks,