similar to: SoftFax question

Only to signal that if you want to compile app_rxfax and app_txfax with last cvsses of asterisk you have to modify the patched versions of the apps directory Makefile to define the symbol _GNU_SOURCE example, the line to compile app_rxfax from: gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c to gcc -D_GNU_SOURCE -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c

I followed the instructions at http://www.opencall.org/instructions.html and http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html I was able to compile spandsp (./configure ; make ; make install), "manually" patched asterisk apps/Makefile (/usr/src/asterisk/apps), as the supplied patch does not fit the actual CVS apps/Makefile After make clean ; make install, I

Hi all, After a long time having no time, I have finally done some fresh work on my software fax machine. I have replaced the original carrier tracking with something more robust. I have also added 4800, and 2400 bits per second modes, and cleaned up a few bugs in areas like superfine mode operation. I apologise for this update taking so long. At ftp://ftp.opencall.org/pub/spandsp you will

Hi, We have no problems sending to HP and Panasonic fax machines in the office. We do have problems when we try to send faxes to services supporting fax, i.e. J2 or our UC platform. The receiving side doesn't recognize fax. To send a fax we drop into /var/spool/asterisk/outgoing: Channel: Zap/g1/<fax number> MaxRetries: 0 WaitTime: 20 Context: webley_txfax Extension: txfax_ext

Hi Steve and all, 1. Faxing from asterisk back to the same asterisk (from one Zap channel to another) doesn't work for us. Txfax called with the 'caller' parameter issues CED, while the receiving side needs CNG in order to switch to fax extension with rxfax. 2. This is probably the reason why J2 and our UC don't recognize incoming fax. Thank you. Alex Zarubin Webley Systems

I am looking for a simple way to send occasional faxes via the FXO port on my SPA3102 -- without having to connect a fax modem to an ATA. In an ideal world, this would be some sort of "softfax" that runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with T.38. This is one of those things that I thought would be relatively straightforward, but a couple of hours of Googling

I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk

Hi, It seems as 'the google' has left me today so I am trying the list. How do I get access to SIP responsecodes from dialplan/agi. Yes I know that I should stay with 'DIALSTATUS' but there are cases where I need the responsecode like '484 adress incomplete' and not just the 'NO ANSWER' DIALSTATUS. Is there a channel variable/function that skipped over by

HP doesn't support Fedora on Proliant hw so you can't just install their ILO and get access to hw info like cpu/mb/temperature,powersupply status,fan info aso. I used the link below to get that access, which enabled me to write a small script that sends snmp-traps to hp-ovo. I did spend quite some time myself until I found this link.

I've got several issues with AGI/FastAGI 1. When an AGI script sends a command to Asterisk via stdin, why does Asterisk block and not return a result until the command is complete? Specifically, the dial command. If I send a Dial command to Asterisk, I don't get a return result until AFTER the call is HUNG UP. Not when it's ringing, not when the call is connected, but when it's

>Freddi Hansen wrote: >> with boards from Aculab, we are replacing Aculab boards with Digium >> boards BUT we would need more >> Digium boards IF we could use both Digium and Aculab cards in the same >> server. The reason being that >> TE410P doesn't support SS7-ISUP so we continue using only Aculab cards >> in the servers that must support >>

I hate to bring this thread back to life, but... > it may be possible to get it supported, do you think the price >point is remotely competitive with Digium hardware? Also as I am not >about to divulge my information to them to look in the downloads >section, what is the licensing of their SDK? What is the licensing of >the driver? > >Steven >On Tue, 2002-11-26 at 14:52,

> > Its up to 5.8G of resident memory with 28321 calls processed. > The OOM killer is going to kill this soon at this rate (8GB RAM machine). > This seems like a pretty serious problem. > It looks like I'll need to restart asterisk every night.... Hi the number of cpu cores that you see with top times 512Mbyte is the level of ram that's needed e.g. a hp-gen8 with 2 octo

I'm really trying to understand this. I have a Sipura 2000, brother MFC, and SpanDSP set up on asterisk. because asterisk softfax was not working well, i set up faxes to goto line2 on the sipura. this is working fine, i have only tested a few short faxes, but already my completion rate was 100% vs 5-10% that spandsp and asterisk gave me. here's the odd part. using the fax machine, I

Hi, okay heres what I want to do .. simple ivr, we take a call, answer it, play a menu, dial out based on options. No problems so far. The CDR always shows the call as answered as I answer the 1st leg to play the prompts, I am actually more interested in if the 2nd leg - the outbound part - has been answered or not before the call is hungup. How can I get this and record the information in

Hi, I got some errors about not being able to create subdir [already existing] on a 'make update' in my zaptel 1.4. I removed the directory and did a new svn co of zaptel 1.4 [ svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel ] now I get: .... /usr/bin/install -c -D -m 644 tonezone.h /usr/include/zaptel/tonezone.h make -C firmware hotplug-install DESTDIR=

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try, database get SIP/Registry/<peername> it gives you a string which contains the info, then pass it to CUT to extract ip-adr and port Freddi > To do that you need to get the remote ip address and port of the sip peer! > > I found the function: > > ${SIPPEER(exten:ip) > > But how can I get the port??? > >

Hi, I started to get some Zaptel compile errors after a 'make update' I did a clean zaptel install with: svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel I am still getting the error, is this someelse seeing this ?. CC [M] /usr/src/zaptel/kernel/zaptel-base.o /usr/src/zaptel/kernel/zaptel-base.c: In function 'zt_reallocbufs':

Hi, Can someone please help me with my problem below. Any suggestions is appreciated. thanks, Carine ----- Original Message ---- From: Carine Liang <carineliang@yahoo.com.sg> To: speex-dev@xiph.org; speex-dev@xiph.org Sent: Tuesday, 31 October 2006 1:05:49 PM Subject: [Speex-dev] Integrating speex with VideoNet application: Constant background noise Hi, I am developing a peer-to-peer