similar to: Not able to receive fax

Hi, I am having an issue with someone accessing our server via a SIP/VOIP connection. I have changed my iptables rules to drop all UDP traffic from and too this IP address, but this traffic seems to still run through my server. These are the iptables rules that I current have on the server. -A INPUT -i eth0 -s 209.61.231.42 -p udp -j DROP -A INPUT -i eth0 -d 209.61.231.42 -p udp -j DROP

Hi, I have a firewall running IPTABLES. I have tried to route sip traffic from my WAN(eth3) interface to a VLAN(eth2.2) interface, however the data will not route to the VLAN it keeps routing to the default interface(eth2). Does anyone have an idea as to what I need to look for? Regards Jennifer Botten ETECH Tel: +2787 150 5285 Fax: 086 638 2412 Mobile: +27 82

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

To provide a reliable fax solution for users connected to a Asterisk 1.6.2.6 server i have tested a few T.38 capable ATA's: - Patton M-ATA - Grandstream HandyTone 486 - Fritz!Box 7170 I have tried Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 and also Asterisk 1.6.2.6 with Fax for Asterisk installed. These Asterisk servers are connected to a Cisco PGW 2200 + AS5400XM. Sending fax

Hi, I seem to have a problem with crontab. I am running CentOS 5.5. If I add my bakup scripts into crontab they take very long to run. If I run my scripts manually they run 100%. I have also tried vixiecron, same issue. Any ideas? Thanks Jenny -------------- next part -------------- An HTML attachment was scrubbed... URL:

Using spandsp-0.0.6-pre17, SendFax on 1.6.1.18 and ReceiveFax on 1.6.2.8. Sip.conf on both sides has t38pt_udptl = yes. -- Executing [s at fax-tx-test:3] SendFAX("SIP/side-sip-00000009", "/var/spool/asterisk/fax/20091113_1455.tif") in new stack [Mar 20 17:05:34] WARNING[6433]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped

New asterisk user, pretty much set up except for spandsp. I get the following when trying to compile: app_rxfax.c app_rxfax.c: In function `phase_e_handler': app_rxfax.c:92: error: structure has no member named `cid' app_rxfax.c:92: error: structure has no member named `cid' app_rxfax.c: In function `rxfax_exec': app_rxfax.c:260: error: structure has no

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Hi I have searched a bit for information regarding the status on the dialplan in lua (pbx_lua.so). I know that 'hint' won't work and has to be put in the regular extensions.conf/ael. Are there any other considerations? Features that are not implemented, or known stability issues? Thanks, Tommy Botten Jensen -----BEGIN PGP

Has anyone seen this before, and know what is happening? USER at HOST:~/asterisk/agx-ast-addons# ./build.sh -- Configuring done -- Generating done -- Build files have been written to: /root/asterisk/agx-ast-addons [ 11%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o Linking C shared module dist/app_devstate.so [ 11%] Built target app_devstate [ 22%] Building C object

Has anybody been successful with compiling the pre3 version of SpanDSP on the current Asterisk CVS? I'm getting: app_rxfax.c: In function `phase_e_handler': app_rxfax.c:77: warning: implicit declaration of function `fax_get_transfer_statistics' app_rxfax.c:78: warning: implicit declaration of function `fax_get_far_ident' app_rxfax.c:79: warning: implicit declaration of

A DID number was dedicated to receive fax, but i have the problem when getting fax call, which call will become a normal phone call and no fax was printed. When fax is detected, the fax extension is executed and dial the extension of the HT486 device (firmware 1.0.5.22). Somehow sending fax out working well. In the mailing lists, i notice some are using HT286 and it work. Could someone share

Receiving a fax pstn - pstn with 1.6.2.6-rc2: -- Executing [s at incoming-pstn-line:1] Answer("DAHDI/4-1", "") in new stack -- Executing [s at incoming-pstn-line:2] Wait("DAHDI/4-1", "3") in new stack -- Executing [s at incoming-pstn-line:3] Dial("DAHDI/4-1", "DAHDI/g0,36") in new stack -- Called g0 --

Hello, Is it possible to detect a hook flash in asterisk. I want to be able to perform some functions an hook flash. I have the following entry in features.conf which executes a Macro on detecting key press '**'. [applicationmap] test => **,caller,Macro,testflash Is it possible to do this action on hook flash? -------------- next part -------------- An HTML attachment was

Hello, Is it possible to use the asterisk manager interface to originate multiple channels? like Action: Originate Channel: SIP/101&SIP/102 So that both extensions 101 and 102 rings simultaneously. I am using asterisk manager interface over http. Thanks

This message is pointed directly to Steve Underwood. I tought it would not be nice to directly email him with a question that interests a good part of the Asterisk community, so here it is. :) Steve, remember a few days ago when we discussed about issues on Asterisk 1.6.1.13 and T.38 fax reception? Well I opened an issue on Mantis (https://issues.asterisk.org/view.php?id=16756) and turns out it

I did cvs update on asterisk, zaptel, libpri as of today (November 5, 2003). I also did 'make clean' on each of them. My previous version of asterisk was cvs of September 15, 2003. No other changes have been made to my system other that these updates. when running 'make asterisk' the following error appears term.c:55: conflicting types for `term_color'

Hi, With 0.0.6pre3: # ./build.sh CMake Warning (dev) in CMakeLists.txt: No cmake_minimum_required command is present. A line of code such as cmake_minimum_required(VERSION 2.6) should be added at the top of the file. The version specified may be lower if you wish to support older CMake versions for this project. For more information run "cmake --help-policy CMP0000".

Hello, I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17. I have context=defcontext set in sip.conf. For each peer I have context=outcontext in the peer definition since I want outgoing calls from registered SIP peers to go through context 'outcontext'. This used to work in the older version (1.6.2.7), but after upgrading this has stopped working. Now outgoing calls are going to

Hi all, Anyone have an idea which version of spandsp, libunicall, libmfcr2, libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the latest asterisk 1.2? Would that be the ones listed below? http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/

Hello, In my scenario, the asterisk machine is installed behind a CISCO mediaGW in order to be able communicate with the PSTN. Asterisk is configured to use T.38 to send and receive faxes. I'm trying to receive a fax from a fax machine located in the PSTN. Apparently everything goes well: the fax machine says the transmission was successfully completed, and the fax file is successfully