similar to: Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4

2009/2/26 Olivier <oza-4h07 at myamail.com> > I must add I tried spandsp0.0.6xxx as a warning message advised me to do so > (using 0.0.4 would be ok for me but current trunk doesn't allow this > anymore, it seems). > > > 2009/2/26 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> With 0.0.6pre3: >> # ./build.sh >> CMake Warning (dev)

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

svn?co?https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons?agx-ast-addons ./build_sh from the trunk. ? -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olivier Sent: 10 February 2009 18:35 To: michael at networkstuff.co.nz; Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Hi All, I am trying to add FAX to my SIP confiig and I am getting some errors, any help would be great. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\"CVS-v1-0-12/23/04-22:36:11\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

somebody know how to compile the rxfax and txfax apps under asterisk 1.4.0?? i get this errors: Generating embedded module rules ... make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. [CC] app_rxfax.c -> app_rxfax.o

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/

How to compile additional module to asterisk? I have app_nv_backgrounddetect.c file and followed instructions below, but "make" did not generate app_nv_backgrounddetect.so or app_nv_backgrounddetect.o (1) Drop the code in your /usr/src/asterisk/apps directory (2) Edit the Makefile in the apps directory. Add the following line: APPS+=app_nv_backgrounddetect.so (3) Go to

Hi, I have recently upgraded from Asterisk 1.2.15 to 1.4.2 and I'm experiencing trouble with rx_fax. I have followed instructions posted by Sems: http://www.sems.org/entry.asp?ENTRY_ID=197 I'm using spandsp-0.0.3pre28 and the app_rxfax and app_txfax from: http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/ rx_fax and tx_fax are both enabled via make

Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, do I have to establish any special setting in Asterisk 1.4 ??? Or the presence status (online,

Hello, I have set pickupgroup and callgroup for zap, sip and iax2 devices. Everything is working good with zap and sip and between these two. Iax2 pickupgroup and callgroup seems to be broken. I cannot pickup a call to IAX2 from SIP. Is there somewhere a bug ? I am running: Asterisk 1.2.9.1 Bartosz

I'm using a Snom 320 in a CAP position and the receptionist wants to do blind transfers. OK, no problem so far. Now she has asked me how to UN-transfer a call, as in, she transfers a call and wants to hook the call back before it connects (she wanted to tell the caller additional information for example) I don't think that this is possible as once my dialplan starts using Dial()

I have googled this and come up empty. Has anyone had any problems compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am getting when I run make. app_nv_faxdetect.c: In function `nv_detectfax_exec': app_nv_faxdetect.c:210: error: structure has no member named `cid' app_nv_faxdetect.c:227: error: structure has no member named `cid' app_nv_faxdetect.c:265: error:

Hi, I found some contradicting infos about pass through of T.38 data. Are there any experiences of just passing T.38 via SIP from one T.38 application or gateway trough asterisk to another T.38 application or gateway? Would asterisk maybe even pass T.38 from chan_oh323 to chan_sip (without understanding the content)? Please tell me, if you have knowledges or experiences on this topic!

When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED Has anybody seen this.

Can anybody share information how to install NVBackgroundDetect? I have the app_nv_backgrounddetect.c but I'm missing: app_nv_backgrounddetect.o and app_nv_backgrounddetect.so and have no idea how to generate them. Wiki point to contact Newman Telecom but all I received was the app_nv_backgrounddetect.c and no instruction how how to install it. The installation instruction from wiki

Anyone know where the sources for app_nv_faxdetect officially live? I couldn't turn them up on a web search, just patched versions for 1.4, etc. Thanks.

After a power interruption, asterisk doesn't seem to be routing calls and there seems to be a premature timeout and hangups occurring. I am clueless where to look. Can someone in the know, look at the following log and enlighten me if there's a problem, or if it looks normal. From the calling phone, it keeps ringing as if never picked up. Thanks soo much. -braman

Hi, maybe a dumb question, but it seems that some calls are directed to our central dial in number despite the extensions the callers say they dialled. E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown extension, where it is right, and redirects the call to the central dial in extension 1234-0. This only seems to happen when the numbers are dialled manually. When

I have a plain POTS line coming into FXO in a Digium card, this is developers kit card with 1 FXO and 1 FXS. The latency is very high, in that, it picks up after 8 rings. I don't know what I can tune to reduce to 2 or 3 rings. If it's of any help , I am posting a section of the log : ==== Apr 30 10:26:50 DEBUG[3050] manager.c: Manager received command 'Command' Apr 30

I seem to be getting a connection from telasip but instead of dialing my default extension, nothing happens. I listen to dead air. I have a fxo card configured and working on both inbound and outbound calls. Telasip is working outbound. I put in the recommended (by telasip) changes to the trunk for incoming, e.g. host=gw4.telasip.com insecure=very qualify=yes type=user context=from-pstn Then