similar to: NVFaxDetect errors on make

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:

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

Hello All, I am looking to get asterisk to receive faxes but I really don't know where to go from here, any help pointing me in the right direction would be great. What I would really like is for this second line to answer faxes, but if a user typing in an extension it goes to that extension but I need to get the fax working first. Error I get when I turn on Fax conf... pbx.c:1945

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".

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!

If you are having problems compiling NVFaxDetect (app_nv_faxdetect.c) or other Newman Telecom applications on Asterisk 1.4, please look at Steve's comments at: http://www.voip-info.org/wiki/view/NewmanTelOnAsterisk14 Several changes to Asterisk prevents NVFaxDetect and other apps from registering. Some changes needed. He also has copies of the code if you need it... Justin Newman

OK, I got my Cisco 7960's to auto-answer on the second line but I can't get the Asterisk to call all the lines at one time. I have 4 phones I would like all of then to answer when I dial x300. Any help would be great Thanks Tuska extensions.conf [conference] exten => 300,1,AGI(callall) exten => 300,2,MeetMe(300,dtqp) ; press # to exit the conference exten =>

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

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:

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)

Any ideas? I edited the Makefile as instructed, ty. Now compiling .... sccp_channel.c 279 lines sccp_channel.c: In function `sccp_channel_send_callinfo': sccp_channel.c:48: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no

We have extra equipment that was over-ordered or unused. All of the equipment is brand new. The equipment has been highly discounted to move quickly - the last set of equipment sold in 48 hours. If this equipment is of interest to you, call or e-mail quickly. Buy on VOXILLA and SAVE $300 each (Cisco routers & switches): http://store.voxilla.com/customer/home.php?cat=259 For Sale (all new):

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

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. --

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

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

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.

Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the "transferer". Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like

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

Hi List, I need to modify the callerID name of the call coming back when a parked call returns to the extension that parked it when it times out. Looking at app_parkandannounce.c /* Now place the call to the extention */ snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars =