similar to: Can't dial out until I dial in once

I have an asterisk server at home. I'd like to test one just installed elsewhere. Both servers are behind firewalls. I can see the session start in CLI, my congratulations is apparently playing and RTP is being sent. Hearing no audio. Can send key presses and see audio playing changed. "Peer audio RTP is at port 198.145.28.177:10180", but that never shows at the client side, behind

Is it possible to force T38 for all invocations ReceiveFAX() ? Receiving fax always worked OK on Callweaver though I could put SipT38Switchover() into the dial plan. I can't with Digium fax, and it always fails at the point it decides to switch to T38.

Folks -- First, apologies for not lurking for weeks or months to get the culture of the list. I read the recent post about improvement to the quality of posts with some amusement and full agreement. The problem is a big and very real one. I hope I'm not deepening it. But my question isn't explicitly asked with this subject line or definitively answered in the archives -- that I have

So, tweaking configs, rebuilding this and that... restarting, twiddling, it works (yeah!), but fails on re-boot to work at all. Consistently, though. I believe it comes down to this: I can call out only *after* I've received a call. So, cold boot. Then: modprobe dahdi modprobe wctc4xxp modprobe wcfxo dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.3

I have a Sangoma b600de analog card using dahdi 2.1.0.4 and I get the following results (same dialplan, config etc): Asterisk 1.6.0.1 => works fine Asterisk 1.6.0.9 => can't dial out unless I dial in once or apply patch >>>==> http://bugs.digium.com/print_bug_page.php?bug_id=14577 Asterisk 1.6.1.0 => can't dial out, regardless of patch or inbound call first.

I'm trying to get my Sangoma B600 working with Asterisk 1.6.1 and DAHDI 2.2. I've tried running ./Setup dahdi (and with --64-bit_4GB flag, running amd64) and the compile fails with this for many versions of the driver (incl. 3.3.16, 3.3.16.22, and 3.4.1): Building modules, stage 2. MODPOST 1 modules CC /usr/src/wanpipe-3.3.16/patches/kdrivers/wanec/wanec.mod.o LD [M]

Hi all, I have tried to install Wanpipe 7.0.20 in a fresh-installed Debian 8.5 (Jessie) server, but when I reach to the point where I have to enter the Linux Headers directory *(installed in default with ?apt-get install linux-headers-$(uname -r)?)*, I get an error, the package refuses to compile and installation aborts. If I try the same process with Debian 7 (Wheezy), installation works just

Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

Hi all, first off all - sorry for the cross posting - i did already posted this message to asterisk-dev - after that i realized that it isn't really a -dev related question - more a -users questions. So ignore it on -dev .... we have the following setup PSTN 3 PRI Lines <---> Asterisk (1.4.22) <---> Siemens HiCom <---> Bosch Integral The Asterisk Machine

Hi, I'm trying to build a SIP / ISDN BRI gateway. I'm using asterisk and zaptel 1.0.6 with the bristuff patches. I have a Billion HFC card connected to a BRI ISDN line. Unfortunately each time asterisk tries to make a call on that channel, I get NOTICE[6236]: app_dial.c:759 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time When I try

All, I was a bit too focused on where I thought the problem was - turns out I wasn't crazy and the dialplan does work as expected. The problem was with dtmf detection - setting relaxdtmf=yes did the trick. Sorry for the premature post for help. Begin forwarded message: > From: Ben Witso <benw@bgwcomp.com> > Date: Mon Jun 21, 2004 7:28:42 PM US/Central > To: Asterisk-Users

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

I keep getting a red alarm when trying to setup asterisk to use my TE420B EC. I only have a blank context setup in my extensions.conf as I haven't started to config that until I can clear this red alarm. I don't have physical access to the server, so I can't go reseat the modules/card/ethernet cable, though I have hands on location that have done this a couple times

Hi! This weekend we had a problem with our Asterisk Box which ran flawlessly for nearly 4 weeks. The Asterisk server sits between the PSTN and a Siemens PBX and bridges 2 BRI lines. No calls, not incoming, not outgoing. The admin rebooted the Dell Box and then everything worked fine again. Now, I'm analyzing log files to find the cause. During the Asterisk outage the logfiles only show

Newest wanpipe (3.3.16) beta drivers do not compile against dahdi-linux 2.2.0-rc2 which is what you get when you get dahdi-linux-current.tar.gz Anyone have a workaround or patch? Error below ==================== Building modules, stage 2. MODPOST CC /usr/src/wanpipe-3.3.16/patches/kdrivers/wanec/wanec.mod.o LD [M] /usr/src/wanpipe-3.3.16/patches/kdrivers/wanec/wanec.ko make[1]:

hello i had configured a Dial plan in which i am using application time base i.e 1 - if? call comes to PSTN line??? from 8pm? evening till morning 8am the call should? automatically forward to guard exten i.e exten 211, and if guard dosent receive call in 30 secs message should be saved in voicemail. 2 - if call comes in working hours than it should be received by ext 112 n from there using

Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-) My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL). Calls come in and are

Preparing to use * for a 'real' installation shortly. Meanwhile, I've got a single port clone thing, 00:06.0 Communication controller: Motorola Wildcard X100P working to answer my landline and send calls to my laptop or voicemail. Sweet! Trying to "call out" from linphone, I set up this: exten => _X.,1,Dial(DAHDI/1,${EXTEN}) Both SIP client and this extension are in

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm testing a telephone connected to FXS port of a Sangoma A200 card. But I'm observing that callerid is not working. The configuration that I'm using in chan_dahdi.conf is the following one: - --------------------------------------------------------------------- ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit

Release Announcements ===================== Samba 3.5.10, 3.4.14 and 3.3.16 are security releases in order to address CVE-2011-2522 (Cross-Site Request Forgery in SWAT) and CVE-2011-2694 (Cross-Site Scripting vulnerability in SWAT). o CVE-2011-2522: The Samba Web Administration Tool (SWAT) in Samba versions 3.0.x to 3.5.9 are affected by a cross-site request forgery. o CVE-2011-2694: