similar to: one way sound with x-lite (sip)

Hi List, Additional with the latest tries from the below I get a nice random seg fault when I hangup on PSTN. (With obviously no sound on x-lite, still!) asterisk -vvvvgc results after hanging up the pstn line in: -- Executing Hangup("SIP/1087997-d79f", "") in new stack == Spawn extension (sip-phone-out, h, 2) exited non-zero on 'SIP/phonenumber-d79f' Segmentation

Hi all, Still having the one way sound problem. Any suggestions how to hunt the problem down ? Regards, Thorsten --------------------------------------------------------------- Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26

Hi, I'm working on an LLVM Pass plugin and I'm running into a problem when loading it into opt. I want to have a custom option for my pass and added an llvm::cl::opt #include'ing "llvm/Support/CommandLine.h" linking the dependant libs causes the following error when loading it with opt: opt: CommandLine Error: Option 'debug-pass' registered more than once! I

Hi Philip, thanks for the quick answer. That makes sense, but when leaving the set LLVM_LINK_COMPONENTS out I get an undefined symbol when loading the plugin: _ZTVN4llvm2cl3optINSt7__cxx1112basic_stringIcSt11char_traitsIcESaIcEEELb0ENS0_6parserIS7_EEEE which boils down to llvm::cl::opt<std::__cxx11::basic_string<char, std::char_traits<char>, std::allocator<char>>,

Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming

i want to use the external LLVM pass that built in out-of-source on windows. and, my sample pass just dump the llvm::Module during compiling some source code. i use this command. opt.exe -load -Xclang mypass.dll -mypass test.bc -o optimized_test.bc so, as my expect, my pass should dump the llvm::Module of 'test.bc', but, i couldn't see anything in my console. one the other hand, i

Hello, I would like to see if this issue is a result of a misunderstanding on my part before I file a bug. I am using LLVM 3.4, built from the source tarballs. My system's uname is "Darwin tyler-air 12.5.0 Darwin Kernel Version 12.5.0: Sun Sep 29 13:33:47 PDT 2013; root:xnu-2050.48.12~1/RELEASE_X86_64 x86_64". All I'm trying to do is add a runtime check after all call

Hello all, Still, is it impossible to use the external llvm custom pass on windows? I can build the 'mypass.dll' on windows using LLVM libraries. And, I use these 2 way to use my pass. clang.exe -Xclang -load -Xclang mypass.dll -c test.c opt.exe -load -Xclang mypass.dll -mypass test.bc -o optimized_test.bc But, Clang and opt didn't working well with my pass. This mean, my pass

Hi. I'm testing out the new 'master users' feature, and I'm not having any luck getting it to work. Anyone out there using it yet? Here's what I have so far: dovecot.conf: ------------ auth_master_user_separator = * auth default { passdb passwd-file { # Master users that can login as anyone else args = /etc/dovecot/dovecot.masterusers master = yes

... of a correctly formatted QUOTA spec in/for a static userdb. That's all I'm askin' for! i've tried a bunch of variants. the most recent two: user at domain.com:{PLAIN}testpass:::::: quota=maildir:storage=4096 user at domain.com:{PLAIN}testpass:::::: maildir:storage=4096 *none* (so far ...) work to override the spec in dovecot.conf. no errors in the logs; rather, simply:

Hi list, I'm in central-europe and signed yesterday a broadvoice account. My Asterisk box is CVS 2005-08-25. Problem I face is: "Failed to authenticate on REGISTER to 'phonenumber@sip.broadvoice.com' (Tries 2)" then "Registration for 'phonenumber@sip.broadvoice.com' timed out" and finaly "Giving up forever to register

Hi, I?m looking for settings to configure net2phone carrier in my asterisk. I found this configurations, but it?s not work. I don?t known if this configuration is for voice line or voice access account. Anybody can help me, with other configuration? Thanks. ---- *sip.conf* [general] useragent = X-Lite release 1103m register => PHONENUMBER:PASSWORD@sip.net2phone.com [net2phone] type = peer

Hi! I have a working asterisk-setup with four sip-clients. Everything works great but when the users call someone the phonenumber shows up on the receiving ends callerid-display. To correct this my provider told me to send #31# before the phonenumber, tried this with: Dial(SIP/#31#${EXTEN}@provider) but my asterisk tells me that it isn't a valid extension. The INVITE looks fine,

Hi ! Did anyone had issues/managed to solve issues with DISA over Zap channels on * 1.0.X (STABLE) ? I have a situatuion where DTMFs that should be recognized in DISA work over SIP channels and do not work over ZAP channels (Zap channels are on TE110P) I have in default context: exten=> 299,1,DISA(no-password|default) and I have SIP extension 200 in [default] and I have Zap trunk which

Hi, We are testing out Samba 4 TP2 release. We have configured Samba4 on an i386 running Linux 2.6.12.6. We are trying to get the SWAT GUI functional in this system, but it refuses login with the message *Login failed: Undetermined error - please try again * Trying as 'root' and root password on the local machine or trying as any another user fails too with this same error message.

Thanks very much for the reply Joshua! So I guess that setting dtmfmode=auto would be the safest choice in order to strip out the DTMFs from the recording, right? Cheers! Patrick Wakano On Tue, 1 May 2018, 19:36 Joshua Colp, <jcolp at digium.com> wrote: > On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote: > > Hello list, > > Hope you are all doing fine! > > >

Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers

Hi all, I configured a linux box (Debian Sarge) with Winbind to authenticate against a windows 2000 server AD. After joining the Domain via net rpc join -U Administrator I can get all AD users and groups using wbinfo -u wbinfo -g getent passwd Everything seems to be really fine, but when I try to authenticate against the AD via wbinfo -a testuser%testpass I get: -x-x-x- plaintext