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

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, 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 chan_capi On the IP side: X-lite (build: 1084) Calling and get calls on PSTN from X-Lite is no problem. We only get sound from PSTN to X-lite. Never from X.-lite to PSTN. The

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>>,

Is it possible to change error messages for Validation helpers? I am writing an app against a existing database (so no control over column names), but when there is validation error (e.g. with validate_presence_of) I would like to customize the field name. For example for telephone whose field name is PhoneNumber I would like to chnage it to "Telephone Number cannot be empty" rather

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 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

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

Hi all, I've been trying to use BroadVoice as a SIP service provider. They don't officially support * but are helpful when it comes to answering questions for setup parameters. They claim they have no firewalls or access lists that need to be set up so I can get access to their servers. However, something's still not quite right when I use the parameters. It looks like our Asterisk

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

I am trying to place an analog outbound call from a Sipura SPA-841 through a * server with a TDM400P and 4 FXO's. When I call in from an analog line everything works fine, I can talk over the SIP phone. When I call out, * says: == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on 'SIP/sipphone-d29d' -- Executing Dial("SIP/sipphone-9eb0",

I have an asterisk 1.8.x box that intermittently returns a 401. Calls come through the same peer all the time, from the same carrier. However intermittently the asterisk box returns a 401. Below is the output of a failed call (1st) and a successful call (2nd). I can't see any difference until we get to these lines. Bad call: --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT)

Hello! My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on Fedora Core 4. I am making automatic call-out campaign with this setup on 4 PRI. The scripts for this: ==================================================================== caller php script write this to outgoung folder: fwrite($outfile,"Channel: Zap/g1/$phonenumber\n"); fwrite($outfile,"MaxRetries:

Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=<phonenumber> authuser=<phonenumber> secret=<registration password> Dan

I''m trying to setup some mildly complex associations for a project we''re working on and can''t seem to find much documentation on n-way has_many :through associations. I have the following models: Person, PhysicalAddress, EmailAddress, PhoneNumber. Each person can have multiple PhysicalAddresses, EmailAddresses, and PhoneNumbers, and multiple people can share the same

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

Is it possible to rewrite caller id's? I would like to have sip phones appear by their local cid (like Henk <208>) but when they call out using the PRI I would like their full DID (MSN) to appear (like 0031201234567) I could ofcourse set callerid to the main phonenumber but surely there must be a better solution? Thanks!! Remco

I am trying to use the simple CID name management script on the wiki. http://www.voip-info.org/wiki-Asterisk+tips+managing+CID+names I can not see what is wrong. The values never get entered in the database. Here are the files: I have asterisk running as the user asterisk also. ---cid-store.php---- <HTML> <HEAD> <TITLE>Storing Asterisk CID data</TITLE> </HEAD>

... 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: