similar to: Asterisk Manager Commands - Timeout

Folks I am getting the following error as of today after updating both asterisk and asterisk-addons. These are both under /usr/src. Any ideas? dora-debian:/usr/local/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c

I'm trying to compile cdr_addon_mysql but keep getting this error. Again, searching the Wiki and the mailing list archive didn't bring up anything useful. Any help? Yes, I'm using MySQL 4.0. Maybe I have to switch back to 3.23? # make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names

Good Afternoon Everyone, I am having a problem with compiling the CVS version of *-addons downloaded today. I am also having problems compiling an older version as well but im ignoring that one for now. I believe I have all the correct libraries, and I have done extensive searches everywhere I just wondered if I was missing something really silly, or if this is a problem other people have

Thanks for the tip, but adding the CFLAGS directive doesn't work either, same error message. I'll try to have a look in -dev, but if anyone comes up with a solution, a reply would be appreciated. -Manuel -----Messaggio originale----- Da: Luckcuck Nick-LCKN001 [mailto:LCKN001@motorola.com] Inviato: lunedì, 21. giugno 2004 13:52 A: asterisk-users@lists.digium.com Oggetto: RE:

Dear All, I am originating the call directly to the SIP Provider using the maganger interface + originate (ASYNC) command. Here is the PHP-AGI Script. $call = $asm->send_request('Originate', array('Channel'=>"SIP/416XXXXXXX at ABC/n", 'Context'=>'ORIG',

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

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

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",

Hi Guys, I am trying to make a web form where a person is allowed to put in $phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof caller ID. There are a few problems that I am facing with Asterisk AMI Originate command. The reason why I want to use the darn AMI Originate is because I am sending calls to mobile phones and I want to have some accountability and to know if a call was

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 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

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>

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