similar to: function declaration isn't a prototype

>Message: 11 >Date: Sun, 28 Aug 2005 11:46:29 +0800 >From: "chris" <chris@fivestartel.com> >Subject: [Asterisk-Users] error compiling on solaris 10 >To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> >Message-ID: <003a01c5ab83$149a9e30$650fa8c0@acid> >Content-Type: text/plain;

Hello all, After reading through the Wiki and archives, I decided to take a stab at installing * on Solaris 9 SPARC. I checked it out via CVS last night as well as about an hour ago, and have run into a compile problem that I can't quite figure out. After running into some unlisted dependencies, such as popt, things are almost compiling. Right now the make bombs when trying to find setenv

----- Original Message ----- From: "chris" <chris@fivestartel.com> To: "Frank Tarczynski" <ftarz@mindspring.com>; <seruser@iptel.org> Sent: Tuesday, July 26, 2005 9:54 AM Subject: Re: [Asterisk-Users] /bin/sh: build_tools/make_version_h: not found > hi Tzafrir > > any idea how we can fix this error? > > include/asterisk/strings.h:232: parse

Hello! I have a problem with build astersik-addons-1.4.7 on Solaris 10. When I tried to do "make" I got such error: * chan_ooh323.c: In function `reload_config': chan_ooh323.c:2053: error: `IPTOS_MINCOST' undeclared (first use in this function) chan_ooh323.c:2053: error: (Each undeclared identifier is reported only once chan_ooh323.c:2053: error: for each function it appears

Hi All, I am trying to get Asterisk up and running on my new Mandrake 9.1 install. I've installed Linux in the "standard" mandrake security mode, and "su" to do my attempts at install. I managed to obtain the source from CVS, and have been able to compile Zaptel. I then ran insmod zaptel, and also make config. I think I have compiled and loaded Zaptel successfully as

Hi all , In my pbx ,there is no zaptel card ,so i loading ztdummy,but problem appear,when i dial the number into the pbx,sometimes i can not listen to the ivr ,and no rtp create. if i unload the ztdummy driver,the proble will disapper. I guess may be the timer source problem, but i dont't know how to solve it . anyone can give me some advices will be appreciated. asteirsk-1.4.21 and

Has anyone succesfully integrated * with a cisco voice gateway ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/3b1ba7b3/attachment.htm

Hi, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at

I did cvs update on asterisk, zaptel, libpri as of today (November 5, 2003). I also did 'make clean' on each of them. My previous version of asterisk was cvs of September 15, 2003. No other changes have been made to my system other that these updates. when running 'make asterisk' the following error appears term.c:55: conflicting types for `term_color'

1.4 was working fine. I thought I would try 1.6 beta 9 from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept the call. [Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite: Call from 'JJ' to extension 'jj_audio' rejected because extension not found. I changed nothing in the config files. I tried setting debug level to 5 and verbose to 5 all

hello i am trying to use res_odbc for sipuser. my connection is working. i have checked using isql. even cdr_odbc is working but i hav problem in res_odbc. i have created user in sip_buddies table but asterisk is no getting user from this sip_buddies table. /etc/asterisk/extconfig.conf [settings] sipusers=>odbc,asterisk,sip_buddies sippeers=>odbc,asterisk,sip_buddies

Question is quite easy: How am I supposed to configure Asteirsk to have the same extension, in 2 differents domains. In the general section of sip.conf, I add the domains, But how to say to Asterisk : user1@domain1 > Pasword1 user2@domain2 > Pasword2 Thanks for your help !!!!! JM -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on

Hi, can i set up my asterisk for: - receive a skinny call in a specific context (yes, i have already compiled asteirsk with h323 support) - forward the call to a sip user A - make the sip user B join the call and create a conference between skinny caller, A and B maky thanks

I have read the wiki and several other internet documents. Can anyone make a comment as to what kind of functionality will you loose if you use Cisco 7940 phones with asterisk 1.4 things like: MWI, call transfer, conference,etc,etc. I have a customer with 6 of those phones that he like to use with the asteirsk PBX. thanks, -- ------------------------------------------------------------ Erick

http://gizmodo.com/gadgets/cellphones/sprintsamsung-instant-cell+to+wi+fi-box-is-official-named-airave-300451.php The above is quite interesting, it would be interesting to see if it uses sip, which I have no reason to believe otherwise, and if it does, can it be hacked to talk to Asteirsk? In which case one could have a very good extension to asterisk using any Sprint Cell phone, or maybe even

I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? -- Thank You Amit Nepal

I have a TDM400P with 1 fxo and 1 fxs module(TDM11B) i just wanted to know how to configure the zaptel and zapata & extension conf to work well Please help me on that

-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/03/04-10:19:04\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\"

I still can't figure out why res_config_mysql module not showing up with many attempt. Anyone have any idea on this? checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... yes configure: creating ./config.status config.status: creating build_tools/menuselect-deps config.status: creating makeopts Sincerely, K -----Original Message----- From: KC