[Asterisk-Users] Web interface
I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX call rejected.....who was trying to reach 's@'
dear All i signed up with an Aussie provider who gives me a DID in Aust... when I call my number I get the following on the console Mar 21 05:54:15 NOTICE[68071]: chan_iax2.c:6123 socket_read: Rejected connect at tempt from 203.13.163.245, who was trying to reach 's@' the s part i can understand by the @nothing ..?!? my iax.conf looks like this register = aa:[EMAIL PROTECTED] [alphanet] type=friend username= auth=plaintext ; ugh plaintext secret= host=proxy.freecall.net.au context=main disallow=all allow=ilbc my extensions.conf looks like this [default] exten = s,1,Answer exten = s,2,Dial(SIP/me,40,tr) also I have the same under [main] aswell anyone got any thoughts on this one its making me nuts Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] blind xfer works atxfer doesn't...help!
Hi all I am having problems with atxfer if I do the extact same thing with blind xfer it works fine when I hit press #2 (defined in conf for atxfer) i get transfer I dial the number I want and i get the following on the console -- Playing 'pbx-transfer' (language 'en') -- Executing Dial(Local/[EMAIL PROTECTED],2, /18005558355) in new stack Mar 15 02:49:15 WARNING[51497]: channel.c:1954 ast_request: No channel type registered for '' Mar 15 02:49:15 NOTICE[51497]: app_dial.c:936 dial_exec_full: Unable to create channel of type '' (cause 66) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Congestion(Local/[EMAIL PROTECTED],2, ) in new stack its missing the channel type etc...how can I make ths work eg IAX2/[EMAIL PROTECTED] etc i *think* I am running CVS-HEAD-03/07/05-08:27:11 Kind Thanks Jeremy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] looking for DID in spain
Dear all I am looking for a per minute DID # in spain..either IAX/SIP Thanks Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] vars for transfered calls
Hi when i try and transfer a call to another party the setvars I have placed in sip.conf do not get passed to my dialplan at all is there a work around for this so I can use them in my dialplan but when i place a normal outbound call from that device the vars get passwd fine I am passing an areacode - used by dial() and a cellnum used to forward the call when it is not answered can someone help Thanks Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PAP2-NA point to poitn calls ??...(Direct IP Dialing)
At 11:40 AM 3/11/2005, you wrote: the only way I found to do this was have them register with a * server and have * connect them Hello, I need to know if there is an option in the PAP2-NA Web Configurator like Enable IP dialing: yes/no I need to make point to point calls with two PAP2-NA by IP address (The PAP2-NA are in the same LAN, no Internet access). Is it possible ? Thank you !! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] diffrent area codes for diffrent phones in dialplan
I have 3 sets of SIP phones all in diff area codes that need to access the PSTN I need to it so that a 7 digit number is converted to a 10 digit with the correct ara code eg a call coming from sip-phone1 needs aera code AAA and a call coming fom sip-phone2 needs BBB how can this be setup in the dialplan is there someway to set a var on a per sip group basis? I thought of the accountcode...since i will not be using it for CDR thoughts Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] looking for cheap 4 port FXS card
Dear list I need to find a way to hook up 4 analog phones to * was wondering if anyone had old hardware not being used and they are willing to sell or know of any places to buy that are cheaper then digium :/ Thanks I need 2 ports but would perfer 4 let me know Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] looking for cheap 4 port FXS card
At 04:43 AM 3/8/2005, you wrote: it doesnt have to be a card it can be a device. long as it has 2-4 ports Jer Dear list I need to find a way to hook up 4 analog phones to * was wondering if anyone had old hardware not being used and they are willing to sell or know of any places to buy that are cheaper then digium :/ Thanks I need 2 ports but would perfer 4 let me know Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer questions
Dear all I am trying to work out how make call trasfer work the way I want is I am the called party I want to transfer a call so I press # and enter the ext but then it disconnects me this is a blind transfer how do I make it so its not a blind transfer so i can talk to the person before i transfer the call...and go backl to the orig caller if the transfered to ext doesnt answer also can the caller hear MOH while I am talking to person I am transfering the call to what would I need to do this just point me in the right direction and i'll go read some more... I using so far is T in dial() Thanks sorry for the noob question Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer questions
At 05:44 AM 3/7/2005, you wrote: Dear all I am trying to work out how make call trasfer work the way I want is I am the called party I want to transfer a call so I press # and enter the ext but then it disconnects me this is a blind transfer how do I make it so its not a blind transfer so i can talk to the person before i transfer the call...and go backl to the orig caller if the transfered to ext doesnt answer also can the caller hear MOH while I am talking to person I am transfering the call to what would I need to do this just point me in the right direction and i'll go read some more... I using so far is T in dial() Thanks sorry for the noob question also tried the following without luck [featuremap] blindxfer = #1; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 it still seems to want to accept only # as transfer I am running Asterisk CVS-v1-0-03/07/05-06:50:06 Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS compile error utils.c
Hi.. I get the following error when compiling the lastest CVS utils.c:405: undefined reference to `__use_ast_pthread_create_instead__' due to the fact I dont know c I thought what the heck and took a look at line 405 return pthread_create(thread, attr, start_routine, data); and changed it to ast_pthread_create(thread, attr, start_routine, data); and it compiledbut when running asterisk it gives me a nasty bus error and dies :/ now i'm feeling stupoidcan someone help??? :) Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cant compile app_meetme2
Dear all I am get the following problem when trying to compile app_meetme2 using mysql...it seems to want to use pgsql.? anyone my Makefile looks like app_meetme2.o: app_meetme2.c #$(CC) -pipe $(CFLAGS) -c -o app_meetme2.o app_meetme2.c $(CC) -pipe -I/usr/local/include/mysql -L/usr/local/lib/mysql $(CFLAGS) -c -o app_meetme2.o app_meetme2.c app_meetme2.so: app_meetme2.o $(CC) $(SOLINK) -o $@ $ -lpq -I/usr/local/include/mysql -L/usr/local/li b/mysql -lmysqlclient # $(CC) $(SOLINK) -o $@ $ -lgdbm app_meetme2.o app_meetme2.c app_meetme2.c: In function `launch_query': app_meetme2.c:138: error: `PGconn' undeclared (first use in this function) app_meetme2.c:138: error: (Each undeclared identifier is reported only once app_meetme2.c:138: error: for each function it appears in.) app_meetme2.c:138: error: `conn' undeclared (first use in this function) app_meetme2.c:139: error: `PGresult' undeclared (first use in this function) app_meetme2.c:139: error: `res' undeclared (first use in this function) app_meetme2.c:151: warning: implicit declaration of function `PQsetdbLogin' app_meetme2.c:152: warning: implicit declaration of function `PQstatus' app_meetme2.c:152: error: `CONNECTION_BAD' undeclared (first use in this functio n) app_meetme2.c:154: warning: implicit declaration of function `PQerrorMessage' app_meetme2.c:154: warning: format argument is not a pointer (arg 6) app_meetme2.c:155: warning: implicit declaration of function `PQfinish' app_meetme2.c:162: warning: implicit declaration of function `PQexec' app_meetme2.c:163: warning: implicit declaration of function `PQresultStatus' app_meetme2.c:163: error: `PGRES_TUPLES_OK' undeclared (first use in this functi on) app_meetme2.c:166: warning: implicit declaration of function `PQclear' app_meetme2.c:171: warning: implicit declaration of function `PQntuples' app_meetme2.c:179: warning: implicit declaration of function `PQgetvalue' app_meetme2.c:179: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:180: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:181: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:182: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:183: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:184: warning: passing arg 1 of `atoi' makes pointer from integer w ithout a cast app_meetme2.c:185: warning: passing arg 2 of `strcpy' makes pointer from integer without a cast app_meetme2.c: In function `launch_query_onefield': app_meetme2.c:254: error: `PGconn' undeclared (first use in this function) app_meetme2.c:254: error: `conn' undeclared (first use in this function) app_meetme2.c:255: error: `PGresult' undeclared (first use in this function) app_meetme2.c:255: error: `res' undeclared (first use in this function) app_meetme2.c:268: error: `CONNECTION_BAD' undeclared (first use in this functio n) app_meetme2.c:270: warning: format argument is not a pointer (arg 6) app_meetme2.c:277: error: `PGRES_COMMAND_OK' undeclared (first use in this funct ion) app_meetme2.c:295: error: `PGRES_TUPLES_OK' undeclared (first use in this functi on) app_meetme2.c:296: warning: format argument is not a pointer (arg 6) app_meetme2.c:302: warning: passing arg 1 of `strlen' makes pointer from integer without a cast app_meetme2.c:307: warning: format argument is not a pointer (arg 3) app_meetme2.c:331: warning: passing arg 1 of `strlen' makes pointer from integer without a cast app_meetme2.c:336: warning: format argument is not a pointer (arg 3) app_meetme2.c: At top level: app_meetme2.c:645: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_I NITIALIZER__' undeclared here (not in a function) app_meetme2.c: In function `count_exec': app_meetme2.c:1547: error: too few arguments to function `ast_say_number' gmake[1]: *** [app_meetme2.o] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cant compile app_meetme2
At 04:34 AM 3/5/2005, you wrote: The error messages are Postgres related. You need to have a special postgres include file (postgres-dev files) to make it compile or disable postpres support somehow. I'm using debian and the the concering include file resided in a subdirectory of what asterisk was told. if this is the case why dont i see a include file missing error someplace? or am I missing something.. Jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iconecthere and *
Hi all I am trying to figuure out how to get iconnecthere incoming calls to work outbound works fine but incoming goes nowhere but to my iconnecthere vocemail if I do a sip show registry it shows up as regg'ed nnn=is my iconnect here number xxx is my secret Thank you Jeremy [general] qualify=no register=NNN:[EMAIL PROTECTED]/N context=default bind = 0.0.0.0 port=5060 bindaddr=192.168.215.5 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=no videosupport=no disallow=all; First disallow all codecs allow=ulaw relaxdtmf=yes nat=yes ; NAT settings externip = 24.172.122.XXX localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks [iconnecthere] type=friend secret= username= I epxect this to be answered by the s ext in the dfault content which is the demo setup a this point or am I missing something all that happens if * never even sees the call (or more to the point i dont see it with sip debug!) THanks Jer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_modem dialout
can a voice modem make an outbound call? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some (lack of) answers regarding the wakeup call application...
At 11:27 AM 6/1/2004, you wrote: Rob I would be very interested Since I only seem to get questions, and no feedback, from the Wiki page, I'll ask here. There seems to be no lack of opinions here... I have a working wakeup call system on my home * system. The architecture is something I'm not perfectly happy with, though. There are two AGI scripts, written in Perl, which handle (a) scheduling, confirming, and cancelling a wakeup call, and (b) the wakeup call itself, with the option to snooze for 5, 15, or 30 minutes. The Perl scripts use the Asterisk::AGI module I came across months ago, but by necessity, can't use the Asterisk/Perl code for creating call files -- it has a habit of creating them right in the outgoing call queue, generating a call immediately. So the Perl code creates call files in a wakeup queue directory, and a cron job (a shell script) runs every minute looking for wakeup calls in the queue that need to be handled, and moves them to the outgoing call queue. It has occurred to me that the two AGI scripts could be rewritten as real compiled asterisk applications, but then it always hits me that without some kind of cron-line built-in scheduler, or changes to the outgoing call queueing that would allow a call to be scheduled for the future, there would still be that external cron-driven shell script. Ugly. What I'm wondering is this: Is there enough interest in the new features I mentioned (either an internal scheduler or scheduled outgoing calls) that I should work on a C version of the wakeup AGI scripts, or should my (impending) next rewrite maintain the current architecture? Anyone with specific questions about using my wakeup app, please email me directly. Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. The superior pilot uses his superior talent and superior judgment to avoid getting into situations where he needs to use his superior skill. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STREAM FILE question
Dear all I was wondering is there a way to advance/rewind in playback?(STREAM FILE) say 5 seconds somehow i don't think so but I thought I' would ask Thanks Jer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI/php script not working
Dear all I am just getting started with AGI so I wrote the following script as a simple test but all that happens is silence before it times out and hangs up can someone help to get me started? yet if i use the agi-test.agi script everything works I don't see the difference Thanks php -q ?php fputs(STDOUT 'SAY NUMBER 123 #*\n'); $lin = fgets(STDIN); ? yet all I get on the console is -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script test.php completed, returning 0 my conf file looks like exten = 4000,1,Wait,1 ; Wait exten = 4000,2,Answer ; Answer exten = 4000,3,AGI,test.php ; run script ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI/php script not working
At 08:47 AM 5/20/2004, you wrote: -vvvc mode I see *CLI -- Executing Wait(Phone/phone0, 1) in new stack -- Executing Answer(Phone/phone0, ) in new stack -- Executing AGI(Phone/phone0, test.php) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script test.php completed, returning 0 my question is why don't I see the output on stderr OK, but I have AGI working and you don't - so please allow me the error since it's a while since I worked on this, Of course, it would help if * used consistent syntax for identical commands in extensions.conf and AGI, but that's another debate. Why not check the logs for php and * and post anything relevant here. Enable the maximum debugging support in *. Iain --On Thursday, May 20, 2004 2:44 pm +0300 Apollon Koutlides [EMAIL PROTECTED] wrote: Iain Stevenson wrote: 'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber followed by a valid string of arguments. Do a show application saynumber in *. In the meantime, you might as well try a show agi yourself :-) Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phone_read Resource temporarily unavailable
Could someone tell me why I would get this using a linejack POTS May 20 17:43:05 WARNING[1217602880]: chan_phone.c:417 phone_read: Error reading: Resource temporarily unavailable May 20 17:43:05 WARNING[1217602880]: app_festival.c:181 send_waveform_to_channel : Null frame == hangup() detected May 20 17:43:05 WARNING[1217602880]: file.c:521 ast_readaudio_callback: Failed tto write frame Thankyou ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk not answering phone
Dear list.. I am trying to use * to answer a call coming in from the PSTN port of a line jack I am using mode=fxo in phone.conf but the line just rings and gins in mode.=dialtone it works fine on the POTS port any ideas what I am doing wrong? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linejack dialout
Dear all I read on the list back in 2003 that * does not support IXJ LineJACK dialout yet is this still the case? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems compiling h323 support
Hi all I am trying to compile Asterisk on RH9 I am have installed pwlib_1.5.2 readline-4.3-5 readline-devel-4.3-5 openssl-devel-0.9.7a-2 openssl-0.9.7a-2 openh323_1.12.2 I read that it has to do with pwlib not being installed correctly so i made sure it was in /usr/local correctly When i try to compile i get the follow errors I was wondering can anyone shed in light on this issue Thanks g++ -Wall -DPNXVER2=2 -DP_LINUX -mcpu=i686 -D_REENTRANT -DP_HAS_SEMAPHORES -O3 - DNDEBUG -DP_SSL -I../include -I/include -I/crypto -DPNX_VERSION=\CVS-05/17/04-1 7:10:07\ -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -I/include/ptlib/unix -I/inc lude -I/home/dictate/openh323/include -DHAS_IXJ -DHAS_IXJ -DHAS_OSS -c chan_h323 .cxx -o chan_h323.o In file included from /usr/include/ptlib/contain.h:222, from /usr/include/ptlib.h:139, from chan_h323.cxx:37: /usr/include/ptlib/object.h:585: parse error before `(' token /usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field /usr/include/ptlib/object.h:1201: parse error before `(' token /usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field /usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL' /usr/include/ptlib/object.h:1201: conflicts with previous declaration `int [ snip ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] problems compiling h323 support
At 06:23 PM 5/17/2004, you wrote: gives the same error... g++ -Wall -DPNXVER2=2 -DP_LINUX -mcpu=i686 -D_REENTRANT -DP_HAS_SEMAPHORES -O3 - DNDEBUG -DP_SSL -I../include -I/include -I/crypto -DPNX_VERSION=\CVS-05/17/04-1 7:10:07\ -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -I/include/ptlib/unix -I/inc lude -I/root/openh323/include -DHAS_IXJ -DHAS_IXJ -DHAS_OSS -c chan_h323.cxx -o chan_h323.o In file included from /usr/include/ptlib/contain.h:222, from /usr/include/ptlib.h:139, from chan_h323.cxx:37: /usr/include/ptlib/object.h:585: parse error before `(' token /usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field /usr/include/ptlib/object.h:1201: parse error before `(' token /usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field /usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL' /usr/include/ptlib/object.h:1201: conflicts with previous declaration `int Try 'export LANG=C' then 'make clean make' -- Matthew Billings | Affordable WWW Internet Solutions foreThought.net | for Small Business [EMAIL PROTECTED] | 910 16th Street, #1220 (303) 228-0070 x821 --The Future is Now!--| Denver, CO 80202(303) 228-0077 fax -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jer Sent: Monday, May 17, 2004 4:07 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] problems compiling h323 support Hi all I am trying to compile Asterisk on RH9 I am have installed pwlib_1.5.2 readline-4.3-5 readline-devel-4.3-5 openssl-devel-0.9.7a-2 openssl-0.9.7a-2 openh323_1.12.2 I read that it has to do with pwlib not being installed correctly so i made sure it was in /usr/local correctly When i try to compile i get the follow errors I was wondering can anyone shed in light on this issue Thanks g++ -Wall -DPNXVER2=2 -DP_LINUX -mcpu=i686 -D_REENTRANT -DP_HAS_SEMAPHORES -O3 - DNDEBUG -DP_SSL -I../include -I/include -I/crypto -DPNX_VERSION=\CVS-05/17/04-1 7:10:07\ -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -I/include/ptlib/unix -I/inc lude -I/home/dictate/openh323/include -DHAS_IXJ -DHAS_IXJ -DHAS_OSS -c chan_h323 .cxx -o chan_h323.o In file included from /usr/include/ptlib/contain.h:222, from /usr/include/ptlib.h:139, from chan_h323.cxx:37: /usr/include/ptlib/object.h:585: parse error before `(' token /usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field /usr/include/ptlib/object.h:1201: parse error before `(' token /usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field /usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL' /usr/include/ptlib/object.h:1201: conflicts with previous declaration `int [ snip ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] failed compile
Hi all I am trying to compile Asterisk on RH9 I am have installed pwlib_1.5.2 readline-4.3-5 readline-devel-4.3-5 openssl-devel-0.9.7a-2 openssl-0.9.7a-2 openh323_1.12.2 I read that it has to do with pwlib not being installed correctly so i made sure it was in /usr/local correctly When i try to compile i get the follow errors I was wondering can anyone shed in light on this issue Thanks g++ -Wall -DPNXVER2=2 -DP_LINUX -mcpu=i686 -D_REENTRANT -DP_HAS_SEMAPHORES -O3 - DNDEBUG -DP_SSL -I../include -I/include -I/crypto -DPNX_VERSION=\CVS-05/17/04-1 7:10:07\ -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -I/include/ptlib/unix -I/inc lude -I/home/dictate/openh323/include -DHAS_IXJ -DHAS_IXJ -DHAS_OSS -c chan_h323 .cxx -o chan_h323.o In file included from /usr/include/ptlib/contain.h:222, from /usr/include/ptlib.h:139, from chan_h323.cxx:37: /usr/include/ptlib/object.h:585: parse error before `(' token /usr/include/ptlib/object.h:1201: `BOOL' declared as a `virtual' field /usr/include/ptlib/object.h:1201: parse error before `(' token /usr/include/ptlib/object.h:1214: `BOOL' declared as a `virtual' field /usr/include/ptlib/object.h:1214: declaration of `int PObject::BOOL' /usr/include/ptlib/object.h:1201: conflicts with previous declaration `int [ snip ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users