search for: ast_max_extension

..., but the default is 0 to be compatible with previous versions. */ @@ -1106,10 +1103,6 @@ static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */ static char global_regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */ static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */ - -static char global_rtpip[AST_MAX_EXTENSION]; - - static char global_sdpsession[AST_MAX_EXTENSION]; /*!< SDP session name for the SIP channel */ static char global_sdpowner[AST_MAX_EXTENSION]; /*!< SDP owner name for the SIP channel */ static in...

...+ char announcemsg[256] = "", *ann; struct localuser *outgoing=NULL, *tmp; struct ast_channel *peer; int to; @@ -344,8 +345,10 @@ int allowredir_out=0; int allowdisconnect=0; int privacy=0; + int announce=0; int resetcdr=0; int clearchannel=0; + int cnt=0; char numsubst[AST_MAX_EXTENSION]; char restofit[AST_MAX_EXTENSION]; char *transfer = NULL; @@ -419,6 +422,16 @@ } else if (strchr(transfer, 'C')) { resetcdr = 1; } + /* XXX ANNOUNCE SUPPORT */ + else if ((ann = strstr(transfer, "A("))) { + announce = 1; + strncpy(announcemsg, ann + 2, sizeof(...

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

...... just seems to work ;) Matteo. -------------- next part -------------- --- asterisk/res/res_parking.c 2003-07-02 16:06:12.000000000 +0200 +++ myasterisk/res/res_parking.c 2003-07-03 23:28:52.000000000 +0200 @@ -49,6 +49,7 @@ /* Extension you type to park the call */ static char parking_ext[AST_MAX_EXTENSION] = "700"; +static char parking_pick[AST_MAX_EXTENSION] = "750"; static char pickup_ext[AST_MAX_EXTENSION] = "*8"; @@ -507,7 +508,23 @@ park = atoi((char *)data); ast_pthread_mutex_lock(&parking_lock); pu = parkinglot; - while(pu) { + if (park == atoi((ch...

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I need to call 20+ devices at the same time so dial strings are very long I cant really use a queue(ringall) which was my original idea as the customer needs different groups for virtually every call so...

I'm getting this message when I start Asterisk chan_capi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81 but when I try and recompile I get this chan_capi.c:60: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) any help would be greatly appreciated. -- Dave Cotton <dcotton@linuxautrement.com>

hey guys I thought I was making progress on my dialplan when I realized that the class features that are available for zap channels aren't available for SIP channels. I see references in the archives to adding pattern matches in the dialplan for CLASS features which has raised a couple questions. 1. Is implementing CLASS like features via the dialplan the currently recommended way to do

...all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I need to call 20+ devices at the same time so dial strings are very long I cant really use a queue(ringall) which was my original idea as the customer needs different groups for virtually every call so...

...lease notice for 13.33.0 received today - can anyone > advise > me the max limit of the string to the Dial Command - see > * [ASTERISK-27946 > <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - > dial (API): Storage of dialed target uses AST_MAX_EXTENSION > when it shouldn't > > I have been fighting with this issue for months trying to find a solution I > need to call 20+ devices at the same time so dial strings are very long I > cant really use a queue(ringall) which was my original idea as the customer > needs different group...

...STERISK-27946&amp;data=02%7C01%7Cf.floimair%40commend.com%7C5a5c413f7d8747dab6c408d7eda07497%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637239145718403259&amp;sdata=JdT9Yvi7ml%2FqzIYMO39ks68rdMKY2P2DFIAGKCCh6a8%3D&amp;reserved=0> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I need to call 20+ devices at the same time so dial strings are very long I cant really use a queue(ringall) which was my original idea as the customer needs different groups for vir...

Hi, I've been looking at the problem of the default caller ID. When a call comes in with no CID or witheld it's always set to 'asterisk' which is what the phone displays. I've been looking for an option to change that. The only place I can find is DEFAULT_CALLERID in chan_sip.c. This is set by the 'callerid' option in the sip.conf. However the documentation

Hi, I've configured my exten.conf for few exten. But I'm curious to know how long can be my exten like (exten => XXXXXXX.....). Is there any limit for this or not. B'coz I've noticed one strange problem. I'm usnig snom300 as my hard phone to make calls. when my exten length is 14 then calls goes immed. without any problem but when I change length from 14 to 15 call goes but

Hi, I'm having trouble dialling the codes for blacklist & un-hide caller ID (*80 & *82) using a TDM400. I get reorder tone as soon as I dial *8, and in fact I can pick up calls with just *8 rather than *8# when there is one to be picked up. Ideas? Thanks, brad -- Brad Bergman <bradley@bergman.ca>

Since www.bkw.org seems not to exist anymore (getting response from some hosting provider), does anyone happend to have a copy of app_valetparking.c from www.bkw.org - the one that should work with * stable 1.0.X ? If so please contact me. One that can be downloaded from www.loligo.com dosn't compile with 1.0.X, and SuperValletParking (www.asterlink.com/svp/) seems to be for * HEAD

Nope ! This is the one that tries to include PRE 1.0.X header file <parking.h>. It cannot compile on * 1.0.X (I have tried also to include <features.h> instead of <parking.h> (as far as I know features.h is successor to parking.h), but still without results). Thanks anyway. Nenad > > Try this > >> Since www.bkw.org seems not to exist anymore (getting

...n this release: Improvements made in this release: ----------------------------------- * ASTERISK-28813 - func_volume: Allow decimal numbers as parameter to improve granularity (Reported by Jean Aunis - Prescom) * ASTERISK-27946 - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't (Reported by Joshua Elson) * ASTERISK-28782 - Add support for Content-Disposition header in multi-part INVITES (Reported by Torrey Searle) Bugs fixed in this release: ----------------------------------- * ASTERISK-28852 - Unprotected access to nochec...

...n this release: Improvements made in this release: ----------------------------------- * ASTERISK-28813 - func_volume: Allow decimal numbers as parameter to improve granularity (Reported by Jean Aunis - Prescom) * ASTERISK-27946 - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't (Reported by Joshua Elson) * ASTERISK-28782 - Add support for Content-Disposition header in multi-part INVITES (Reported by Torrey Searle) Bugs fixed in this release: ----------------------------------- * ASTERISK-28852 - Unprotected access to nochec...

...-- * ASTERISK-28853 - Missing include on FreeBSD (Reported by Guido Falsi) * ASTERISK-28813 - func_volume: Allow decimal numbers as parameter to improve granularity (Reported by Jean Aunis - Prescom) * ASTERISK-27946 - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't (Reported by Joshua Elson) * ASTERISK-28782 - Add support for Content-Disposition header in multi-part INVITES (Reported by Torrey Searle) * ASTERISK-28787 - res_pjsip_session: Decide more intelligently when to add video (Reported by Joshu...

...-- * ASTERISK-28853 - Missing include on FreeBSD (Reported by Guido Falsi) * ASTERISK-28813 - func_volume: Allow decimal numbers as parameter to improve granularity (Reported by Jean Aunis - Prescom) * ASTERISK-27946 - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't (Reported by Joshua Elson) * ASTERISK-28782 - Add support for Content-Disposition header in multi-part INVITES (Reported by Torrey Searle) * ASTERISK-28787 - res_pjsip_session: Decide more intelligently when to add video (Reported by Joshu...

...-- * ASTERISK-28853 - Missing include on FreeBSD (Reported by Guido Falsi) * ASTERISK-28813 - func_volume: Allow decimal numbers as parameter to improve granularity (Reported by Jean Aunis - Prescom) * ASTERISK-27946 - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't (Reported by Joshua Elson) * ASTERISK-28782 - Add support for Content-Disposition header in multi-part INVITES (Reported by Torrey Searle) * ASTERISK-28787 - res_pjsip_session: Decide more intelligently when to add video (Reported by Joshu...