[asterisk-users] Problem compiling Zaptel 1.4.5.1
Hi. Please help. When trying to compile Zaptel 1.4.5.1 I get the following: /build/include/linux/modversions.h -DSTANDALONE_ZAPATA -I.. -o base.o -c base.c base.c:48:29: linux/workqueue.h: No such file or directory base.c:292: warning: `vpm150m_firmware' defined but not used make[2]: *** [base.o] Error 1 make[2]: Leaving directory `/usr/src/zaptel-1.4.5.1/wctdm24xxp' make[1]: *** [wctdm24xxp/wctdm24xxp.o] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4.5.1' make: *** [all] Error 2 Can anybody help me with this? I run make distclean, configure and then make. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OriginateResponse 'reason' property.
Hi all. I'm trying to determine the reason for call failure (busy, no answer, no such number, etc...). Calls are made via the Manager API using the Originate manager command. Originally I thought that the 'reason' property within the OriginateResponse could be used for this purpose, but with Asterisk 1.2.* versions the reason always returned a '1' for all types of failures (busy, no answer, no such number, etc...) and a '4' in the event of success/call was answered. I have asked around on this mailing list about this issue before and got a reply that the reason code is a bit of mess and that it is set according to the last communication frame read from the originated channel. I recently installed Asterisk 1.4.2 and noticed that its returning somewhat different reason values for different types of failure. From my limited tests (using only SIP channels) I got the following: 0 = No such extension / number. 1 = No answer. 4 = Call answered. 8 = Congested / unavailable. I can't seem to find any formal documentation on the 'reason' property and the values it can take on. Can somebody please tell me whether one can indeed use this 'reason' property to determine type of failure? If this is the case what values can it take on and what are their meanings? Thank you. Regards, Jan. ___ --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] chan_zap not compiling.
Hi all. chan_zap not compiling, yes yes I know this sounds trivial but here me out... This morning I decided to upgrade to Asterisk 1.4.2 and Zaptel 1.4.1. I successfully installed zaptel 1.4.1 and the card is picked up and correctly configured. ztcfg shows the following: [EMAIL PROTECTED] root]# ztcfg -vv Zaptel Version: 1.4.1 Echo Canceller: MG2 Configuration == Channel map: Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. And zttool shows no alarms. But when I compile Asterisk it does not compile chan_zap. The funny thing is that 'make menuselect' shows that one can compile it - [*] 16. chan_zap I load the drivers (modprobe zaptel and modprobe wctdm) and I run ./configure before running make. But still no chan_zap.so What am I doing wrong? Any suggestions? Regards, Jan. ___ --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] func_curl fails to compile, asterisk1.4
Hi. After successfully running ./configure I run make. When running make I get the following error: func_curl.c: In function `curl_internal': func_curl.c:95: `CURLOPT_NOSIGNAL' undeclared (first use in this function) func_curl.c:95: (Each undeclared identifier is reported only once func_curl.c:95: for each function it appears in.) make[1]: *** [func_curl.o] Error 1 make: *** [funcs] Error 2 This happens in the beta3 release as well as the HEAD revision in SVN. Has anybody else came across this problem? Please help. ___ --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] Re: func_curl fails to compile, asterisk1.4
I got it sorted by myself in the meantime. I had version 7.9.8 of CurlLib installed. I upgraded to 7.16.0 and everything compiled just fine. Why didn't the configure script check for this version dependency of CurlLib? ___ --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] Cannot find ptlib-config, installing 1.4-beta3
Hi When trying to install asterisk1.4-beta3 I get the following error when running ./configure: Cannot find ptlib-config - please install and try again What is this ptlib-config? Can't seem to find it on google. Where can I find it and how can I install it? Moreover do I really need it, can I force a bypass? I have successfully installed zaptel 1.4.0-beta2 and libpri 1.4.0-beta1 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] OriginateEvent reason codes.
As said by Moises the reason in the orginate events is not working in version 1.2.12.1. Does anyobdy know in what version it is working, preferably one later than 1.2.12.1 and not prior to 1.2.5? Is it working correctly in asterisk 1.4? ___ --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] Re: OriginateEvent reason codes.
As said by Moises the reason in the orginate events is not working in version 1.2.12.1. Does anyobdy know in what version it is working, preferably one later than 1.2.12.1 and not prior to 1.2.5? Is it working correctly in asterisk 1.4? ___ --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] OriginateEvent reason codes.
Hi. I'm making calls via the Manager OriginateAction. My action is set to be async and therefore I receive originiate events. Within the originate event that I receive there is a reason code. In the event of failure I need to dermine why the call failed (no pickup, rejected, no such number, circuit busy, ect) and inform the user with a meaningful message. I assume that one is suppose to determine the failure cause by interpreting the reason code. But the reason is always 1. If the callee does not pickup the reason is 1, if the callee rejects the call the reason is 1, if the number does not exist the reason is 1. If the call was successful the reason is 4. Is this correct behaviour? Am I doing something wrong? What are all the different reason codes? Where can I find a list that explains what all the different codes mean? Are they the same as the hangup causes? If the reason code is not meant to determine failure causes, how else can I determine this? Thanks. Regards, Jan. ___ --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] Re: Where is the PlayDTMF command?
http://bugs.digium.com/view.php?id=6682 Thanks I patch my installation with the patch on the above URL. It works fine now. Thanks Moises. ___ --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] Re: Where is the PlayDTMF command?
Benny Amorsen benny+usenet at amorsen.dk writes: JdT == Jan du Toit jan.du.toit at decisionworx.com writes: JdT PS: This reply will probably go under a new thread with the same JdT subject. I receive the digest mode of the mails on this list, and JdT replying to it breaks the thread. How can I avoid this in the JdT future? Thanks. Switch to a newsreader and use gmane.org... Thanks for the tip about using a newsreader. My problem of getting an error while executing the Manager PlayDTMF action still persist. Oct 6 09:31:06 WARNING[3449]: channel.c:1610 ast_waitfor_nandfds: Thread 294931 Blocking 'SIP/Jan-081ba140', already blocked by thread 360468 in procedure ast_waitfor_nandfds Can somebody please help me with this. ___ --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] Re: Where is the PlayDTMF command?
So I patch my asterisk (version 1.2.12.1) with the patch given by Moises. http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch Thanks Moises. When I type in show manager command PlayDTMF it is their. With the show manager commands it is not within the list containing all the commands. When I execute the manager PlayDTMF action, the manager response says DTMF successfully queued. I don't hear anything on the phone, when I look at the CLI I see the following warning message. Its produced everytime I execute the PlayDTMF action. Oct 6 09:31:06 WARNING[3449]: channel.c:1610 ast_waitfor_nandfds: Thread 294931 Blocking 'SIP/Jan-081ba140', already blocked by thread 360468 in procedure ast_waitfor_nandfds Am I doing something wrong? Is this a bug? Please help, I need this to work as soon as possible... Thanks for all the help. ___ --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] Re: Where is the PlayDTMF command?
So I patch my asterisk (version 1.2.12.1) with the patch given by Moises. http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch Thanks Moises. When I type in show manager command PlayDTMF it is their. With the show manager commands it is not within the list containing all the commands. When I execute the manager PlayDTMF action, the manager response says DTMF successfully queued. I don't hear anything on the phone, when I look at the CLI I see the following warning message. Its produced everytime I execute the PlayDTMF action. Oct 6 09:31:06 WARNING[3449]: channel.c:1610 ast_waitfor_nandfds: Thread 294931 Blocking 'SIP/Jan-081ba140', already blocked by thread 360468 in procedure ast_waitfor_nandfds Am I doing something wrong? Is this a bug? Please help, I need this to work as soon as possible... Thanks for all the help. PS: This reply will probably go under a new thread with the same subject. I receive the digest mode of the mails on this list, and replying to it breaks the thread. How can I avoid this in the future? Thanks. Regards, Jan. ___ --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] Where is the PlayDTMF command?
So I patch my asterisk (version 1.2.12.1) with the patch given by Moises. http://galileo.ivsol.net/play_dtmf-1.2.12.1.patch Thanks Moises. When I type in show manager command PlayDTMF it is their. With the show manager commands it is not within the list containing all the commands. When I execute the manager PlayDTMF action, the manager response says DTMF successfully queued. I don't hear anything on the phone, when I look at the CLI I see the following warning message. Its produced everytime I execute the PlayDTMF action. Oct 6 09:31:06 WARNING[3449]: channel.c:1610 ast_waitfor_nandfds: Thread 294931 Blocking 'SIP/Jan-081ba140', already blocked by thread 360468 in procedure ast_waitfor_nandfds Am I doing something wrong? Is this a bug? Please help, I need this to work as soon as possible... Thanks for all the help. PS: This reply will probably go under a new thread with the same subject. I receive the digest mode of the mails on this list, and replying to it breaks the thread. How can I avoid this in the future? Thanks. Regards, Jan. ___ --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] Where is the PlayDTMF command?
Hi all. I'm confused. On http://www.voip-info.org/wiki-Asterisk+manager+API it says that the PlayDTMF command is available since version 1.2.8. I upgraded to version 1.2.12.1 but I cant find it if I type in show manager commands there is no PlayDTMF command. According to resources on the internet this action links to the send dtmf application. I checked the source code under the apps folder and it is their! |apps/app_senddtmf.c |Is it not compiling? Why is this function not available to me? Please help. 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] Changing the recording dir of MeetMe recordings.
Hi.
I'm using the default recording file name for meetme recordings of
meetme-conf-rec-${CONFNO}-${UNIQUEID}
The recordings are created in /var/lib/asterisk/sounds.
I want to change this direcrory to /var/lib/asterisk/sounds/storagedrive.
Settig the ${MEETME_RECORDINGFILE} variable doesn't help. It uses the
correct directory name but then just create the file as
meetme-conf-rec--.wav as it doesn't know the values of CONFNO and
UNIQUEID at that stage.
Can somebody please help me to achieve this directory change.
Thank you very much.
Regards.
___
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[asterisk-users] Extension presedence.
Hi.
I have the following two extensions:
exten = _71405XXX,1,Dial(Zap/g1/${EXTEN:5}|20,tr)
exten = _71.,1,Dial(Zap/g2/${EXTEN:2}|20,tr)
I have an external application that generates dialstrings, it generates
the 71 prefix so that the call can go through the T1 cards. As you can
see the 71 is cut of when passing it to the cards itself (EXTEN:2).
On the T1 card we have 4 circuits. One for local (group one is
configured for that) and three long disctance (group 2,3 and 4).
When I dial a 405 number it goes to the second exetension, group 2. I
want it to go through the first group.
I thought that since the first extension is more specific it will have
presedence over the second general one.
Is this right? Am I doing something wrong?
Moreover, how can I define the long distance extension to use the first
available channel on group 2, 3 and 4. Currently it only uses the first
avaialable channel in group 2. If all 23 bank channels in group 2 are in
use then the call fails, I want it then to try group 3 and so on.
Please help.
Thank you very much. I appreciate it.
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[asterisk-users] Configuring meetme recording quality (8kHz to 32kHz or higher)
Hi. At the moment we record our meetme conference to wav format, after which we have a script in place converting the wav to mp3 format. I need it to be encoded with MPEG1-layer3, so that I can play it back via our application using the Java Media Framework, but the wave files are encoded to MPEG2.5-Layer3. This is because asterisk records the files at 8kHz. You can't encode a 8kHz wave file to MPEG1-layer 3, only to MPEG2.5-Layer 3. In order to encode a wave file to MPEG1-layer3 you need either 32, 44.1 or 48 kHz. So my question is can you/how configure asterisk to record at 32kHz and not 8kHz? Thanks for all the help. ___ --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] How to check if channel varaible have been set/not empty?
Hi. How can you check if a channel variable has been set in the dialplan? I have some logic in the dialplan that must do a ExecIF() if a specific channel variable has been set, otherwise it must do nothing. Thank you very much. ___ --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] MeetMe recordings in mp3 format.
Hi. We have lots of long conferences using the meetme application, resulting in the default recording format (wav) being way too BIG to work with. I have installed the astersik addon pack, in the hope of getting mp3 recording support. But it only gives mp3 playback support. How can I get the recordings to be in mp3 format? Moreover, what other recording formats gives me the benifit of small size and good sound quality of mp3? Thank you very much. Cheers. ___ --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
Re: [asterisk-users] MeetMe recordings in mp3 format.
We have lots of long conferences using the meetme application, resulting in the default recording format (wav) being way too BIG to work with. I have installed the astersik addon pack, in the hope of getting mp3 recording support. But it only gives mp3 playback support. How can I get the recordings to be in mp3 format? Moreover, what other recording formats gives me the benifit of small size and good sound quality of mp3? What most people do for this is they execute a shell script afterwards to convert it into the MP3 format. As for another format that will work for you I can't think of any. You can try to use lame. Perhaps set up a cron/shell script that finds and converts new .wav files to mp3. Thanks for the help. I see the Monitor application has a variable called MONITOR_EXEC, this variable contains a the name of a script which are executed everytime the monitor application exits. The script is usually used for audio editing/mixing/conversion. I thought problem solved, because in the name of consistency I was sure that the MeetMe application will have a MEETME_EXEC variable, which executes a script at the end of a meetme conference. But I couldn't seem to find it. As Pierre du Plessis pointed out one can have a cron/shell script that finds and converts new .wav files to mp3, but it feels a bit iffy to me. I want the conversion to take place the moment when the meetme finish, thus the end of the meetme must trigger the script. How will one go about doing this? Thanks a lot. Cheers. ___ --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] TDM01B -1 FXO card not working.
Hi. I'm trying to install and configure a TDM01B -1 FXO card. I'm getting the following errors when starting up asterisk: Jul 25 08:48:40 WARNING[1775]: chan_zap.c:923 zt_open: Unable to specify channel 1: No such device Jul 25 08:48:40 ERROR[1775]: chan_zap.c:6879 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Jul 25 08:48:40 ERROR[1775]: chan_zap.c:10311 setup_zap: Unable to register channel '1' Jul 25 08:48:40 WARNING[1775]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Jul 25 08:48:40 WARNING[1775]: loader.c:554 load_modules: Loading module chan_zap.so failed! My /etc/zaptel.conf looks as follows: fxsks=1 loadzone = us defaultzone=us The relevant stuff in my /etc/asterisk/zapata.conf looks as follows: signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=sgs channel = 1 (I have attached the whole zapata.conf file) The card is properly installed in the box and the wctdm driver loads. When I type in modprobe wctdm the driver loads and the light on the card goes on. Typing in ztcf -vv gives the following: Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Typing in cat /proc/zaptel/1 produces the following: Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 When I comment out the channel creation part of the zapata.conf asterisk loads fine and the CLI command "Zap show status" produces the following: localhost*CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 But, when I type in "cat /dev/zap/1" I get the following: cat: /dev/zap/1: No such device Should it show sometging else. The output given by "ztcf -vv" and "cat /proc/zaptel/1" seems fine to me, but I'm not an expert. Is it fine? Please, any help will be appreciated. Thank you. Jan. ; ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [trunkgroups] ; ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ;group = trunkgroup,dchannel[,backup1...] ; ;trunkgroup is the numerical trunk group to create ;dchannelis the zap channel which will have the ;d-channel for the trunk. ;backup1 is an optional list of backup d-channels. ; ;trunkgroup = 1,24,48 ;trunkgroup = 1,24 ; ; Spanmap: Associates a span with a trunk group ;spanmap = zapspan,trunkgroup[,logicalspan] ; ;zapspan is the zap span number to associate ;trunkgroup is the trunkgroup (specified above) for the mapping ;logicalspan is the logical span number within the trunk group to use. ;if unspecified, no logical span number is used. ; ;spanmap = 1,1,1 ;spanmap = 2,1,2 ;spanmap = 3,1,3 ;spanmap = 4,1,4 [channels] ; Default language ; ;language=en ; ; Default context ; context=sgs ; ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: ATT 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN ; ni1:Old National ISDN 1 ; qsig: Q.SIG ; switchtype=national ; ; Some switches (ATT especially) require network specific facility IE ; supported values are currently 'none', 'sdn', 'megacom', 'accunet' ; ;nsf=none ; ; PRI Dialplan: Only RARELY used for PRI. ; ; unknown:Unknown ; private:Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; ;pridialplan=national ; ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's numbering plan) ; ; unknown:Unknown ; private:Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; ;prilocaldialplan=national ; ; PRI callerid prefixes based on the given TON/NPI (dialplan) ; This is especially needed for euroisdn E1-PRIs ; ; sample 1 for Germany ;internationalprefix = 00 ;nationalprefix = 0 ;localprefix = 0711 ;privateprefix = 07115678 ;unknownprefix = ; ; sample 2 for Germany ;internationalprefix = + ;nationalprefix = +49 ;localprefix = +49711 ;privateprefix = +497115678 ;unknownprefix = ; ; PRI resetinterval: sets the time in seconds between restart of unused ; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like ; channel restarts. so set the interval to a very long interval e.g. 1 ; or 'never' to disable *entirely*. ; ;resetinterval = 3600 ; ; Overlap dialing mode (sending overlap digits) ; ;overlapdial=yes ; ; PRI Out of band indications. ;
[asterisk-users] TDM01B -1 FXO card not working.
Hi. I'm trying to install and configure a TDM01B -1 FXO card. I'm getting the following errors when starting up asterisk: Jul 25 08:48:40 WARNING[1775]: chan_zap.c:923 zt_open: Unable to specify channel 1: No such device Jul 25 08:48:40 ERROR[1775]: chan_zap.c:6879 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Jul 25 08:48:40 ERROR[1775]: chan_zap.c:10311 setup_zap: Unable to register channel '1' Jul 25 08:48:40 WARNING[1775]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Jul 25 08:48:40 WARNING[1775]: loader.c:554 load_modules: Loading module chan_zap.so failed! My /etc/zaptel.conf looks as follows: fxsks=1 loadzone = us defaultzone=us The relevant stuff in my /etc/asterisk/zapata.conf looks as follows: signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived group=1 context=sgs channel = 1 (I have attached the whole zapata.conf file) The card is properly installed in the box and the wctdm driver loads. When I type in modprobe wctdm the driver loads and the light on the card goes on. Typing in ztcf -vv gives the following: Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Typing in cat /proc/zaptel/1 produces the following: Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 When I comment out the channel creation part of the zapata.conf asterisk loads fine and the CLI command "Zap show status" produces the following: localhost*CLI zap show status Description Alarms IRQ bpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 But, when I type in "cat /dev/zap/1" I get the following: cat: /dev/zap/1: No such device Should it show sometging else. The output given by "ztcf -vv" and "cat /proc/zaptel/1" seems fine to me, but I'm not an expert. Is it fine? Please, any help will be appreciated. Thank you. Jan. ___ --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] Internet exposed asterisk server.
Hi. I have a soft phone (X-Lite) which registers with a asterisk server that can only be accessible once we have some virtual private network software up and running. With the above scenario everything works fine. In the mean time the asterisk server was exposed to the internet, thus the virtual private network software is no longer needed. But when I try and register it gives me the following: Registration from 'User sip:[EMAIL PROTECTED]' failed for 'yyy.yyy.yyy.yyy' - Wrong password Were xxx.xxx.xxx.xxx is the internal ip of the asterisk server, not the ip the external ip (the ip on the internet) and were yyy.yyy.yyy.yyy is my external ip address as seen on the internet. I trippled check all the authentication details. Why is it not working on the exposed server? Did I do something wrong? Is there special confugurations when exposing an asterisk server? Thanks in advance. ___ --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] Meetme volume increase/decrease
Hi. The UPGRADE.txt of asterisk distribution contains the following snippet under the MeetMe heading: "MeetMe: * The conference application now allows users to increase/decrease their speaking volume and listening volume (independently of each other and other users); the 'admin' and 'user' menus have changed, and new sound files are included with this release. However, if a user calling in over a Zaptel channel that does NOT have hardware DTMF detection increases their speaking volume, it is likely they will no longer be able to enter/exit the menu or make any further adjustments, as the software DTMF detector will not be able to recognize the DTMF coming from their device. " My question is... How do you increase/decrease your speaking/listening volume while in a meetme room? Thankxs. ___ --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] Problems loading res_odbc.so and cdr_odbc.so
Hi. I am having troubles loading the res_ and cdr_odbc modules, they fail because they cannot find libodbc.so.1 I have unixODBC properly installed and the needed DNS setup correctly. Any ideas why I am having this troubles? Where is asterisk looking for the libodbc.so.1 file? And were can I configure this path? Thanks in advance. ___ --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] How to install the cdr_odbc module.
Hi. I want to the CDR detail to be logged to a database backend. For that I need the cdr_odbc module. Executing the CLI command show modules I realized I don't have the cdr_odbc module loaded. So, I tried to load it using load cdr_odbc.so, but it didn't work because I don't have it. Where can I get this module and how do I install it? Thanks in advance. ___ --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] Missing meetme recordings.
Hi. I am recording conferences taking place via the meetme application by using the 'r' option. When I start the conference I get the message in the CLI : Starting recording of MeetMe Conference 8000 into file meetme-conf-rec-8000-1138265171.201.wav. No additional warnings or errors is displayed in the CLI during and after the conference. This tells me everything is fine. But I can't seem to find the recording. I have looked under /var/spool/asterisk/meetme but it is not there. Were is the recording stored? Am I doing something wrong? Thanks in advance. ___ --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] Mute a given channel.
Hi. Does asterisk support muting per a specific channel? (like the soft hangup command, were you specify a channel and then asterisks hangs it up). If it does not, how will one go about to do something like this? Thank you in advance. ___ --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] No sound problem, chan_sip.c:3451
Hi. I'm trying to hook up the SIP-Communicator softphone. When I dial their is no sound and on the asterisk box the following is displayed: chan_sip.c:3451 process_sdp: Unknown SDP media type in offer: video 2 RTP/AVP 26 34 31 How can I fix this problem? Thanks in advance. ___ --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
