[Asterisk-Dev] IAX2 no compatible codecs
Hi All I have a strange problem using IAX2. When placing a call to my IAX clients (firefly) via the Asterisk dialplan all works great. However trying to initiate a call via the manager interface to the IAX client using the following command results in an error: Action: Originate Channel: IAX2/7000 Extension: 7000 Context: local Priority: 1 ActionID: 1 The error I get in the CLI is "Jun 17 08:18:36 WARNING[180236]: chan_iax2.c:4534 socket_read: Call rejected by #IP: No compatible Codecs" Does anyone have any ideas. Thanks in advance Jason ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Channel Status
Hi all, I've seen a problem with the CVS from June. Could someone please try this... You need a tdm400p + FXO. You need a SIP phone. - Make an outgoing call from SIP phone - While the call is active, do CLI> show channels For the last 2 weeks, when I do this, 'show channels' reports the status as 'Dialing'. -- Called g0/2442790 show channels Channel (ContextExtensionPri ) State Appl. Data Zap/3-1 (from-pstn s1 ) Dialing AppDial (Outgoing Line) SIP/204-64eb (from-internal 92442790 1 )Ring Dial ZAP/g0/2442790|| 2 active channel(s) If I go back to a May build, it properly reports the state as 'Up'. Can someone try to prove me wrong please? Thanks -- .. Ryan Courtnage Coalescent Systems 403.244.8089 www.voxbox.ca ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Modified Prepaid App Error
I am trying to install the Modified Prepaid App. I have installed PostgeSQL, created the tables, etc. Make Install runs ok. The when I try to launch asterisk (asterisk -vgc), it fails to run. I get the following errors, 1st error: [app_prepaid.so] => (Prepaid Application) == Parsing '/etc/asterisk/prepaid.conf': Found Jun 16 14:27:27 ERROR[-1085267840]: app_prepaid.c:127 check_connected: app_prepaid: cannot connect to database server localhost. Calls will not be logged == Registered application 'Prepaid' 2nd error: [app_datetime.so] => (Date and Time) Jun 16 14:27:28 WARNING[-1085267840]: pbx.c:2240 ast_register_application: Already have an application 'DateTime' Jun 16 14:27:28 WARNING[-1085267840]: loader.c:326 ast_load_resource: app_datetime.so: load_module failed, returning -1 == Unregistered application 'DateTime' Jun 16 14:27:28 WARNING[-1085267840]: loader.c:421 load_modules: Loading module app_datetime.so failed! I would highly appreciate any help. Thanks __ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] appl Dial variables behavior
Hi, I've modified app_dial to create some variables to get call duration and other stuff needed in some agi applications. But, when I exec the Dial command inside agi application, if the destination hung up the call, I am able to get channels variables (example. DIALEDPEERNUMBER, etc) however if the originator hung up the call the channels variables doesn't exists. I've tested ZAP and SIP channels. This is a normal behavior or a bug? Regards, Danilo Lotina F. Gerente Técnico NetGlobalis S.A <[EMAIL PROTECTED]> ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
RE: [Asterisk-Dev] cdr_odbc and SQLite
I still haven't found a good answer on this one... OH the joy of ODBC! :) bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-dev- > [EMAIL PROTECTED] On Behalf Of David Creemer > Sent: Wednesday, June 16, 2004 3:27 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Dev] cdr_odbc and SQLite > > Hi- > > I've been looking at cdr_odbc.c (lastest * CVS HEAD). I've tried > binding to a SQLite DB, with some errant behavior -- character string > do not make it into the database (but integers do). I installed > unixODBC from latest source, and have verified that I can connect to > and read/write to my data source using for example python-odbc. > > In looking at the cdr_odbc.c source, I notice that calls to > SQLBindParameter look like this: > > SQLBindParameter(ODBC_stmt, 2, SQL_PARAM_INPUT, SQL_C_CHAR, > SQL_CHAR, sizeof(cdr->clid), 0, cdr->clid, 0, NULL); > > I've looked at some of the sample ODBC drivers in the unixODBC package > and this doesn't seem correct. See the txt driver for example. > > The following seems to work with txt and SQLite backends: > > SQLLEN flen = 80; > SQLBindParameter(ODBC_stmt, 2, SQL_PARAM_INPUT, SQL_C_CHAR, > SQL_CHAR, sizeof(cdr->clid), 0, cdr->clid, sizeof(cdr->clid), &flen); > > > It seems that at least some ODBC drivers interpret the final field as > the actual length of the data, and the second to last field as the size > of the passed in field buffer. > > I also think that sizeof( X ) only works as intended above if X is > defined a char X[80], not if X is "char *X = (char*)malloc(80)". I > haven't checked all of the cdr-> fields to be sure they meet that > restriction. > > Is my analysis here correct? If so (sorry for newbie question) should I > just file a bug? > > regards, > -- David > > ___ > Asterisk-Dev mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] ast_data, mysql, md5secret
RE: checking for .conf file existence Good idea RE: username updating to "name" This is by design in the original chan_sip.c code, and is mirrored in ast_data. The username from the client is what is saved in the username field. Not sure why, but thats how it was in the past. Rob - Original Message - From: "Gunnar Schaller" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, June 16, 2004 5:58 AM Subject: Re[2]: [Asterisk-Dev] ast_data, mysql, md5secret > One more question: I saved a sip-user in the database and set name to > "" and username to "abc". When the client logs in asterisk makes > an update to the database and sets username to "". Why that? Any > reasons? > > Gunnar > > ___ > Asterisk-Dev mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] cdr_odbc and SQLite
Hi- I've been looking at cdr_odbc.c (lastest * CVS HEAD). I've tried binding to a SQLite DB, with some errant behavior -- character string do not make it into the database (but integers do). I installed unixODBC from latest source, and have verified that I can connect to and read/write to my data source using for example python-odbc. In looking at the cdr_odbc.c source, I notice that calls to SQLBindParameter look like this: SQLBindParameter(ODBC_stmt, 2, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->clid), 0, cdr->clid, 0, NULL); I've looked at some of the sample ODBC drivers in the unixODBC package and this doesn't seem correct. See the txt driver for example. The following seems to work with txt and SQLite backends: SQLLEN flen = 80; SQLBindParameter(ODBC_stmt, 2, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, sizeof(cdr->clid), 0, cdr->clid, sizeof(cdr->clid), &flen); It seems that at least some ODBC drivers interpret the final field as the actual length of the data, and the second to last field as the size of the passed in field buffer. I also think that sizeof( X ) only works as intended above if X is defined a char X[80], not if X is "char *X = (char*)malloc(80)". I haven't checked all of the cdr-> fields to be sure they meet that restriction. Is my analysis here correct? If so (sorry for newbie question) should I just file a bug? regards, -- David ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] RE: [Asterisk-Users] Simplified Voicemail app / keeping peace withcohabitants
Has anyone done any work on making the voicemail interface user-configurable (and/or does anyone want to)? It's something I've given some thought to and I have some ideas for and wouldn't mind working on, but I wouldn't want to step on anyone's toes and would like to have some people to collaborate with. On Fri, 11 Jun 2004, Jay Milk wrote: > You just put into words exactly what I've been contemplating over the > last few weeks. All voicemail systems have pretty much the same set of > functions, just different keys to access them. Wouldn't it be great if > we could configure the voicemail menus and prompts? If ever I find time > again (my 2-wk old son is my newest "project"), I'll take a look at the > sources and see what it would take. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Robert > Withrow > Sent: Friday, June 11, 2004 11:01 AM > To: Asterisk-users > Subject: Re: [Asterisk-Users] Simplified Voicemail app / keeping peace > withcohabitants > > > On Fri, 2004-06-11 at 11:54, Doug Kennedy wrote: > > > I have modified the VoiceMailMain application to satisfy the request > > of > > the "local users", i.e., my wife. > > Seems like the VoiceMail app should take a .conf file that is analogous > with the extensions.conf that would specify a voicmailplan instead of a > dialplan. ;-) > > ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] app_dial - Feature or bug ?
Hi, I have an Agi script that uses the app_dial to setup calls. After the call is terminated I do some clean up work updating records using information about the call from the CDR etc. All of this works well except when the called party hangs up. In this case my script exists before the CDR record is populated. Is this expected behaviour. Any comments ? suggesstions. Thanks Umar. ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Eicon diva server 2M problem
I am trying to configure an Eicon Diva Server 2M with Asterisk (I am a newbi!). I downloaded from melware.de the packages "divas_kernel24.tgz", "kernel24capi.tgz", "diva_isdn_make.diff" and "divactrl_2.1.tar.gz". I patched my kernel (2.4.26 downloaded from kernel.org) and compiled it. When i execute "make modules_install" I get this error (no errors before): > find kernel -path '*/pcmcia/*' -name '*.o' | xargs -i -r ln -sf ../{} pcmcia > if [ -r System.map ]; then /sbin/depmod -ae -F System.map 2.4.26-pbx; fi > depmod: *** Unresolved symbols in > /lib/modules/2.4.26-pbx/kernel/crypto/autoload.o > depmod: crypto_alg_lookup > depmod: *** Unresolved symbols in > /lib/modules/2.4.26-pbx/kernel/crypto/proc.o > depmod: crypto_alg_sem > depmod: crypto_alg_list > depmod: *** Unresolved symbols in > /lib/modules/2.4.26-pbx/kernel/drivers/isdn/eicon/diva_mnt.o > depmod: copy_to_user > depmod: copy_from_user Can anyone help me to understand why this happens? Regards. ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] RTSP channel
Title: RTSP channel We are developing a new * plugin to make it interoperable with RTSP Streamming servers like Darwin. In this way it would be possible, eg, to make any * video enabled terminal (like sip and h323) to access Real-time multimedia content. For that we are developing an * RTSP channel. We are having the following approach: 1- we have an application that handles the RTSP messages exchanged with RTSP server (eg Darwin) that will initialise an ast_channel struct (only with the values needed for the RTP/RTCP session) to link the terminal with RTSP server. 2- we will move the RTSP messages from the application to a new RTSP channel module and complete the ast_channel struct initialization. This way we may reuse any existing * application like Dial to redirect * extensions to RTSP addresses to have services like simple playvideo or new applications for more complex services like Video portals (to add video to existing voice portals) where we can associate dtmf commands to RTSP commands like pause/play. What do you think about this approach? Anyone interested to collaborate on this ? Regards, PCh & Neutel
Re[2]: [Asterisk-Dev] ast_data, mysql, md5secret
One more question: I saved a sip-user in the database and set name to "" and username to "abc". When the client logs in asterisk makes an update to the database and sets username to "". Why that? Any reasons? Gunnar ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re[2]: [Asterisk-Dev] ast_data, mysql, md5secret
Wow! Great work. It works fine on my asterisk. One comment on README.txt: When runnning it everytime my data.conf is overwritten. I'm forgetfully and so I don't comment it everytime out when redownloaded ast_data :o) Would be fine if you add something like if(!exists(/etc/asterisk/data.conf)) cp data.conf.sample ${ASTETC}/data.conf to the file. Thanks again for your help! Gunnar > Oddly enough, some code was in the modules for md5secret support, but I had > commented it out a while back because it was not made optional at the time. > It is now supported in all data_xxx.c modules as well as in the > chan_sip.c.patch.txt file. > Thanks for the hint Gunnar! > Rob > - Original Message - > From: "Gunnar Schaller" <[EMAIL PROTECTED]> > To: "Gunnar Schaller" <[EMAIL PROTECTED]> > Sent: Tuesday, June 15, 2004 1:45 PM > Subject: Re: [Asterisk-Dev] ast_data, mysql, md5secret >> Ups, just wanted to say "everything" works fine ... >> >> > I am running asterisk cvs version patched with newest ast_data. My >> > database is in MySQL, and there is my problem. Anything works fine, >> -- >> ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev