[Asterisk-Users] Unexpected hangups when calling Dialogic D/41JTC-LS
Has anyone tried attaching calling a Dialogic D/41JTC-LS (analog) device on another system from an asterisk system with TDM10B? Calling to asterisk from the outside, asterisk correctly dials the internal line and makes the connection to the Dialogic system. A few seconds later Asterisk debug info says it had an On Hook event and hangs up Zap-2-1. I have worked on this problem for over a week--no luck so far. Rollin Weeks ___ 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] Unexpected hangups when calling Dialogic D/41JTC-LS
Thanks Eric, I tried the changes to zapata.conf. I still get the hangup. It makes me wonder if the Dialogic card is sending a hangup tone to the FXO module. It seems to work OK if I use an analog phone instead of linking to the Dialogic card. RollinOn 8/19/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Rollin Weeks wrote: Has anyone tried attaching calling a Dialogic D/41JTC-LS (analog) device on another system from an asterisk system with TDM10B? Calling to asterisk from the outside, asterisk correctly dials the internal line and makes the connection to the Dialogic system. A few seconds later Asterisk debug info says it had an On Hook event and hangs up Zap-2-1.set busydetect=no and callprogress=no___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://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
Re: [Asterisk-Users] Realtime + MYSQL
Damon, You may be querying the wrong table, because the following fields in your Select statement do not exit in the table, voicemail_users, that you created: category, var_name, var_val, cat_metric, filename, commented Every item mentioned in a Select query must exist in the table that is being queried. Rollin Weeks On 8/10/05, Damon Estep [EMAIL PROTECTED] wrote: I'm having a few issues with the MySQL realtime configuration in CVS-HEAD. I tested it initially with realtime extensions (realtime_ext = mysql,asterisk,extensions) and a realtime switch in extensions.conf and that works fine, So I though I'd go back and test a static configuration mapping. I used the table structure from the asterisk guru postgres howto to create something similar in MySQL (shown below) and included the following in extconfig; voicemail.conf = mysql,asterisk,voicemail_users The result is that app_voicemail fails to load and it appears from the debug that it is not happy with the table structure... however the names it has for the fields seem strange (to me that is :)) If anyone has gone through the process of creating the correct tablesin MySQL and doesn't mind sharing I would be most appreciative. Regards, Nathan. MySQL Table CREATE TABLE voicemail_users ( id int NOT NULL auto_increment, customer_id varchar(255) NOT NULL default '0', context varchar(255) NOT NULL default '', mailbox varchar(255) NOT NULL default '', password varchar(4) NOT NULL default '0', fullname varchar(50) NOT NULL default '', email varchar(50) NOT NULL default '', pager varchar(50) NOT NULL default '', stamp datetime NOT NULL default '-00-00 00:00:00', PRIMARY KEY(`id`) ); ### res_mysql.conf [general] dbhost = localhost dbname = asterisk dbuser = asterisk dbpass = dbport = 3306 dbsock = /var/run/mysqld/mysqld.sock Debug Log Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Static SQL: SELECT category, var_name, var_val, cat_metric FROMvoicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Everything is fine. Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Query: SELECT category, var_name, var_val, cat_metric FROM voicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id Aug 11 01:16:23 DEBUG[1028] res_config_mysql.c: MySQL RealTime: Query Failed because: Unknown column 'category' in 'field list' ___ This works for voicemail in CVS-HEADCREATE TABLE `voicemail` (`uniqueid` int(11) NOT NULL auto_increment,`customer_id` int(11) NOT NULL default '0',`context` varchar(50) NOT NULL default '',`mailbox` varchar(10) NOT NULL default '0', `password` varchar(4) NOT NULL default '0',`fullname` varchar(50) NOT NULL default '',`email` varchar(50) NOT NULL default '',`pager` varchar(50) NOT NULL default '',`stamp` timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP,PRIMARY KEY(`uniqueid`),KEY `mailbox_context` (`mailbox`,`context`)) ENGINE=MyISAM DEFAULT CHARSET=latin1;___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://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] Unexpected On Hook event
I have an asterisk configuration in which the external analog line dials in through an FXO. The dial plan has Zap1-1 dial Zap2-1 after playing the congratulations message. A few seconds after Zap2 answers and begins to send audio messages across the line, an On Hook(1) event for Zap2-1 seems to occur out of the blue, and the connection goes away. How could this happen? Rollin Weeks ___ 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] error compiling asterisk on solaris
Chris, The problem is that your compiler can't find a library called libcrypt.so.0.9.7. This library is apparently needed by libssl.so. These are both runtime, shared libraries. The result is that you end up with undefined symbols (probably variables used in services the libraries provide). You need to find the encryption library for Solaris 9. Rollin WeeksOn 8/9/05, chris [EMAIL PROTECTED] wrote: hello, can anyone help me? im gettitng this error when i tried runnin make on solaris 9 rm -f include/asterisk/version.h.tmpmake[1]: `ast_expr.a' is up to date.make[1]: Leaving directory `/export/home/fst/chris/cvs/asterisk'gcc -g -o asterisk io.o sched.o logger.o frame.o loader.o config.o channel.o t ranslate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmod em.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt. o aestab.o aeskey.o utils.o config_old.o plc.o jitterbuf.o dnsmgr.o devicestate. o netsock.o slinfactory.o strcompat.o ast_expr.a editline/libedit.a db1-ast/libd b1.a stdtime/libtime.a -lncurses -lm -lpthread -ldl -lnsl -lsocket -lresolv -L/u sr/local/ssl/lib -lssl/usr/local/sparc-sun-solaris2.8/bin/ld: warning: libcrypto.so.0.9.6, needed by / usr/local/ssl/lib/libssl.so, not found (try using -rpath or -rpath-link)utils.o: In function `vasprintf':/export/home/fst/chris/cvs/asterisk/utils.c:623: undefined reference to `va_copy '/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_DigestInit'/usr/local/ssl/lib/libssl.so: undefined reference to `BIO_find_type'/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_enc_null'/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_CIPHER_CTX_init'/usr/local/ssl/lib/libssl.so: undefined reference to `X509_NAME_dup'/usr/local/ssl/lib/libssl.so: undefined reference to `COMP_compress_block'/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_rc2_cbc'/usr/local/ssl/lib/libssl.so: undefined reference to `sk_new_null'/usr/local/ssl/lib/libssl.so: undefined reference to `X509_STORE_get_by_subject'/usr/local/ssl/lib/libssl.so: undefined reference to `lh_free'/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_VerifyFinal'/usr/local/ssl/lib/libssl.so: undefined reference to `COMP_CTX_new'/usr/local/ssl/lib/libssl.so: undefined reference to `sk_dup'/usr/local/ssl/lib/libssl.so: undefined reference to `X509_STORE_CTX_set_ex_data '/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_DigestFinal'/usr/local/ssl/lib/libssl.so: undefined reference to `X509_free'/usr/local/ssl/lib/libssl.so: undefined reference to `CRYPTO_get_ex_data'/usr/local/ssl/lib/libssl.so: undefined reference to `BN_bin2bn'/usr/local/ssl/lib/libssl.so: undefined reference to `CRYPTO_get_ex_new_index'/usr/local/ssl/lib/libssl.so: undefined reference to `PEM_read_bio_RSAPrivateKey '/usr/local/ssl/lib/libssl.so: undefined reference to `BN_bn2bin'/usr/local/ssl/lib/libssl.so: undefined reference to `RAND_add'/usr/local/ssl/lib/libssl.so: undefined reference to `BIO_s_socket'/usr/local/ssl/lib/libssl.so: undefined reference to `asn1_add_error'/usr/local/ssl/lib/libssl.so: undefined reference to `d2i_RSAPrivateKey'/usr/local/ssl/lib/libssl.so: undefined reference to `sk_num'/usr/local/ssl/lib/libssl.so: undefined reference to `BIO_free_all'/usr/local/ssl/lib/libssl.so: undefined reference to `BIO_get_retry_reason'/usr/local/ssl/lib/libssl.so: undefined reference to `X509_STORE_new'/usr/local/ssl/lib/libssl.so: undefined reference to `SHA1_Init'/usr/local/ssl/lib/libssl.so: undefined reference to `HMAC_Final'/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_md5'/usr/local/ssl/lib/libssl.so: undefined reference to `ASN1_object_size'/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_get_cipherbyname'/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_rc4'/usr/local/ssl/lib/libssl.so: undefined reference to `EVP_add_cipher'/usr/local/ssl/lib/libssl.so: undefined reference to `ASN1_get_object'/usr/local/ssl/lib/libssl.so: undefined reference to `BIO_s_file'/usr/local/ssl/lib/libssl.so: undefined reference to `COMP_expand_block'/usr/local/ssl/lib/libssl.so: undefined reference to `BIO_snprintf'/usr/local/ssl/lib/libssl.so: undefined reference to `d2i_RSAPrivateKey_bio'/usr/local/ssl/lib/libssl.so: undefined reference to `ASN1_dup'/usr/local/ssl/lib/libssl.so: undefined reference to `RSA_sign'/usr/local/ssl/lib/libssl.so: undefined reference to `ERR_peek_error'/usr/local/ssl/lib/libssl.so: undefined reference to `PEM_read_bio_PrivateKey'/usr/local/ssl/lib/libssl.so: undefined reference to `lh_retrieve'/usr/local/ssl/lib/libssl.so: undefined reference to `X509_get_pubkey'/usr/local/ssl/lib/libssl.so: undefined reference to `CRYPTO_dup_ex_data'/usr/local/ssl/lib/libssl.so: undefined reference to `DH_generate_key'/usr/local/ssl/lib/libssl.so: undefined reference to
Re: [Asterisk-Users] Build on Itanium fails
One must keep in mind that the config files specify how hardware is to be handled. If config files are present, the defaults in them are adequate to keep really bad things from happening. If not . . . . . . By the nature of this beast, it can easily seg fault if hardware drivers don't have proper controls. I had seg faults several times WITH config files until I got the parameters right. I did not consider that a huge bug, but rather part of my learning curve with asterisk. As far as the compile problem with the Itaniam system, are you sure you have a compiler version that fully supports this hardware/OS combination? Rollin Weeks On 8/9/05, Jonas Arndt [EMAIL PROTECTED] wrote: Ben,This is an enormous help. This is exactly what I was looking for.THANKS,// JonasAsterisk wrote:Jose,It might help to have a look at the debian SOURCE package for Asterisk. Here is the Debian DIFF Filehttp://ftp.debian.org/debian/pool/main/a/asterisk/asterisk_1.0.7.dfsg.1-2.diff.gz They've obviously been successful in compiling it for Itanium - maybe something obvious will jump out.I wish I had time to look it over myself, as I'd normally be happy to help, but unfortunately, today I'm just too busy. Best RegardsBenAs they have build Debian packages for Itanium I was hoping thatsomebody would have experience with compiling on Itanium and could give me some pointers.This message was checked by MailScan for WorkgroupMail.www.govarion.com ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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
Re: [Asterisk-Users] Build on Itanium fails
Hi Jonas, You may know this already, a codec is an algorithm for compressing and uncompressing some signal. Often the signal was originally analog, but has been digitized to reduce size/bandwidth and to store it in files. The GSM codec is important in asterisk, because most or all music playback is done using it. It is also one of the codec options for VoIP. I think Windows .wav files are translated to the gsm codec format. Asterisk will be hurting if it doesn't have the GSM codec available. Compiling code is a challenging problem. You have to know the correct or preferred compiler options. You need the matching libraries, and you need the include files (in this case the .h files for the C compilation). The make file has to tell the compiler where to find everything. If you have more than one version, sometimes it doesn't access the right one. If multiple sets of code are involved, you often have to compile them in the right order. I seem to recall that libpri and zapata and zaptec have to be ompiled before you can do asterisk itself. It's frustating, but you have to follow through each of the errors until you have eliminated all of them. Good luck! Rollin On 8/9/05, Jonas Arndt [EMAIL PROTECTED] wrote: Hi Rollin,I am using SuSE's SLE 9.0, which is built for Itanium. The compilerworks for other 32 and 64 bits applications. There could still be aproblem with my environment though. I have not excluded that. I can make it compile if I exclude the GSM codec. Now, how will thataffect the functionality? I know, I would know that if I learned moreabout the product.I agree that there could be a scenario where you, as a programmer, could cause a segmentation fault by incorrectly using pointers that were neverinitialized because of a problem in an external file (not existing orsyntax errors). I also agree that you as a programmer could cause segmentation faults by incorrectly trying to communicate with hardwarethat is not there. All those scenarios are bugs though. They should beavoided by proper error control and handling. Still, those are bugs I can live with as long as I have my stuff together when it comes to buildit on Itanium. If it comes out that the coredumps are indeed caused bybad config files, it would be really good news to me, as I really want to use the Itanium hardware. I did try make samples and there is nodifference, it still segfaultsI will now dig into the compilation options and the link I got from Joseto see if I can find anything useful. When I have a successful build without coredumps I will focus on learning the product. From what I haveseen so far it seems to be a really cool product.Thanks for all the help,// JonasRollin Weeks wrote: One must keep in mind that the config files specify how hardware is to be handled. If config files are present, the defaults in them are adequate to keep really bad things from happening.If not . . . . . . By the nature of this beast, it can easily seg fault if hardware drivers don't have proper controls.I had seg faults several times WITH config files until I got the parameters right.I did not consider that a huge bug, but rather part of my learning curve with asterisk. As far as the compile problem with the Itaniam system, are you sure you have a compiler version that fully supports this hardware/OS combination? Rollin Weeks On 8/9/05, *Jonas Arndt* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ben, This is an enormous help. This is exactly what I was looking for. THANKS, // Jonas Asterisk wrote: Jose, It might help to have a look at the debian SOURCE package for Asterisk. Here is the Debian DIFF File http://ftp.debian.org/debian/pool/main/a/asterisk/asterisk_1.0.7.dfsg.1-2.diff.gz They've obviously been successful in compiling it for Itanium - maybe something obvious will jump out. I wish I had time to look it over myself, as I'd normally be happy to help, but unfortunately, today I'm just too busy. Best Regards Ben As they have build Debian packages for Itanium I was hoping that somebody would have experience with compiling on Itanium and could give me some pointers. This message was checked by MailScan for WorkgroupMail. www.govarion.com http://www.govarion.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto: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 http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk-to-IVR Problem
This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line -X100P-asterisk-TDM10B- phone cord-Dialogic analog port-IVR system. Asterisk should dial the IVR system, which should answer and play back its IVR scenario script to the caller. However, when a call comes in, asterisk answers on Zap1-1 and dials Zap2-1. The IVR system answers the call and begins to play back its scenario. After 5 to 15 seconds, asterisk apparently senses an on-hook condition (exception 17?) and disconnects the connection bridge. The logs on the IVR system shows that it is not initially aware of the hangup, and continues playing its scenario. Going to an analog phone in the TDM10B instead of the IVR system appears to work OK, with the exception that asterisk is still sending dial tones when the analog phone is answered. The phone stays connected to asterisk until it really does hang up. What causes the hangup? What generates the exception? I have looked at the chan_zap.c code and can not see how zt_exception gets into the picture. Is there a TDMF incompatibility problem? Is this a case where DAX should be used? My config files and sample debug output are given below. Thanks for anyone's help. Rollin Weeks # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # fxsks=1 # For the X100P fxoks=2 # For the TDM400P (TDM10B) # loadzone = us # defaultzone=us ; ; Zapata telephony interface ; ; Configuration file [channels] ; language=en context=incoming switchtype=national signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; Whether or not to use caller ID ; usecallerid=asreceived ; Whether or not to hide outgoing caller ID (Override with *67 or *82) ; hidecallerid=no ; callwaitingcallerid=yes ; for default voicemail context, the example below is fine: ; mailbox=1234 echocancel=yes ; If you are having trouble with DTMF detection, you can relax the ; DTMF detection parameters. Relaxing them may make the DTMF detector ; more likely to have talkoff where DTMF is detected when it ; shouldn't be. ; relaxdtmf=yes ; You may also set the default receive and transmit gains (in dB) ; rxgain=0.0 txgain=0.0 immediate=no ; On trunk interfaces (FXS) and EM interfaces (EM, Wink, Feature Group D ; etc, it can be useful to perform busy detection either in an effort to ; detect hangup or for detecting busies ; busydetect=yes ; ; If busydetect is enabled, is also possible to specify how many ; busy tones to wait before hanging up. The default is 4, but ; better results can be achieved if set to 6 or even 8. Mind that ; higher the number, more time is needed to hangup a channel, but ; lower is probability to get random hangups ; busycount=40 ; Select which class of music to use for music on hold. If not specified ; then the default will be used. ; musiconhold=default channel = 1 signalling=fxo_ks context=internal channel = 2 ; Home grown extension file [globals] ;RECEPTIONIST=Zap/1 ; [incoming] exten = s,1,Wait(1) exten = s,2,Answer() exten = s,3,Playback(demo-congrats) ; Plays the demo-congrats file after answering the line exten = s,4,Dial,Zap/2/1000\20 exten = s,5,Hangup [internal] exten = 1000,1,Answer() exten = 1000,2,Wait(2) exten = 1000,3,Hangup --- -- Starting simple switch on 'Zap/1-1' Urgent handler Aug 4 15:43:39 DEBUG[5059]: pbx.c:1274 pbx_extension_helper: Launching 'Wait' -- Executing Wait(Zap/1-1, 1) in new stack Urgent handler Aug 4 15:43:40 DEBUG[5059]: pbx.c:1274 pbx_extension_helper: Launching 'Answer' -- Executing Answer(Zap/1-1, ) in new stack Urgent handler Aug 4 15:43:40 DEBUG[5059]: chan_zap.c:2301 zt_answer: Took Zap/1-1 off hook Aug 4 15:43:40 DEBUG[5059]: chan_zap.c:1231 zt_enable_ec: Enabled echo cancellation on channel 1 Aug 4 15:43:40 DEBUG[5059]: chan_zap.c:1250 zt_train_ec: No echo training requested Aug 4 15:43:40 DEBUG[5059]: pbx.c:1274 pbx_extension_helper: Launching 'Playback' -- Executing Playback(Zap/1-1, demo-congrats) in new stack Urgent handler Aug 4 15:43:40 DEBUG[5059]: channel.c:1719 ast_set_write_format: Set channel Zap/1-1 to write format gsm Aug 4 15:43:40 DEBUG[5059]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'demo-congrats' (language 'en') Urgent handler Aug 4 15:44:08 DEBUG[5059]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Aug 4 15:44:08 DEBUG[5059]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Aug 4 15:44:08 DEBUG[5059]: channel.c:1719 ast_set_write_format: Set channel Zap/1-1 to write format ulaw Aug 4 15:44:08 DEBUG[5059]: pbx.c:1274
