[asterisk-users] OT: Upgrade Addpac AP200C
Hi guys, I have made a upgrade to my addpac ap200c, however it does not upload complete, now I can load addpac. Is there anyway that can I upload the old firwmare? Any help is appreciated. System Boot Loader, Version 2.2.5/DUAL(852) Copyright (c) by AddPac Technology Co., Ltd. Since 1999. System Bootstrap, Version 1.2 Decompressing the image: # ___ -- 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
Re: [asterisk-users] OT: Upgrade Addpac AP200C
Hi guys, I have made a upgrade to my addpac ap200c, however it does not upload complete, now I can not load addpac. Is there anyway that I can upload the old firwmare? Any help is appreciated. System Boot Loader, Version 2.2.5/DUAL(852) Copyright (c) by AddPac Technology Co., Ltd. Since 1999. System Bootstrap, Version 1.2 Decompressing the image: # ___ -- 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] Issues with making calls
Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) My sources are: libpri-1.4.1.tar.gz zaptel-1.4.5.1.tar.gz asterisk-1.4.11.tar.gz asterisk-addons-1.4.2.tar.gz asterisk-perl-0.10.tar.gz I have 1/2 E1 from my provider telephony, my configuration is [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us #cat /etc/asterisk/zapata.conf [channels] context=default switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no amaflags=documentation musiconhold=default ;Configure Channels group=0 callgroup=0 pickupgroup=0 channel = 1-4 group=1 callgroup=1 pickupgroup=1 channel = 5-8 group=2 callgroup=2 pickupgroup=2 channel = 9-12 group=3 callgroup=3 pickupgroup=3 channel = 13-14 group=4 callgroup=4 pickupgroup=4 channel = 15 I have could make calls but, after of some minutes my server is hung, suggestions are welcome. Thanks for any help in advance. ___ --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
Re: [asterisk-users] Issues with making calls
Yes, the module is load # asterisk -r ippbx*CLI module show like chan_zap.so Module Description Use Count chan_zap.soZapata Telephony 0 1 modules loaded ippbx*CLI ippbx*CLI 2007/10/18, Brian West [EMAIL PROTECTED]: Make sure chan_zap.so is loaded. /b On Oct 18, 2007, at 9:34 AM, Pablo Almido wrote: Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) My sources are: libpri-1.4.1.tar.gz zaptel-1.4.5.1.tar.gz asterisk-1.4.11.tar.gz asterisk-addons-1.4.2.tar.gz asterisk-perl-0.10.tar.gz I have 1/2 E1 from my provider telephony, my configuration is [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone=us #cat /etc/asterisk/zapata.conf [channels] context=default switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no amaflags=documentation musiconhold=default ;Configure Channels group=0 callgroup=0 pickupgroup=0 channel = 1-4 group=1 callgroup=1 pickupgroup=1 channel = 5-8 group=2 callgroup=2 pickupgroup=2 channel = 9-12 group=3 callgroup=3 pickupgroup=3 channel = 13-14 group=4 callgroup=4 pickupgroup=4 channel = 15 I have could make calls but, after of some minutes my server is hung, suggestions are welcome. Thanks for any help in advance. ___ --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 ___ --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 ___ --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
Re: [asterisk-users] Issues with making calls
I have unload and load the module, it is output ippbx*CLI module unload chan_zap.so == Unregistered application 'ZapSendKeypadFacility' ippbx*CLI module load chan_zap.so == Registered application 'ZapSendKeypadFacility' == Parsing '/etc/asterisk/zapata.conf': Found [Oct 18 10:46:38] WARNING[2790]: chan_zap.c:903 zt_open: Unable to specify channel 1: No such device or address [Oct 18 10:46:38] ERROR[2790]: chan_zap.c:7160 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 [Oct 18 10:46:38] ERROR[2790]: chan_zap.c:10466 build_channels: Unable to register channel '1-4' ippbx*CLI 2007/10/18, Brian West [EMAIL PROTECTED]: Why would a config error stop the module from loading? That seems like a suboptimal behavior. /b On Oct 18, 2007, at 9:50 AM, Jared Smith wrote: That would seem to indicate that the chan_zap.so module isn't being loaded. What happens if you type module unload chan_zap.so and then module load chan_zap.so from the Asterisk CLI? I'll bet you'll find that either there's a problem in your zapata.conf file, or that chan_zap hasn't been compiled for some reason. ___ --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 ___ --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
Re: [asterisk-users] Issues with making calls
I run this command [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCT1/0 Wildcard TE12xP Card 0 IRQ misses: 40 1 WCT1/0/1 2 WCT1/0/2 3 WCT1/0/3 4 WCT1/0/4 5 WCT1/0/5 6 WCT1/0/6 7 WCT1/0/7 8 WCT1/0/8 9 WCT1/0/9 10 WCT1/0/10 11 WCT1/0/11 12 WCT1/0/12 13 WCT1/0/13 14 WCT1/0/14 15 WCT1/0/15 16 WCT1/0/16 17 WCT1/0/17 18 WCT1/0/18 19 WCT1/0/19 20 WCT1/0/20 21 WCT1/0/21 22 WCT1/0/22 23 WCT1/0/23 24 WCT1/0/24 25 WCT1/0/25 26 WCT1/0/26 27 WCT1/0/27 28 WCT1/0/28 29 WCT1/0/29 30 WCT1/0/30 31 WCT1/0/31 Then I run ztcfg [EMAIL PROTECTED] ~]# ztcfg -vv Zaptel Version: 1.4.5.1 Echo Canceller: MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels configured. Changing signalling on channel 1 from Unused to Clear channel Changing signalling on channel 2 from Unused to Clear channel Changing signalling on channel 3 from Unused to Clear channel Changing signalling on channel 4 from Unused to Clear channel Changing signalling on channel 5 from Unused to Clear channel Changing signalling on channel 6 from Unused to Clear channel Changing signalling on channel 7 from Unused to Clear channel Changing signalling on channel 8 from Unused to Clear channel Changing signalling on channel 9 from Unused to Clear channel Changing signalling on channel 10 from Unused to Clear channel Changing signalling on channel 11 from Unused to Clear channel Changing signalling on channel 12 from Unused to Clear channel Changing signalling on channel 13 from Unused to Clear channel Changing signalling on channel 14 from Unused to Clear channel Changing signalling on channel 15 from Unused to Clear channel Changing signalling on channel 16 from Unused to HDLC with FCS check Changing signalling on channel 17 from Unused to Clear channel Changing signalling on channel 18 from Unused to Clear channel Changing signalling on channel 19 from Unused to Clear channel Changing signalling on channel 20 from Unused to Clear channel Changing signalling on channel 21 from Unused to Clear channel Changing signalling on channel 22 from Unused to Clear channel Changing signalling on channel 23 from Unused to Clear channel Changing signalling on channel 24 from Unused to Clear channel Changing signalling on channel 25 from Unused to Clear channel Changing signalling on channel 26 from Unused to Clear channel Changing signalling on channel 27 from Unused to Clear channel Changing signalling on channel 28 from Unused to Clear channel Changing signalling on channel 29 from Unused to Clear channel Changing signalling on channel 30 from Unused to Clear channel Changing signalling on channel 31 from Unused to Clear channel Then I back to run this command , but I can not understand why it change in channel 16, However I can not make calls [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCT1/0 Wildcard TE12xP Card 0 HDB3/CCS/CRC4 IRQ misses: 40 1 WCT1/0/1 Clear 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear
[asterisk-users] Change verbose level
Hi folks, How I can change default level in asterisk from 3 level to 7level, using the script/etc/init.d/asterisk ___ --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] E1 Card recommendation
Hi guys, I have an asterisk running only as sip server in my job, can anyone recommend me a good E1 card with interface RJ48. I am going to integrate my Asterisk to PSTN with a half of channels of E1´s from my provider Telephone Service Provider. ___ --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] Problem HandyTone 488 does not call transfer
Hi I have a analog phone connected to my Gateway Handytone and registered to Asterisk 1.4 I have configured my HandyTone 488 (in the section FXS Port) for make and receive calls, however I can not transfer a call when it come via PSTN. But, when a call come from via IP I can transfer it. [phoneanalog] type=friend secret=XXX context=local nat=no qualify=yes host=dynamic dtmfmode=rfc2833 canreinvite=no disallow=all allow=gsm allow=alaw callerid=Krix Altrust 2041 [EMAIL PROTECTED] ___ --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] Asterisk 1.4 and CDR
Hi guys, I have recently installed a Asterisk Server with CDR Call Detail Records. I have installed it over a Asterisk 1.2 , but now It do not run . I have installed it with the following procedure: # yum install ncurses #yum install openh323-devel # yum install mysql-server # yum install mysql # yum install php-gd # yum install php-mysql # yum install mysqlclient10 # yum install zlib # yum install zlib-devel # yum install ncurses-devel Install perl support perl -MCPAN -e install DBD::mysql I compile /usr/src/asterisk-addons as follows: # ./configure # make clean # make install In the file /etc/asterisk/cdr_mysql.conf [global] hostname=localhost dbname=asteriskcdrdb table=cdr password=strongpass user=asterisk port=3306 userfield=1 In the File asterisk-stat define (WEBROOT, http://192.168.190.10/asterisk-stat/;); define (FSROOT, /var/www/html/asterisk-stat-v2/); define (LIBDIR, FSROOT.lib/); define (HOST, localhost); define (PORT, 3306); define (USER, asterisk); define (PASS, strongpass); define (DBNAME, asteriskcdrdb); define (DB_TYPE, mysql); // mysql or postgres define (DB_TABLENAME, cdr); When I compile asterisk-addons it pass very good, but I do not build the file *cdr_addon_mysql.so* Do you have similar problem ?Thanks for your response. Excuseme for my english, it is not my native language. ___ --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] Asterisk 1.4 and CDR
I have solved the problem, I have already install mysql-devel and then # cd asterisk-addons-1.4.0 # make distclean # ./configure # make # make install # make samples My Call Detail Records is running. 2007/1/17, Savoy, Kevin - Williston, ND [EMAIL PROTECTED]: I had the same issue. I needed to install #yum install mysql-devel. Once I did this the addons compiled the file fine. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Pablo Almido *Sent:* Wednesday, January 17, 2007 9:43 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Asterisk 1.4 and CDR Hi guys, I have recently installed a Asterisk Server with CDR Call Detail Records. I have installed it over a Asterisk 1.2 , but now It do not run . I have installed it with the following procedure: # yum install ncurses #yum install openh323-devel # yum install mysql-server # yum install mysql # yum install php-gd # yum install php-mysql # yum install mysqlclient10 # yum install zlib # yum install zlib-devel # yum install ncurses-devel Install perl support perl -MCPAN -e install DBD::mysql I compile /usr/src/asterisk-addons as follows: # ./configure # make clean # make install In the file /etc/asterisk/cdr_mysql.conf [global] hostname=localhost dbname=asteriskcdrdb table=cdr password=strongpass user=asterisk port=3306 userfield=1 In the File asterisk-stat define (WEBROOT, http://192.168.190.10/asterisk-stat/;); define (FSROOT, /var/www/html/asterisk-stat-v2/); define (LIBDIR, FSROOT.lib/); define (HOST, localhost); define (PORT, 3306); define (USER, asterisk); define (PASS, strongpass); define (DBNAME, asteriskcdrdb); define (DB_TYPE, mysql); // mysql or postgres define (DB_TABLENAME, cdr); When I compile asterisk-addons it pass very good, but I do not build the file *cdr_addon_mysql.so* Do you have similar problem ?Thanks for your response. Excuseme for my english, it is not my native language. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Calling to PSTN newbie question
Hi list, Ihave a asterisk and sopthonesworking well, and I make a configuration for it work with a voice gatewayAddpac 2120 (4port FXO y 4 ports FXS), I have connected my gateway to my PBX, when Itry to call to PSTN from my softphone, I have a trouble that Asterisk add the number 9overthe number that I dial. How can ignore that number 9 when I dial from my softphone? ___ --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] Asterisk with Addpac 2120
Hi list, I am a newbie with Asterisk I setupa asterisk withsoftphonesalsoIhaveconfigure2addpac200A,Icanmakecallstosoftphonesandgatewaysaddpac.NowIwanttojointoPSTNIwantthat my softphones can make calls via Addpac 2120 (4 FXS y 4 FXO ports). How I can do it?I have a PBX with 2 lines and 8 anexs ___ --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
