[asterisk-users] I can't receive fax
Hi: How can I see the communication between hylafax and iaxmodem and the console of them?I only can see the console of asterisk.It shows: 'IAX2/iaxmodem-2 is ringing' when I dial the fax number. and nothing else.I can't receive fax. I installed asterisk 1.4.18 and iaxmodem-1.2.0 and hylafax-5.2.9-1.fc9.i386.rpm and these are my main configurations: extensions.conf: [from-pstn] exten = 9711315,1,Answer exten = 9711315,2,Dial(IAX2/iaxmodem) iax.conf: [iaxmodem] type=friend secret=password port=4570 host=dynamic context=from-pstn disallow=all allow=alaw /etc/iaxmodem/ttyIAX0 device /dev/ttyIAX0 owner uucp:uucp mode 660 port 4570 refresh 50 server 127.0.0.1 peername iaxmodem secret password codec alaw and this is my modem type in /var/spool/hylafax/etc/config.tty ModemType: Class1 /etc/inittab: id:5:initdefault: IA00:23:respawn:/usr/bin/iaxmodem ttyIAX0 IA00:23:respawn:/usr/bin/iaxmodem ttyIAX0 mo00:23:respawn:/usr/sbin/faxgetty ttyIAX0 I started hylafax and run iaxmodem. I see below lines when I run faxstat: HylaFAX scheduler on localhost.localdomain: Running Modem ttyIAX0 (9711315): Waiting for modem to come ready I'd appreciate any help,idea. ~ ___ -- 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] I can't receive fax
thank you Dear doug But,I don't have any file in /var/spool/hylafax/log directory. ___ -- 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] receive fax problem
Hi: I want to receive a fax with an E1 link connected to A102d card from a fax machine,but after dialling the phone number, it connects then will be busy.In fact asterisk can't detect the fax.These are zapata.conf, extensions.conf filels and debug in console: extensions.conf: [from-pstn] exten = 9711315,1,Answer() exten = 9711315,2,Wait(10) exten = fax,1,SetVar(FAXFILE=/tmp/test.tif) exten = fax,2,rxfax(FAXFILE) 9711315 is E1 number. zapata.conf [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes faxdetect=from-pstn relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A102 port 1 [slot:6 bus:1 span:1] wanpipe1 switchtype=national context=from-pstn group=0 signalling=pri_cpe channel =1-15,17-31 -- Executing [9711...@from-pstn:1] Answer(Zap/6-1, ) in new stack -- Accepting call from '3318545' to '9711315' on channel 0/6, span 1 == Auto fallthrough, channel 'Zap/6-1' status is 'UNKNOWN' -- Hungup 'Zap/6-1' I've installed spandsp and app_rxfax. I'd appreciate any help and idea. ___ -- 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] asterisk 1.2 and openser 1.4
Hi: Can asterisk 1.2 and openser 1.4 work togather ? Regards. ___ -- 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] music on hold
See: http://astrecipes.net/index.php?q=AstRecipes/Music-on-hold%20without%20MPG123 ___ -- 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] Dial outside number using the E1 Link
Hi: I've configured an asterisk server with A102d sangoma's card and the E1 link.I want to dial outside number using the E1 Link.How can I dial a phone number?Is this true? exten = 123,1,Dial(ZAP/1/phone number) I'd appreciate any help. ___ -- 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] multiple passwords for one meetme!
Hi: Can one conference room have multiple passwords for example 10 passwords for one meetme room ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gsm files instead mp3 files in a conference room!
Hi: I want to asterisk play gsm files instead mp3 files when only one person is in a conference room with 'M' option in Meetme application.Is it possible? (I place 2 gsm files in mohmp3 folder and didn't install mpg123) I'd appreciate any help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sendmail file
Hi: How can I configure sendmail file to asterisk send voicemails to my mail.sendmail file in /usr/sbin is a read only file. I'd appreciate any help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mpg123 problem
Hi: I want to install mpg123-0.59r on my asterisk server.I downloaded it in /usr/src then untared it and I typed these command : #cd /usr/src/mpg123-0.59r #make linux after run make linux ,I saw 2 errors in terminal: make CC=gcc LDFLAGS= \ OBJECTS='decode_i386.o dct64_i386.o decode_i586.o \ audio_oss.o term.o' \ CFLAGS='-DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX \ -DREAD_MMAP -DOSS -DTERM_CONTROL\ -Wall -O2 -m486 \ -fomit-frame-pointer -funroll-all-loops \ -finline-functions -ffast-math' \ mpg123-make make[1]: Entering directory `/usr/src/mpg123-0.59r' make[2]: Entering directory `/usr/src/mpg123-0.59r' make[2]: *** No rule to make target `\ ', needed by `mpg123'. Stop. make[2]: Leaving directory `/usr/src/mpg123-0.59r' make[1]: *** [mpg123-make] Error 2 make[1]: Leaving directory `/usr/src/mpg123-0.59r' make: *** [linux] Error 2 What is the problem? Please guide me. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi : asterisk didn't send voice message to my mail([EMAIL PROTECTED]).My main configured files are: extensions.conf: [from-pstn] exten = 9711315,1,Dial(SIP/3000,30) exten = 9711315,2,VoiceMail([EMAIL PROTECTED]) exten = 9711315,3,PlayBack(vm-goodbye) exten = 9711315,4,HangUp() sip.conf: [3000] type=friend username=3000 secret=1234567 host=dynamic context=from-pstn [EMAIL PROTECTED] voicemail.conf: [ff_tutorial] 3000 = 1234567,3000,[EMAIL PROTECTED] And these are in console: Jun 29 12:08:11 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 -- Accepting call from '3322000' to '9711315' on channel 0/1, span 1 Jun 29 12:08:15 DEBUG[24207]: chan_zap.c:1554 zt_enable_ec: Enabled echo cancellation on channel 1 -- Executing Dial(Zap/1-1, SIP/3000|30) in new stack Jun 29 12:08:15 DEBUG[24257]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on RTP to 0 Jun 29 12:08:15 DEBUG[24257]: chan_sip.c:2085 sip_call: Outgoing Call for 3000 -- Called 3000 Jun 29 12:08:16 DEBUG[24203]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found -- SIP/3000-08941d28 is ringing Jun 29 12:08:16 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication 3 on channel Zap/1-1 Jun 29 12:08:21 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:08:31 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:08:41 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 -- Nobody picked up in 3 ms Jun 29 12:08:47 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication -1 on channel Zap/1-1 Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:2450 sip_hangup: update_call_counter(3000) - decrement call limit counter Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:1392 __sip_ack: Acked pending invite 102 Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 29 12:08:47 DEBUG[24257]: app_dial.c:1661 dial_exec_full: Exiting with DIALSTATUS=NOANSWER. -- Executing VoiceMail(Zap/1-1, [EMAIL PROTECTED]) in new stack Jun 29 12:08:47 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-intro' (language 'en') Jun 29 12:08:47 DEBUG[24203]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 29 12:08:47 DEBUG[24203]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Not Found Jun 29 12:08:51 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:08:52 DEBUG[24257]: app.c:1235 ast_lock_path: Locked path '/var/spool/asterisk/voicemail/ff_tutorial/3000/INBOX' Jun 29 12:08:52 DEBUG[24257]: app.c:1256 ast_unlock_path: Unlocked path '/var/spool/asterisk/voicemail/ff_tutorial/3000/INBOX' Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'beep' (language 'en') Jun 29 12:08:53 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:08:53 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals -- Recording the message Jun 29 12:08:53 DEBUG[24257]: app.c:568 ast_play_and_record_full: play_and_record: None, /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9, 'wav49|gsm|wav' Jun 29 12:08:53 DEBUG[24257]: app.c:585 ast_play_and_record_full: Recording Formats: sfmts=wav49 -- x=0, open writing: /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: wav49, 0x88b0f48 -- x=1, open writing: /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: gsm, 0x88b12a0 -- x=2, open writing: /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: wav, 0x88b15e0 Jun 29 12:08:53 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication 18 on channel Zap/1-1 Jun 29 12:09:01 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:09:11 DEBUG[24257]: chan_zap.c:3669 zt_handle_dtmfup: DTMF digit: # on Zap/1-1 -- User ended message by pressing # Jun 29 12:09:11 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'auth-thankyou' (language 'en') Jun 29 12:09:11 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:09:12 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:09:12 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:09:12
[asterisk-users] voicemail didn't send voice message to my email
Hi : asterisk didn't send voice message to my mail([EMAIL PROTECTED]).My main configured files are: extensions.conf: [from-pstn] exten = 9711315,1,Dial(SIP/3000,30) exten = 9711315,2,VoiceMail([EMAIL PROTECTED]) exten = 9711315,3,PlayBack(vm-goodbye) exten = 9711315,4,HangUp() sip.conf: [3000] type=friend username=3000 secret=1234567 host=dynamic context=from-pstn [EMAIL PROTECTED] voicemail.conf: [ff_tutorial] 3000 = 1234567,3000,[EMAIL PROTECTED] And these are in console: Jun 29 12:08:11 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 -- Accepting call from '3322000' to '9711315' on channel 0/1, span 1 Jun 29 12:08:15 DEBUG[24207]: chan_zap.c:1554 zt_enable_ec: Enabled echo cancellation on channel 1 -- Executing Dial(Zap/1-1, SIP/3000|30) in new stack Jun 29 12:08:15 DEBUG[24257]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on RTP to 0 Jun 29 12:08:15 DEBUG[24257]: chan_sip.c:2085 sip_call: Outgoing Call for 3000 -- Called 3000 Jun 29 12:08:16 DEBUG[24203]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found -- SIP/3000-08941d28 is ringing Jun 29 12:08:16 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication 3 on channel Zap/1-1 Jun 29 12:08:21 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:08:31 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:08:41 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 -- Nobody picked up in 3 ms Jun 29 12:08:47 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication -1 on channel Zap/1-1 Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:2450 sip_hangup: update_call_counter(3000) - decrement call limit counter Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:1392 __sip_ack: Acked pending invite 102 Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 29 12:08:47 DEBUG[24257]: app_dial.c:1661 dial_exec_full: Exiting with DIALSTATUS=NOANSWER. -- Executing VoiceMail(Zap/1-1, [EMAIL PROTECTED]) in new stack Jun 29 12:08:47 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-intro' (language 'en') Jun 29 12:08:47 DEBUG[24203]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 29 12:08:47 DEBUG[24203]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Not Found Jun 29 12:08:51 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:08:52 DEBUG[24257]: app.c:1235 ast_lock_path: Locked path '/var/spool/asterisk/voicemail/ff_tutorial/3000/INBOX' Jun 29 12:08:52 DEBUG[24257]: app.c:1256 ast_unlock_path: Unlocked path '/var/spool/asterisk/voicemail/ff_tutorial/3000/INBOX' Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'beep' (language 'en') Jun 29 12:08:53 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:08:53 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals -- Recording the message Jun 29 12:08:53 DEBUG[24257]: app.c:568 ast_play_and_record_full: play_and_record: None, /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9, 'wav49|gsm|wav' Jun 29 12:08:53 DEBUG[24257]: app.c:585 ast_play_and_record_full: Recording Formats: sfmts=wav49 -- x=0, open writing: /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: wav49, 0x88b0f48 -- x=1, open writing: /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: gsm, 0x88b12a0 -- x=2, open writing: /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: wav, 0x88b15e0 Jun 29 12:08:53 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication 18 on channel Zap/1-1 Jun 29 12:09:01 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:09:11 DEBUG[24257]: chan_zap.c:3669 zt_handle_dtmfup: DTMF digit: # on Zap/1-1 -- User ended message by pressing # Jun 29 12:09:11 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'auth-thankyou' (language 'en') Jun 29 12:09:11 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:09:12 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:09:12 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:09:12
[asterisk-users] voicemail problem
Hi: I configured asterisk for voicemail service.My main configuration files are: extensions.conf [from-pstn] exten =gt; 9711315,1,Dial(SIP/3000,30) exten =gt; 9711315,2,VoiceMail([EMAIL PROTECTED]) exten =gt; 9711315,3,PlayBack(vm-goodbye) exten =gt; 9711315,4,HangUp() voicemail.conf [ff_tutorial] 555 =gt; 1234567,3000,[EMAIL PROTECTED] sip.conf [3000] type=friend username=3000 secret=1234567 host=dynamic context=from-pstn [EMAIL PROTECTED] But when I dial 9711315, after 30s I hear goodbye and call hangups. in console: -- Accepting call from '3322000' to '9711315' on channel 0/2, span 1 -- Executing Dial(Zap/2-1, SIP/3000|30) in new stack -- Called 3000 -- SIP/3000-08f18698 is ringing Jun 24 11:55:32 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 24 11:55:42 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 24 11:55:52 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 -- Nobody picked up in 3 ms -- Executing VoiceMail(Zap/2-1, [EMAIL PROTECTED]) in new stack Jun 24 11:55:53 WARNING[5188]: app_voicemail.c:2461 leave_voicemail: No entry in voicemail config file for '555' -- Executing Playback(Zap/2-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') -- Executing Hangup(Zap/2-1, ) in new stack == Spawn extension (from-pstn, 9711315, 4) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' Jun 24 11:56:02 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 what's problem? should I do something in sip phone for voicemail? I'd appreciate any help. ___ -- 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] asterisk was discunnected suddenly
Hi: I configured an asterisk server with 2.4G cpu and 1G ram for conference call service but when 5 peoples about 20 minutes were talking together suddenly asterisk was disconnected.May it has happened because low cpu or ram?I saw var/log/asterisk/messages file but everything was going well apparently,asterisk was disconnected suddenly.What 'snbsp; your idea?Pleasenbsp; guide me. ___ -- 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] cpu and ram requirements
Hi: How much cpu and ram is required for conference call service with 1 E1 link(30 zaptel ports) and 50 sip phones,to asterisk won't be disconnected during conference? Regards.nbsp; ___ -- 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] meetme recording with security?
Hi: I configured an asterisk server for conference call service but I have a problem now :Does asterisk have an option to secure and warranty meetme,in the other word,How can I play up users that their conference won't hear by us in spite of asterisk can record meetme ? I'd appreciate any help. ___ -- 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] asterisk1.2.24
Hi: Can asterisk1.2.24 work with zaptel1.2.20.1 and libpri1.2.5?And How is asterisk1.2.24? - Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. ___ --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] asterisk1.2
Hi: I want to use asterisk1.2 but I don't know which version of asterisk1.2 and zaptel1.2 is best.Please offer me one version of asterisk and zaptel and libpri.How about asterisk1.2.24 and zaptel1.2.20.1 and libpri1.2.5?And do they work togather well? Best regards. - Pinpoint customers who are looking for what you sell. ___ --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] Fwd: Asterisk doesn't answer to incoming call from pstn.
Note: forwarded message attached. - Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. ---BeginMessage--- Hi: I installed A102d sangoma's card successfully but Asterisk doesn't answer to incoming call from pstn and console doesn't show any message of incoming call in the other word when I dial the number of E1 I can't connect to asterisk and dial the number of extension. I'd apreciate any idea. My configuration files: zaptel.conf: Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:3 bus:1 span: 1] span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 #Sangoma A102 port 2 [slot:3 bus:1 span: 2] span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 zapata.conf: ;autogenerated by /usr/local/sbin/config-zaptel do not hand edit ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A102 port 1 [slot:3 bus:1 span: 1] switchtype=national context=from-pstn group=0 signalling=pri_cpe channel = 1-15,17-31 ;Sangoma A102 port 2 [slot:3 bus:1 span: 2] switchtype=euroisdn context=from-pstn group=0 signalling=pri_cpe channel = 32-46,48-62 extensions.conf: [from-pstn] exten = 611,1,Answer() exten = 611,2,Echo() - Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us.---End Message--- ___ --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] Asterisk doesn't answer to incoming call
Hi: I installed A102d sangoma's card successfully but Asterisk doesn't answer to incoming call from pstn and console doesn't show any message of incoming call in the other word when I diall the number of E1 I can't connect to asterisk and dial the number of extension. I'd apreciateany idea. - Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. ___ --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] configuration of wanpipe for asterisk.
Hi: I install A102 sangoma's card and connect E1 link it now for configuring wanpipe which one should I select for dial plan context:from pstn?or from internal? Best regards. - Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] extensions for conference call
Hi: Can I set 1 extension(i.e.6000) in extensions.conf file for several room for conference call service ? Or for every room I should set 1 special extension. Regards. - Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] unnumbered priorities
Hi: When should we use unnumbered priorities(n) in extensions.What is the different between these 2 forms of extensions.conf? and ,Are both true? extensions.conf: form1: [Conferencerooms] exten = 333,1,Answer exten = 333,n,meetme(8000|cim) exten = 333,n,playback(vm-goodbye) exten = 333,n,hangup form2: [Conferencerooms] exten = 333,1,Answer exten = 333,2,meetme(8000|cim) exten = 333,3,playback(vm-goodbye) exten = 333,4,hangup I'd appreciate any help. Regards. - Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search.___ --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] enter menu
For user and administrator enter menu when *-key is pressed we should use 's' option or nothing(asterisk does it automatic). I'd appreciate any idea. Regards. - Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. ___ --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] A102d sangoma's card and ztdummy
Hi: I want to have conference call service and I use A102d sangoma's card.Do I should install ztdummy or app-conference? Best regards. - Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. ___ --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] asterisk 1.2 or 1.4 for conference call service
Hi: I want to have conference call service and I have A102d sangoma's card so I install asterisk 1.2.x or 1.4.x? Best regards. - Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. ___ --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] app-conference
Hi: I think app-conference is used where there isn't zaptel hardware,in the other word when we use zaptel hardware we shouldn't use app-conference for conference call sevice and we should use meetme application and load ztdummy.Is it true? Best regards. - Pinpoint customers who are looking for what you sell. ___ --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] app-conference
Hi: I think app-conference is used where there isn't zaptel hardware,in the other word when we use zaptel hardware we shouldn't use app-conference for conference call sevice and we should use meetme application and load ztdummy.Is it true? Best regards. - Shape Yahoo! in your own image. Join our Network Research Panel today!___ --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] Which interface?
Hi: If any body use meetmemanager or conman or web-meetme please say how about is it.I'd appreciated any idea. Regards. - Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. ___ --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] How do I configure asterisk?
Hi: Which one is better and easier for configure asterisk,directly or by GUI ? I'd appreciate any idea. Regards. - Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.___ --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] asterisk1.2.24 or asterisk1.4.10.1
Hi: You offer me use asterisk1.2.24 or asterisk1.4.10.1.How's it if I want to use astbill? Best Regards. - Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us.___ --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] The Future of Telephony, by Leif Madsen, Jared Smith, and Jim Van Meggelen
Hi: Which was released for free download under a Creative Commons license for The Future of Telephony, by Leif Madsen, Jared Smith, and Jim Van Meggelen. Regards. - Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. ___ --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 or analog line
Hi: I want to have conference call(meetme) service with asterisk and 30 users.Now do I use 1E1 or 30 analog lines with due attention to high price of E1 line?And which interface card do I use? Best regards. - Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. ___ --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] A102d samgoma's card
Hi: Please every that work with A102d say how about is it?Is it really difficult to install card for me new in asterisk? Best regards. - Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.___ --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] TE212 or TE220
Hi: I want to have conference call with asterisknow and need 2 ports E1.Which Digium card is better?TE212 or TE220.I haven't problem with motherboard. Regards. - Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. ___ --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] asterisk or asterisknow
Hi: I want to have conference call service.You offer me use asterisk or asterisknow. Regards. - Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.___ --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
