[asterisk-users] Dailing any number PSTN or MObile number
Hello I have a -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Dailing any number PSTN or MObile number
On 24/08/09 6:27 PM, ABBAS SHAKEEL wrote: Hello I have a :) hmmm might need a little more info -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Dailing any number PSTN or MObile number
Sorry that was sent by mistake The question is . I have configured Asterisk with TDM400P i can recieve calls every thing goes fine... But one is unclear to me. If i want to intiate a call to a PSTN number or any mobile number as we normally do with our phones How can i do that with asterisk . I am not getting what would be right key word to search for this ...initiate a call / calling out from asterisk etc etc are use less .. Please guide -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Dailing any number PSTN or MObile number
On 24/08/09 6:31 PM, ABBAS SHAKEEL wrote: Sorry that was sent by mistake The question is . I have configured Asterisk with TDM400P i can recieve calls every thing goes fine... But one is unclear to me. If i want to intiate a call to a PSTN number or any mobile number as we normally do with our phones How can i do that with asterisk . I am not getting what would be right key word to search for this ...initiate a call / calling out from asterisk etc etc are use less .. Probably search for: Asterisk Dial Application You might want to read Asterisk: The future of telephony - you can either buy a copy (and support future editions) or download a free copy. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Dailing any number PSTN or MObile number
Sorry that was sent by mistake i pressed ctrl+s unintentionally The question is . I have configured Asterisk with TDM400P i can recieve calls every thing goes fine... But one is unclear to me. If i want to intiate a call to a PSTN number or any mobile number as we normally do with our phones How can i do that with asterisk . I am not getting what would be right key word to search for this ...initiate a call / calling out from asterisk etc etc are use less .. Please guide On Mon, Aug 24, 2009 at 11:30 AM, Matt Riddell li...@venturevoip.comwrote: On 24/08/09 6:27 PM, ABBAS SHAKEEL wrote: Hello I have a :) hmmm might need a little more info -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Dailing any number PSTN or MObile number
At 11:31 PM 8/23/2009, you wrote:
If i want to intiate a call to a PSTN number or any mobile number as
we normally do with our phones How can i do that with asterisk .
Unless I messed up this uses outgoing DHADI channel 1 to dial 10
digit local numbers with the required 1310 prefix when a 7 digit
number is dialed
exten = _NXXX,1,dial(DAHDI/1/1310${EXTEN})
Ira
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Re: [asterisk-users] Dailing any number PSTN or MObile number
Thanks Matt and Ira this worked for me... exten = 123,2,Dial(DAHDI/2/thephonenumber) Any body let me know . I want to know if i can say some thing. It do dail but dont say any thing . ie the bell rings at dailed number but when the call is picked nothing happens How can it say some thing if the call is picked by other end (the number dailed). -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Dailing any number PSTN or MObile number
On 24/08/09 7:31 PM, ABBAS SHAKEEL wrote: Thanks Matt and Ira this worked for me... exten = 123,2,Dial(DAHDI/2/thephonenumber) Any body let me know . I want to know if i can say some thing. It do dail but dont say any thing . ie the bell rings at dailed number but when the call is picked nothing happens How can it say some thing if the call is picked by other end (the number dailed). How are you dialing 123? You might want to replace that extension temporarily with exten = 123,2,Echo() And see if you can hear yourself when you talk -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Dailing any number PSTN or MObile number
I am not dailing 123 but i have changed it when shown on forums i just want to illustrate the dial function :) Thanks On Mon, Aug 24, 2009 at 12:37 PM, Matt Riddell li...@venturevoip.comwrote: On 24/08/09 7:31 PM, ABBAS SHAKEEL wrote: Thanks Matt and Ira this worked for me... exten = 123,2,Dial(DAHDI/2/thephonenumber) Any body let me know . I want to know if i can say some thing. It do dail but dont say any thing . ie the bell rings at dailed number but when the call is picked nothing happens How can it say some thing if the call is picked by other end (the number dailed). How are you dialing 123? You might want to replace that extension temporarily with exten = 123,2,Echo() And see if you can hear yourself when you talk -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Dailing any number PSTN or MObile number
On 24/08/09 7:45 PM, ABBAS SHAKEEL wrote: I am not dailing 123 but i have changed it when shown on forums i just want to illustrate the dial function :) Thanks Yeah, I just meant to try echo instead of dial - it should echo back to you what you send to it - that way you can check that your phone is working ok. Bear in mind that if the latency on the network is low, you might not be able to hear your voice. The other option would be to use the Record application to record something and then the Playback application to play it back. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Dailing any number PSTN or MObile number
Thanks Matt for the value able info Echo dont works for me :( One more thing what i want to do is making a call to some one and playing a sound like (I Shakeel is calling you ..if you want to be friend press 1 or ..) thanks -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Dailing any number PSTN or MObile number
On 24/08/09 8:07 PM, ABBAS SHAKEEL wrote: Thanks Matt for the value able info Echo dont works for me :( One more thing what i want to do is making a call to some one and playing a sound like (I Shakeel is calling you ..if you want to be friend press 1 or ..) Use the A(soundfile) option to the Dial command - for more info type: core show application dial -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Request Pending retransmitions
Hi, im trying to build a UAC and I'm coming up with some trouble whenever I receive a SIP 491 Request Pending Response. This happens because I try to place a call on hold using an Invite request rigth before Asterisk sends me a Re-Invite for the same call. I respond to the 491 response with an ACK however for some strange reason Asterisk doesn't accept the ACK and insists on retransmitting the 491 Response. Asterisk replies with the following 491 response: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 10.110.7.89:5070;branch=z9hG4bK5a668c33f196837c3602266b23b389e0;received=10.110.7.89 From: sip:30...@10.110.7.20:5070;tag=SIPTester To: sip:30...@10.110.7.20;tag=as2ea72122 Call-ID: 0dd43bb5a64eb5a2fb0114193821f...@10.110.7.89 CSeq: 5 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal And I send the following ACK: ACK sip:30...@10.110.7.20 SIP/2.0 Call-ID: 0dd43bb5a64eb5a2fb0114193821f...@10.110.7.89 Max-Forwards: 70 From: sip:30...@10.110.7.20:5070;tag=SIPTester To: sip:30...@10.110.7.20;tag=as2ea72122 Via: SIP/2.0/UDP 10.110.7.89:5070;branch=z9hG4bK5a668c33f196837c3602266b23b389e0 CSeq: 5 ACK Content-Length: 0 However this doesn't seem to be valid because Asterisk insists be resending the same 491 Response until it sends 6 messages and the decides to destroy the dialog: Aug 21 11:21:06 http://www.voip-info.org/boards/Aug%2021%2011:21:06 WARNING9686 http://www.voip-info.org/boards/9686 : chan_sip.c:1967 retrans_pkt: Maximum retries exceeded on transmission 0dd43bb5a64eb5a2fb0114193821f...@10.110.7.89 for seqno 5 (Critical Response) - See doc/sip-retransmit.txt. Aug 21 11:21:06 http://www.voip-info.org/boards/Aug%2021%2011:21:06 WARNING9686 http://www.voip-info.org/boards/9686 : chan_sip.c:1989 retrans_pkt: Hanging up call 0dd43bb5a64eb5a2fb0114193821f...@10.110.7.89 - no reply to our critical packet (see doc/sip-retransmit.txt). Does anyone have a clue of what it is I'm doing wrong? Do I have to send a CANCEL request of the hold's INVITE? Thanks in advance! Joaquín ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Core dump gets created while accessing voicemail
I am also having the same issue with voicemail. I have the exact setup you have described. Additionally, I am getting segmentation faults during an asterisk reload. This is happening intermittently. Below is the core dump backtrace. Not sure how to determine if it an asterisk or odbc issue. Program terminated with signal 11, Segmentation fault. #0 0x00322b417649 in SQLFreeEnv () from /usr/lib64/libodbc.so.1 (gdb) bt #0 0x00322b417649 in SQLFreeEnv () from /usr/lib64/libodbc.so.1 #1 0x00322b417b5c in SQLFreeHandle () from /usr/lib64/libodbc.so.1 #2 0x2aaabcd6187b in odbc_unload_module () from /usr/lib/asterisk/modules/cdr_odbc.so #3 0x2aaabcd619d9 in reload () from /usr/lib/asterisk/modules/cdr_odbc.so #4 0x004647ea in ast_module_reload (name=0x0) at loader.c:597 #5 0x00444713 in handle_reload_deprecated (fd=50, argc=0, argv=0x322b00f858) at cli.c:182 #6 0x00443a2d in ast_cli_command (fd=50, s=0x421ae7b9 reload) at cli.c:1992 #7 0x00472d06 in action_command (s=0x16a0f50, m=0x421ae810) at manager.c:1753 #8 0x0047135b in process_message (s=0x16a0f50, m=0x421ae810) at manager.c:2214 #9 0x0047281c in do_message (s=0x16a0f50) at manager.c:2310 #10 0x0047284d in session_do (data=value optimized out) at manager.c:2326 #11 0x004aef5c in dummy_start (data=value optimized out) at utils.c:912 #12 0x00322b006307 in start_thread () from /lib64/libpthread.so.0 #13 0x00322a4d1ded in clone () from /lib64/libc.so.6 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Core dump gets created while accessing voicemail
On 24/08/09 8:54 PM, John Riek wrote: I am also having the same issue with voicemail. I have the exact setup you have described. Additionally, I am getting segmentation faults during an asterisk reload. This is happening intermittently. Below is the core dump backtrace. Not sure how to determine if it an asterisk or odbc issue. Program terminated with signal 11, Segmentation fault. #0 0x00322b417649 in SQLFreeEnv () from /usr/lib64/libodbc.so.1 Looks like an issue with ODBC - Tilghman might disagree though :) Are you using the latest version? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Dailing any number PSTN or MObile number
thanks Alot it worked for me using *G(context^exten^pri)*: If the call is answered, transfer both parties to the specified context and extension. The calling party is transferred to priority x, and the called party to priority x+1. This allows the dialplan to distinguish between the calling and called legs of the call (new in v1.2). Thanks for help Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Core dump gets created while accessing voicemail
On 24/08/09 8:54 PM, John Riek wrote: I am also having the same issue with voicemail. I have the exact setup you have described. Additionally, I am getting segmentation faults during an asterisk reload. This is happening intermittently. Below is the core dump backtrace. Not sure how to determine if it an asterisk or odbc issue. Program terminated with signal 11, Segmentation fault. #0 0x00322b417649 in SQLFreeEnv () from /usr/lib64/libodbc.so.1 Looks like an issue with ODBC - Tilghman might disagree though :) Are you using the latest version? -- Cheers, Matt Riddell Director Matt, I am using the following configuration. centos 5.2 64bit unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 mysql-server-5.0.45-7.el5 asterisk 1.4.22.1 Hopefully upgrading to the latest 1.4 version of asterisk will fix it. Thanks, John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory
On Fri, 21 Aug 2009, Olivier wrote: So basically it's harmless, unless you actually have such a card. Yes, but as you mentioned, most don't have a transcoder card. My opinion is such message shouldn't be send at all for those environments where there is no transcoder card, (as it will remain, IMHO, normal behaviour to care about ERROR messages). I agree, or at worst reduce it to a message that just notifies that no transcoder card was found. When i see error i always think i did something wrong :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Core dump gets created while accessing voicemail
Hopefully upgrading to the latest 1.4 version of asterisk will fix it. Yeah, that'd be the best bet for the moment - if it doesn't you'll need to open a bug on http://issues.asterisk.org I'd make a backup of everything before you upgrade - just in case :) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] exchanging CDR data between Asterisk servers
Hi! I have the following setup: PSTN--Asterisk-SIP--Asterisk GW/LCR \ \ ... \ \ ... \ --SIP--Asterisk \ ... ---Asterisk The GW-Asterisk just does the gatewaying stuff and writes the CDRs for the billing system. The other Asterisk servers handle all the services (IVR, REGISTER, ...) Some scenarios require to write some additional data to the CDRs. For outgoing calls this is not a problem (I signal the extra data in a SIP header and set a CDR() variable in the GW asterisk). My problem are incoming calls. The extra data is only known to the service Asterisk, but the CDR is written by the GW Asterisk. Does anybody know a method how to signal the extra data during the call from the server Asterisk back to the GW Asterisk und put it into a CDR() variable? Regards Klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] problem on compiling asterisk-addons-1.6.2.0-rc1
hello, I tried to compil asterisk-addons-1.6.2.0-rc1, and I have that error: [CC] res_config_mysql.c - res_config_mysql.o res_config_mysql.c:1367: error: unknown field âupdate2_funcâ specified in initializer res_config_mysql.c: In function âparse_configâ: res_config_mysql.c:1432: error: âCONFIG_STATUS_FILEMISSINGâ undeclared (first use in this function) res_config_mysql.c:1432: error: (Each undeclared identifier is reported only once res_config_mysql.c:1432: error: for each function it appears in.) res_config_mysql.c:1436: error: âCONFIG_STATUS_FILEINVALIDâ undeclared (first use in this function) Here, all my commands to do it: apt-get install curl doxygen libnewt-dev mysql-client php5 php5-cli libmysqlclient15-dev libncurses5 libncurses5-dev openssl libssl-dev libssl0.9.8 mpg123 make g++ subversion subversion-tools newt-tcl linux-headers-`uname -r` php5-memcache php-pear DAHDI #cd /usr/src #wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linu x-complete-2.2.0.2+2.2.0.tar.gz http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux -complete-2.2.0.2+2.2.0.tar.gz [ http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linu x-complete-2.2.0.2+2.2.0.tar.gz ^] #tar zxfv dahdi-linux-complete-2.2.0.2+2.2.0.tar.gz #cd dahdi-linux-complete-2.2.0.2+2.2.0/ #make all make install make config LIBPRI #wget http://downloads.asterisk.org/pub/telephony/libpri/releases/libpri-1.4.10.t ar.gz http://downloads.asterisk.org/pub/telephony/libpri/releases/libpri-1.4.10.ta r.gz [ http://downloads.asterisk.org/pub/telephony/libpri/releases/libpri-1.4.10.t ar.gz ^] #tar zxfv libpri-1.4.10.tar.gz #cd libpri-1.4.10 #make make install ASTERISK #wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6. 1.5-rc1.tar.gz http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1 .5-rc1.tar.gz [ http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6. 1.5-rc1.tar.gz ^] #tar zxfv asterisk-1.6.1.5-rc1.tar.gz #ln -s asterisk-1.6.1.5-rc1 asterisk #cd asterisk #./configure make menuselect -- attention à vérifier que chan_dahdi soit sélectionné #make make install make samples #cp /usr/src/asterisk/contrib/init.d/rc.debian.asterisk /etc/init.d/asterisk #/usr/sbin/update-rc.d asterisk defaults 99 99 #groupadd asterisk #useradd asterisk -g asterisk #vim /etc/init.d/asterisk Décommenter : AST_USER=asterisk AST_GROUP=asterisk #vim /etc/asterisk/asterisk.conf Effacer le (!) Changer: astrundir = /var/run/asterisk #mkdir /var/run/asterisk #chown asterisk /var/run/asterisk/ #chown asterisk /etc/asterisk -R #chown asterisk /var/lib/asterisk -R #chmod 777 /var/log/asterisk #chown asterisk /usr/lib/asterisk/ -R #chown asterisk /var/spool/asterisk/ -R #chown asterisk /etc/dahdi -R #chown asterisk /lib/modules/2.6.26-2-686/dahdi -R #chown asterisk /usr/share/dahdi ADDONS #wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-addo ns-1.6.2.0-rc1.tar.gz http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-addon s-1.6.2.0-rc1.tar.gz [ http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-addo ns-1.6.2.0-rc1.tar.gz ^] #tar zxfv asterisk-addons-1.6.2.0-rc1.tar.gz #cd asterisk-addons-1.6.2.0-rc1 #./configure make menuselect make make install make samples thank you for help Cordialement, BERGANZ François cid:image001.gif@01C8F7CD.6BC1D2C0 http://www.acropolistelecom.net/ http://www.acropolistelecom.net P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Need to now my Asterisk User ID
From voip-info.org : Asterisk will look for these files in the /var/lib/asterisk/keys directory, so copy them there and make sure only the asterisk user id can read the keys and that no one can write over them. How do I know my Asterisk User ID ?? Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Follow me IVR sounds
Hellos, I am looking for the sounds used in this ivr example http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me. The one with 6900. Any assistance is welcome. -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] SIP doesn't recognize hangup
Hi at all !
I've a problem and I don't know how to solve it.
My configuration is the following:
ISDN LINE --- PATTON (SIP) --- ASTERISK
in asterisk my sip.conf for sip patton account is the following:
[general]
port=5060
bindaddr=0.0.0.0
context=default
language=it
limitonpeers=yes
notifyringing=yes
[acc1]
context=fromPSTN_Ext1
type=friend
qualifiy=yes
host=dynamic
username=acc1
secret=1234
qualify=yes
Now I want to receive a call on acc1 and then redirect it again on acc1
through PSTN, in the following way:
[fromPSTN_Ext1]
exten = _X.,1,Noop(start call and redirect call through PSTN)
exten = _X.,n,Background(${SoundsPath}/message)
exten = _X.,n,WaitExten(2)
exten = i,n,Monitor(wav,${MONITORFILENAME},m)
exten = i,n,Dial(SIP/numbertoc...@acc1,10,r)
ISDN LINE --- PATTON (SIP acc1) --- ASTERISK --- PATTON (SIP acc1) ---
ISDN line
But if the external caller hang up the call ... the call to NUMBERTOCALL on
acc1 continue to ring until the called answer, but the call is out.
Someone can help me ?!?!?
Thanks to all
Marco
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Re: [asterisk-users] Request Pending retransmitions
Are you using newest Asterisk versions? There were some similar problems fixed recently: https://issues.asterisk.org/view.php?id=13849 https://issues.asterisk.org/view.php?id=14239 https://issues.asterisk.org/view.php?id=14584 regards klaus Guillén Melo, Joaquin schrieb: Hi, im trying to build a UAC and I'm coming up with some trouble whenever I receive a SIP 491 Request Pending Response. This happens because I try to place a call on hold using an Invite request rigth before Asterisk sends me a Re-Invite for the same call. I respond to the 491 response with an ACK however for some strange reason Asterisk doesn't accept the ACK and insists on retransmitting the 491 Response. Asterisk replies with the following 491 response: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 10.110.7.89:5070;branch=z9hG4bK5a668c33f196837c3602266b23b389e0;received=10.110.7.89 From: sip:30...@10.110.7.20:5070;tag=SIPTester To: sip:30...@10.110.7.20;tag=as2ea72122 Call-ID: 0dd43bb5a64eb5a2fb0114193821f...@10.110.7.89 CSeq: 5 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal And I send the following ACK: ACK sip:30...@10.110.7.20 SIP/2.0 Call-ID: 0dd43bb5a64eb5a2fb0114193821f...@10.110.7.89 Max-Forwards: 70 From: sip:30...@10.110.7.20:5070;tag=SIPTester To: sip:30...@10.110.7.20;tag=as2ea72122 Via: SIP/2.0/UDP 10.110.7.89:5070;branch=z9hG4bK5a668c33f196837c3602266b23b389e0 CSeq: 5 ACK Content-Length: 0 However this doesn't seem to be valid because Asterisk insists be resending the same 491 Response until it sends 6 messages and the decides to destroy the dialog: Aug 21 11:21:06 http://www.voip-info.org/boards/Aug%2021%2011:21:06 WARNING9686 http://www.voip-info.org/boards/9686: chan_sip.c:1967 retrans_pkt: Maximum retries exceeded on transmission 0dd43bb5a64eb5a2fb0114193821f...@10.110.7.89 for seqno 5 (Critical Response) — See doc/sip-retransmit.txt. Aug 21 11:21:06 http://www.voip-info.org/boards/Aug%2021%2011:21:06 WARNING9686 http://www.voip-info.org/boards/9686: chan_sip.c:1989 retrans_pkt: Hanging up call 0dd43bb5a64eb5a2fb0114193821f...@10.110.7.89 - no reply to our critical packet (see doc/sip-retransmit.txt). Does anyone have a clue of what it is I'm doing wrong? Do I have to send a CANCEL request of the hold's INVITE? Thanks in advance! **Joaquín ** ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Show queue-name near the callerId
Hi, I want to show the QueueName to my Queue Member. I try to find the solution to show the QueueName near the callerid of each call. Can somebody help me ? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Show queue-name near the callerId
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Thalassoline - Service technique wrote:
I want to show the QueueName to my Queue Member.
I try to find the solution to show the QueueName near the callerid of
each call.
Can somebody help me ?
Set the CALLERID(name) variable prior to sending the call to the Queue.
e.g.:
exten = s,1,Set(CALLERID(name)=${QueueName}${CALLERID(name)});
exten = s,n,Queue(${QueueName});
Adjust to taste...
Barry
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Version: GnuPG v1.4.5 (GNU/Linux)
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Re: [asterisk-users] Follow me IVR sounds
Since the tutorial is 4+ years old and the Wiki author wasn't nice enough to include the source for /var/lib/asterisk/sounds/portable-number-ivr, the simplest solution I can offer is to either record these sounds yourself or to install swift and pipe out these files per the quoted values. Since my swift isn't licensed, I can't create them without the 8 second nag on front. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Mutuku Sent: Monday, August 24, 2009 6:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Follow me IVR sounds Hellos, I am looking for the sounds used in this ivr example http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me. The one with 6900. Any assistance is welcome. -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Problems sending voicemail emails
Hi everybody, I'm trying my Asterisk to send emails when a new message arribes to a voicemail user but no email arribes. my voicemail configuration is the following: VOICEMAIL.CONF: [general] format=wav serveremail=aster...@mydomain.com attach=yes maxmsg=20 maxsecs=180 minsecs=3 maxsilence=10 silencethreshold=128 maxlogins=3 fromstring=My Asterisk When I look at maillog file, this is what I get: * n7OCivth003603: from=root, size=5340, class=0, nrcpts=1, msgid= asterisk-1-227856683-222-3...@myserver, relay=r...@localhost * n7OCiw9W003604: from=r...@localhost.localdomain, size=5473, class=0, nrcpts=1, msgid=asterisk-1-227856683-222-3...@prosima, proto=ESMTP, daemon=MTA, relay=MYSERVER [127.0.0.1] * n7OCivth003603: to=Test User 1 testu...@mydomain.com, ctladdr=root (0/0), delay=00:00:06, xdelay=00:00:05, mailer=relay, pri=35340, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (n7OCiw9W003604 Message accepted for delivery) * n7OCiw9W003604: to=testu...@mydomain.com, ctladdr= r...@localhost.localdomain (0/0), delay=00:00:01, xdelay=00:00:01, mailer=esmtp, pri=125473, relay=mx1.datagrama.net. [212.9.65.111], dsn=5.1.8, stat=User unknown * n7OCiw9W003604: n7OCj49W003606: DSN: User unknown * n7OCj49W003606: to=r...@localhost.localdomain, delay=00:00:01, xdelay=00:00:00, mailer=local, pri=36710, dsn=2.0.0, stat=Sent I don't understand why it looks as the message is been sent (Message accepted for delivery) but then I get the message dsn=5.1.8, stat=User unknown and fiinally I get the message Sent but I don't receive any email. do I have to change any configuration? Many thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] how to install asterisk
I have a small script that do the trick for you. At you terminal use the follow wget http://www.fastway.com.br/wp-content/uploads/2009/02/install_asterisk.sh chmod 755 install_asterisk.sh sudo ./install_asterisk.sh Valter 2009/8/21 aster...@opensourcesolution.in hello friends, i have to configures asterisk n my hardware details are O.S - Ubuntu 8.04 Lts Memory - 1 GB Proccessor- core 2 duo is any one having a good link or how to related asterisk. any help,support will be higly appreciated thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
Start with simple mail testing (forget asterisk) Does mx1.datagrama.net accept messages for testu...@mydomain.com ? Try a telnet session first... _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 9:56 AM To: Asterisk Users List Subject: [asterisk-users] Problems sending voicemail emails Hi everybody, I'm trying my Asterisk to send emails when a new message arribes to a voicemail user but no email arribes. my voicemail configuration is the following: VOICEMAIL.CONF: [general] format=wav serveremail=aster...@mydomain.com attach=yes maxmsg=20 maxsecs=180 minsecs=3 maxsilence=10 silencethreshold=128 maxlogins=3 fromstring=My Asterisk When I look at maillog file, this is what I get: * n7OCivth003603: from=root, size=5340, class=0, nrcpts=1, msgid=asterisk-1-227856683-222-3...@myserver, relay=r...@localhost * n7OCiw9W003604: from=r...@localhost.localdomain, size=5473, class=0, nrcpts=1, msgid=asterisk-1-227856683-222-3...@prosima, proto=ESMTP, daemon=MTA, relay=MYSERVER [127.0.0.1] * n7OCivth003603: to=Test User 1 testu...@mydomain.com, ctladdr=root (0/0), delay=00:00:06, xdelay=00:00:05, mailer=relay, pri=35340, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (n7OCiw9W003604 Message accepted for delivery) * n7OCiw9W003604: to=testu...@mydomain.com, ctladdr=r...@localhost.localdomain (0/0), delay=00:00:01, xdelay=00:00:01, mailer=esmtp, pri=125473, relay=mx1.datagrama.net. [212.9.65.111], dsn=5.1.8, stat=User unknown * n7OCiw9W003604: n7OCj49W003606: DSN: User unknown * n7OCj49W003606: to=r...@localhost.localdomain, delay=00:00:01, xdelay=00:00:00, mailer=local, pri=36710, dsn=2.0.0, stat=Sent I don't understand why it looks as the message is been sent (Message accepted for delivery) but then I get the message dsn=5.1.8, stat=User unknown and fiinally I get the message Sent but I don't receive any email. do I have to change any configuration? Many thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] asterisk 1.6.1.1 + Asterisk GUI v2.0
Hi Has anyone already use asterisk 1.6.1.1 with asterisk GUI last release ? I'm trying it but I have this problem : Just after I logged, I have system status main page but no other links where I can click to go to other pages (remember left panel!) Regards Harry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
Make sure regular sendmail is working; asterisk voicemail uses this unless you reconfigure it. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Monday, August 24, 2009 9:09 AM To: 'Asterisk Users List' Subject: Re: [asterisk-users] Problems sending voicemail emails Start with simple mail testing (forget asterisk) Does mx1.datagrama.net accept messages for testu...@mydomain.com ? Try a telnet session first... _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 9:56 AM To: Asterisk Users List Subject: [asterisk-users] Problems sending voicemail emails Hi everybody, I'm trying my Asterisk to send emails when a new message arribes to a voicemail user but no email arribes. my voicemail configuration is the following: VOICEMAIL.CONF: [general] format=wav serveremail=aster...@mydomain.com attach=yes maxmsg=20 maxsecs=180 minsecs=3 maxsilence=10 silencethreshold=128 maxlogins=3 fromstring=My Asterisk When I look at maillog file, this is what I get: * n7OCivth003603: from=root, size=5340, class=0, nrcpts=1, msgid=asterisk-1-227856683-222-3...@myserver, relay=r...@localhost * n7OCiw9W003604: from=r...@localhost.localdomain, size=5473, class=0, nrcpts=1, msgid=asterisk-1-227856683-222-3...@prosima, proto=ESMTP, daemon=MTA, relay=MYSERVER [127.0.0.1] * n7OCivth003603: to=Test User 1 testu...@mydomain.com, ctladdr=root (0/0), delay=00:00:06, xdelay=00:00:05, mailer=relay, pri=35340, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (n7OCiw9W003604 Message accepted for delivery) * n7OCiw9W003604: to=testu...@mydomain.com, ctladdr=r...@localhost.localdomain (0/0), delay=00:00:01, xdelay=00:00:01, mailer=esmtp, pri=125473, relay=mx1.datagrama.net. [212.9.65.111], dsn=5.1.8, stat=User unknown * n7OCiw9W003604: n7OCj49W003606: DSN: User unknown * n7OCj49W003606: to=r...@localhost.localdomain, delay=00:00:01, xdelay=00:00:00, mailer=local, pri=36710, dsn=2.0.0, stat=Sent I don't understand why it looks as the message is been sent (Message accepted for delivery) but then I get the message dsn=5.1.8, stat=User unknown and fiinally I get the message Sent but I don't receive any email. do I have to change any configuration? Many thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
Hi Michelle, If I try telnet mx1.datagrama.net I have no answer, I get: Trying 212.9.65.110... ¿? 2009/8/24 Michelle Dupuis supp...@ocg.ca Start with simple mail testing (forget asterisk) Does mx1.datagrama.net accept messages for testu...@mydomain.com ? Try a telnet session first... -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Joan Antoni Terre *Sent:* Monday, August 24, 2009 9:56 AM *To:* Asterisk Users List *Subject:* [asterisk-users] Problems sending voicemail emails Hi everybody, I'm trying my Asterisk to send emails when a new message arribes to a voicemail user but no email arribes. my voicemail configuration is the following: VOICEMAIL.CONF: [general] format=wav serveremail=aster...@mydomain.com attach=yes maxmsg=20 maxsecs=180 minsecs=3 maxsilence=10 silencethreshold=128 maxlogins=3 fromstring=My Asterisk When I look at maillog file, this is what I get: * n7OCivth003603: from=root, size=5340, class=0, nrcpts=1, msgid= asterisk-1-227856683-222-3...@myserver, relay=r...@localhost * n7OCiw9W003604: from=r...@localhost.localdomain, size=5473, class=0, nrcpts=1, msgid=asterisk-1-227856683-222-3...@prosima, proto=ESMTP, daemon=MTA, relay=MYSERVER [127.0.0.1] * n7OCivth003603: to=Test User 1 testu...@mydomain.com, ctladdr=root (0/0), delay=00:00:06, xdelay=00:00:05, mailer=relay, pri=35340, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (n7OCiw9W003604 Message accepted for delivery) * n7OCiw9W003604: to=testu...@mydomain.com, ctladdr= r...@localhost.localdomain (0/0), delay=00:00:01, xdelay=00:00:01, mailer=esmtp, pri=125473, relay=mx1.datagrama.net. [212.9.65.111], dsn=5.1.8, stat=User unknown * n7OCiw9W003604: n7OCj49W003606: DSN: User unknown * n7OCj49W003606: to=r...@localhost.localdomain, delay=00:00:01, xdelay=00:00:00, mailer=local, pri=36710, dsn=2.0.0, stat=Sent I don't understand why it looks as the message is been sent (Message accepted for delivery) but then I get the message dsn=5.1.8, stat=User unknown and fiinally I get the message Sent but I don't receive any email. do I have to change any configuration? Many thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
2009/8/24 Danny Nicholas da...@debsinc.com Danny, sorry if it's a silly question but how can I check if sendmail is working' Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
On Mon, Aug 24, 2009 at 9:25 AM, Joan Antoni Terrenebh...@gmail.com wrote: Hi Michelle, If I try telnet mx1.datagrama.net I have no answer, I get: Trying 212.9.65.110... ¿? telnet mx1.datagrama.net 25 that's a space, then the port, in this case, 25. is ms1.datagrama.net what you really want though? It looks like you're using mydomain.com as the domain in your asterisk configuration. Do you really intend to use mydomain.com ? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
You also need to specify the port so telnet mx1.datagrama.net 25 return is the command to use. db From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems sending voicemail emails Hi Michelle, If I try telnet mx1.datagrama.net I have no answer, I get: Trying 212.9.65.110... ¿? 2009/8/24 Michelle Dupuis supp...@ocg.ca Start with simple mail testing (forget asterisk) Does mx1.datagrama.net http://mx1.datagrama.net/ accept messages for testu...@mydomain.com ? Try a telnet session first... _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 9:56 AM To: Asterisk Users List Subject: [asterisk-users] Problems sending voicemail emails Hi everybody, I'm trying my Asterisk to send emails when a new message arribes to a voicemail user but no email arribes. my voicemail configuration is the following: VOICEMAIL.CONF: [general] format=wav serveremail=aster...@mydomain.com attach=yes maxmsg=20 maxsecs=180 minsecs=3 maxsilence=10 silencethreshold=128 maxlogins=3 fromstring=My Asterisk When I look at maillog file, this is what I get: * n7OCivth003603: from=root, size=5340, class=0, nrcpts=1, msgid=asterisk-1-227856683-222-3...@myserver, relay=r...@localhost * n7OCiw9W003604: from=r...@localhost.localdomain, size=5473, class=0, nrcpts=1, msgid=asterisk-1-227856683-222-3...@prosima, proto=ESMTP, daemon=MTA, relay=MYSERVER [127.0.0.1] * n7OCivth003603: to=Test User 1 testu...@mydomain.com, ctladdr=root (0/0), delay=00:00:06, xdelay=00:00:05, mailer=relay, pri=35340, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (n7OCiw9W003604 Message accepted for delivery) * n7OCiw9W003604: to=testu...@mydomain.com, ctladdr=r...@localhost.localdomain (0/0), delay=00:00:01, xdelay=00:00:01, mailer=esmtp, pri=125473, relay=mx1.datagrama.net http://mx1.datagrama.net/ . [212.9.65.111], dsn=5.1.8, stat=User unknown * n7OCiw9W003604: n7OCj49W003606: DSN: User unknown * n7OCj49W003606: to=r...@localhost.localdomain, delay=00:00:01, xdelay=00:00:00, mailer=local, pri=36710, dsn=2.0.0, stat=Sent I don't understand why it looks as the message is been sent (Message accepted for delivery) but then I get the message dsn=5.1.8, stat=User unknown and fiinally I get the message Sent but I don't receive any email. do I have to change any configuration? Many thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Need to now my Asterisk User ID
Hi Jonas Type 'id asterisk' at your command line. It should return uid gid and all groups the asterisk user belongs to. Cheers Bails jonas kellens wrote: From voip-info.org : /Asterisk will look for these files in the /var/lib/asterisk/keys directory, so copy them there and make sure only the asterisk user id can read the keys and that no one can write over them./ How do I know my Asterisk User ID ?? Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
Danny, I've done as you explained and get the same messages at /var/log/maillog What I've seen at /var/spool/mail/root the following Subject: Returned mail: see transcript for details Auto-Submitted: auto-generated (failure) This is a MIME-encapsulated message --n7OCj49W003606.1251117904/localhost.localdomain The original message was received at Mon, 24 Aug 2009 14:45:03 +0200 from MYSERVER [127.0.0.1] - The following addresses had permanent fatal errors - j.gall...@prosima.es (reason: 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found) - Transcript of session follows - ... while talking to mx1.datagrama.net.: DATA 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found 550 5.1.1 testu...@mydomain.com... User unknown 554 5.5.1 Error: no valid recipients --n7OCj49W003606.1251117904/localhost.localdomain Content-Type: message/delivery-status Reporting-MTA: dns; localhost.localdomain Received-From-MTA: DNS; MYSERVER Arrival-Date: Mon, 24 Aug 2009 14:45:03 +0200 Final-Recipient: RFC822; testu...@mydomain.com Action: failed Status: 5.1.8 Remote-MTA: DNS; mx1.datagrama.net Diagnostic-Code: SMTP; 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found Last-Attempt-Date: Mon, 24 Aug 2009 14:45:04 +0200 --n7OCj49W003606.1251117904/localhost.localdomain Content-Type: message/rfc822 Return-Path: r...@localhost.localdomain Received: from localhost.localdomain (MYSERVER [127.0.0.1]) by localhost.localdomain (8.13.8/8.13.8) with ESMTP id n7OCiw9W003604 for testu...@mydomain.com; Mon, 24 Aug 2009 14:45:03 +0200 Received: (from r...@localhost) by localhost.localdomain (8.13.8/8.13.8/Submit) id n7OCivth003603; Mon, 24 Aug 2009 14:44:57 +0200 Date: Mon, 24 Aug 2009 14:44:57 +0200 From: =?ISO-8859-1?Q?=22Asterisk=22?= aster...@mydomain.com To: Test User testu...@mydomain.com Subject: =?ISO-8859-1?Q?=5BPBX=5DRecibido_mensaje_numero_1_en_su_buzon_de?= =?ISO-8859-1?Q?_voz?= Message-ID: asterisk-1-227856683-222-3...@myserver X-Asterisk-CallerID: 222 X-Asterisk-CallerIDName: Test User MIME-Version: 1.0 Content-Type: multipart/mixed; boundary=voicemail_122234791383843144 This is a multi-part message in MIME format. --voicemail_122234791383843144 Content-Type: text/plain; charset=ISO-8859-1 Content-Transfer-Encoding: 8bit Notice: Diagnostic-Code: SMTP; 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found By configuring serveremail = aster...@mydomain.com, sendmail should use this email adress as sender, is it right? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
Not silly at all. Voicemail.conf specifies that /usr/bin/sendmail -t will be used to send the mail, so doing this command: Sendmail -t To: nebh...@gmail.com followed by enter and ctrl-d should pop an empty email from root into your inbox if all is well. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems sending voicemail emails 2009/8/24 Danny Nicholas da...@debsinc.com Danny, sorry if it's a silly question but how can I check if sendmail is working' Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
Jonathan, now I've done telnet mx1.datagrama.net 25 And I've got: Trying 212.9.65.110... Connected to mx1.datagrama.net (212.9.65.110). Escape character is '^]'. 220 mailhub03.datagrama.net ESMTP Datagrama It looks as it has connected but has not asked for any user / Password mx1.datagrama.net is my ISP ESMT server. 2009/8/24 Jonathan Moore supermegat...@gmail.com On Mon, Aug 24, 2009 at 9:25 AM, Joan Antoni Terrenebh...@gmail.com wrote: Hi Michelle, If I try telnet mx1.datagrama.net I have no answer, I get: Trying 212.9.65.110... ¿? telnet mx1.datagrama.net 25 that's a space, then the port, in this case, 25. is ms1.datagrama.net what you really want though? It looks like you're using mydomain.com as the domain in your asterisk configuration. Do you really intend to use mydomain.com ? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
The receiving server does not ask for any user id or password. The protocal says, the sender has to just send the user or pass command with the data required. Try reading /var/log/mail(if you have access), at least that's where the outgoing mail logs on my servers are. Lyle Joan Antoni Terre wrote: Jonathan, now I've done telnet mx1.datagrama.net http://mx1.datagrama.net 25 And I've got: Trying 212.9.65.110... Connected to mx1.datagrama.net http://mx1.datagrama.net (212.9.65.110). Escape character is '^]'. 220 mailhub03.datagrama.net http://mailhub03.datagrama.net ESMTP Datagrama It looks as it has connected but has not asked for any user / Password mx1.datagrama.net http://mx1.datagrama.net is my ISP ESMT server. 2009/8/24 Jonathan Moore supermegat...@gmail.com mailto:supermegat...@gmail.com On Mon, Aug 24, 2009 at 9:25 AM, Joan Antoni Terrenebh...@gmail.com mailto:nebh...@gmail.com wrote: Hi Michelle, If I try telnet mx1.datagrama.net http://mx1.datagrama.net/ I have no answer, I get: Trying 212.9.65.110... ¿? telnet mx1.datagrama.net http://mx1.datagrama.net/ 25 that's a space, then the port, in this case, 25. is ms1.datagrama.net http://ms1.datagrama.net/ what you really want though? It looks like you're using mydomain.com http://mydomain.com/ as the domain in your asterisk configuration. Do you really intend to use mydomain.com http://mydomain.com/ ? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
Danny, my sendmail.cf is not at /etc but at /etc/mail. These lines you mention are not at the end of this file. Enclosed you'll find my /etc/mail/sendmail.cf Many thanks 2009/8/24 Danny Nicholas da...@debsinc.com In /etc/sendmail.cf, do you have these lines at the end? Tuser Asterisk Tuser asterisk -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Joan Antoni Terre *Sent:* Monday, August 24, 2009 9:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Problems sending voicemail emails Danny, I've done as you explained and get the same messages at /var/log/maillog What I've seen at /var/spool/mail/root the following Subject: Returned mail: see transcript for details Auto-Submitted: auto-generated (failure) This is a MIME-encapsulated message --n7OCj49W003606.1251117904/localhost.localdomain The original message was received at Mon, 24 Aug 2009 14:45:03 +0200 from MYSERVER [127.0.0.1] - The following addresses had permanent fatal errors - j.gall...@prosima.es (reason: 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found) - Transcript of session follows - ... while talking to mx1.datagrama.net.: DATA 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found 550 5.1.1 testu...@mydomain.com... User unknown 554 5.5.1 Error: no valid recipients --n7OCj49W003606.1251117904/localhost.localdomain Content-Type: message/delivery-status Reporting-MTA: dns; localhost.localdomain Received-From-MTA: DNS; MYSERVER Arrival-Date: Mon, 24 Aug 2009 14:45:03 +0200 Final-Recipient: RFC822; testu...@mydomain.com Action: failed Status: 5.1.8 Remote-MTA: DNS; mx1.datagrama.net Diagnostic-Code: SMTP; 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found Last-Attempt-Date: Mon, 24 Aug 2009 14:45:04 +0200 --n7OCj49W003606.1251117904/localhost.localdomain Content-Type: message/rfc822 Return-Path: r...@localhost.localdomain Received: from localhost.localdomain (MYSERVER [127.0.0.1]) by localhost.localdomain (8.13.8/8.13.8) with ESMTP id n7OCiw9W003604 for testu...@mydomain.com; Mon, 24 Aug 2009 14:45:03 +0200 Received: (from r...@localhost) by localhost.localdomain (8.13.8/8.13.8/Submit) id n7OCivth003603; Mon, 24 Aug 2009 14:44:57 +0200 Date: Mon, 24 Aug 2009 14:44:57 +0200 From: =?ISO-8859-1?Q?=22Asterisk=22?= aster...@mydomain.com To: Test User testu...@mydomain.com Subject: =?ISO-8859-1?Q?=5BPBX=5DRecibido_mensaje_numero_1_en_su_buzon_de?= =?ISO-8859-1?Q?_voz?= Message-ID: asterisk-1-227856683-222-3...@myserver X-Asterisk-CallerID: 222 X-Asterisk-CallerIDName: Test User MIME-Version: 1.0 Content-Type: multipart/mixed; boundary=voicemail_122234791383843144 This is a multi-part message in MIME format. --voicemail_122234791383843144 Content-Type: text/plain; charset=ISO-8859-1 Content-Transfer-Encoding: 8bit Notice: Diagnostic-Code: SMTP; 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found By configuring serveremail = aster...@mydomain.com, sendmail should use this email adress as sender, is it right? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 sendmail.mc Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] asterisk 1.6.1.1 + Asterisk GUI v2.0
harry R wrote: Hi Has anyone already use asterisk 1.6.1.1 with asterisk GUI last release ? I'm trying it but I have this problem : Just after I logged, I have system status main page but no other links where I can click to go to other pages (remember left panel!) I have Asterisk 1.6.1.4 and GUI 2.0 (Latest). I never had such problem (my main Asterisk server is at 1.6.0.6 with latest GUI as well). I would reinstall the GUI. But I can tell you it *SHOULD* work. Oh! While I'm writing this mail. I just looked at one of my client's Asterisk: 1.6.1.1 with GUI 2.0, brand new install. It's working great (unless we're takling about controlling a TDM400P, for which nobody seem to have an answer for me :-( -- Christian... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
In /etc/sendmail.cf, do you have these lines at the end? Tuser Asterisk Tuser asterisk _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems sending voicemail emails Danny, I've done as you explained and get the same messages at /var/log/maillog What I've seen at /var/spool/mail/root the following Subject: Returned mail: see transcript for details Auto-Submitted: auto-generated (failure) This is a MIME-encapsulated message --n7OCj49W003606.1251117904/localhost.localdomain The original message was received at Mon, 24 Aug 2009 14:45:03 +0200 from MYSERVER [127.0.0.1] - The following addresses had permanent fatal errors - j.gall...@prosima.es (reason: 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found) - Transcript of session follows - ... while talking to mx1.datagrama.net.: DATA 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found 550 5.1.1 testu...@mydomain.com... User unknown 554 5.5.1 Error: no valid recipients --n7OCj49W003606.1251117904/localhost.localdomain Content-Type: message/delivery-status Reporting-MTA: dns; localhost.localdomain Received-From-MTA: DNS; MYSERVER Arrival-Date: Mon, 24 Aug 2009 14:45:03 +0200 Final-Recipient: RFC822; testu...@mydomain.com Action: failed Status: 5.1.8 Remote-MTA: DNS; mx1.datagrama.net Diagnostic-Code: SMTP; 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found Last-Attempt-Date: Mon, 24 Aug 2009 14:45:04 +0200 --n7OCj49W003606.1251117904/localhost.localdomain Content-Type: message/rfc822 Return-Path: r...@localhost.localdomain Received: from localhost.localdomain (MYSERVER [127.0.0.1]) by localhost.localdomain (8.13.8/8.13.8) with ESMTP id n7OCiw9W003604 for testu...@mydomain.com; Mon, 24 Aug 2009 14:45:03 +0200 Received: (from r...@localhost) by localhost.localdomain (8.13.8/8.13.8/Submit) id n7OCivth003603; Mon, 24 Aug 2009 14:44:57 +0200 Date: Mon, 24 Aug 2009 14:44:57 +0200 From: =?ISO-8859-1?Q?=22Asterisk=22?= aster...@mydomain.com To: Test User testu...@mydomain.com Subject: =?ISO-8859-1?Q?=5BPBX=5DRecibido_mensaje_numero_1_en_su_buzon_de?= =?ISO-8859-1?Q?_voz?= Message-ID: asterisk-1-227856683-222-3...@myserver X-Asterisk-CallerID: 222 X-Asterisk-CallerIDName: Test User MIME-Version: 1.0 Content-Type: multipart/mixed; boundary=voicemail_122234791383843144 This is a multi-part message in MIME format. --voicemail_122234791383843144 Content-Type: text/plain; charset=ISO-8859-1 Content-Transfer-Encoding: 8bit Notice: Diagnostic-Code: SMTP; 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found By configuring serveremail = aster...@mydomain.com, sendmail should use this email adress as sender, is it right? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Core dump gets created while accessing voicemail
On Monday 24 August 2009 04:44:18 am Matt Riddell wrote: Hopefully upgrading to the latest 1.4 version of asterisk will fix it. Yeah, that'd be the best bet for the moment - if it doesn't you'll need to open a bug on http://issues.asterisk.org I'd make a backup of everything before you upgrade - just in case :) Actually, I'd say that the best bet is upgrading UnixODBC to 2.2.14 and MySQL-Connector-ODBC to the latest (whatever source version is on the MySQL site) is the best bet. This is a crash within ODBC code, and only a modification of the ODBC code is likely to fix it. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Need to now my Asterisk User ID
Bails, thanks for your reply. bash-3.2# id asterisk id: asterisk: No such user So how do I know my Asterisk User ID ?? Greetingz, Jonas. On Mon, 2009-08-24 at 15:49 +0100, bails wrote: Hi Jonas Type 'id asterisk' at your command line. It should return uid gid and all groups the asterisk user belongs to. Cheers Bails ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] problem on compiling asterisk-addons-1.6.2.0-rc1
On Monday 24 August 2009 05:17:24 am BERGANZ François wrote: I tried to compil asterisk-addons-1.6.2.0-rc1, and I have that error: [CC] res_config_mysql.c - res_config_mysql.o res_config_mysql.c:1367: error: unknown field âupdate2_funcâ specified in initializer res_config_mysql.c: In function âparse_configâ: res_config_mysql.c:1432: error: âCONFIG_STATUS_FILEMISSINGâ undeclared (first use in this function) res_config_mysql.c:1432: error: (Each undeclared identifier is reported only once res_config_mysql.c:1432: error: for each function it appears in.) res_config_mysql.c:1436: error: âCONFIG_STATUS_FILEINVALIDâ undeclared (first use in this function) You need to have Asterisk 1.6.2 beta already installed on your system before attempting to compile the 1.6.2 addons. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
This one is getting a little over my pay grade. I'd try adding the two lines at the end and restarting the mail daemon. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems sending voicemail emails Danny, my sendmail.cf is not at /etc but at /etc/mail. These lines you mention are not at the end of this file. Enclosed you'll find my /etc/mail/sendmail.cf Many thanks 2009/8/24 Danny Nicholas da...@debsinc.com In /etc/sendmail.cf http://sendmail.cf/ , do you have these lines at the end? Tuser Asterisk Tuser asterisk _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems sending voicemail emails Danny, I've done as you explained and get the same messages at /var/log/maillog What I've seen at /var/spool/mail/root the following Subject: Returned mail: see transcript for details Auto-Submitted: auto-generated (failure) This is a MIME-encapsulated message --n7OCj49W003606.1251117904/localhost.localdomain The original message was received at Mon, 24 Aug 2009 14:45:03 +0200 from MYSERVER [127.0.0.1] - The following addresses had permanent fatal errors - j.gall...@prosima.es (reason: 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found) - Transcript of session follows - ... while talking to mx1.datagrama.net http://mx1.datagrama.net/ .: DATA 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found 550 5.1.1 testu...@mydomain.com... User unknown 554 5.5.1 Error: no valid recipients --n7OCj49W003606.1251117904/localhost.localdomain Content-Type: message/delivery-status Reporting-MTA: dns; localhost.localdomain Received-From-MTA: DNS; MYSERVER Arrival-Date: Mon, 24 Aug 2009 14:45:03 +0200 Final-Recipient: RFC822; testu...@mydomain.com Action: failed Status: 5.1.8 Remote-MTA: DNS; mx1.datagrama.net http://mx1.datagrama.net/ Diagnostic-Code: SMTP; 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found Last-Attempt-Date: Mon, 24 Aug 2009 14:45:04 +0200 --n7OCj49W003606.1251117904/localhost.localdomain Content-Type: message/rfc822 Return-Path: r...@localhost.localdomain Received: from localhost.localdomain (MYSERVER [127.0.0.1]) by localhost.localdomain (8.13.8/8.13.8) with ESMTP id n7OCiw9W003604 for testu...@mydomain.com; Mon, 24 Aug 2009 14:45:03 +0200 Received: (from r...@localhost) by localhost.localdomain (8.13.8/8.13.8/Submit) id n7OCivth003603; Mon, 24 Aug 2009 14:44:57 +0200 Date: Mon, 24 Aug 2009 14:44:57 +0200 From: =?ISO-8859-1?Q?=22Asterisk=22?= aster...@mydomain.com To: Test User testu...@mydomain.com Subject: =?ISO-8859-1?Q?=5BPBX=5DRecibido_mensaje_numero_1_en_su_buzon_de?= =?ISO-8859-1?Q?_voz?= Message-ID: asterisk-1-227856683-222-3...@myserver X-Asterisk-CallerID: 222 X-Asterisk-CallerIDName: Test User MIME-Version: 1.0 Content-Type: multipart/mixed; boundary=voicemail_122234791383843144 This is a multi-part message in MIME format. --voicemail_122234791383843144 Content-Type: text/plain; charset=ISO-8859-1 Content-Transfer-Encoding: 8bit Notice: Diagnostic-Code: SMTP; 550 5.1.8 r...@localhost.localdomain: Sender address rejected: Domain not found By configuring serveremail = aster...@mydomain.com, sendmail should use this email adress as sender, is it right? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] how to install asterisk
2009/8/21 aster...@opensourcesolution.in
i have to configures asterisk n my hardware details are
Is it just me, or would you think someone from a domain named like
Open Source Solution should be able to figure this one out...
On Mon, 24 Aug 2009, Valter Nogueira wrote:
I have a small script that do the trick for you.
http://www.fastway.com.br/wp-content/uploads/2009/02/install_asterisk.sh
Just a suggestion...
If you define the version numbers as variables your script will be
easier to maintain. For example:
ADDONS_VERSION=1.4.7
ASTERISK_VERSION=1.4.23.1
LIBPRI_VERSION=1.4.9
ZAPTEL_VERSION=1.4.12
cd /usr/src
wget
http://downloads.digium.com/pub/libpri/releases/libpri-${LIBPRI_VERSION}.tar.gz
wget
http://downloads.digium.com/pub/zaptel/releases/zaptel-${ZAPTEL_VERSION}.tar.gz
wget
http://downloads.digium.com/pub/asterisk/releases/asterisk-${ASTERISK_VERSION}.tar.gz
wget
http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-${ADDONS_VERSION}.tar.gz
tar -zxvf libpri-${LIBPRI_VERSION}.tar.gz
tar -zxvf zaptel-${ZAPTEL_VERSION}.tar.gz
tar -zxvf asterisk-${ASTERISK_VERSION}.tar.gz
tar -zxvf asterisk-addons-${ADDONS_VERSION}.tar.gz
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
___
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Re: [asterisk-users] Problems sending voicemail emails
Hi Danny, I've tryed it but still the same. It looks that my ISP SMTP server doesn't like the sender, which I guess is r...@localhost.localdomain. Do you know how to change this sender? Many thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Need to now my Asterisk User ID
Un-top-posting... jonas kellens wrote: How do I know my Asterisk User ID ?? On Mon, 24 Aug 2009, bails wrote: Type 'id asterisk' at your command line. It should return uid gid and all groups the asterisk user belongs to. This assumes you have a user named asterisk. Also assumes that Asterisk is running as the user named asterisk. There's probably a more proper way, but this works: ~$ ps -ef | grep /sbin/asterisk | grep -v grep You should get something like: root 12477 12476 0 Aug03 ?00:02:09 /usr/sbin/asterisk -f -g -n -p -q -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Autodial not waiting for voicemail
Hi All - I'm setting up a corporate emergency broadcast system that uses an autodialer to contact all company employees. Everything works fine except if the auto-dialed calls go to the end users' voicemail. If that happens, asterisk starts playback of the emergency message while the voicemail system on the other end is playing its outgoing message. The result is that the beginning (or all) of my emergency message is clipped off. It seems like overkill to try and use DSP to detect if the call has reached voicemail (detect the beep?), but I can't think of any other reasonable way to get the full message to the end users' voicemail. I guess I could prepend a welcome message just to kill some time while the user's outgoing greeting is playing, but that's still somewhat unreliable, especially if the user has a long outgoing message. Has anybody found a way to deal with this? Thanks! Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
That's what the MASQUERADE(localhost.localdomain) is supposed to do; Don't know why it is not. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems sending voicemail emails Hi Danny, I've tryed it but still the same. It looks that my ISP SMTP server doesn't like the sender, which I guess is r...@localhost.localdomain. Do you know how to change this sender? Many thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] install the digium card TC400P howto?
Hello, I need help to install a digium card TC400P. I compiled the dahdi source, but dahdi dont find the card! debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# lspci 11:03.0 Ethernet controller: Digium, Inc. Wildcard TC400P transcoder base card (rev 11) debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# /etc/init.d/dahdi restart Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp: done wcfxo: done wctdm: done wcb4xxp: done wctc4xxp: done xpp_usb: done No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: done. Have you an idea? Thank you P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] install the digium card TC400P howto?
BERGANZ François wrote: debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# /etc/init.d/dahdi restart Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp: done wcfxo: done wctdm: done wcb4xxp: done wctc4xxp: done xpp_usb: done wctc4xxp: done -- That's the module for the card. So it is loading. -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] install the digium card TC400P howto?
What have I to do? Cordialement, Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Sean Bright Envoyé : lundi 24 août 2009 18:21 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] install the digium card TC400P howto? BERGANZ François wrote: debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# /etc/init.d/dahdi restart Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp: done wcfxo: done wctdm: done wcb4xxp: done wctc4xxp: done xpp_usb: done wctc4xxp: done -- That's the module for the card. So it is loading. -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] how to install asterisk
I will consider it and to change it to DAHDI too.
It woul be great if repository had files named CURRENT like
asterisk_1.4.CURRENT so would have no need to change any script.
Thanks,
Valter
2009/8/24 Steve Edwards asterisk@sedwards.com
2009/8/21 aster...@opensourcesolution.in
i have to configures asterisk n my hardware details are
Is it just me, or would you think someone from a domain named like
Open Source Solution should be able to figure this one out...
On Mon, 24 Aug 2009, Valter Nogueira wrote:
I have a small script that do the trick for you.
http://www.fastway.com.br/wp-content/uploads/2009/02/install_asterisk.sh
Just a suggestion...
If you define the version numbers as variables your script will be
easier to maintain. For example:
ADDONS_VERSION=1.4.7
ASTERISK_VERSION=1.4.23.1
LIBPRI_VERSION=1.4.9
ZAPTEL_VERSION=1.4.12
cd /usr/src
wget
http://downloads.digium.com/pub/libpri/releases/libpri-${LIBPRI_VERSION}.tar.gzhttp://downloads.digium.com/pub/libpri/releases/libpri-$%7BLIBPRI_VERSION%7D.tar.gz
wget
http://downloads.digium.com/pub/zaptel/releases/zaptel-${ZAPTEL_VERSION}.tar.gzhttp://downloads.digium.com/pub/zaptel/releases/zaptel-$%7BZAPTEL_VERSION%7D.tar.gz
wget
http://downloads.digium.com/pub/asterisk/releases/asterisk-${ASTERISK_VERSION}.tar.gzhttp://downloads.digium.com/pub/asterisk/releases/asterisk-$%7BASTERISK_VERSION%7D.tar.gz
wget
http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-${ADDONS_VERSION}.tar.gzhttp://downloads.digium.com/pub/asterisk/releases/asterisk-addons-$%7BADDONS_VERSION%7D.tar.gz
tar -zxvf libpri-${LIBPRI_VERSION}.tar.gz
tar -zxvf zaptel-${ZAPTEL_VERSION}.tar.gz
tar -zxvf asterisk-${ASTERISK_VERSION}.tar.gz
tar -zxvf asterisk-addons-${ADDONS_VERSION}.tar.gz
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
___
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AstriCon 2009 - October 13 - 15 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
___
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Register Now: http://www.astricon.net
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Re: [asterisk-users] how to install asterisk
If I do #dmesg, I have it :
[ 101.994189] Unregistered codec translator 'DTE Decoder' with 92
transcoders (srcs=0101, dsts=000c)
[ 101.994189] Unregistered codec translator 'DTE Encoder' with 92
transcoders (srcs=000c, dsts=0101)
[ 101.999162] dahdi_transcode: Unloaded.
[ 102.011417] dahdi_transcode: Loaded.
[ 102.011418] wctc4xxp: tc400b0: Attached to device at :11:03.0.
[ 102.011418] firmware: requesting dahdi-fw-tc400m.bin
[ 107.427805] wctc4xxp: tc400b0: (G.729a / G.723.1) Transcoder support
LOADED (firm ver = 6.12)
[ 107.427805] wctc4xxp: tc400b0: Installed a Wildcard TC: Wildcard
TC400P+TC400M
[ 107.427805] dahdi_transcode: Registered codec translator 'DTE Encoder'
with 92 transcoders (srcs=000c, dsts=0101)
[ 107.427805] dahdi_transcode: Registered codec translator 'DTE Decoder'
with 92 transcoders (srcs=0101, dsts=000c)
[ 107.574569] dahdi_dummy: Trying to load High Resolution Timer
[ 107.574569] dahdi_dummy: Initialized High Resolution Timer
[ 107.574569] dahdi_dummy: Starting High Resolution Timer
[ 107.574569] dahdi_dummy: High Resolution Timer started, good to go
It know my card but why it dont load it!?
Cordialement,
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Valter
Nogueira
Envoyé : lundi 24 août 2009 18:30
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] how to install asterisk
I will consider it and to change it to DAHDI too.
It woul be great if repository had files named CURRENT like
asterisk_1.4.CURRENT so would have no need to change any script.
Thanks,
Valter
2009/8/24 Steve Edwards asterisk@sedwards.com
2009/8/21 aster...@opensourcesolution.in
i have to configures asterisk n my hardware details are
Is it just me, or would you think someone from a domain named like
Open Source Solution should be able to figure this one out...
On Mon, 24 Aug 2009, Valter Nogueira wrote:
I have a small script that do the trick for you.
http://www.fastway.com.br/wp-content/uploads/2009/02/install_asterisk.sh
Just a suggestion...
If you define the version numbers as variables your script will be
easier to maintain. For example:
ADDONS_VERSION=1.4.7
ASTERISK_VERSION=1.4.23.1
LIBPRI_VERSION=1.4.9
ZAPTEL_VERSION=1.4.12
cd /usr/src
wget
http://downloads.digium.com/pub/libpri/releases/libpri-${LIBPRI_VERSION}.tar
.gz
http://downloads.digium.com/pub/libpri/releases/libpri-$%7BLIBPRI_VERSION%7
D.tar.gz
wget
http://downloads.digium.com/pub/zaptel/releases/zaptel-${ZAPTEL_VERSION}.tar
.gz
http://downloads.digium.com/pub/zaptel/releases/zaptel-$%7BZAPTEL_VERSION%7
D.tar.gz
wget
http://downloads.digium.com/pub/asterisk/releases/asterisk-${ASTERISK_VERSIO
N}.tar.gz
http://downloads.digium.com/pub/asterisk/releases/asterisk-$%7BASTERISK_VER
SION%7D.tar.gz
wget
http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-${ADDONS_V
ERSION}.tar.gz
http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-$%7BADDON
S_VERSION%7D.tar.gz
tar -zxvf libpri-${LIBPRI_VERSION}.tar.gz
tar -zxvf zaptel-${ZAPTEL_VERSION}.tar.gz
tar -zxvf asterisk-${ASTERISK_VERSION}.tar.gz
tar -zxvf asterisk-addons-${ADDONS_VERSION}.tar.gz
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
___
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http://www.api-digital.com/ --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net http://www.astricon.net/
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Register Now: http://www.astricon.net
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Re: [asterisk-users] install the digium card TC400P howto?
BERGANZ François wrote: What have I to do? Nothing. The system recognizes the card, and the appropriate module is loading. What is happening that makes you think it isn't working? -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
Do a quick search for SMTP commands - to simulate a complete session via telnet. Most MTA's will check sender and recipient for validity, relaying, etc. Be sure both are reasonable and acceptable to host using telnet first. If you are new to sendmail.cf, read the instructions at the top of the file. You have to re-make the config file (mc vs cf), restart sendmail, etc... _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, August 24, 2009 12:13 PM To: Asterisk Users List Subject: Re: [asterisk-users] Problems sending voicemail emails That's what the MASQUERADE(localhost.localdomain) is supposed to do; Don't know why it is not. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems sending voicemail emails Hi Danny, I've tryed it but still the same. It looks that my ISP SMTP server doesn't like the sender, which I guess is r...@localhost.localdomain. Do you know how to change this sender? Many thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] install the digium card TC400P howto?
On Mon, Aug 24, 2009 at 12:49 PM, Sean Bright sean.bri...@gmail.com wrote: BERGANZ François wrote: What have I to do? Nothing. The system recognizes the card, and the appropriate module is loading. What is happening that makes you think it isn't working? -- Sean Bright sean.bri...@gmail.com No hardware timing source found in /proc/dahdi, loading dahdi_dummy would make me think it is not loading correctly. Have you setup your configs? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] install the digium card TC400P howto?
Steve Totaro wrote: No hardware timing source found in /proc/dahdi, loading dahdi_dummy would make me think it is not loading correctly. The TC400P is a transcoder card. It is not a timing source. -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
In voicemail.conf you can set serveremail=asterisk so some other address and this will be used as the sender's email address, at least as I understand things. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Aug 24, 2009, at 9:50 AM, Michelle Dupuis wrote: Do a quick search for SMTP commands - to simulate a complete session via telnet. Most MTA's will check sender and recipient for validity, relaying, etc. Be sure both are reasonable and acceptable to host using telnet first. If you are new to sendmail.cf, read the instructions at the top of the file. You have to re-make the config file (mc vs cf), restart sendmail, etc... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com ] On Behalf Of Danny Nicholas Sent: Monday, August 24, 2009 12:13 PM To: Asterisk Users List Subject: Re: [asterisk-users] Problems sending voicemail emails That’s what the MASQUERADE(localhost.localdomain) is supposed to do; Don’t know why it is not. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com ] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems sending voicemail emails Hi Danny, I've tryed it but still the same. It looks that my ISP SMTP server doesn't like the sender, which I guess isr...@localhost.localdomain. Do you know how to change this sender? Many thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] install the digium card TC400P howto?
On Mon, Aug 24, 2009 at 1:06 PM, Sean Bright sean.bri...@gmail.com wrote: Steve Totaro wrote: No hardware timing source found in /proc/dahdi, loading dahdi_dummy would make me think it is not loading correctly. The TC400P is a transcoder card. It is not a timing source. -- Sean Bright sean.bri...@gmail.com Silly me. I forgot about those overpriced transcoder cards. Dollar for dollar, I will go for bogomips! -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] IPKall and FWD
Oh my god.. Today its saying there is NOONE to take your call.I am using IdeaSIP What could be the reasons ? It was working perfectly till saturday . On Thu, Aug 20, 2009 at 11:54 PM, David @ULC ucoms2...@gmail.com wrote: IdeaSIP worked perfect for me. On Thu, Aug 20, 2009 at 11:27 PM, David @ULC ucoms2...@gmail.com wrote: We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Need to now my Asterisk User ID
At 9:04 AM on 24 Aug 2009, Steve Edwards wrote: Un-top-posting... jonas kellens wrote: How do I know my Asterisk User ID ?? On Mon, 24 Aug 2009, bails wrote: Type 'id asterisk' at your command line. It should return uid gid and all groups the asterisk user belongs to. This assumes you have a user named asterisk. Also assumes that Asterisk is running as the user named asterisk. There's probably a more proper way, but this works: ~$ ps -ef | grep /sbin/asterisk | grep -v grep You should get something like: root 12477 12476 0 Aug03 ?00:02:09 /usr/sbin/asterisk -f -g -n -p -q $ ps -fC asterisk Or for the uid: $ ps --no-headers -o uid -C asterisk -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 signature.asc Description: PGP signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Need to now my Asterisk User ID
For a newbie, this would be preferable; ps --no-headers -o uname -C asterisk -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Chad Wallace Sent: Monday, August 24, 2009 2:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Need to now my Asterisk User ID At 9:04 AM on 24 Aug 2009, Steve Edwards wrote: Un-top-posting... jonas kellens wrote: How do I know my Asterisk User ID ?? On Mon, 24 Aug 2009, bails wrote: Type 'id asterisk' at your command line. It should return uid gid and all groups the asterisk user belongs to. This assumes you have a user named asterisk. Also assumes that Asterisk is running as the user named asterisk. There's probably a more proper way, but this works: ~$ ps -ef | grep /sbin/asterisk | grep -v grep You should get something like: root 12477 12476 0 Aug03 ?00:02:09 /usr/sbin/asterisk -f -g -n -p -q $ ps -fC asterisk Or for the uid: $ ps --no-headers -o uid -C asterisk -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] IPKall and FWD
A quick look at the system shows you're not logged in, which is why you're getting that message. N. David @ULC wrote: Oh my god.. Today its saying there is NOONE to take your call.I am using IdeaSIP What could be the reasons ? It was working perfectly till saturday . On Thu, Aug 20, 2009 at 11:54 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: IdeaSIP worked perfect for me. On Thu, Aug 20, 2009 at 11:27 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] E1 w/ TE420B EC
I keep getting a red alarm when trying to setup asterisk to use my TE420B EC. I only have a blank context setup in my extensions.conf as I haven't started to config that until I can clear this red alarm. I don't have physical access to the server, so I can't go reseat the modules/card/ethernet cable, though I have hands on location that have done this a couple times already. Please help. I'm quite frustrated at this point. Thank you in advance for any help. /etc/dahdi/system.conf # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Global data loadzone= nl defaultzone = nl /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] ; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) group=1 context=frompstn switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 context = default cat /proc/dahdi/1 Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 RED 1 TE4/0/1/1 Clear (In use) RED(SWEC: MG2) 2 TE4/0/1/2 Clear (In use) RED(SWEC: MG2) 3 TE4/0/1/3 Clear (In use) RED(SWEC: MG2) 4 TE4/0/1/4 Clear (In use) RED(SWEC: MG2) 5 TE4/0/1/5 Clear (In use) RED(SWEC: MG2) 6 TE4/0/1/6 Clear (In use) RED(SWEC: MG2) 7 TE4/0/1/7 Clear (In use) RED(SWEC: MG2) 8 TE4/0/1/8 Clear (In use) RED(SWEC: MG2) 9 TE4/0/1/9 Clear (In use) RED(SWEC: MG2) 10 TE4/0/1/10 Clear (In use) RED(SWEC: MG2) 11 TE4/0/1/11 Clear (In use) RED(SWEC: MG2) 12 TE4/0/1/12 Clear (In use) RED(SWEC: MG2) 13 TE4/0/1/13 Clear (In use) RED(SWEC: MG2) 14 TE4/0/1/14 Clear (In use) RED(SWEC: MG2) 15 TE4/0/1/15 Clear (In use) RED(SWEC: MG2) 16 TE4/0/1/16 HDLCFCS (In use) RED 17 TE4/0/1/17 Clear (In use) RED(SWEC: MG2) 18 TE4/0/1/18 Clear (In use) RED(SWEC: MG2) 19 TE4/0/1/19 Clear (In use) RED(SWEC: MG2) 20 TE4/0/1/20 Clear (In use) RED(SWEC: MG2) 21 TE4/0/1/21 Clear (In use) RED(SWEC: MG2) 22 TE4/0/1/22 Clear (In use) RED(SWEC: MG2) 23 TE4/0/1/23 Clear (In use) RED(SWEC: MG2) 24 TE4/0/1/24 Clear (In use) RED(SWEC: MG2) 25 TE4/0/1/25 Clear (In use) RED(SWEC: MG2) 26 TE4/0/1/26 Clear (In use) RED(SWEC: MG2) 27 TE4/0/1/27 Clear (In use) RED(SWEC: MG2) 28 TE4/0/1/28 Clear (In use) RED(SWEC: MG2) 29 TE4/0/1/29 Clear (In use) RED(SWEC: MG2) 30 TE4/0/1/30 Clear (In use) RED(SWEC: MG2) 31 TE4/0/1/31 Clear (In use) RED(SWEC: MG2) ~T___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Bottlenecks with my asterisk setup.
Hello. I've been seting up a small VoIP setup, with roughly 5 persons, doing essentially some Meetme conferences. People have been experiencing some quality problems with the sound. Essentially delay, and some tolerable echo. I'd appreciate advice on how to troubleshoot this issue. What could be the most common reasons behind this? Please feel free to ask for more relevant details. All the best, Guillaume Yziquel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Bottlenecks with my asterisk setup.
Guillaume Yziquel a écrit : Hello. I've been seting up a small VoIP setup, with roughly 5 persons, doing essentially some Meetme conferences. People have been experiencing some quality problems with the sound. Essentially delay, and some tolerable echo. I'd appreciate advice on how to troubleshoot this issue. What could be the most common reasons behind this? Please feel free to ask for more relevant details. Another question: should I expect these issues to be less important if I switch to a Zaptel configuration instead of only VoIP? All the best, Guillaume Yziquel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] IPKall and FWD
My IPKall number is sending a CLID of a different number!! Since I was looking for the correct CLID, my Asterisk rejected it, and it went to, I assume, IPKall asterisk VM!!! Sounds like IPKall is really wrapped around the axle shaft today! John Novack David @ULC wrote: Oh my god.. Today its saying there is NOONE to take your call.I am using IdeaSIP What could be the reasons ? It was working perfectly till saturday . On Thu, Aug 20, 2009 at 11:54 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: IdeaSIP worked perfect for me. On Thu, Aug 20, 2009 at 11:27 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 Checked by AVG - www.avg.com Version: 8.5.409 / Virus Database: 270.13.65/2322 - Release Date: 08/23/09 18:03:00 -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] IPKall and FWD
you're not logged in means ? On Mon, Aug 24, 2009 at 11:39 PM, David @ULC ucoms2...@gmail.com wrote: Oh my god.. Today its saying there is NOONE to take your call.I am using IdeaSIP What could be the reasons ? It was working perfectly till saturday . On Thu, Aug 20, 2009 at 11:54 PM, David @ULC ucoms2...@gmail.com wrote: IdeaSIP worked perfect for me. On Thu, Aug 20, 2009 at 11:27 PM, David @ULC ucoms2...@gmail.com wrote: We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] IPKall and FWD
John Novack wrote: My IPKall number is sending a CLID of a different number!! Since I was looking for the correct CLID, my Asterisk rejected it, and it went to, I assume, IPKall asterisk VM!!! Sounds like IPKall is really wrapped around the axle shaft today! John Novack Searching their support forum, posted today is the fact they are discontinuing any VM, and the same problem we have experienced, calls from certain mobile and VOIP providers come in with a CLID of 206-204-0232, regardless of your IPKall number. Calling that number gives RNA. IPKall is probably short lived John Novack David @ULC wrote: Oh my god.. Today its saying there is NOONE to take your call.I am using IdeaSIP What could be the reasons ? It was working perfectly till saturday . On Thu, Aug 20, 2009 at 11:54 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: IdeaSIP worked perfect for me. On Thu, Aug 20, 2009 at 11:27 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 Checked by AVG - www.avg.com Version: 8.5.409 / Virus Database: 270.13.65/2322 - Release Date: 08/23/09 18:03:00 -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 Checked by AVG - www.avg.com Version: 8.5.409 / Virus Database: 270.13.65/2322 - Release Date: 08/23/09 18:03:00 -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] IPKall and FWD
Means your username is not registered on the IdeaSIP system (your client/phone is not logged into IdeaSIP). N. David @ULC wrote: you're not logged in means ? On Mon, Aug 24, 2009 at 11:39 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: Oh my god.. Today its saying there is NOONE to take your call.I am using IdeaSIP What could be the reasons ? It was working perfectly till saturday . On Thu, Aug 20, 2009 at 11:54 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: IdeaSIP worked perfect for me. On Thu, Aug 20, 2009 at 11:27 PM, David @ULC ucoms2...@gmail.com mailto:ucoms2...@gmail.com wrote: We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
On Mon, 2009-08-24 at 17:56 +0200, Joan Antoni Terre wrote: [] I've tryed it but still the same. It looks that my ISP SMTP server doesn't like the sender, which I guess is r...@localhost.localdomain. Do you know how to change this sender? Set a sane/correct hostname (which is resolvable via DNS by the ISPs SMTP server). Bernd -- Firmix Software GmbH http://www.firmix.at/ mobil: +43 664 4416156 fax: +43 1 7890849-55 Embedded Linux Development and Services ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
On Mon, 2009-08-24 at 11:12 -0500, Danny Nicholas wrote: That’s what the MASQUERADE(localhost.localdomain) is supposed to do; Don’t know why it is not. It's commented out. The dnl at the begin of the line means delete 'til newline for m4 (which processes that file and produces the .cf file). You need a DNS-resolvable hostname also there (and not mydomain.com). And you will probably learn a little bit about SMTP and http://www.sendmail.org/m4/masquerading.html. Bernd -- Firmix Software GmbH http://www.firmix.at/ mobil: +43 664 4416156 fax: +43 1 7890849-55 Embedded Linux Development and Services ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Problems sending voicemail emails
Check your hostname settings, hosts file, and order of name resolution... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bernd Petrovitsch Sent: Monday, August 24, 2009 5:26 PM To: Asterisk Users List Subject: Re: [asterisk-users] Problems sending voicemail emails On Mon, 2009-08-24 at 17:56 +0200, Joan Antoni Terre wrote: [] I've tryed it but still the same. It looks that my ISP SMTP server doesn't like the sender, which I guess is r...@localhost.localdomain. Do you know how to change this sender? Set a sane/correct hostname (which is resolvable via DNS by the ISPs SMTP server). Bernd -- Firmix Software GmbH http://www.firmix.at/ mobil: +43 664 4416156 fax: +43 1 7890849-55 Embedded Linux Development and Services ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] LDAP Get for Asterisk 1.6.x
I'd appreciate it if someone could give me an answer to using LDAP in Asterisk 1.6.x From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn Sent: Thursday, 20 August 2009 4:01 PM To: 'Asterisk Users Mailing List' Subject: [asterisk-users] LDAP Get for Asterisk 1.6.x What is everyone using in Asterisk 1.6.x to retrieve data from LDAP. The version of app_ldap I have only works with Asterisk 1.4.x http://www.mezzo.net/asterisk/app_ldap-2.0rc1.tgz Without a way to get data from LDAP stop me from using Asterisk 1.6.x Regards David. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] LDAP Get for Asterisk 1.6.x
Not an LDAP user but perhaps using AGI to access your LDAP server may be a solution? Looks like it may require some work but I wouldn't think it to be too hard. sl From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn Sent: Monday, August 24, 2009 5:53 PM To: 'Asterisk Users Mailing List' Subject: Re: [asterisk-users] LDAP Get for Asterisk 1.6.x I'd appreciate it if someone could give me an answer to using LDAP in Asterisk 1.6.x From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn Sent: Thursday, 20 August 2009 4:01 PM To: 'Asterisk Users Mailing List' Subject: [asterisk-users] LDAP Get for Asterisk 1.6.x What is everyone using in Asterisk 1.6.x to retrieve data from LDAP. The version of app_ldap I have only works with Asterisk 1.4.x http://www.mezzo.net/asterisk/app_ldap-2.0rc1.tgz Without a way to get data from LDAP stop me from using Asterisk 1.6.x Regards David. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] E1 w/ TE420B EC
trebaum schreef: I keep getting a red alarm when trying to setup asterisk to use my TE420B EC. I only have a blank context setup in my extensions.conf as I haven't started to config that until I can clear this red alarm. I don't have physical access to the server, so I can't go reseat the modules/card/ethernet cable, though I have hands on location that have done this a couple times already. Please help. I'm quite frustrated at this point. Thank you in advance for any help. */etc/dahdi/system.conf* # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Global data loadzone= nl defaultzone = nl */etc/asterisk/chan_dahdi.conf* [trunkgroups] [channels] ; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) group=1 context=frompstn switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 context = default *cat /proc/dahdi/1* Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 RED 1 TE4/0/1/1 Clear (In use) RED(SWEC: MG2) 2 TE4/0/1/2 Clear (In use) RED(SWEC: MG2) 3 TE4/0/1/3 Clear (In use) RED(SWEC: MG2) 4 TE4/0/1/4 Clear (In use) RED(SWEC: MG2) 5 TE4/0/1/5 Clear (In use) RED(SWEC: MG2) 6 TE4/0/1/6 Clear (In use) RED(SWEC: MG2) 7 TE4/0/1/7 Clear (In use) RED(SWEC: MG2) 8 TE4/0/1/8 Clear (In use) RED(SWEC: MG2) 9 TE4/0/1/9 Clear (In use) RED(SWEC: MG2) 10 TE4/0/1/10 Clear (In use) RED(SWEC: MG2) 11 TE4/0/1/11 Clear (In use) RED(SWEC: MG2) 12 TE4/0/1/12 Clear (In use) RED(SWEC: MG2) 13 TE4/0/1/13 Clear (In use) RED(SWEC: MG2) 14 TE4/0/1/14 Clear (In use) RED(SWEC: MG2) 15 TE4/0/1/15 Clear (In use) RED(SWEC: MG2) 16 TE4/0/1/16 HDLCFCS (In use) RED 17 TE4/0/1/17 Clear (In use) RED(SWEC: MG2) 18 TE4/0/1/18 Clear (In use) RED(SWEC: MG2) 19 TE4/0/1/19 Clear (In use) RED(SWEC: MG2) 20 TE4/0/1/20 Clear (In use) RED(SWEC: MG2) 21 TE4/0/1/21 Clear (In use) RED(SWEC: MG2) 22 TE4/0/1/22 Clear (In use) RED(SWEC: MG2) 23 TE4/0/1/23 Clear (In use) RED(SWEC: MG2) 24 TE4/0/1/24 Clear (In use) RED(SWEC: MG2) 25 TE4/0/1/25 Clear (In use) RED(SWEC: MG2) 26 TE4/0/1/26 Clear (In use) RED(SWEC: MG2) 27 TE4/0/1/27 Clear (In use) RED(SWEC: MG2) 28 TE4/0/1/28 Clear (In use) RED(SWEC: MG2) 29 TE4/0/1/29 Clear (In use) RED(SWEC: MG2) 30 TE4/0/1/30 Clear (In use) RED(SWEC: MG2) 31 TE4/0/1/31 Clear (In use) RED(SWEC: MG2) ~T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 As I see you have specified nl as defaultzone so I expect that you are using a ISDN-30/15 line from provider KPN in the Netherlands. If so then remove the crc4 option from the span line in /etc/dahdi/system.conf. */etc/dahdi/system.conf* # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Global data loadzone= nl defaultzone = nl KPN is not using the crc4 checksum and therefore the card is not getting the wrong checksum on the lines and so they get a red alarm status. After the change reload dahdi and your lines should change colours. If this is working for you please answer to the mailing list so people in the future will find it. The next time please specify the type of telephoneline and provider. Regards, Michel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
