Re: [Asterisk-Users] zap show channels - no such command
On Mon, 2004-08-23 at 15:53, Imran Akbar wrote: Thanks, I seem to have done the zaptel installation - what am I missing - i still don't have a chan_zap.so file? In the asterisk src directory, do make clean install It won't even build the zaptel stuff unless you have installed zaptel first. Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
Hi Imran, I seem to have done the zaptel installation - what am I missing - i still don't have a chan_zap.so file? Did you rebuild Asterisk after installing Zaptel? chan_zap.so is built as part of the actually Asterisk build, AFAIK. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. If you don't have Zaptel installed when you build Asterisk, it might not build chan_zap.so. On my system the asterisk modules are in /usr/lib/asterisk/modules/. Regards Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
dmesg: Zapata Telephony Interface Registered on major 196 ... pci: fOUND irq 11 for device 00 wcfxo: daa mode is 'FCC' Found a wildcard fxo: wildcard x101p PCI: found iraq 11 for device pci: sharing irq 11 with 0 wcfxo: DAA modeis 'FCC' Found a Wildcard FXO: Wildcard X101p that's for two FXO cards. Thanks el Flynn wrote: Imran Akbar wrote: edited the zaptel.conf, zapata.conf, extensions.conf to proper settings. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. Why don't you post a snippet of the zaptel stuff as reported by dmesg? That may help. Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
wo, i have to rebuild asterisk after i install zaptel? where did that come from? let me try... thanks Imran Darryl Ross wrote: Hi Imran, I seem to have done the zaptel installation - what am I missing - i still don't have a chan_zap.so file? Did you rebuild Asterisk after installing Zaptel? chan_zap.so is built as part of the actually Asterisk build, AFAIK. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. If you don't have Zaptel installed when you build Asterisk, it might not build chan_zap.so. On my system the asterisk modules are in /usr/lib/asterisk/modules/. Regards Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
Tried recompiling asterisk after the zaptel installation... still don't have a chan_zap.so file. help anyone? Thanks, Imran Imran Akbar wrote: wo, i have to rebuild asterisk after i install zaptel? where did that come from? let me try... thanks Imran Darryl Ross wrote: Hi Imran, I seem to have done the zaptel installation - what am I missing - i still don't have a chan_zap.so file? Did you rebuild Asterisk after installing Zaptel? chan_zap.so is built as part of the actually Asterisk build, AFAIK. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. If you don't have Zaptel installed when you build Asterisk, it might not build chan_zap.so. On my system the asterisk modules are in /usr/lib/asterisk/modules/. Regards Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with mysql and with asterisk
Hi Every one and Lerale Erwan I have briefly describe my problem and I have provide the steps as follows: I have intalled redhat properly and from the konsole I checked with mysql. rpm -qa | grep mysql and the konsole provide me the message: mysql-3.23.54a-11 mysql-server-3.23.54a-11 Then I have download the asterisk and addons: By the using of : cd /usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login - the password is anoncvs. cvs checkout asterisk Then cd/usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login - the password is anoncvs. cvs checkout asterisk-addons Compile /usr/src/asterisk-addons as follows: cd asterisk-addons make clean make install But the system send me an error messge like ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory for x in ; do install -m 755 $x /usr/lib/asterisk/modules ; done The I have used to use mysql-vm-routines, set USE_MYSQL_VM_INTERFACE to 1 in asterisk/apps/Makefile , then put this file into asterisk/apps/ and (re)build asterisk. Then use make from the /usr/src/asterisk/ Then system have give me this type of error message: make[1]: Entering directory `/usr/src/asterisk/apps' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\CVS-HEAD-08/23/04-11:39:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -DUSEMYSQLVM -c -o app_voicemail.o app_voicemail.c app_voicemail.c:45:25: mysql/mysql.h: No such file or directory In file included from app_voicemail.c:372: mysql-vm-routines.h:7: parse error before '*' token mysql-vm-routines.h:7: warning: type defaults to `int' in declaration of `dbhandler' mysql-vm-routines.h:7: warning: data definition has no type or storage class mysql-vm-routines.h: In function `mysql_login': mysql-vm-routines.h:18: warning: implicit declaration of function `mysql_init' mysql-vm-routines.h:18: warning: assignment makes pointer from integer without a cast mysql-vm-routines.h:19: warning: implicit declaration of function `mysql_real_connect' mysql-vm-routines.h: In function `mysql_logout': mysql-vm-routines.h:29: warning: implicit declaration of function `mysql_close' mysql-vm-routines.h: In function `find_user': mysql-vm-routines.h:35: `MYSQL_RES' undeclared (first use in this function) mysql-vm-routines.h:35: (Each undeclared identifier is reported only once mysql-vm-routines.h:35: for each function it appears in.) mysql-vm-routines.h:35: `result' undeclared (first use in this function) mysql-vm-routines.h:36: `MYSQL_ROW' undeclared (first use in this function) mysql-vm-routines.h:36: parse error before rowval mysql-vm-routines.h:37: `MYSQL_FIELD' undeclared (first use in this function) mysql-vm-routines.h:37: `fields' undeclared (first use in this function) mysql-vm-routines.h:68: warning: implicit declaration of function `mysql_query' mysql-vm-routines.h:69: warning: implicit declaration of function `mysql_store_result' mysql-vm-routines.h:70: `rowval' undeclared (first use in this function) mysql-vm-routines.h:70: warning: implicit declaration of function `mysql_fetch_row' mysql-vm-routines.h:71: warning: implicit declaration of function `mysql_num_fields' mysql-vm-routines.h:72: warning: implicit declaration of function `mysql_fetch_fields' mysql-vm-routines.h:89: warning: implicit declaration of function `mysql_free_result' make[1]: *** [app_voicemail.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Please help me. I had totaly upset to install the asterisk with cdr. Please help me because i am now helpless. With best regards. Dipak Kumar Paul Tryarc LLC _ Claim your Citibank Ready Cash today. http://go.msnserver.com/IN/54177.asp ItÂ’s fast, easy and affordable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with mysql and with asterisk
try installing mysql-devel -Adam DIPAK PAUL wrote: Hi Every one and Lerale Erwan I have briefly describe my problem and I have provide the steps as follows: I have intalled redhat properly and from the konsole I checked with mysql. rpm -qa | grep mysql and the konsole provide me the message: mysql-3.23.54a-11 mysql-server-3.23.54a-11 Then I have download the asterisk and addons: By the using of : cd /usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login - the password is anoncvs. cvs checkout asterisk Then cd/usr/src export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login - the password is anoncvs. cvs checkout asterisk-addons Compile /usr/src/asterisk-addons as follows: cd asterisk-addons make clean make install But the system send me an error messge like ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` cdr_addon_mysql.c:33:19: mysql.h: No such file or directory cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory for x in ; do install -m 755 $x /usr/lib/asterisk/modules ; done The I have used to use mysql-vm-routines, set USE_MYSQL_VM_INTERFACE to 1 in asterisk/apps/Makefile , then put this file into asterisk/apps/ and (re)build asterisk. Then use make from the /usr/src/asterisk/ Then system have give me this type of error message: make[1]: Entering directory `/usr/src/asterisk/apps' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\CVS-HEAD-08/23/04-11:39:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -DUSEMYSQLVM -c -o app_voicemail.o app_voicemail.c app_voicemail.c:45:25: mysql/mysql.h: No such file or directory In file included from app_voicemail.c:372: mysql-vm-routines.h:7: parse error before '*' token mysql-vm-routines.h:7: warning: type defaults to `int' in declaration of `dbhandler' mysql-vm-routines.h:7: warning: data definition has no type or storage class mysql-vm-routines.h: In function `mysql_login': mysql-vm-routines.h:18: warning: implicit declaration of function `mysql_init' mysql-vm-routines.h:18: warning: assignment makes pointer from integer without a cast mysql-vm-routines.h:19: warning: implicit declaration of function `mysql_real_connect' mysql-vm-routines.h: In function `mysql_logout': mysql-vm-routines.h:29: warning: implicit declaration of function `mysql_close' mysql-vm-routines.h: In function `find_user': mysql-vm-routines.h:35: `MYSQL_RES' undeclared (first use in this function) mysql-vm-routines.h:35: (Each undeclared identifier is reported only once mysql-vm-routines.h:35: for each function it appears in.) mysql-vm-routines.h:35: `result' undeclared (first use in this function) mysql-vm-routines.h:36: `MYSQL_ROW' undeclared (first use in this function) mysql-vm-routines.h:36: parse error before rowval mysql-vm-routines.h:37: `MYSQL_FIELD' undeclared (first use in this function) mysql-vm-routines.h:37: `fields' undeclared (first use in this function) mysql-vm-routines.h:68: warning: implicit declaration of function `mysql_query' mysql-vm-routines.h:69: warning: implicit declaration of function `mysql_store_result' mysql-vm-routines.h:70: `rowval' undeclared (first use in this function) mysql-vm-routines.h:70: warning: implicit declaration of function `mysql_fetch_row' mysql-vm-routines.h:71: warning: implicit declaration of function `mysql_num_fields' mysql-vm-routines.h:72: warning: implicit declaration of function `mysql_fetch_fields' mysql-vm-routines.h:89: warning: implicit declaration of function `mysql_free_result' make[1]: *** [app_voicemail.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Please help me. I had totaly upset to install the asterisk with cdr. Please help me because i am now helpless. With best regards. Dipak Kumar Paul Tryarc LLC _ Claim your Citibank Ready Cash today. http://go.msnserver.com/IN/54177.asp Its fast, easy and affordable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug in recording uavarible
From samles config. When user change uavarible message from firefly login by IAX got 3 file .WAV .gsm . wav But no sound record. When login by antek 804 GW (SIP mode) record success. but message can't play when someone need to leave voice mail to this box. But when delete .WAV . gsm and leave only .wav its' work. Is problem in asterisk Or client Dome C. -- This mail sent through Msarn Mail Solution - Full Spam-Vitus Protect - For more information Please contact [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with asterisk and postgresql
Hi Every one and Fabio Donaggio Fabio Donaggio you have faced same type of problem with postgresql with asterisk. Did you solved your problem. Please help me. The problem is as follows: When i reload the asterisk I had got this error == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Aug 23 14:22:33 ERROR[1076245120]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! Aug 23 14:22:33 ERROR[1076245120]: cdr_pgsql.c:300 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? And I have configure the cdr_pgsql.conf file as: [global] hostname=localhost port=5432 dbname=asterisk password= user=postgres Please help me. With best regards. Dipak Kumar Paul Tryarc LLC _ Sports, sports and more sports! Keep up with all the action! http://www.msn.co.in/sports/ Stay connected with MSN Sports! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 servers
Good day all I've tried my iax conf and I'm struggling.So I want to know If someone else got this working and if they can pleas send my their configs I have to asterisk server,in different tows,both offices connected wit a direct line so both servers are on the same network running SIP.Each town got different extension register to each sever.Town A=100+ town B=200+ How do I get town A people to dial 201 and it will go to sown B's server's 201 SIP users Please not that I'm only a newbie and my terms may be wrong but I'm really having a bod time with this Please help Thanks ALtus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Choosing between TE405P and TE410P
Is there anything to choose, in performance, between a TE405P and a TE410P? I understand the difference between the PCI bus voltages, and certainly don't intend to try Andrew's hacksaw operation :-). But if I choose the card first, and a compatible mobo second, does it make any difference which? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] False Hangups on Asterisk
Hi, -Original Message- What I get is an almost random hung up during calls (both incoming and dialed). Sometimes tha call can last for 30 minutes without any problems sometimes 1min. In all cases the phone switches to the busy tone and the caller hears nothing, The line does not hang-up though. I tried various tests trying to pin down the problem but I was unsuccessful so far. My latest theory was that this may happen when the ATA re-registers with Asterisk but as you dont have any ATAs it seems that my theory is wrong :-( Does anybody else have similar problems or knows a procedure to try in order to identify the source of the problem? I've never experienced this myself, but I have one user with a box with an X100P that complained about this. Could be busydetect=yes, or some version issue ? Would be nice to have a notice of when busydetect=yes kicked in or something ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] using ChanIsAvail
Title: using ChanIsAvail
Hi
I am trying to use ChanIsAvail to decide if a particular extension is available in the sip channel
I am using MySQL to hold my SIP friends.
and wy cvs version shows Asterisk CVS-08/02/04
my intention is, that if the extension is not available in Sip channel, I will send the call somewhere else
my extensions file contains the following:
exten = _[123]XX,1,ChanIsAvail(sip/${EXTEN})
exten = _[123]XX,2,dial(sip/${EXTEN},30)
exten = _[123]XX,102,Dial(IAX2/sip01-xx:[EMAIL PROTECTED]/${EXTEN})
if I understand ChanIsAvail correctly this should give med following:
if i dial extension 111, and that is a local extension, it dials the sip channel
on the other hand, if extension 111 is not avaliable in the local sip channel, it dials on IAX2
But it does not work, if 111 is not a local extension the dial in priority 2 returns with -1, in my opinion it should never have been executed
when I have all SIP frinds in sip.conf it works, but it does not when using MySQL
is this a bug, or is ChanIsAvail not intended to work when SIP frinds are in MySQL ??
Kind regards
Poul Pedersen
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Re: [Asterisk-Users] MusicOnHold problem
It can't find mpg123 or you don't have any mp3 files in your moh directory. - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 23, 2004 12:58 AM Subject: [Asterisk-Users] MusicOnHold problem Hi, I had music on hold working but now don't know what happened. I get : WARNING[...] res_musiconhold.c:334 moh0_exec: Unable to start music on hold (class '') on channel SIP... Any ideas what is wrong ? Regards, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to recover the problem with pgsql and asterisk
Hi Please help to store datas of call history into the pgsql. When I have run ./asterisk -c then I have faced == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Aug 23 16:11:38 ERROR[1076245120]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! Aug 23 16:11:38 ERROR[1076245120]: cdr_pgsql.c:300 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? Thanks in advanced Dipak Kumar Paul Tryarc LLC _ Post Classifieds on MSN classifieds. http://go.msnserver.com/IN/44045.asp Buy and Sell on MSN Classifieds. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adit 600 FXO FXS
Gurus, I am missing something with Adit 600 channel back. - FXS cards designed for connecting phonesets or telephony lines? Thanks for your advice ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Adit 600 FXO FXS
FXS is for stations FXO is for exchange lines cheers greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vasyl Rublyov Sent: Monday, 23 August 2004 9:39 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Adit 600 FXO FXS Gurus, I am missing something with Adit 600 channel back. - FXS cards designed for connecting phonesets or telephony lines? Thanks for your advice ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Choosing between TE405P and TE410P
Except for the PCI bus voltages, there does not appear to be any difference. I've load-tested both extensively and they perform about the same. I did have an issue with a TE410P getting stuck, ie not responding after a re-boot (but not a power down), but that seems to have resolved itself when running the same board in another chassis, so not sure if that was a design issue or not. I think the TE405P is a slightly newer design, but I'll bet they are virtually identical. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Monday, August 23, 2004 2:25 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Choosing between TE405P and TE410P Is there anything to choose, in performance, between a TE405P and a TE410P? I understand the difference between the PCI bus voltages, and certainly don't intend to try Andrew's hacksaw operation :-). But if I choose the card first, and a compatible mobo second, does it make any difference which? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 2 servers
Quoting [EMAIL PROTECTED]: How do I get town A people to dial 201 and it will go to sown B's server's 201 SIP users Please not that I'm only a newbie and my terms may be wrong but I'm really having a bod time with this Please help Thanks ALtus I have a doc on it. (Sorry was going to copy/paste but my mail reader didn't like the columns from the doc.) If you want it drop me a line and I'll send you the file. (Should also probably put it in the wiki :-) dbc. -- David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about dial out via Zap
Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1 answered SIP/6149236651-eafb Urgent handler -- Hungup 'Zap/1-1' Urgent handler Urgent handler -- Hungup 'Zap/2-1' Urgent handler Urgent handler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Choosing between TE405P and TE410P
Scott Stingel [EMAIL PROTECTED] wrote: Except for the PCI bus voltages, there does not appear to be any difference. I've load-tested both extensively and they perform about the same. Thanks Scott, that's useful to know. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adit 600 FXO FXS
Thank you. Does anyone by any chance have Adit 600 FXO spare modules to sell? Greg Smith wrote: FXS is for stations FXO is for exchange lines cheers greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vasyl Rublyov Sent: Monday, 23 August 2004 9:39 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Adit 600 FXO FXS Gurus, I am missing something with Adit 600 channel back. - FXS cards designed for connecting phonesets or telephony lines? Thanks for your advice ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel installation
Hi, the problems i previously had with zap show channels seems to be from an incorrect zaptel installation which is why I don't have a chan_zap.so file. I compile and do a make clean, make, make install of zaptel and do my modprobe's, and I was told to reinstall asterisk after that. I do so however, but it makes no difference. any hints? thanks Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Phone recommendation for Receptionist
The expansion module is NOT supported with SIP. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Bogan Sent: Sunday, August 22, 2004 7:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Phone recommendation for Receptionist I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. What would you guys recommend? A Cisco 7960 with the 7914 expansion module [ http://www.cisco.com/en/US/products/hw/phones/ps379/ps1856/ind ex.html ] -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] routing telephone calls via switchboard/asterisk.
I'm new to this list.Reading the asterisk handbook pdf (good work)but but still have some questions. Using Trustix 2.1 and installed Asterisk via CVS, zaptel and libpri. We have a dedicated server which is connected to our telephone company. It makes us able to call ordinary phones via VOIP using Ericsson DRG22. Would like to make people able to call me - and get a message "dial 1 for Hans, 2 for Eric, 3 for Hanna." Can I set up such a recording/playback software with the asterisk system ? And how can I route the calls onto the right number ? (guessing that I need to run mysql and storing all the phonenumber, IP, etc) Regards, Stig Henning ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-bitrate codecs
Simone Ricci wrote: Anyone knows if there's a way to select the bitrate of those codecs supporting multiple bitrates (eg. g.726)? I've tried searching and googling a lot, but without useful results... I don't think there's an API for this, other than defining mutiple codec(s) each of which is handled by different codec handlers which are identical other than their settings. It would be useful for many codecs, including also speex. -SteveK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] routing telephone calls via switchboard/asterisk.
Yes, it's very likely that you can perform these IVR functions within asterisk. If the realtime switching decisions are simple, they can probably be stored in the asterisk dialplan itself. Alternatively, you could retrieve them from a DB. Have you read the background material in the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stig Thune Sent: Monday, August 23, 2004 6:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] routing telephone calls via switchboard/asterisk. I'm new to this list. Reading the asterisk handbook pdf (good work) but but still have some questions. Using Trustix 2.1 and installed Asterisk via CVS, zaptel and libpri. We have a dedicated server which is connected to our telephone company. It makes us able to call ordinary phones via VOIP using Ericsson DRG22. Would like to make people able to call me - and get a message dial 1 for Hans, 2 for Eric, 3 for Hanna. Can I set up such a recording/playback software with the asterisk system ? And how can I route the calls onto the right number ? (guessing that I need to run mysql and storing all the phonenumber, IP, etc) Regards, Stig Henning ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940 Question
Hi all, I know this is a stupid question, but it is one I've been trying answer for quite some time. Exactly how many simultaneous calls can the Cisco 7940 have, considering you can be talking to one, and have XXX others on hold? Using SIP, is XXX only 1? I've found documents in various places indicating different values in regard to the max number of calls the phone can handle. I'm just trying to nail down the exact number when the phone is only assigned one directory number (extension). Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC-S in NT mode, wiring?
I've got an old HFC-S card to play with, and I would like to use it in NT mode. I've a problem only: wiring. I can't fully understand the instructions I was able to find online. Someone can point me to a site which explains the whole procedure clearly (like with some schematics, even in ASCII)? TIA, Simone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] routing telephone calls viaswitchboard/asterisk.
Thank you for fast answer! I will read up on this, thank you for link! Regards, Stig Henning - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Monday, August 23, 2004 4:02 PM Subject: RE: [Asterisk-Users] routing telephone calls viaswitchboard/asterisk. Yes, it's very likely that you can perform these IVR functions within asterisk. If the realtime switching decisions are simple, they can probably be stored in the asterisk dialplan itself. Alternatively, you could retrieve them from a DB. Have you read the background material in the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stig Thune Sent: Monday, August 23, 2004 6:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] routing telephone calls via switchboard/asterisk. I'm new to this list. Reading the asterisk handbook pdf (good work) but but still have some questions. Using Trustix 2.1 and installed Asterisk via CVS, zaptel and libpri. We have a dedicated server which is connected to our telephone company. It makes us able to call ordinary phones via VOIP using Ericsson DRG22. Would like to make people able to call me - and get a message dial 1 for Hans, 2 for Eric, 3 for Hanna. Can I set up such a recording/playback software with the asterisk system ? And how can I route the calls onto the right number ? (guessing that I need to run mysql and storing all the phonenumber, IP, etc) Regards, Stig Henning ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions.conf
Hey guys, Is there a way to make asterisk return the calling point after the last command is completed in a context? The goto statement doesn't seem to work as I expected it too. For Example: [iax-demo] exten = s,1,Playback(demo-abouttotry) exten = s,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = s,3,Playback(demo-nogo) [some-menu] exten = s,1,Playback(Some-file) exten = s,2,Playback(another-file) exten = s,3,Playback(another-file) exten = s,4,WaitExten(4) exten = s,5,Queue(some-queue) exten = 300,1,Goto(iax-demo,s,1) exten = 300,2,goto(s,3) [some-other-menu] exten = s,1,Playback(Some-different-file) exten = s,2,Playback(some-other-file) exten = s,3,WaitExten(4) exten = s,4,Queue(different-queue) exten = 300,1,Goto(iax-demo,s,1) exten = 300,2,Goto(s,1) You see, I want the iax-demo in both contexts, but after the demos done, I want the caller to be returned to the context, but where depends on the calling context. Any help ? Thank you, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel installation
On Mon, Aug 23, 2004 at 06:00:10PM +0500, Imran Akbar wrote: the problems i previously had with zap show channels seems to be from an incorrect zaptel installation which is why I don't have a chan_zap.so file. I compile and do a make clean, make, make install of zaptel and do my modprobe's, and I was told to reinstall asterisk after that. I do so however, but it makes no difference. any hints? What kernel version? If 2.6, you need make linux26, you know... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP unphones
Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of money to have all those fancy displays and keys tucked away behind a speakergrille and drywall. BTW, I'm not dead-set on SIP, but it seems to be the most logical protocol for this app (NOTIFY msg can carry directions on mike/speaker/two-way, etc) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Swissvoice MGCP Error 502
I have 1 IP phone (Swissvoice IP10S) and 1 POTS phone. When I dial the number for the IP phone off the POTS phone, the IP phone rings. But when I pick up the handset on the IP phone, I get a busy signal and this message on *: Aug 23 09:38:57 NOTICE[1142106560]: chan_mgcp.c:2243 handle_response: Terminating on result 502 from [EMAIL PROTECTED] Here is the entire session. svip10 is the 1 and only line on the swissvoice phone. -- Executing Dial(SIP/64.72.107.2-0811d658, MGCP/[EMAIL PROTECTED]) in new stack -- MGCP mgcp_request([EMAIL PROTECTED]) -- MGCP cw: -1, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/[EMAIL PROTECTED]) created in state: Down -- Called [EMAIL PROTECTED] -- MGCP/[EMAIL PROTECTED] is ringing -- Endpoint '[EMAIL PROTECTED]' observed 'hd' -- MGCP/[EMAIL PROTECTED] answered SIP/64.72.107.2-0811d658 -- Attempting native bridge of SIP/64.72.107.2-0811d658 and MGCP/[EMAIL PROTECTED] Aug 23 09:38:57 NOTICE[1142106560]: chan_mgcp.c:2243 handle_response: Terminating on result 502 from [EMAIL PROTECTED] == Spawn extension (exten, 2815699913, 1) exited non-zero on 'SIP/64.72.107.2-0811d658' -- Endpoint '[EMAIL PROTECTED]' observed 'hu' -- MGCP handle_request([EMAIL PROTECTED]) ast_channel already destroyed -- MGCP handle_request([EMAIL PROTECTED]) set vmwi(-) mgcp.conf [00059002042b] context=matthew host=dynamic callerid = John Doe 123 callgroup=0 pickupgroup=0 nat=yes threewaycalling=yes transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer callwaiting=yes ; this might be a cause of trouble for ip10s cancallforward=yes line = svip10 extensions.conf [general] exten = 1115551212,1,Dial(MGCP/[EMAIL PROTECTED]) (1115551212 is not the real #; replaced for privacy) [matthew] exten = 4,1,Dial(MGCP/[EMAIL PROTECTED]) Any ideas on this? Thanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP / IAX provider in the Netherlands.
You can use http://www.voipgate.nl -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: woensdag 18 augustus 2004 10:59 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP / IAX provider in the Netherlands. Hi all. Can you reccomend a SIP / IAX provider in the Netherlands ? I need a few Numbers, and of course cheap rates :) /Regards Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about dial out via Zap
Found out something strange.. In zapata.conf if I change the signalling from featd to em_w I'm able to dial out without a problem. But I'm unable to get calls in because of the featd data sent. Change it back to featd and I'm now able to call in but unable to call out. So my question is do I need to do something when calling out for featd? It looks to me like a problem with featd. Below is a copy of my zapata.conf file. zapata.conf [channels] context=from-analog signalling=featd ;signalling=em_w group=1 channel = 1-12 usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes useincomingcalleridonzaptransfer=yes callerid=asreceived echocancel=yes echocancelwhenbridged=yes rxgain=1.0 txgain=1.0 musiconhold=default Thanks Eric -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Monday, August 23, 2004 8:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Question about dial out via Zap Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1 answered SIP/6149236651-eafb Urgent handler -- Hungup 'Zap/1-1' Urgent handler Urgent handler -- Hungup 'Zap/2-1' Urgent handler Urgent handler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telnet and Root
neil wrote: Sorry if this is posted to the wrong forum but as it is related to a problem I have with Asterisk it may just scrape through!! I am running Fedora 1 and I can telnet in to my asterisk box as any user except root and am using the same credentials as logging in locally. I am new to Linux and any help would be gratefully appreciated. OpenSSH is much easier to secure than Telnet because most telnet servers and clients expect to pass the passwords to eachother in plain site. If you MUST use telnet, please set up Kerberos and configure it to encrypt the entire session, not just the login. you must use the telnet server and client that comes with the kerberos distribution as well. However, in general it is easier to set up SSH than to set up kerberized telnet in a secure way. Due to its vulnerability, most telnet servers will not allow root to log in via telnet. OpenSSH has a configuration option for this, and can be set either way. You can get OpenSSH from http://www.openssh.org. It depends on openssl which is available from http://www.openssl.org. Thanks Neil begin:vcard fn:Chris Travers n:Travers;Chris email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telnet and Root
Steve Szmidt wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Friday 20 August 2004 06:02 am, Thomas Kuepper wrote: use ssh instead of telnet. telnet is a bad idea. And the reason telnet is a bad idea, is because it sends the password in clear text. Today there's no valid reason to use telnet over ssh. First of all, Kerberos comes with a telnet server which can be as secure as OpenSSH. Also, I wouldn't be surprised if Microsoft starts using kerberized telnet as part of their SFU (last time I asked, they were concerned about licensing issues with OpenSSH and had no plans to include it). So telnet might not be as dead as one might think. However, One must take care when using Kerberized telnet servers for important administration because they can be easily misconfigured not to encrypt the session or to fall back on plain text transfers. Also, many binary distributions of openssh don't support kerberos, which makes kerberized telnet more scalable in many instances. Best Wishes, Chris travers Metatron Technology Consulting begin:vcard fn:Chris Travers n:Travers;Chris email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] telnet and Root
Mark Woods wrote: Chris Shaw wrote: If you really want to be able to telnet in as root, locate telnetd.conf or somesuch and it should be in there somewhere as a yes/no. (It is for ssh anyway..) No, not under any distro I'm familiar with... It's under /etc/securetty... You add the tty of the device you want to allow root access to, like pts/0... DON'T DO THIS THOUGH, unless you don't care that your root password will be sent PLAINTEXT over the internet... He may not be telneting to it across the internet. He may only be doing it from his local network. That being said, I like almost everyone else, recommend ssh *and* su, though I'm guilty of logging in as root across the internet with ssh. Nah. Private key authentication is probably more secure for this. I have my ssh servers generally set up to require key authentication and deny password authentication. This does effectively force me to su with ssh because I haven't set up the key authentication for the root account, but I am still not sure that there is that much more to be gained. Best Wishes, Chris Travers Metatron Technology Consulting begin:vcard fn:Chris Travers n:Travers;Chris email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using ChanIsAvail
Title: using ChanIsAvail
Looks correct to me, I'm using a similar setup...
Sounds like maybe it's a bug in the ChanIsAvailApp, like maybe it's
hardcoded to look in sip.conf...
-Chris
- Original Message -
From:
Poul Pedersen
To: [EMAIL PROTECTED]
Sent: Monday, August 23, 2004 3:15
AM
Subject: [Asterisk-Users] using
ChanIsAvail
Hi
I am trying to use ChanIsAvail to decide if a particular
extension is available in the sip channel
I am using MySQL to hold my SIP friends. and wy cvs version shows Asterisk CVS-08/02/04
my intention is, that if the extension is not available in Sip
channel, I will send the call somewhere else
my extensions file contains the following:
exten = _[123]XX,1,ChanIsAvail(sip/${EXTEN})
exten = _[123]XX,2,dial(sip/${EXTEN},30) exten =
_[123]XX,102,Dial(IAX2/sip01-xx:[EMAIL PROTECTED]/${EXTEN})
if I understand ChanIsAvail correctly this should give med
following:
if i dial extension 111, and that is a local extension, it
dials the sip channel on the other hand, if extension
111 is not avaliable in the local sip channel, it dials on IAX2
But it does not work, if 111 is not a local extension the dial
in priority 2 returns with -1, in my opinion it should never have been
executed
when I have all SIP frinds in sip.conf it works, but it does
not when using MySQL
is this a bug, or is ChanIsAvail not intended to work when SIP
frinds are in MySQL ??
Kind regards
Poul Pedersen
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Re: [Asterisk-Users] determining what number was dialed?
- Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 22, 2004 8:12 PM Subject: Re: [Asterisk-Users] determining what number was dialed? On Sat, 2004-08-21 at 16:37, Paul Concepcion wrote: well, that's our setup (8 analog lines - channel bank - t100P), so it looks like DNIS is out of the question. We do have 8 phone numbers though. Could we have a 1-800 number direct to each of those, then do what you suggested with contexts? What would happen if two people dialed 1-800-a if 1-800-a was pointed to just one phone number? Depends on hunt groups and such. If you have rollover/hunt groups, pointing a 1800 to a number is not very useful for getting DID or DNIS functionality. The different context solution was based on the idea of making each incoming analog line have it's own logical seperation in the dialplan. The trouble is, as you roll from one busy line to the next, there is no information about what group the person dialed into. If you where to split your hunt group into 2 - 4 line groups without talking to the telco, you could fill group 1 up and then be rolling into group b. Same works the other way with wrap around hunting. If you don't have hunt group functionality, and you point a 1800 number to a analog line, then the second phone call will hit a busy signal. I'm using a similar setup here, we have 3 companies in this building. We're using a Merlin Legend PBX with FXO modules. Our incoming lines come from a T1 which terminates on an ADIT 600. It is then split into lines through FXS cards in the ADIT... Company A has 5 lines, the first of which has the 1-800 number pointed to it. It is set up on a linear hunt group to the other 4 lines. No matter what line the call comes in on, since it's in that first set of 5 lines, the PBX answers with Company 'A' IVR... * can do the same thing, I would group the first 5 channels into 'g1' for example, then place them in a context like [companyA]... Company B has 3 lines, same thing only set up on a separate linear hunt group so that it doesn't roll into the first 5 lines or the next 8 lines... Company C has 8 lines... you get the idea... I'm not sure how many companies you have or how many 1-800 numbers you're using... Obviously this is not the ideal setup because it requires the different companies to have a fixed amount of lines whether they use them all or not... A better solution would be a PRI with DNIS but this is what we have to work with and it seems to work well... Like Steven said if you don't have hunt groups, then when someone calls a number and another person calls that same number, the 2nd person will get a busy signal... At least with the way our hunt groups work, the hunt will keep looking in a linear fashion until a line becomes free (resulting in the person hearing ringing)... Hope this helps! -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 Question
You can have two calls per line appearance. If you assign both line appearances (both can be the same extension) you are allowed four calls. On Mon, 23 Aug 2004, Christopher L. Wade wrote: Hi all, I know this is a stupid question, but it is one I've been trying answer for quite some time. Exactly how many simultaneous calls can the Cisco 7940 have, considering you can be talking to one, and have XXX others on hold? Using SIP, is XXX only 1? I've found documents in various places indicating different values in regard to the max number of calls the phone can handle. I'm just trying to nail down the exact number when the phone is only assigned one directory number (extension). Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jeremy Parr Senior Engineer, Network Services BGC Ltd. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk T100P to Merlin Legend
Spectro, This configuration works fine to me. I have Merlin Legend with 2 DS1 100D cards - one goes to upstream PRI (Verizon) and the second one goes to Asterisk with T100P card. The timing is going from Verizon and asterisk configured as pri_net. Crossover cable used between Merling Legend and Asterisk. Merling Legend is configured to Tandem PRI, so Asterisk can dial all extenstion and external numbers and all routing has been done using UDP - which is quite simple and work as charm. There is a sample for Merlin Legend confiuraration (related to connecting between ML and (*) ) - Slot 10 = Asterisk, Slot 11 = Verizon PRI A T1/PRI/BRI Clock Synchronization: A Primary Secondary Tertiary A 11/1 Loop10/1 Loop ASlot # 10: 100D ASlot # 11: 100D-U A 817 10/ 1 Dedicated 890 Yes Long 441 A 818 10/ 2 Dedicated 890 Yes Long 441 A 819 10/ 3 Dedicated 890 Yes Long 441 A 820 10/ 4 Dedicated 890 Yes Long 441 A 821 10/ 5 Dedicated 890 Yes Long 441 A 822 10/ 6 Dedicated 890 Yes Long 441 A 823 10/ 7 Dedicated 890 Yes Long 441 A 824 10/ 8 Dedicated 890 Yes Long 441 A 825 10/ 9 Dedicated 890 Yes Long 441 A 826 10/10 Dedicated 890 Yes Long 441 A 827 10/11 Dedicated 890 Yes Long 441 A 828 10/12 Dedicated 890 Yes Long 441 A 829 10/13 Dedicated 890 Yes Long 441 A 830 10/14 Dedicated 890 Yes Long 441 A 831 10/15 Dedicated 890 Yes Long 441 A 832 10/16 Dedicated 890 Yes Long 441 A 833 10/17 Dedicated 890 Yes Long 441 A 834 10/18 Dedicated 890 Yes Long 441 A 835 10/19 Dedicated 890 Yes Long 441 A 836 10/20 Dedicated 890 Yes Long 441 A 837 10/21 Dedicated 890 Yes Long 441 A 838 10/22 Dedicated 890 Yes Long 441 A 839 10/23 Dedicated 890 Yes Long 441 A 840 10/24 DedicatedYes Long 4 ADS1 SLOT ATTRIBUTES ASlot Type Format Supp Signal LineComp A 10 PRI ESF B8ZS DMI-MOS 2 A 11 PRI ESF B8ZS DMI-MOS 4 A PRI INFORMATION A Slot 10 Switch: Legend-Ntwk A Slot 11 Switch: DMS-100 A System: By extension Base number: A BchnlGrp #: Slot: TestTelNum: NtwkServ:Incoming Routing: A 11 11 DMS-FX By Dial Plan A Channel ID: 23 22 21 20 19 18 17 16 15 14 A 13 12 11 10 9 8 7 6 5 4 A3 2 1 A BchnlGrp #: Slot: TestTelNum: NtwkServ:Incoming Routing: A 80 10 ElecTandNtwk Route Directly to UDP A Channel ID: 23 22 21 20 19 18 17 16 15 14 A 13 12 11 10 9 8 7 6 5 4 A3 2 1 A PRI INFORMATION A NtwkServ: VirtPrivNet A NON-LOCAL DIALPLAN A Range Ptn Dgt Range Ptn Dgt Range Ptn Dgt A1) 1000-1999 01 0418) - 35) - A2) 2150-2169 05 0419) - 36) - A3) 2600-2999 02 0420) - 37) - A4) 3000-3999 03 0421) - 38) - A5) 4800-4899 06 0422) - 39) - A6) 7200-7399 04 0423) - 40) - A7) - 24) - 41) - A8) - 25) - 42) - A9) - 26) - 43) - A 10) - 27) - 44) - A 11) - 28) - 45) - A 12) - 29) - 46) - A 13) - 30) - 47) - A 14) - 31) - 48) - A 15) - 32) - 49) - A 16) - 33) - 50) - A 17) - 34) - APattern 1: APool Absorb Other Digits FRL Call type A1)890- 00 3BOTH A2) -- -- A3) -- -- A4)
Re: [Asterisk-Users] extensions.conf
On Mon, 2004-08-23 at 09:40, Steve Maroney wrote: Hey guys, Is there a way to make asterisk return the calling point after the last command is completed in a context? The goto statement doesn't seem to work as I expected it too. Goto is a unconditional jump. If you expected anything else from it, your expectations where flawed. I have an idea you where expecting a gosub type of operation. For Example: [iax-demo] exten = s,1,Playback(demo-abouttotry) exten = s,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = s,3,Playback(demo-nogo) [some-menu] exten = s,1,Playback(Some-file) exten = s,2,Playback(another-file) exten = s,3,Playback(another-file) exten = s,4,WaitExten(4) exten = s,5,Queue(some-queue) exten = 300,1,Goto(iax-demo,s,1) exten = 300,2,goto(s,3) [some-other-menu] exten = s,1,Playback(Some-different-file) exten = s,2,Playback(some-other-file) exten = s,3,WaitExten(4) exten = s,4,Queue(different-queue) exten = 300,1,Goto(iax-demo,s,1) exten = 300,2,Goto(s,1) You see, I want the iax-demo in both contexts, but after the demos done, I want the caller to be returned to the context, but where depends on the calling context. Any help ? You can accomplish your intended function by using either Macros, channel variables, or an include. Specifically, you could make the IAX demo be on a specific extension and include it appropriately. [iax-demo] exten = 300,1,Playback exten = 300,2,Dial exten = 300,3,Playback [some-menu] exten = . normal stuff include = iax-demo [some-other-menu] exten = .. normal stuff include = iax-demo In this case, you never leave the context [iax-demo] exten = s,1,playback exten = s,2,dial exten = s,3,playback exten = s,4,goto($STORED_CONTEXT,s,3) [some-stuff] exten = s,1,SetVar(STORED_CONTEXT,'some-stuff') exten = ... normal stuff exten = 300,1,goto(iax-demo,s,1) [some-more-stuff[ exten = s,1,setvar(STORED_CONTEXT,'some-more-stuff) exten = ... normal stuff exten = 300,1,goto(iax-demo,s,1) Pretty easy when you think about what is provided to you. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error compiling meetme2
Geoff Nordli wrote:
I am trying to compile the meetme2 application with the latest CVS head and
it fails. Here is the error message that I get. Can someone point me in
the right direction?
[snipped lengthy error message]
Here's a patch. You can apply it by hand, or save it to a text file and
use the 'patch' command to apply it.
It's two simple changes.
-Steve
--- app_meetme2.c 2004-04-07 06:37:18.0 -0500
+++ /usr/src/asterisk/apps/app_meetme2.c2004-08-20
07:05:41.0 -0500
@@ -643,7 +643,7 @@
} *confs;
-static ast_mutex_t conflock = AST_MUTEX_INITIALIZER;
+AST_MUTEX_DEFINE_STATIC(conflock);
#include enter.h
#include leave.h
@@ -1545,7 +1545,7 @@
}else{
if (chan-_state != AST_STATE_UP)
ast_answer(chan);
- res = ast_say_number(chan, cnt, , chan-language);
+ res = ast_say_number(chan, cnt, , chan-language,
(char *)NULL);
}
LOCAL_USER_REMOVE(u);
return res;
___
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[Asterisk-Users] Re: skinny or sccp?
Hi, I got following error, so I made changes in chan_sccp.c and sccp_device.c and replace pthread_create function with with ast_pthread_create the module now loading fine with asterisk :o) PJ [EMAIL PROTECTED] chan_sccp]# make Now compiling chan_sccp.c 706 lines chan_sccp.c: In function `reload_config': chan_sccp.c:552: warning: implicit declaration of function `__use_ast_pthread_create_instead__' Now compiling sccp_actions.c 790 lines Now compiling sccp_channel.c 279 lines Now compiling sccp_device.c641 lines sccp_device.c: In function `sccp_dev_allocate_channel': sccp_device.c:444: warning: implicit declaration of function `__use_ast_pthread_create_instead__' Now compiling sccp_line.c 61 lines ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: skinny or sccp?
Hi, I got following error,so I made changes in chan_sccp.c and sccp_device.c and replace "pthread_create" function with "ast_pthread_create"the module now loading fine with asterisk :o)PJ[EMAIL PROTECTED] chan_sccp]# makeNow compiling chan_sccp.c 706 lineschan_sccp.c: In function `reload_config':chan_sccp.c:552: warning: implicit declaration of function `__use_ast_pthread_create_instead__'Now compiling sccp_actions.c 790 linesNow compiling sccp_channel.c 279 linesNow compiling sccp_device.c 641 linessccp_device.c: In function `sccp_dev_allocate_channel':sccp_device.c:444: warning: implicit declaration of function `__use_ast_pthread_create_instead__'Now compiling sccp_line.c 61 lines ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firmware Update IAXy
Hi ! Just found the beta-firmware-release for iaxy. Which way to get it into IAXy ? Thx for help. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP unphones
I recently saw something just like this and I had it bookmarked... It looks like what you're talking about, but I don't think it uses SIP. Rather some proprietary protocol that transmit RTP... I could be wrong... Check it out... http://www.digitalacoustics.com/lanplay.htm I would agree that it really should be SIP, you wouldn't want to have to rip it out of the wall when the protocol becomes obsolete or when a SIP-Compliant alternative comes out... -Chris - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 23, 2004 7:55 AM Subject: [Asterisk-Users] SIP unphones Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of money to have all those fancy displays and keys tucked away behind a speakergrille and drywall. BTW, I'm not dead-set on SIP, but it seems to be the most logical protocol for this app (NOTIFY msg can carry directions on mike/speaker/two-way, etc) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Guide to getting started with Asterisk on MacOSX
For the record, I have put up a Wiki page with a 'Guide to (Bgetting started with Asterisk on MacOSX' (B (Bhttp://www.voip-info.org/wiki-Asterisk+Getting+Started+on+MacOSX (B (BSome sections are still incomplete, still working on them. (B (BNote that this is based on using the various GUI tools we (Breleased, so there is intentionally nothing about (Bconfiguration files nor CLI commands and it should stay (Bthis way. (B (BIn other words, this is less interesting for the Asterisk (Bsavvy readership of this list, but it will be helpful for (Bnewcomers searching the archives. (B (BAlso, you may want to use this as a pointer to give to the (Bkind of newbie user who is too busy to learn Asterisk. (BTell them to get an old iMac from the junkyard or a second (Bhand shop. I picked one up for 80 bucks last week and it's (Balready been repurposed as a family PBX for a guy who (Balways bothered me but never followed the pointers I gave (Bhim to read. It took 1 minute to install and 3 minutes to (Bconfigure. (B (Brgds (Bbenjk (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BGANBARE! NIPPON! (BYahoo! JAPAN JOC OFFICIAL INTERNET PORTAL SITE (Bhttp://mail.ganbare-nippon.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk in india
I dont think they are using Cable. They may be having ISDN lines. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Totaro Sent: Monday, August 16, 2004 10:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk in india - Original Message - From: usedcanon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 16, 2004 6:18 PM Subject: RE: [Asterisk-Users] asterisk in india I am a fan of Mohammad Rafi, so could do with some calls from India ;-) Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kanuri, Seshu Sent: 13 August 2004 21:14 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk in india There is one more twist to this. The broadband connection in India really sucks as it carries heavy load of Hindi Films and Hindi Film Songs. We have a few Broadband connections running from India. When the call comes from India to USA, the VOIP Broadband Phone turns into a Juke Box and all that our users in US can listen to is the Voice Overs of melodies of Lata Mangeshkar, Manna Dey and Mohammed Rafi. The VOIP Voice is critically low. Another factor is that the noise eliminates any real use of advanced codecs like G729. The Internet bandwidth is so expensive to run from India, there is no real use of hosting your Asterisk box there, Vis-a-Vis the same in USA. And offcourse you cannot legally interconnect VOIP with Analog or Digital PRI lines there. so how does dell do it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: HFC-S in NT mode, wiring?
On Mon, Aug 23, 2004 at 04:15:25PM +0200, Simone Ricci wrote: I've got an old HFC-S card to play with, and I would like to use it in NT mode. I've a problem only: wiring. I can't fully understand the instructions I was able to find online. Someone can point me to a site which explains the whole procedure clearly (like with some schematics, even in ASCII)? This site may be usefull, although it is not an asterisk page ;-) http://home.foni.net/~jolly2/download/PBX4Linux-2.3.html -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Phone recommendation for Receptionist
Try the netweb-301 Hard IP Phone attached as a Pic Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James H. Thompson Sent: Sunday, August 22, 2004 11:54 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP Phone recommendation for Receptionist el Flynn wrote: Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Other than the incoming lines, the receptionist would need the normal keyphone type stuff -- call pickup, park, hold, forward etc. What would you guys recommend? How about a touch screen LCD display running the Asterisk Flash Operator Panel? Or mabe a Tablet PC running Asterisk Flash Operator Panel? Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users attachment: products_301.jpg___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions.conf
On Mon, Aug 23, 2004 at 09:40:54AM -0500, Steve Maroney wrote: For Example: [iax-demo] exten = s,1,Playback(demo-abouttotry) exten = s,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = s,3,Playback(demo-nogo) [some-menu] exten = s,1,Playback(Some-file) exten = s,2,Playback(another-file) exten = s,3,Playback(another-file) exten = s,4,WaitExten(4) exten = s,5,Queue(some-queue) exten = 300,1,Goto(iax-demo,s,1) exten = 300,2,goto(s,3) [some-other-menu] exten = s,1,Playback(Some-different-file) exten = s,2,Playback(some-other-file) exten = s,3,WaitExten(4) exten = s,4,Queue(different-queue) exten = 300,1,Goto(iax-demo,s,1) exten = 300,2,Goto(s,1) You see, I want the iax-demo in both contexts, but after the demos done, I want the caller to be returned to the context, but where depends on the calling context. Any help ? Since you have the extensions the same in both contexts, why don't you just change it to be like this: [iax-demo] exten = 300,1,Playback(demo-abouttotry) exten = 300,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = 300,3,Playback(demo-nogo) [some-menu] exten = s,1,Playback(Some-file) exten = s,2,Playback(another-file) exten = s,3,Playback(another-file) exten = s,4,WaitExten(4) exten = s,5,Queue(some-queue) include = iax-demo wxten = 300,4,goto(s,3) [some-other-menu] exten = s,1,Playback(Some-different-file) exten = s,2,Playback(some-other-file) exten = s,3,WaitExten(4) exten = s,4,Queue(different-queue) include = iax-demo exten = 300,4,Goto(s,1) -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with asterisk and postgresql
Dipak; Is the postmaster running with the -i options? Is localhost defined in HOSTS? Is localhosts allowed access to postgres via pg_hba.conf? I am assuming that the postgres package installed as you were able to compile * with postgres support, so check out your postgres install. The error stats that * can not connect to postgres. Lyle - Original Message - From: DIPAK PAUL [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 23, 2004 4:03 AM Subject: [Asterisk-Users] Problem with asterisk and postgresql Hi Every one and Fabio Donaggio Fabio Donaggio you have faced same type of problem with postgresql with asterisk. Did you solved your problem. Please help me. The problem is as follows: When i reload the asterisk I had got this error == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Aug 23 14:22:33 ERROR[1076245120]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! Aug 23 14:22:33 ERROR[1076245120]: cdr_pgsql.c:300 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on host localhost and accepting TCP/IP connections on port 5432? And I have configure the cdr_pgsql.conf file as: [global] hostname=localhost port=5432 dbname=asterisk password= user=postgres Please help me. With best regards. Dipak Kumar Paul Tryarc LLC _ Sports, sports and more sports! Keep up with all the action! http://www.msn.co.in/sports/ Stay connected with MSN Sports! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP and dialing out
I have recently found out that * is very strict about dialing out. If a number isn't listed in extensions.conf, good luck trying to dial it. I had to put in a line for each of our area codes with XXX's before I could dial local numbers. Anyway..now that I 'can' dial them, as soon as the other party picks up the phone I get a busy signal on my end. Also..just tried an IpPhone to IpPhone call. Works fine. However, when the other person hung up, I (on the swissvoice) got that busy signal again until I hung up. Any ideas on that busy signal plague? Thanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about dial out via Zap
You probably have a mistake in your dialplan in extensions.conf and/or zapata.conf. And my mind reader failed to compile today, so I have no more guesses for you as you did not post anything about the pertiant configs or type of phones involved. Lyle - Original Message - From: Hall, Eric M. [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 23, 2004 7:32 AM Subject: [Asterisk-Users] Question about dial out via Zap Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1 answered SIP/6149236651-eafb Urgent handler -- Hungup 'Zap/1-1' Urgent handler Urgent handler -- Hungup 'Zap/2-1' Urgent handler Urgent handler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: 2 servers
Dave, I am implementing this solution and would appreciate if you can send me the doc at this email address - [EMAIL PROTECTED] Thanks Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Cook Sent: Monday, August 23, 2004 8:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: 2 servers Quoting [EMAIL PROTECTED]: How do I get town A people to dial 201 and it will go to sown B's server's 201 SIP users Please not that I'm only a newbie and my terms may be wrong but I'm really having a bod time with this Please help Thanks ALtus I have a doc on it. (Sorry was going to copy/paste but my mail reader didn't like the columns from the doc.) If you want it drop me a line and I'll send you the file. (Should also probably put it in the wiki :-) dbc. -- David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 Question
On Mon, 23 Aug 2004 09:07:26 -0500, Christopher L. Wade [EMAIL PROTECTED] wrote: Hi all, I know this is a stupid question, but it is one I've been trying answer for quite some time. Exactly how many simultaneous calls can the Cisco 7940 have, considering you can be talking to one, and have XXX others on hold? Using SIP, is XXX only 1? I've found documents in various places indicating different values in regard to the max number of calls the phone can handle. I'm just trying to nail down the exact number when the phone is only assigned one directory number (extension). The Cisco 7940 can handle 2 directory numbers, not one. The phone can handle 2 simultaneous calls per line/DN plus an extra call on each line for call transfer. Thanks, Chris -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue Monitor
-Original Message- From: Kevin [mailto:[EMAIL PROTECTED] Sent: Saturday, August 21, 2004 5:16 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Queue Monitor I understand that putting monitor-format in the queues.conf file will start monitor recording of an active queue call. Is there a way to automatically do the post call processing like the 'm' option like when specifying the use of the monitor command? Try monitor-join=yes in queues.conf after the monitor-format line. This seems to join the two files together for me. Hope this helps, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)
Get an HFC-PCI card instead That leaves all the stuff to Zaptel, and the card costs you 50 or less On 20. aug. 2004, at 06.21, Shaun Ewing wrote: Hello all, I was wondering if anybody knows where one might obtain a PCI ISDN card supporting a single BRI for use with Asterisk in Australia (and using something like chan_capi). Because of the Isdn4Linux DTMF issue, I don't want one of those cards. I've already spent too much time messing about with my current card. I'm after something like the AVM Fritz! cards. I found one place that sells them, but when comparing to the prices they're available for overseas the pricing seemed a bit high. Other than that - if anybody has any recommendations for reasonably priced cards, I'm open to suggestions. If it's not possible - does anybody have any recommendations for retailers overseas? Some searching has found retailers in Germany with the information in German, but as I've only just started learning German I wouldn't be too comfortable constructing a query/etc. Thanks, Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Install on Kernel 2.6.x
i know asterisk itself will install on a linux kernel 2.6.x, but i've seen places say that the zaptel drivers wont? is this still true? is it possible to build asterisk/zaptel on a linux 2.6.x kernel? -- Shawn Parker Network Administrator Cumulus Broadcasting, LLC. Columbia-Jefferson City, Missouri 1.573.449.4141 [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk ------- Quintum SIP Registration
Hi All I'm trying with no luck to connected the Quintum D series Gateway with the new SIP release to asterisk. Have anyone done this? If yes then how should I configure the sip.conf to accept the registration? maybe a sample config? Thanks /Krystian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP unphones
At $65, the Grandstream Budgetone series can auto answer right to the speakerphone. I doubt you'll find a SIP device for much cheaper than that. When set in that mode, it does give relatively little flexibility as far as two way, speaker settings, etc, but it does work. One of my customers needed an intercom mic into their office, and one of those worked like a charm. Brian On Mon, 23 Aug 2004 09:55:48 -0500, Jay Milk [EMAIL PROTECTED] wrote: Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of money to have all those fancy displays and keys tucked away behind a speakergrille and drywall. BTW, I'm not dead-set on SIP, but it seems to be the most logical protocol for this app (NOTIFY msg can carry directions on mike/speaker/two-way, etc) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HFC-S in NT mode, wiring?
Hi Simone, I think the pbx4linux site offers good instructions what you have to do to connect a phone directly to your hfc-s based card or even to connect it using a NT1 (NTBA). Just read chapter 2 of this documentation: http://home.foni.net/~jolly2/download/PBX4Linux-2.3.html bye bo At 16:15 23.08.2004, you wrote: I've got an old HFC-S card to play with, and I would like to use it in NT mode. I've a problem only: wiring. I can't fully understand the instructions I was able to find online. Someone can point me to a site which explains the whole procedure clearly (like with some schematics, even in ASCII)? TIA, Simone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple X100P config?
Can somebody with more than one X100P card in their setup post its zaptel.conf, zapata.conf and extensions.conf files (only relevant parts for group dial-out needed)? I added a second X100P to my * and it shows as an available channel but when I Dial,zap/2 I get unable to create channel error. Thanks in advance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP unphones
Jay Milk wrote: Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of money to have all those fancy displays and keys tucked away behind a speakergrille and drywall. BTW, I'm not dead-set on SIP, but it seems to be the most logical protocol for this app (NOTIFY msg can carry directions on mike/speaker/two-way, etc) Grandstream phone has this feature. Would be hard to find anything cheaper. Alternative would be to use an analog auto-answer phone attached to a SIP ATA. Would cost more than a Grandstream though. For more information see: http://www.voip-info.org/wiki-Asterisk+paging+and+intercom http://www.voip-info.org/wiki-Asterisk+phone+door http://www.voip-info.org/wiki-Analog+Telephone+Information Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
The wiki page on Asterisk + Nat (http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions) lists the possible types of server/client relationships with one most probably interesting to us being #3. snip 3. Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk snip Then it goes on to say: * #3 Works with port forwarding and some header mangling magic Can somebody explain a little more about the header mangling magic as it is not discussed anywhere else in the document. Currently I have my firewall port forwarding 5060 to my asterisk server and the UDP port range forwarded as well. Registration works, but no audio. Obviously the RTP stuff is not happy with the forwarding. dbc. -- David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uniden UIP200 Review
On Sun, 22 Aug 2004 18:37:41 -0600, Ryan Courtnage [EMAIL PROTECTED] wrote: Tim // NCS wrote: - You MUST use nat=never in sip.conf. The UIP200 does not like rfc3581 (rport), and will not reply to requests that contain it. Using 'nat=never' in sip.conf disables *'s support for this rfc. Uniden has acknowledged the issue (DR#60). Are you running RC1 or RC2?. We are running a pre-RC1 version of * and nat=yes works fine. After we upgraded to RC2 we were unable to call the UIP200 extension. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp make error mmx.h
Hello, I receive an error during the make of spandsp on a sparc machine. I have tried it with libtiff 3.5.7 and 3.6.0, but get the same error. spandsp/mmx.h: In function `mm_support': spandsp/mmx.h:72: error: unknown register name `edx' in `asm' spandsp/mmx.h:72: error: unknown register name `ecx' in `asm' spandsp/mmx.h:72: error: unknown register name `ebx' in `asm' spandsp/mmx.h:72: error: unknown register name `eax' in `asm' make[2]: *** [echo.lo] Error 1 make[2]: Leaving directory `/usr/src/fax/spandsp-0.0.1/src' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/fax/spandsp-0.0.1/src' make: *** [all-recursive] Error 1 thanks Adrian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BT Communicator (SIP???) and Asterisk
gARetH baBB wrote: On Fri, 20 Aug 2004, Robert Boardman wrote: BT are providing a SIP gateway for PSTN through the BT communicator with Yahoo Messenger, I have done an ethereal trace and found that the BT Communicator side of the software is using SIP, so in theory I could add more PSTN lines to Asterisk for BT using SIP, but I am having problems deciphering the trace so my question is has anyone else tried to get BT Communicator work with Asterisk, or would someone be willing to help get this SIP provider to work? The only issue with it working with Asterisk is the current lack of reasonable Outbound Proxy support - or BT telling you where a direct SIP regitration server is (I've looked for one and failed). Otherwise it's easy, I've used Communicator with a range of the usual soft phones (X-lite etc.). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Gareth Heartened by your that you have got x-lite working, I have been trying, but failing to now get x-lite working, don suppose you could send me a quick screen shot of you x-lite settings? thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960G, Skinny.conf, and reboots
I could use some skinny/Cisco help here. Was finally able to get the phone registered to * but whenever someone tries to call that phone it freezes and reboots itself. Same thing happens when you pick the handset up off the 7960G; it locks and reboots about 5 sec later. Here is what * shows when I plug the phone in: -- Starting Skinny session from 64.72.107.1 Device SEP000F3442E4A7 is attempting to register -- Device 'Matthew' successfuly registered Requesting capabilities RECEIVED UNKNOWN MESSAGE TYPE: 2b Received CapabilitiesRes RECEIVED UNKNOWN MESSAGE TYPE: 2b Buttontemplate requested Recieved SoftKey Template Request Received SoftKeySetReq Received LineStateReq RECEIVED UNKNOWN MESSAGE TYPE: 2d Received Time/Date Request And when I pick up the hand set: -- Starting simple switch on '[EMAIL PROTECTED]' No dialtone heard. 5 seconds later phone reboots. -- Model: CP-7960G App Load ID: P00306000400 Boot Load ID: PC030301001 Asterisk CVS-HEAD-07/14/04-12:20:42 skinny.conf -- [Matthew] device=SEP000F3442E4A7 version=P00306000400 nat=yes callerid=Matthew Boehm 3044 transfer=1 callwaiting=1 threewaycalling=1 context=matthew linelabel=my line lable mailbox=100 line = 100 dtmfmode=rfc2833 Any help on this would be appreciated. Thanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange problem with Dial
On Fri, Aug 20, 2004 at 07:14:06PM -0400, Michael George wrote:
I'm trying to add an emergency dial to my context. However, when I try to
dial it, I get caught in an endless loop.
For debugging, I have pared out nearly all the control flow and just have
ChanIsAvail() and Dial() called. Using two different extensions to call teh
same number, I get two different actions by *.
Here is the vvverbose output:
-- Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1, Zap/1/5932336||T) in new stack
-- Called 1/5932336
-- Zap/1-1 answered Zap/3-1
-- Attempting native bridge of Zap/3-1 and Zap/1-1
-- Hungup 'Zap/1-1'
== Spawn extension (internal, 95932336, 1) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
-- Starting simple switch on 'Zap/3-1'
-- Executing ChanIsAvail(Zap/3-1, Zap/1) in new stack
-- Hungup 'Zap/1-1'
-- Executing NoOp(Zap/3-1, avail: Zap/1-1) in new stack
-- Executing Dial(Zap/3-1, Zap/1/5932336||T) in new stack
Aug 20 19:09:27 NOTICE[294926]: app_dial.c:714 dial_exec: Unable to create
channel of type 'Zap'
== Everyone is busy/congested at this time
-- Executing NoOp(Zap/3-1, busy) in new stack
-- Hungup 'Zap/3-1'
the first way, I'm matching this context:
exten = _9NXX,1,Dial(${TrunkLocal}/${EXTEN:${TrunkMSD}},,T)
exten = _9NXX,2,Congestion
exten = _9517XXX,1,Dial(${TrunkLocal}/${EXTEN},,T})
exten = _9517XXX,2,Congestion
The second way I'm mathing this one:
exten = 911,1,ChanIsAvail(Zap/1)
exten = 911,2,NoOp(avail: ${AVAILCHAN})
exten = 911,3,Dial(Zap/1/5932336,,T)
exten = 911,102,NoOp(None Avail)
exten = 911,104,NoOp(busy)
Why does the latter fail at the Dial()?
I am still having a problem with this call flow. I just updated my * source
and rebuilt and reinstalled.
I want to implement the feature I saw in Tips Tricks where before calling an
emergency number, the outgoing channel(s) are checked for availability so one
can be cleared before trying to dial. The example code is this:
exten = 911,1,ChanIsAvail(Zap/1)
exten = 911,2,Dial(Zap/1/911)
exten = 911,3,Hangup()
exten = 911,102,SoftHangup(Zap/1-1)
exten = 911,103,Wait(1)
exten = 911,104,Goto(1)
However, every time I try this flow, the Dial() called immediately after the
ChanIsAvail() will fail as busy (return to prio+101). I know the channel is
available because I see that ChanIsAvail() went to prio+1.
Has there been a change in the code that might cause this? Perhaps an issue
in the zaptel driver?
Thanks!
--
-M
There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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