Re: [Asterisk-Users] Polycom Reboot Script
Matthew T. O'Connor wrote: Kristian Kielhofner wrote: Matthew T. O'Connor wrote: Any Ideas? Have a look at /etc/asterisk/sip_notify.conf look for: [polycom-check-cfg] So, from the CLI: asterisk -r sip notify polycom-check-cfg [name] Isn't sip_notify.conf just an Asterisk 1.2 thing? I'm running 1.0.9. I'm trying to setup a production system for my company, do you think 1.2 is ready for that? Thanks, Matt Matt, It sure is! You should be testing it! :) Test it and see, but 1.2 will be STABLE pretty soon here... -- Kristian Kielhofner ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk addons
folks, I am doing an install of AMP from the AMP PDF file. I get to the part where I need to install the addons and I get the folloing error on a make. I have done a make clean before I did a make. I can see the errors, common.o is in place as far as I can tell. format_mp3.so is no where to be found. I am on page 10 of this manual http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.4.pdf Any help would be greatly aprecieated. Thank you for your time, Tommy ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:15:27: asterisk/file.h: No such file or directory app_addon_sql_mysql.c:16:29: asterisk/logger.h: No such file or directory app_addon_sql_mysql.c:17:30: asterisk/channel.h: No such file or directory app_addon_sql_mysql.c:18:26: asterisk/pbx.h: No such file or directory app_addon_sql_mysql.c:19:29: asterisk/module.h: No such file or directory app_addon_sql_mysql.c:20:34: asterisk/linkedlists.h: No such file or directory app_addon_sql_mysql.c:21:31: asterisk/chanvars.h: No such file or directory app_addon_sql_mysql.c:22:27: asterisk/lock.h: No such file or directory cdr_addon_mysql.c:17:29: asterisk/config.h: No such file or directory cdr_addon_mysql.c:18:30: asterisk/options.h: No such file or directory cdr_addon_mysql.c:19:30: asterisk/channel.h: No such file or directory cdr_addon_mysql.c:20:26: asterisk/cdr.h: No such file or directory cdr_addon_mysql.c:21:29: asterisk/module.h: No such file or directory cdr_addon_mysql.c:22:29: asterisk/logger.h: No such file or directory cdr_addon_mysql.c:23:26: asterisk/cli.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/root/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o common.o common.c common.c:1:29: asterisk/logger.h: No such file or directory common.c: In function `decode_header': common.c:93: warning: implicit declaration of function `ast_log' common.c:93: error: `LOG_WARNING' undeclared (first use in this function) common.c:93: error: (Each undeclared identifier is reported only once common.c:93: error: for each function it appears in.) make[1]: *** [common.o] Error 1 make[1]: Leaving directory `/root/asterisk-addons/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Reboot Script
Kristian Kielhofner wrote: Matthew T. O'Connor wrote: Isn't sip_notify.conf just an Asterisk 1.2 thing? I'm running 1.0.9. I'm trying to setup a production system for my company, do you think 1.2 is ready for that? It sure is! You should be testing it! :) Test it and see, but 1.2 will be STABLE pretty soon here... I'm happy to help out and test out 1.2 beta, but I don't think pretty soon will be soon enough. We are opening our new office in less than two weeks. I can't imagine that 1.2 will be out of Beta by then. Thanks for you help. Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FAX with Asterisk
Hi, I want to do FAX through Asterisk with the following scenario: Fax Machine --Nortel PBX --- E1 (euro-isdn) --- Asterisk - SIP -Asterisk E1 (euro-isdn)-Nortel PBX-- Fax Machine Is there anyone who can help me to configure the above scenario without any extra application/software. I would appreciate if anyone help me. Regards Nahid ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FAX with Asterisk
Hi Nahid, I think youll want a fax on-ramp and off-ramp on your asterisk boxes instead of trying to send a fax using VoIP (SIP). I believe it is possible but not recommended. There are technical reasons for this that you can find online in many places. Basically asterisk answers the fax and sends it to your fax program. the fax progam receives the fax, turns it into a TIFF file and emails it to the other end of your network. then the process is reversed and the TIFF is faxed via software out the T1 at the other end. This process has a standard called T.37. Im not sure if there is currently support for this in asterisk or not (search the archieves) but its what I do with our Cisco router and a very neat little windows fax program called T37FSP from Sandler Consulting. You could prolly use the free version for testing. hope this helps, cheers, Mick Nahid Hossain [EMAIL PROTECTED] wrote in message news:!~!UENERkVCMDkAAQACABgABittf/[EMAIL PROTECTED] Hi, I want to do FAX through Asterisk with the following scenario: Fax Machine --àNortel PBX ---à E1 (euro-isdn) ---à Asterisk -à SIP -àAsteriskà E1 (euro-isdn)-àNortel PBX--à Fax Machine Is there anyone who can help me to configure the above scenario without any extra application/software. I would appreciate if anyone help me. Regards Nahid ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm04b hangup problem
Hi, I'm sorry about the false information. It seems after the crash, the problems is still exist. Anyone can help me? Could it be IRQ issue? Here is output from cat /proc/interrupt: CPU0 0: 92807252 XT-PIC timer 1: 8 XT-PIC i8042 2: 0 XT-PIC cascade 5: 92732654 XT-PIC ohci_hcd, wctdm 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi, ehci_hcd 10: 92730937 XT-PIC SiS SI7012, ohci_hcd, wctdm 11: 95761662 XT-PIC eth0, ohci_hcd, wctdm 12: 66 XT-PIC i8042 14: 163276 XT-PIC ide0 15: 997605 XT-PIC ide1 NMI: 0 ERR: 0 Best Regards, Stevanus stevanus wrote: Hi, Yesterday, the asterisk machine was crash :S. But after the crash, it seems the previous problems were eliminated. I will notice it in about a week or two. If it's stable now, so the recommended solution when there are problems with asterisk is to restart the machine? Weird. Does nobody like to share any comments? Just curious :P Best Regards, Stevanus stevanus wrote: Any thought anyone? stevanus wrote: Hi, I have severe problem here.. My asterisk server use tdm04b from digium and is often incapable of detecting hangup signal. It is happened occasionally in incoming call so I have to watch fop all the time and hangup the channel manually there. Another problem is when an outgoing call was placed and the caller ended the conversation, the tdm04b did not hangup the channel. So when the caller does off hook too fast and interpreted by asterisk as hold, both zap channel will be connected by asterisk as the caller hangup the second call. Anyone experiences this issue? Is it possible that this is caused by improper setting in rxgain or txgain? Currently, I set rxgain = 15.0 and txgain = 5.0.. Thanks.. Best Regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error compiling on solaris 10
hi frank, i was able to find gmake at /usr/sfw/bin, however, i got this new error : gmake[1]: Leaving directory `/export/home/fst/ice/cvs/asterisk/stdtime' cd editline unset CFLAGS LIBS test -f config.h || ./configure creating cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc ) works... yes checking whether the C compiler (gcc ) is a cross-compiler... no checking whether we are using GNU C... yes checking whether gcc accepts -g... yes checking how to run the C preprocessor... gcc -E checking host system type... sparc-sun-solaris2.10 checking for a BSD compatible install... install checking for ranlib... : checking for ar... no checking for tgetent in -ltermcap... yes checking for termcap.h... no checking for term.h... yes checking for curses.h... yes checking for sys/cdefs.h... no checking for vis.h... no checking for issetugid... yes checking for strlcat... yes checking for strlcpy... yes checking for fgetln... no checking for strvis... no checking for strunvis... no updating cache ./config.cache creating ./config.status creating Makefile creating config.h gmake -C editline libedit.a gmake[1]: Entering directory `/export/home/fst/ice/cvs/asterisk/editline' /bin/sh makelist -h common.c common.h /bin/sh makelist -h emacs.c emacs.h /bin/sh makelist -h vi.c vi.h /bin/sh makelist -fh common.h emacs.h vi.h fcns.h sed: command garbled: ccygwin /bin/sh makelist -fc common.h emacs.h vi.h fcns.c if [ = cygwin ]; then cat fcns.c | sed -e s/sys\.h/config.h/g fcns.c.copy; mv --force fcns.c.copy fcns.c; fi /bin/sh makelist -bh common.c emacs.c vi.c help.h sed: command garbled: ccygwin /bin/sh makelist -bc common.c emacs.c vi.c help.c if [ = cygwin ]; then cat help.c | sed -e s/sys\.h/config.h/g help.c.copy; mv --force help.c.copy help.c; fi /bin/sh makelist -e common.c emacs.c vi.c chared.c el.c hist.c key.c map.c parse.c prompt.c read.c refresh.c search.c sig.c term.c tty.c fcns.c help.c editline.c gcc -c -Wall -pipe -g3 -O -DSOLARIS -I../include/solaris-compat '-D__RCSID(x)=' '-D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. editline.c -o editline.o_a In file included from editline.c:18: term.c: In function `term_set': term.c:913: warning: implicit declaration of function `tgetent' term.c:931: warning: implicit declaration of function `tgetflag' term.c:940: warning: implicit declaration of function `tgetnum' term.c:943: warning: implicit declaration of function `tgetstr' term.c:943: warning: passing arg 3 of `term_alloc' makes pointer from integer without a cast term.c: In function `term_echotc': term.c:1441: warning: assignment makes pointer from integer without a cast gcc -c -Wall -pipe -g3 -O -DSOLARIS -I../include/solaris-compat '-D__RCSID(x)=' '-D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. np/fgetln.c -o np/fgetln.o_a gcc -c -Wall -pipe -g3 -O -DSOLARIS -I../include/solaris-compat '-D__RCSID(x)=' '-D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. np/vis.c -o np/vis.o_a np/vis.c: In function `svis': np/vis.c:204: warning: implicit declaration of function `alloca' gcc -c -Wall -pipe -g3 -O -DSOLARIS -I../include/solaris-compat '-D__RCSID(x)=' '-D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. np/unvis.c -o np/unvis.o_a gcc -c -Wall -pipe -g3 -O -DSOLARIS -I../include/solaris-compat '-D__RCSID(x)=' '-D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. history.c -o history.o_a gcc -c -Wall -pipe -g3 -O -DSOLARIS -I../include/solaris-compat '-D__RCSID(x)=' '-D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. tokenizer.c -o tokenizer.o_a gcc -c -Wall -pipe -g3 -O -DSOLARIS -I../include/solaris-compat '-D__RCSID(x)=' '-D__COPYRIGHT(x)=' '-D__RENAME(x)=' '-D_DIAGASSERT(x)=' -I. readline.c -o readline.o_a readline.c: In function `_history_expand_command': readline.c:396: warning: implicit declaration of function `alloca' cru libedit.a editline.o_a np/fgetln.o_a np/vis.o_a np/unvis.o_a history.o_a tokenizer.o_a readline.o_a gmake[1]: cru: Command not found gmake[1]: *** [libedit.a] Error 127 gmake[1]: Leaving directory `/export/home/fst/ice/cvs/asterisk/editline' gmake: *** [editline/libedit.a] Error 2 pls advise on how i can fix this. thnks - Original Message - From: Frank Tarczynski [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, August 29, 2005 3:08 AM Subject: RE: [Asterisk-Users] error compiling on solaris 10 Message: 11 Date: Sun, 28 Aug 2005 11:46:29 +0800 From: chris [EMAIL PROTECTED] Subject: [Asterisk-Users] error compiling on solaris 10 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 hello, i change my OS from solaris 9 to solaris 10, tried running make to install asterisk but i'm getting the error below: make -C editline libedit.a To start try using gmake. It's there, just add it to your PATH. Frank
Re: [Asterisk-Users] mrtg+manager.conf
Dear friends , Through the a bit more probeing i found out that we need to use the username and the password which we have given in the manager.conf as the parameters to the perl script file . even though i get the error messages as follows [EMAIL PROTECTED] root]# ./a.out -h localhost -u vrk -p vrk -1 SIP Constant subroutine POLLIN redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine POLLPRI redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine POLLOUT redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine POLLRDNORM redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine POLLWRNORM redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine POLLRDBAND redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine POLLWRBAND redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine POLLERR redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine POLLHUP redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine POLLNVAL redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine _IOFBF redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine _IOLBF redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine _IONBF redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine SEEK_SET redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine SEEK_CUR redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Constant subroutine SEEK_END redefined at /usr/lib/perl5/5.8.0/i386-linux-thread-multi/DynaLoader.pm line 65535. Error: (Missing channels) Syntax: ./a.out -h host -u username -p password [-cwv] * --username -u Username * --password -p Password * --host -h Host --port -P n Port (if not using 6060) --chan1 -1 xxx Display channel xxx as 1. --chan2 -2 xxx Display channel xxx as 2. --verbose -vVerbose --help -H This help cat /etc/asterisk/manager.conf ; ; Asterisk Call Management support ; [general] enabled = yes port = 5038 bindaddr = *.*.*.* (my ip) [vrk] secret = vrk ;deny=0.0.0.0/0.0.0.0 permit=*.*.*.*/255.255.255.254 ( my ip ) read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user your help will be immensely appreciated with regards rk --- rkvalmiki [EMAIL PROTECTED] wrote: Dear freinds, The pl script file which is available in the asterisk monitoring section of the voip-info.com expects username ,password and host parameters . Which one we should provied is the acconts we registered for asterisk or any thing else your help will be immensely appreciated . with regards rk ___ Too much spam in your inbox? Yahoo! Mail gives you the best spam protection for FREE! http://in.mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ How much free photo storage do you get? Store your friends 'n family snaps for FREE with Yahoo! Photos http://in.photos.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to use * and # as part of number in dialcommand
Damon Estep wrote: I did not see an actual error message in your first post, what is the error message? Damon, Well, it is not a 'real' error message, asterisk logs it as a 'warning' , but for me it looks like it is linked to the problem. See my comments in the logs between [ ]. -- Executing Dial(Zap/2-1, Zap/4/*31*040268000) in new stack -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called 4/*31*040268000 -- Zap/4-1 is making progress passing it to Zap/2-1 [thus far it looks okay] -- Channel 0/1, span 2 got hangup [hmm, it seems that the channel was hangup, so it failed] Aug 27 23:32:28 WARNING[17591]: app_dial.c:412 wait_for_answer: Unable to forward voice [this warning indicates that asterisk was unable to forward voice, I think this is because of the *31* in the dial string, because when I leave the *31* out, the warning is not there and the connection is made without problems] -- Hungup 'Zap/4-1' == No one is available to answer at this time -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' Thank you for your time trying to help me out! Regards, Michel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIALSTATUS for Originate
On Sun, 2005-08-28 at 12:45 -0700, Geoff Karl wrote: If you are using Async and the action ID for some reason the Event: Newstate doesn't respond with the ActionID, but only a automatically generated Uniqueid. When using Async you receive an OriginateSuccess or OriginateFailure event. These events contain the proper ActionID (i.e. the one you set with the Originate action) and they contain an integer field reason, that indicates the reason for the failure. =Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference and HFC card conflict: no solution??
Hi, I'm using a HFC card on my asterisk box. I tried to make a conference but it doesn't work. I read on internet to use ztdummy but my server has no uhci (only ohci but it doesn't work) so I cannot use it. I tried zaprtc but after loading the module (it appears when typing lsmod) nothing has changed. Should I buy a x100p to get the right timing? Or there is another solution? TIA Giorgio -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digi QuadMicro ISDN adapter with asterisk?
Hi all, Has anybody used this card (Digi QuadMicro) with asterisk or can anybody tell me the likelyhood of it working out OK? I need a multiport BRI adapter for use with asterisk in Japan and this card seems to support INS64 (Japanese BRI standard) and also CAPI 2.0. here is a link to the datasheet: http://www.digi.com/pdf/prd_mca_datafirequad.pdf Ive only used asterisk with Cisco SIP gateways so Im not sure if this is enough information. thanks again for any help, cheers, Mick Hastings ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc troubles
Hi, you are right!!! I tried zaprtc but even if it doesn't give me errors and I loaded it as a module, it is not working: with or without is the same, conference doesn't work with asterisk and HFC card. Giorgio -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Japanese ISDN BRI card for asterisk (INS64) where to start?
Hi It looks to me that the intel board is the same as the dialogic board. Clive On 29 Aug 2005 at 11:43, Mick Hastings wrote: Hi All, I currently run asterisk in our office (in Japan) and use a cisco PRI gateway for connection to the PSTN. I would like to setup some more systems for our smaller offices (in Japan) that would use BRI and preferably using a PCI card in the asterisk box and not a seperate Cisco gateway (expensive). HOWEVER, Japan has this INS64 protocol for their BRI lines and im not sure what cards are available that are compatible with asterisk and Japanese BRI (INS64). I know that it is supported by Cisco (like they support Japanese T1 PRI (INS1500)) but it just adds to the cost and is another piece of hardware. I tried searching the archives and only found a few references to INS64 and it didnt sound too promising. I then searched the net and found this Intel/Dialogic board: BRI/80-PCI BRI/PCI Series High-Density ISDN Basic Rate Interface Boards (for details see: http://www.intel.com/network/csp/products/7007web.htm) It seems to support INS64 but appears to only have windows drivers. Has anybody used this cards with asterisk? is it possible? or even likely that it would be supported by any of the linux ISDN drivers? I also noticed some other mentions of 'ISDN protocol converters' What are these specifically? (im guessing they convert between US BRI standards and INS64), how much are they? where do I get one? Has anybody out there got an asterisk system running with INS64 connections to their box? If so could you please let me know how you are doing it, else can anybody offer any information as to where I should start to look for more informaion this topic? I really appreciate the help. cheers, Mick Hastings ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: chan_unical-MFC/R2 CPU usage problem
Hi, My variant is standard ITU, I tried almost all versions I could put my hand on to no avail. I tried also to profile the channel and related libraries to no avail as my profiling skills on linux are abit lacking. If anybody with this problem and knows how to profile multithreaded apps on linux then we might at least pinpoint the location of the error. I cant realy put the machine into active duty if I cant solve the problem. Btw, what version of libtiff are you using? It difficult to believe that it might be related as I don’t need the fax functionality. Mine is the version that comes with CentOS 4.1 which is 3.6.1. Hadi. Message: 21 Date: Wed, 24 Aug 2005 13:01:15 -0300 From: Leonardo Gomes Figueira [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_unical-MFC/R2 CPU usage problem To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=UTF-8; format=flowed Hi, Hadi Jadallah wrote: I have installed chan_unicall and MFC/R2 successfully, and is runnign fine. But I noticed that once unicall is installed, asterisk CPU usage as reported by 'top', jumps to 99% every few seconds. I have no incoming calls, and I have even removed the E1 lines from card and I tried almost everything possible but I was not successful in determining the cause of this high cpu utilization. It happens here too. But only when there is at least one Unicall channel up. It does not happen on every call and I couldn't find a pattern yet. My setup includes: asterisk 1.0.9, libpri 1.0.9, and zaptel 1.0.9.1 Unicall 0.0.3pre3 and tried unicall-0.0.2c Digium TE410p Intel SE7520BD2 with Xeon 3.4GHz, 2 Gig Ram Almost the same setup here. The only difference is hardware. Soyo + P4 2.8 512MB. You didn't specify your R2 variant. Here it's the brazilian and the Asterisk box is connected on an Ericsson MD110. I'll upgrade to 0.0.3pre4 now. Maybe it's fixed in this version ? Bye, Leonardo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] realtime and include
Hi, is there any support for include statement in the database when using realtime configurations? I would like to have as much as possible configuration in my postgres db but we have different access controls for different user contexts (allow international, national etc). Today we have different contexts for access rules e.g. [allow_international] exten = _00.,1,Dial... and for users we just include the allow_xxx and deny_xxx contexts. This makes it easier since we don't need to change each users dialplan just include the right contexts. Is this possible with realtime? The only way I see is to add/remove switch statements in extensions.conf and then we back to make the changes in extensions.conf and not in the database... /urban ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk truncate my FAX !!!
Hi all, i've a problem receiving faxes. I'm using AMP and i hope that all work well without big changes. However i've done some tests on .tif file created by asterisk and i've noticed that it truncates my fax almost after 5-6 seconds. As results my pdf are corrupted and i receive a mail with empty pdf :-( someone can help me ? Thanks !!! Oz -- O-Zone ! No (C) 2005 WEB @ http://www.zerozone.it HOBBY @ http://peggy.altervista.org Call me with FWD: 692329 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Register Asterisk with Gatekeeper - oh323
I have tried everything. to register with this gatekeeper to make and receive calls These are the details I received from the voip provider: protocol H.323 Gatekeeper Address - [EMAIL PROTECTED] Port - 1719 RAS - 53 Q931 - 80 h245 - 1722 RTP - 1722 Username - H323 I have 2 phone number/accounts with this gatekeeper that I need to register to. ID - HMA0200.10szxn- e.164 - 22xx2912 ID - HMA0200.10szxn- e.164 - 22xx2913 Here is my oh323.conf: [general] listenAddress=0.0.0.0 listenPort=1720 [EMAIL PROTECTED] gatekeeperTTL=600 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout userInputMode=TONE amaFlags=default accountCode=H323 language=en context=voip-h323 [register] alias=ASTERISK [codecs] codec=G711A frames=20 [22xx2912] type=friend [EMAIL PROTECTED] port=1720 alias=HMA0200.10szxn- e164=22xx2912 context=default disallow=all allow=ulaw dtmfmode=rfc2833 [22xx2913] type=friend [EMAIL PROTECTED] port=1720 alias=HMA0200.10szxn- e164=22xx2913 context=default disallow=all allow=ulaw dtmfmode=rfc2833 All I get from Asterisk is the following: Aug 29 10:00:57 WARNING[9715]: chan_oh323.c:4228 oh323_gk_check: Failed to register with gatekeeper '[EMAIL PROTECTED]'. -- Retrying gatekeeper registration. Am I on the right track or have I missed the point. I do not want Asterisk to be the gatekeeper, I simply want Asterisk to register with the gatekeeper so I can receive calls from it and then use this gatekeeper to make calls to it. Any help would be appreciated. Thanks Steve.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using * in number to chose outgoing peer.
I want to dial for example 1* to set a different peer Ie: ;1* gives: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,tT) ;2* gives: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,tT) How can I do this? Regards, Arne Morten ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 presence
Hi All, Just wondering if anyone has managed to get line presence working on the 7 indicator lights on a grandstream gxp-2000 with asterisk? If so, what is the trick? :) I have simple presence working with my polycom phones but cant seem to get it working with the gxp-2000 - is it available in the latest firmware or is it something that will be released later on? Or is there something tricky i need to do on teh * side? Cheers, Ben ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Using * in number to chose outgoing peer.
Ok. I figured it out.
exten = _2*X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,tT) ;
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Arne Morten
Johansen
Sendt: 29. august 2005 11:21
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] Using * in number to chose outgoing peer.
I want to dial for example 1* to set a different peer
Ie:
;1* gives:
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,tT)
;2* gives:
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,tT)
How can I do this?
Regards,
Arne Morten
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[Asterisk-Users] Call file always redials (grrrrr)
Hi list! Our CRM app is creating call files for outgoing calls which is working great I just have one problem. I am using this as my call file: Channel: SIP/228(my phone) MaxRetries: 0 Context: from-internal (the context to dial from) Extension: 003120531234 (the phone number) Priority: 1 Callerid: Myfinecustomer 003120531234 so the external number is connected to my sip phone. However after speaking for approx 5 minuted, Asterisk always does a retry and I see the external number in my display on the second line. It does this on every call. When I'm finished I also see 2 records in the log files. Any idea why Asterisk is trying to place the call again even though the first attempt was succesful and the call is still in progress? I didn't specify a redial anywhere. I'm running the latest cvs stable (of this morning), Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Register Asterisk with Gatekeeper - oh323
Hi Steve, Your [general] section looks fine. In the [register] section remove everything else and leave these lines. context=incoming-h323-calls alias=HMA0200.10szxn- alias=22xx2912 alias=HMA0200.10szxn- alias=22xx2913 Now all H.323 calls will enter in 'incoming-h323-call' context. Try this and see if it works. Michael. Steve Ducat wrote: I have tried everything. to register with this gatekeeper to make and receive calls These are the details I received from the voip provider: protocol H.323 Gatekeeper Address - [EMAIL PROTECTED] Port - 1719 RAS - 53 Q931 - 80 h245 - 1722 RTP - 1722 Username - H323 I have 2 phone number/accounts with this gatekeeper that I need to register to. ID - HMA0200.10szxn- e.164 - 22xx2912 ID - HMA0200.10szxn- e.164 - 22xx2913 Here is my oh323.conf: [general] listenAddress=0.0.0.0 listenPort=1720 [EMAIL PROTECTED] gatekeeperTTL=600 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout userInputMode=TONE amaFlags=default accountCode=H323 language=en context=voip-h323 [register] alias=ASTERISK [codecs] codec=G711A frames=20 [22xx2912] type=friend [EMAIL PROTECTED] port=1720 alias=HMA0200.10szxn- e164=22xx2912 context=default disallow=all allow=ulaw dtmfmode=rfc2833 [22xx2913] type=friend [EMAIL PROTECTED] port=1720 alias=HMA0200.10szxn- e164=22xx2913 context=default disallow=all allow=ulaw dtmfmode=rfc2833 All I get from Asterisk is the following: Aug 29 10:00:57 WARNING[9715]: chan_oh323.c:4228 oh323_gk_check: Failed to register with gatekeeper '[EMAIL PROTECTED]'. -- Retrying gatekeeper registration. Am I on the right track or have I missed the point. I do not want Asterisk to be the gatekeeper, I simply want Asterisk to register with the gatekeeper so I can receive calls from it and then use this gatekeeper to make calls to it. Any help would be appreciated. Thanks Steve.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Japanese ISDN BRI card for asterisk (INS64)where to start?
Hi Clive, Thank you for your response to my posting. It looks to me that the intel board is the same as the dialogic board Can you please tell what that means? I haven't worked with any BRI cards before so I don't know if it's a good thing or a bad thing. Is / was the dialogic board compatible with asterisk? Using CAPI drivers? Can you please point me in the right direction for more information for this card? Im sure you get the picture here, I really don't know where to start J and really appreciate your help. Thanks Mick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT and SIP.conf update.
Title: Message I'm assuming no apps/scriptsexist which completes this? Can someone please confirm thatif I use a FQDN in sip.conf for my external IP, the FQDN is only resolved at the time of loading, therefore if my IP changes after sip is loaded, I will have to manually reload asterisk/sip? Regards, Ray - Ray Originally Wrote: I have a standard BT home DSL, which meansI cannot have a static IP address, therefore i'm forced to use NAT,I subscribe to a DDNS service and have written a VB app which polls the router every 10 seconds and updates the DDNS if appropriate. This is fine but I need to be able to modify my sip.conf (externip = w.x.y.z) and reload sip, does anyone know of a script/appwhich does an nslookup and modifies the conf file, then reloads sip? Regards, Ray ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk: Unable to read password.
Hi,
I changed my phones settings to inband and then
changed and then changed the settings in sip.conf to
dtmfmode=inband. It didnt work. I tried rfc and sip
info method too. I dont think its a problem with the
phone because audio works perfectly when I am leaving
a message, the problem is playing them back.
Any further ideas?
--- Anthony Rodgers [EMAIL PROTECTED] wrote:
Hi Pat,
I would check the DTMF settings on your phone - I
had a similar problem
until I switched to RFC from Inband.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Aug 26, 2005, at 4:56 AM, pat newham wrote:
Hello,
I am using asterisk as voicemail for my sip proxy.
When a user (1234)dials , the call is
forwarded to
asterisk. However I receive the following error:
--Executing VoiceMailMain(SIP/1234-9afc, 1234)
in
new stack
--Playing 'vm-password' (language 'en')
[WARNING]: app_voicemail.c:3359 vm-execmain:
Unable to
read password
==Spawn extension (default, , 1) exited
non-zero
on 'SIP/1234-9afc'
My configs are as follows:
;sip.conf
[1234]
type=friend
host=dynamic
context=default
mailbox=1234
;extensions.conf
[default]
exten=1234, 1, Voicemail(u${EXTEN})
exten=1234, 2, Hangup
exten=, 1, VoicemailMain(${CALLERIDNUM})
;voicemail.conf
1234=1234, P, [EMAIL PROTECTED]
Please advise if possible as i have looked through
the
asterisk mail archives but cannot see what would
be
wrong with the configuration.
many thanks.
___
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Re: [Asterisk-Users] OT: Are you using a Lucent?
Hi I am using a Lucent MAX TNT to terminate 11 PRIs and using a single Asterisk box to handle all calls --- Andrew Thrift [EMAIL PROTECTED] wrote: We have the ability to do this on a large scale, but want to do it on a smaller scale for 1 to maybe a maximum of 5 TNT's. Andrew Thrift wrote: Hi Mathew, We are interested in doing this too, is it possible you can share the information with us? We are looking at using a TNT MAX to terminate 8 E1's from the Telco, but we need a way of receiving the SS7 signalling and passing it to the TNT's via IPDC or whatever. Regards, Andy Matthew Boehm wrote: Is anyone out there using Lucent brand equipment to handle an incomming DS3, converting all 672 calls to SIP (as G729) and sending those to Asterisk/SER over ethernet? If you are and are willing to speak to my boss about your experiences (over the phone) with it, please contact me off list. We have a possible contract with a local CLEC to handle their long distance, and they want to send to us using DS3 and SS7. I'm trying to convince my boss to use a $9K Lucent, but he wants to spend much more by breaking out the DS3 into DS1's and stack up 6 asterisk boxes with 1 4-port card in each. Again, if you are using Lucent and are willing to speak to my boss about your experiences, please contact me off list so I can setup a call. Thanks, Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Custom Application For Asterisk
Hi no i write this application for my custom needs, but anybody of you can use it or customized it according to your needs cheers --- Matt Riddell [EMAIL PROTECTED] wrote: Gulzar Hussain wrote: Hi All I just completed a custom application for Asterisk (i m not a C guru so i just copy codes from other application and alter according to my needs) attached files is the source file this application is working fine but still i need you people to give suggestion to improve it Primary task of this application is to get a parameter from extensions.conf, query sql server and play a files according to the result Is this GPL? Is there a site where people can read about it and download it? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared
I'd suggest turning off echotraining on the FXS altogether, and perhaps even killing the echocanceller on FXS entirely. (you won't be getting significant echo from the FXS, and the FXO should be handling it anyway) -- echocancelwhenbridged might be an interesting thing to play with as well. e.g. (assuming port 1-3 are FXO and port 4-7 are FXS) echocancel=64 echocancelwhenbridged=yes echotraining=800 channel = 1-3 echocancelwhenbridged=no channel = 4-7 Andrew, I am sure you know that in zapata.conf parameter settings are in effect until specifically overridden later on in the file. In the first paragraph you suggest that I turn off both echotraining and echocanceler on FXS channels, so may I correct your example, that is, do you mean something like the following?: echocancel=64 echocancelwhenbridged=yes echotraining=800 channel = 1-3 echocancel=no echocancelwhenbridged=no echotraining=no channel = 4-7 Please correct me if I'm wrong, in your example echocanceler would still run on connections other than TDM (such as FXS-SIP). Did you knowingly mean it? With my additions above, FXS channels would never use echocanceler. Right? Thank you guys for all the help and comments. Rich's last comments were quite enlighthening, as always. I never knew echocanceler could be used on FXS channels. Sorry for my ignorance (but nowhere in docs or wiki could I see this information, I should have thought about it, my bad). I'll try and post the results. There are lots of things like this that aren't documented and probably never will be given the constant upgrades, code additions, etc. It will be interesting to hear your results. :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detect Dialtone
On Mon, 2005-08-29 at 01:23 +0200, Goran Dj. wrote: Dialtone detection should be an option in .conf for zap channel, i agree with that. Are you trying to play with the case where you have an analog phone bridged on your fxo line, and detect the lack of dialtone when someone is using that analog phone? Belive or not, but at some places on the world are still in use some old (non-digital) ATC-es which do now provide dial-tone instantly. For example, when ATC ARF-102 is very congested with outgoing calls, you must wait some (unknown) time to get dialtone (10sec, 1min, 5min...) Couldn't you just do 2 stage dialing? Dial the outgoing Zap channel and then wait for exchange to give dialtone. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low handset microphone volume with Sipura SPA-841
The firmware on the phones is version 3.1.3(a). I will try today using the 3.1.4 firmware. The size of the display could be better, but the lack of a backlight is what really bothers me. On Sunday 28 August 2005 11:46, John Novack wrote: I have not experienced that problem, but earlier firmware resulted in an unusable speakerphone. Check if you have the latest firmware, then ask Sipura support for help. The one time I E-mailed them they were quite responsive. the 841 still has a worthless display though, doesn't it? Lack of backlightimg and too small isn't going to be fixed by a firmware change! John Novack Juan Jose Comellas wrote: I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem? There are only two holes for the microphone in the handset and they are really small. I was thinking that myabe this is the cause. Any thoughts? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low handset microphone volume with Sipura SPA-841
I tried changing the gain settings and also the volume settings in the User tab, Audio Volume section. I didn't notice any change in the microphone output volume. On Sunday 28 August 2005 18:20, Rob Lith wrote: In Admin/Advanced have you tried the Handset Input Gain: settings? Rob On 8/28/05, Juan Jose Comellas [EMAIL PROTECTED] wrote: I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem? There are only two holes for the microphone in the handset and they are really small. I was thinking that myabe this is the cause. Any thoughts? -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
Should it be in half duplex or full duplex? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Sunday, August 28, 2005 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards Adam Robins wrote: We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. Check if the network card is in half duplex mode. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: chan_unical-MFC/R2 CPU usage problem
Hi, Hadi Jadallah wrote: My variant is standard ITU, I tried almost all versions I could put my hand on to no avail. I tried also to profile the channel and related libraries to no avail as my profiling skills on linux are abit lacking. If anybody with this problem and knows how to profile multithreaded apps on linux then we might at least pinpoint the location of the error. I cant realy put the machine into active duty if I cant solve the problem. Btw, what version of libtiff are you using? It difficult to believe that it might be related as I don’t need the fax functionality. Mine is the version that comes with CentOS 4.1 which is 3.6.1. libtiff 3.5.7. Leonardo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HDLC/Zaptel/Kernel 2.6.11(.9)
Oh meaning it won't work w/ a Cisco? :-) -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Thursday, August 25, 2005 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HDLC/Zaptel/Kernel 2.6.11(.9) Matt Schulte wrote: Forgive my ignorance, what encapsulation would you use on the ISP end of the T1? This is for data also, correct? This is only relevant for data. The ISP end is no different from the client end; the same encapsulation has to be used on both ends. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detect Dialtone
yes thats one issue the other issue is that sometimes the pstn line is dead due to some technical problems so people trying to make calls will just listen silence and they'll never know whats going on... -- Original Message -- From: Rich Adamson [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Sun, 28 Aug 2005 08:52:45 -0600 i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say all lines are busy/congested how can i configure that?? i already done (immediate=no) and still it opens the zap trunk even when theres no dialtone and shows that zap/3 answered - ;Specify whether the channel should be answered immediately or ; if the simple switch should provide dialtone, read digits, etc. ; immediate=no -- I might be way off base here, but the immediate=no parameter is oriented towards incoming zap calls (not outgoing calls), and the callprogress and busy detect stuff was intended to detect busy tones (not dial tone). I don't think there is any logic in the zap channels to listen for dial tone before dialing. (But, I could be wrong.) What are you using for the zap fxo channel (eg, channel bank, tdm, x100p)? Are you trying to play with the case where you have an analog phone bridged on your fxo line, and detect the lack of dialtone when someone is using that analog phone? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Powered by Hellacious Riders - http://www.hriders.com Want to be able to access your mail via POP 3? Please view: http://www.hriders.com/web_page.cfm?web_pageID=94 for more info. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards
Adam Robins wrote: Should it be in half duplex or full duplex? Full. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Client
-- Original Message -- From: bodra [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Sun, 28 Aug 2005 02:35:01 -0700 Hi all i am developing a client for the asterisk that controls ur phone from an Xp c# application what functions in Asterisk that will allow you to put someone on hold but not park calls and bring them back, without using flash hook cuz it will be a button in that application and i think i couldnt send a flash hook signal to the server.. Regards Powered by Hellacious Riders - http://www.hriders.com Want to be able to access your mail via POP 3? Please view: http://www.hriders.com/web_page.cfm?web_pageID=94 for more info. Powered by Hellacious Riders - http://www.hriders.com Want to be able to access your mail via POP 3? Please view: http://www.hriders.com/web_page.cfm?web_pageID=94 for more info. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Reboot Script
Anything like this for grandstream phones?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Matthew T. O'Connor
|Sent: Lunes, 29 de Agosto de 2005 12:22 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: [Asterisk-Users] Polycom Reboot Script
|
|Hello, I'm trying to setup the revised Polycom remote reboot
|script as found on:
|http://www.voip-info.org/tiki-index.php?page=Polycom+reboot+har
dphone+script
|
|I'm not sure how to use this script, it's just a perl script,
|so I tried creating an executable perl script and running it,
|but I get the following:
|
|[EMAIL PROTECTED] agi-bin]# ./polycom_reboot.pl 192.168.3.205
|Checking ARP table.
|192.168.3.205 is reachable.
|checking for polycom config name...
|touching config file /home/polycom/0004f201d398.cfg Use of
|uninitialized value in concatenation (.) or string at
|./polycom_reboot.pl line 97, ARP line 3.
|Use of uninitialized value in concatenation (.) or string at
|./polycom_reboot.pl line 99, ARP line 3.
|Use of uninitialized value in concatenation (.) or string at
|./polycom_reboot.pl line 99, ARP line 3.
|reboot of phone 192.168.3.205 was successful
|
|While it does say it is successful, I can tell you the phone
|does NOT reboot.
|
|line 97 looks like this:
|$call_id = $tm . msgto$sip_to;
|
|It's part of this sub routine:
|
|sub reboot_sip_phone {# Send the phone a check-sync to reboot it
|$phone_ip = shift;
|
|$local_ip = shift;
|$sip_to = shift;
|$sip_from = asterisk;
|$tm = time();
|$call_id = $tm . msgto$sip_to;
|$httptime = `date -R`;
|$MESG = NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
|Via: SIP/2.0/UDP $local_ip
|From: sip:[EMAIL PROTECTED]
|To: sip:[EMAIL PROTECTED]
|Event: check-sync
|Date: $httptime
|Call-ID: [EMAIL PROTECTED]
|CSeq: 1300 NOTIFY
|Contact: sip:[EMAIL PROTECTED]
|Content-Length: 0
|
|;
|
|Any Ideas?
|
|Thanks,
|
|Matt O'Connor
|
|___
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|
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|Asterisk-Users@lists.digium.com
|http://lists.digium.com/mailman/listinfo/asterisk-users
|To UNSUBSCRIBE or update options visit:
| http://lists.digium.com/mailman/listinfo/asterisk-users
|
|
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RE: [Asterisk-Users] 1.2.0 Beta1
I doubt my cvs was that current so... It's a clena install then... |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Kevin P. Fleming |Sent: Domingo, 28 de Agosto de 2005 09:56 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] 1.2.0 Beta1 | |Anton Krall wrote: | I upgraded from cvs head 1.0.x which in my case was cvs head about 2 | months ago. | |There is no such thing as cvs head 1.0.x. You could mean 'CVS v1-0 |(whatever was current in the 1.0.x branch at the time) or 'CVS HEAD' |(the current development branch at the time). | | Do you recommend doing a clean install vs. installing on top? | |Unless you were running a recent CVS HEAD already, yes, a |clean install is a good idea. |___ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards
Matt Riddell wrote: Adam Robins wrote: Should it be in half duplex or full duplex? Full. AFAIK, depends... If you have your switches doing autonegotiation, you can't disable autoneg in the NIC and hardcode it to do 100/Full-duplex, or you WILL have a duplex mismatch. This is as per the standard. A duplex mismatch is really bad, is in fact worse than having segments doing halfduplex (properly). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_sms: using * as an smsc
Tobias Wolf ha scritto: Let us assume that i have a couple of phones which should be able to receive SMS directly from my * box ( and not from an SMSC from BT or Deutsche Telekom ), So all these phones have the phone number of the * as Service Center configured. I recognized that the numbers of other SMSCs differs for outgoing and incoming SMS. I tried that successfully with my own SMS rig a couple of years ago. As far as I could tell from experimenting and from the ETSI docs, the phone knows it shouldn't ring, but it should answer and talk FSK to the SMSC, by looking at the caller ID; so, yes, you should set the correct caller ID in * to talk to your phone. Regards, -- Emanuele ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
Everything is set to autoneg, NICs, switches and router -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julio Arruda Sent: Monday, August 29, 2005 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards Matt Riddell wrote: Adam Robins wrote: Should it be in half duplex or full duplex? Full. AFAIK, depends... If you have your switches doing autonegotiation, you can't disable autoneg in the NIC and hardcode it to do 100/Full-duplex, or you WILL have a duplex mismatch. This is as per the standard. A duplex mismatch is really bad, is in fact worse than having segments doing halfduplex (properly). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] static noise - follow up
Hi two weeks ago I posted a message concerning static noise on our asterisk system we have made a bunch of tests and these are the results We use a TDM card revision I and on the card there is a sticker that says revision G If we put one fxo modules there is no noise if we put two fxo modules there is no noise if we put three fxo modules on the lines 1-2-3, we have noise on line 1 (Zap/1). line 2 and 3 have no noise if we put three fxo modules on the lines 2-3-4 we have no noise if we put 4 fxo modules we have noise on the line 1 if we use an older TDM card, revision E/F, there is no noise problem. Digium has no explanation for now and have asked for a RMA of one of the cards. I will keep you informed. If someone else has seen this behaviour, tell me if there is an explanation. Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Detect Dialtone
bodra wrote: yes thats one issue the other issue is that sometimes the pstn line is dead due to some technical problems so people trying to make calls will just listen silence and they'll never know whats going on... Which should be less of a problem, given that the FXO card gives a red alarm when there is no battery on the line. Priority + 101 could play an error message in that case. John Novack -- Original Message -- From: Rich Adamson [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Date: Sun, 28 Aug 2005 08:52:45 -0600 i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say all lines are busy/congested how can i configure that?? i already done (immediate=no) and still it opens the zap trunk even when theres no dialtone and shows that zap/3 answered - ;Specify whether the channel should be answered immediately or ; if the simple switch should provide dialtone, read digits, etc. ; immediate=no -- I might be way off base here, but the immediate=no parameter is oriented towards incoming zap calls (not outgoing calls), and the callprogress and busy detect stuff was intended to detect busy tones (not dial tone). I don't think there is any logic in the zap channels to listen for dial tone before dialing. (But, I could be wrong.) What are you using for the zap fxo channel (eg, channel bank, tdm, x100p)? Are you trying to play with the case where you have an analog phone bridged on your fxo line, and detect the lack of dialtone when someone is using that analog phone? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: cvs update error?
Hi, I am trying to update Asterisk from cvs as I think it might solve a secondary problem that I am experiencing (see below). In the /usr/src/asterisk directory I typed make upgrade. However I get an error: Makefile:16: *** missing separator. Stop. Make[2]L Leaving directory /usr/src/asterisk Make: *** [depend] Error 1 Has anyone come across this or does anyone know a way of solving this? Many thanks -Original Message- From: Aisling [mailto:[EMAIL PROTECTED] Sent: 26 August 2005 15:44 To: 'asterisk-users@lists.digium.com' Subject: cvs update error? Hi, Im experiencing a problem with playing back my voicemail. (Failed to write frame). It has been indicated in the archives that this is problem can be solved by updating asterisk from the cvs. I did make update in the /usr/src//asterisk directory to resolve this. However I got a message saying The following files have conflicts: channels/MakeFileCould someone advise me on what I need to do now to resolve these issues? Many thanks. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] When 486 ATA crashes, asterisk does not disconnect the call
Hi, On several occasions one or more of our grandstream Handy tone 486 ATA would crash. If for some reason that ATA is not rebooted immediately, asterisk would not disconnect the call, even though the party on the other end of the call have already hung up the call. The call would continue via my asterisk server and my sip termination provider indefinitely until I either reboot the ATA device or restart asterisk. It even ignores the timeout setting for the call. Can anyone explain why that would happen and how I can resolve that problem. Thanks Joel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards
You may want to check if the autonegotiation agreed in both sides. Older nic/drivers/switches would have problems with autonegotiation. Also, statistics can tell you something about this.. Example, if you have shorts/runts in one port, and late-collisions in the L1 'peer' port (the other side of the cable), you may have one side in full and the other in half. (the late-collisions would be counted in the half duplex side, and shorts/runts in the full-duplex side) Adam Robins wrote: Everything is set to autoneg, NICs, switches and router -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julio Arruda Sent: Monday, August 29, 2005 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 Softphone Quality Network Cards Matt Riddell wrote: Adam Robins wrote: Should it be in half duplex or full duplex? Full. AFAIK, depends... If you have your switches doing autonegotiation, you can't disable autoneg in the NIC and hardcode it to do 100/Full-duplex, or you WILL have a duplex mismatch. This is as per the standard. A duplex mismatch is really bad, is in fact worse than having segments doing halfduplex (properly). ___ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to use * and # as part of number indialcommand
Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? It looks as if the PSTN switch accepts the *31* and then hangs up so you can make the NEXT call with the *31* feature enabled. If so I assume the *31* feature will be enabled for the next call on the ENTIRE SPAN if it is an ISDN trunk group. If that is the case try putting two dials in sequence; Dial(zap/xx/*31*) Dial(zap/xx/restofnumber) Check to see if the *31* feature was activated on the line. I am not sure if the asterisk dialplan will stay sequential execution after you get the line hung up status. At any rate, the keypad protocol looks to me like a way to allow end user equipment to activate and deactivate features that are normally controlled in the setup, display IE, and facility IE, which are all elements of the ISDN signalling protocol. I would be very surprised if the same methods used in the US did not work in the Netherlands in place of the keypad protocol. That would be; SetCallerID(calleridvalue|a) And SetCallerIDNumber(caleridvalue) And SetCallerPres (presentation) - this one is in newer code and allows setting of the presentation flags in the ISDN setup message. Try setting these plags if you are using any type of isdn signalling and see if they are accepted (that is the feature is activated for he call). Here are the values used with SetCallerPres for 'show application setcallerpres' [Synopsis] Set CallerID Presentation [Description] SetCallerPres(presentation): Set Caller*ID presentation on a call. Always returns 0. Valid presentations are: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable Have a look at this doc for more info on keypad protocol http://www.ecma-international.org/publications/files/ECMA-ST/Ecma-156.pd f Damon -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michel Koenen Sent: Monday, August 29, 2005 1:55 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] How to use * and # as part of number indialcommand Damon Estep wrote: I did not see an actual error message in your first post, what is the error message? Damon, Well, it is not a 'real' error message, asterisk logs it as a 'warning' , but for me it looks like it is linked to the problem. See my comments in the logs between [ ]. -- Executing Dial(Zap/2-1, Zap/4/*31*040268000) in new stack -- Requested transfer capability: 0x10 - 3K1AUDIO -- Called 4/*31*040268000 -- Zap/4-1 is making progress passing it to Zap/2-1 [thus far it looks okay] -- Channel 0/1, span 2 got hangup [hmm, it seems that the channel was hangup, so it failed] Aug 27 23:32:28 WARNING[17591]: app_dial.c:412 wait_for_answer: Unable to forward voice [this warning indicates that asterisk was unable to forward voice, I think this is because of the *31* in the dial string, because when I leave the *31* out, the warning is not there and the connection is made without problems] -- Hungup 'Zap/4-1' == No one is available to answer at this time -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' Thank you for your time trying to help me out! Regards, Michel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
Everything is set to autoneg, NICs, switches and router To ensure reasonable performance, key devices (eg, routers, servers) should _always_ have duplex settings statically defined. Speed is less of an issue as the 10/100 negotiation is hard to get wrong. Part of the duplex negotiation problem is that consistent standards have not been implemented by all manufacturers (and nic card drivers). The two ends of a cat5 cable will often times try to auto negotiate the duplex settings at roughly the same time, and 50% of the time it will be wrong (eg, mismatched). As someone mentioned previously, mismiatched duplex settings will seriously impact performance and throughput. Keep in mind that opening the cat5 cable at either end (eg, unplug and replug the rj45) will cause a re-nogitation, as will a reboot, etc. There are a lot of systems and drivers that don't include the code to tell you what the actual duplex setting is after a re-negotiation. MS-based products are poor, and finding the actual setting in many of the linux distro's is not necessarily easy. For an asterisk server _always_ statically define the duplex setting on both the switch and the nic card. On sip phones and workstations, the duplex setting is less important, but should still match at both ends of the cable. (FWIW, my company does professional network performance assessments and you couldn't even guess how many large small corporate admins don't have a clue. That's based on 12 years of experience at sites in over 40 US states.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
For an asterisk server _always_ statically define the duplex setting on both the switch and the nic card. On sip phones and workstations, Can you give an example of how to check the duplex setting and statically define it for, say, RedHat9 On Mon, 29 Aug 2005, Rich Adamson wrote: Everything is set to autoneg, NICs, switches and router To ensure reasonable performance, key devices (eg, routers, servers) should _always_ have duplex settings statically defined. Speed is less of an issue as the 10/100 negotiation is hard to get wrong. Part of the duplex negotiation problem is that consistent standards have not been implemented by all manufacturers (and nic card drivers). The two ends of a cat5 cable will often times try to auto negotiate the duplex settings at roughly the same time, and 50% of the time it will be wrong (eg, mismatched). As someone mentioned previously, mismiatched duplex settings will seriously impact performance and throughput. Keep in mind that opening the cat5 cable at either end (eg, unplug and replug the rj45) will cause a re-nogitation, as will a reboot, etc. There are a lot of systems and drivers that don't include the code to tell you what the actual duplex setting is after a re-negotiation. MS-based products are poor, and finding the actual setting in many of the linux distro's is not necessarily easy. For an asterisk server _always_ statically define the duplex setting on both the switch and the nic card. On sip phones and workstations, the duplex setting is less important, but should still match at both ends of the cable. (FWIW, my company does professional network performance assessments and you couldn't even guess how many large small corporate admins don't have a clue. That's based on 12 years of experience at sites in over 40 US states.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compile problem with 1.2 beta 1
Has anyone else got 1.2 compiled from cvs ? I've posted the question below to the -dev list but got no answers: 1) No-one else is trying beta 1 2) No-one else is having any issues (I must be the idiot) 3) No-one else saw my message :) I have been trying to compile 1.2 beta 1 on a centos 4 box, to no avail. The make command seems to compile ok, but make install simply keeps looping. (see below). After this, no make command (clean/install/update etc) works. CVS head compiles and installs with no problems on the same machine. Julian. +- Asterisk Build Complete -+ + Asterisk has successfully been built, but + + cannot be run before being installed by + + running: + + + + make install+ +---+ [EMAIL PROTECTED] asterisk]# make install build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c config_old.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c config_old.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c config_old.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
For an asterisk server _always_ statically define the duplex setting on both the switch and the nic card. On sip phones and workstations, Can you give an example of how to check the duplex setting and statically define it for, say, RedHat9 Multiple ways... try 'dmesg | grep duplex' or use 'mii-tool'. Be careful with assumptions relative to what happens after a reboot on any system. Static use of the mii-tool within your system startup scripts may be necessary to ensure full duplex operation. On Mon, 29 Aug 2005, Rich Adamson wrote: Everything is set to autoneg, NICs, switches and router To ensure reasonable performance, key devices (eg, routers, servers) should _always_ have duplex settings statically defined. Speed is less of an issue as the 10/100 negotiation is hard to get wrong. Part of the duplex negotiation problem is that consistent standards have not been implemented by all manufacturers (and nic card drivers). The two ends of a cat5 cable will often times try to auto negotiate the duplex settings at roughly the same time, and 50% of the time it will be wrong (eg, mismatched). As someone mentioned previously, mismiatched duplex settings will seriously impact performance and throughput. Keep in mind that opening the cat5 cable at either end (eg, unplug and replug the rj45) will cause a re-nogitation, as will a reboot, etc. There are a lot of systems and drivers that don't include the code to tell you what the actual duplex setting is after a re-negotiation. MS-based products are poor, and finding the actual setting in many of the linux distro's is not necessarily easy. For an asterisk server _always_ statically define the duplex setting on both the switch and the nic card. On sip phones and workstations, the duplex setting is less important, but should still match at both ends of the cable. (FWIW, my company does professional network performance assessments and you couldn't even guess how many large small corporate admins don't have a clue. That's based on 12 years of experience at sites in over 40 US states.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER + ASTERISK voicemail
Hello,
Thanks for help it's ok with static file
voicemail.conf
However something is wrong with ARA .
app_voicemail search entries in voicemail.conf ?!
I set apps/Makefile for USE_ODBC_STORAGE.
Regards
Harry
//
Connected to Asterisk CVS-HEAD currently running on
serveur1 (pid = 2584)
Verbosity is at least 3
-- Executing VoiceMail(SIP/asterisk-8db8, b84)
in new stack
Aug 29 16:11:40 WARNING[7947]: app_voicemail.c:2602
leave_voicemail: No entry in voicemail config file
for '84'
Aug 29 16:11:50 WARNING[7947]: pbx.c:2336
__ast_pbx_run: Timeout, but no rule 't' in context
'loc al'
serveur1*CLI odbc show
Name: asterisk
DSN: asterisk
Connected: yes
serveur1*CLI
///
--- Steve Blair [EMAIL PROTECTED] a écrit :
You'll want some rules in your sip.conf to handle
the connection from
SER. A
starting point might be:
[ser ip addr:ser port ?= 5060]
type=peer
context=my sip context name
tos=lowdelay; tos delay
allow=ulaw ; dtmfmode=inband
only works with ulaw
or alaw!
dtmfmode=inband; Choices are
inband, rfc2833, or info
You'll then want some rules in extensions.conf to
accept the call and
redirect it
to mailboxes defined in your voicemail.conf or in
MySQL. Something like:
[general]
context=my sip context name
switch = Realtime/my sip context
name@extensions
static=yes
[my sip context name]
exten = _uX,1,VoiceMail(${EXTEN}@my sip
context name)
exten = _X,1,VoiceMail(${EXTEN}@my sip
context name)
exten = _bX,1,VoiceMail(${EXTEN}@my sip
context name))
exten = #,2,Hangup ; Hang
them up.
Steve
harry gaillac wrote:
Hello,
I try set Ua---SERAsterisk (voicemail/ARA)
|
Ua
ser stable
asterisk cvs head
I read
http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
to forward unavailable or busy sip agents to
asterisk
voicemail in failure route.
How may I configure extensions.conf and ser.cfg ?
I have been trying without success!
Regards
Harry
___
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nouveau Yahoo! Messenger
Téléchargez cette version sur
http://fr.messenger.yahoo.com
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Re: [Asterisk-Users] Compile problem with 1.2 beta 1
Julian Lyndon-Smith wrote: Has anyone else got 1.2 compiled from cvs ? I've posted the question below to the -dev list but got no answers: Mine complies fine under Mandrake and a kernel downloaded from kernel.org, ztdummy won't load, but other then that no issues. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
If nic is loaded using modprobe - you can set options for duplex - depending on the nic... See /etc/modules.conf -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, August 29, 2005 11:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards For an asterisk server _always_ statically define the duplex setting on both the switch and the nic card. On sip phones and workstations, Can you give an example of how to check the duplex setting and statically define it for, say, RedHat9 Multiple ways... try 'dmesg | grep duplex' or use 'mii-tool'. Be careful with assumptions relative to what happens after a reboot on any system. Static use of the mii-tool within your system startup scripts may be necessary to ensure full duplex operation. On Mon, 29 Aug 2005, Rich Adamson wrote: Everything is set to autoneg, NICs, switches and router To ensure reasonable performance, key devices (eg, routers, servers) should _always_ have duplex settings statically defined. Speed is less of an issue as the 10/100 negotiation is hard to get wrong. Part of the duplex negotiation problem is that consistent standards have not been implemented by all manufacturers (and nic card drivers). The two ends of a cat5 cable will often times try to auto negotiate the duplex settings at roughly the same time, and 50% of the time it will be wrong (eg, mismatched). As someone mentioned previously, mismiatched duplex settings will seriously impact performance and throughput. Keep in mind that opening the cat5 cable at either end (eg, unplug and replug the rj45) will cause a re-nogitation, as will a reboot, etc. There are a lot of systems and drivers that don't include the code to tell you what the actual duplex setting is after a re-negotiation. MS-based products are poor, and finding the actual setting in many of the linux distro's is not necessarily easy. For an asterisk server _always_ statically define the duplex setting on both the switch and the nic card. On sip phones and workstations, the duplex setting is less important, but should still match at both ends of the cable. (FWIW, my company does professional network performance assessments and you couldn't even guess how many large small corporate admins don't have a clue. That's based on 12 years of experience at sites in over 40 US states.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, -- -- Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] realtime and include
Urban wrote: Hi, is there any support for include statement in the database when using realtime configurations? I would like to have as much as possible configuration in my postgres db but we have different access controls for different user contexts (allow international, national etc). Today we have different contexts for access rules e.g. [allow_international] exten = _00.,1,Dial... and for users we just include the allow_xxx and deny_xxx contexts. This makes it easier since we don't need to change each users dialplan just include the right contexts. Is this possible with realtime? The only way I see is to add/remove switch statements in extensions.conf and then we back to make the changes in extensions.conf and not in the database... If you store the extensions.conf in database, then it will work. If you want to use the switch, then no. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] plainvoip provider problem
Hi: Is there anybody familliar with www.plainvoip.com voip provider. I sent them money through paypal and they didn't add the money to my account and they didn't respond to my request to send the money back to paypal. Is there anything I can do besides disputing the charge with paypal? Regards; Chawki Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Softphone Quality Network Cards
If nic is loaded using modprobe - you can set options for duplex - depending on the nic... See /etc/modules.conf I assume you really meant /etc/modprobe.conf ;) -Original Message- For an asterisk server _always_ statically define the duplex setting on both the switch and the nic card. On sip phones and workstations, Can you give an example of how to check the duplex setting and statically define it for, say, RedHat9 Multiple ways... try 'dmesg | grep duplex' or use 'mii-tool'. Be careful with assumptions relative to what happens after a reboot on any system. Static use of the mii-tool within your system startup scripts may be necessary to ensure full duplex operation. On Mon, 29 Aug 2005, Rich Adamson wrote: Everything is set to autoneg, NICs, switches and router To ensure reasonable performance, key devices (eg, routers, servers) should _always_ have duplex settings statically defined. Speed is less of an issue as the 10/100 negotiation is hard to get wrong. Part of the duplex negotiation problem is that consistent standards have not been implemented by all manufacturers (and nic card drivers). The two ends of a cat5 cable will often times try to auto negotiate the duplex settings at roughly the same time, and 50% of the time it will be wrong (eg, mismatched). As someone mentioned previously, mismiatched duplex settings will seriously impact performance and throughput. Keep in mind that opening the cat5 cable at either end (eg, unplug and replug the rj45) will cause a re-nogitation, as will a reboot, etc. There are a lot of systems and drivers that don't include the code to tell you what the actual duplex setting is after a re-negotiation. MS-based products are poor, and finding the actual setting in many of the linux distro's is not necessarily easy. For an asterisk server _always_ statically define the duplex setting on both the switch and the nic card. On sip phones and workstations, the duplex setting is less important, but should still match at both ends of the cable. (FWIW, my company does professional network performance assessments and you couldn't even guess how many large small corporate admins don't have a clue. That's based on 12 years of experience at sites in over 40 US states.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Variuos hangup codes in Manager API for failover
On 8/28/05, Matt Riddell [EMAIL PROTECTED] wrote:
Steve Edwards wrote:
Normally the way I do it is to program the failover into the dialplan
and then
send the call to Local/[EMAIL PROTECTED] to initiate it.
How about a snippet? (Local channels somewhat escape me.)
Ok,
If you had something like this (we're assuming +101 jumping for arguments sake
here):
[outbound]
exten = _9X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9X.,102,Dial(IAX/myiaxprovider/${EXTEN:1})
exten = _9X.,203,Dial(IAX/myiaxprovider/${EXTEN:1})
Then you could originate a call with the following channel:
Local/[EMAIL PROTECTED]
which would do the whole failover thing for you.
Note that this is slightly simplified. The jumping behaviour has now been
changed and will require the 'j' option in the latest versions unless you use
gotoif and check the dialstatus.
Normally you'd want to connect the originated call with an extension/context
so that once that number answers it is connected to say an agent or an
application. This part should be pretty self explanatory.
Make sense now? Feel free to ask if it doesn't!
:)
--
Cheers,
Matt Riddell
___
Thanks Matt, that is a good strategy.
Any idea on how to pass the reason a call failed back through the
Asterisk Manager Interface? It would be great to send something back
like Busy, NoAnswer, etc...
Geoff
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[Asterisk-Users] Return code of txfax
Hi, I have asterisk 1.0.7 and spandsp-0.0.2_pre18. txfax return a non-zero return code only if the fax file is not found. Unfortunately I can't get any information, whether the fax was transmitted completely or not. Will an update to a newer version change this? Thanks for telling me your experience! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: cvs update error?
I am trying to update Asterisk from cvs as I think it might solve a secondary problem that I am experiencing (see below). In the /usr/src/asterisk directory I typed “make upgrade”. However I get an error: Makefile:16: *** missing separator. Stop. Are you on FreeBSD (or not Linux)? You need to be using gmake. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile problem with 1.2 beta 1
On 8/29/05, Doug Lytle [EMAIL PROTECTED] wrote: Julian Lyndon-Smith wrote: Has anyone else got 1.2 compiled from cvs ? I've posted the question below to the -dev list but got no answers: Mine complies fine under Mandrake and a kernel downloaded from kernel.org, ztdummy won't load, but other then that no issues. Doug I get the same compile errors on Debian Sarge. I have been compiling previous CVS HEAD versions. I was able to compile the tarball. Geoff ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: cvs update error?
I'm using suse linux. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Bockman Sent: 29 August 2005 16:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: cvs update error? I am trying to update Asterisk from cvs as I think it might solve a secondary problem that I am experiencing (see below). In the /usr/src/asterisk directory I typed make upgrade. However I get an error: Makefile:16: *** missing separator. Stop. Are you on FreeBSD (or not Linux)? You need to be using gmake. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 and Phone does not 'ring'
I have a running * with a TDM40B board in it. I have 3 analog phones that works (rings) perfectly when connected to a Telco POTS line. When connected to the Digium TDM40B (with FXS port), I have problems with 'ringing': 1 phone 'ringes' normally 1 phone 'ringes' a bit cripled (instead of ring-ring... ring-ring..., it does 'ring-ri... ri ring... ri...) and the 3rd one does not ring at all when Asterisk says 'Ringing Zap/6'. However, when I do an 'off-hook' on this phone, I get tone signal and can dial and talk perfectly. I have phones compliant to the Belgium (Belgacom) Telco specs. Are there differences in 'Ring Voltage' ? Anyone with a suggestion ? Thanks Alex -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] text till answer
hello! i'm looking for a feature to play a sound-file containing a text until the called party picks up the phone. i've already tried with the 'special' musiconhold-feature by adding the m-option at the end of DIAL but it is not exactly what i want. the problem with the m-option is that the file is played to a second caller at the same position as it was played to the first caller, so when the second person calls, the text is played somewhere in the middle of the track instead of the beginning. i only works when the file has already been fully played(e.g. when there was enough time between the first and the second call). i want to play a text for every caller(that is played from the beginning) until the called party picks up the phone, is that possible? thank you for all suggestions! regards chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sqlite + stable asterisk
hi, i have problem with compiling cdr_sqlite rhel4(gcc3.4.3) + sqlite3 (from fc4 - rebuilded) any ideas? gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-08/11/05-19:35:03\ -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-c -o cdr_sqlite.o cdr_sqlite.c cdr_sqlite.c:38: error: syntax error before '*' token cdr_sqlite.c:38: warning: type defaults to `int' in declaration of `db' cdr_sqlite.c:38: warning: data definition has no type or storage class cdr_sqlite.c: In function `sqlite_log': cdr_sqlite.c:92: warning: implicit declaration of function `sqlite_exec_printf' cdr_sqlite.c: In function `unload_module': cdr_sqlite.c:153: warning: implicit declaration of function `sqlite_close' cdr_sqlite.c: In function `load_module': cdr_sqlite.c:166: warning: implicit declaration of function `sqlite_open' cdr_sqlite.c:166: warning: assignment makes pointer from integer without a cast cdr_sqlite.c:174: warning: implicit declaration of function `sqlite_exec' make[1]: *** [cdr_sqlite.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/cdr' make: *** [subdirs] Error 1 --- Marek Cervenka === ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER NAT any additional requirement
Hello
i am trying to use this exmple with SER-0.9.3
but still NATED Clients are not working any other
requirement
http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper
---
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
#
# simple quick-start config script
#
# --- global configuration parameters
debug=3 # debug level (cmd line:
-dd)
fork=yes
log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo=/tmp/ser_fifo
alias=mydomain.dyndns.org
# -- module loading
--
loadmodule /usr/local/lib/ser/modules/nathelper.so
loadmodule /usr/local/lib/ser/modules/textops.so
loadmodule /usr/local/lib/ser/modules/sl.so
loadmodule /usr/local/lib/ser/modules/tm.so
loadmodule /usr/local/lib/ser/modules/rr.so
loadmodule /usr/local/lib/ser/modules/maxfwd.so
loadmodule /usr/local/lib/ser/modules/usrloc.so
loadmodule /usr/local/lib/ser/modules/registrar.so
# - setting module-specific
parameters ---
# -- usrloc params --
modparam(usrloc, db_mode, 0)
# -- rr params --
# add value to ;lr param to make some broken UAs
happy
modparam(rr, enable_full_lr, 1)
# - request routing logic
---
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
break;
};
if (len_gt( max_len )) {
sl_send_reply(513, Message too big);
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy;
that's
# particularly good if upstream and downstream
entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following
command
# with proper names and addresses in it)
if (uri==myself) {
if (method==REGISTER) {
save(location);
break;
};
# native SIP destinations are handled using our
USRLOC DB
if (!lookup(location)) {
sl_send_reply(404, Not Found);
break;
};
};
#inserted by klaus
if (method==INVITE) {
record_route();
force_rtp_proxy();
/* set up reply processing */
t_on_reply(1);
};
# forward to current uri now; use stateful
forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
#inserted by klaus
# all incoming replies for t_onrepli-ed transactions
enter here
onreply_route[1] {
if (status=~[12][0-9][0-9])
force_rtp_proxy();
}
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Re: [Asterisk-Users] FW: cvs update error?
On Mon, 2005-08-29 at 14:04 +0100, Aisling wrote: Hi, I am trying to update Asterisk from cvs as I think it might solve a secondary problem that I am experiencing (see below). In the /usr/src/asterisk directory I typed “make upgrade”. However I get an error: Makefile:16: *** missing separator. Stop. Make[2]L Leaving directory ‘/usr/src/asterisk’ Make: *** [depend] Error 1 Has anyone come across this or does anyone know a way of solving this? Look at your Makefile it looks like there was a conflict during your make upgrade. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Compile problem with 1.2 beta 1
Had the same issue, tried to submit the bug and the bug tracker would not take bugs for versions other than CVS head. I did a little more research and found a directory /usr/src/asterisk/asterisk! I did not create the folder above! CVS Head compiled on the same machine without issues There has been a new tarball posted since I downloaded mine. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Monday, August 29, 2005 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Compile problem with 1.2 beta 1 Has anyone else got 1.2 compiled from cvs ? I've posted the question below to the -dev list but got no answers: 1) No-one else is trying beta 1 2) No-one else is having any issues (I must be the idiot) 3) No-one else saw my message :) I have been trying to compile 1.2 beta 1 on a centos 4 box, to no avail. The make command seems to compile ok, but make install simply keeps looping. (see below). After this, no make command (clean/install/update etc) works. CVS head compiles and installs with no problems on the same machine. Julian. +- Asterisk Build Complete -+ + Asterisk has successfully been built, but + + cannot be run before being installed by + + running: + + + + make install+ +---+ [EMAIL PROTECTED] asterisk]# make install build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c config_old.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c config_old.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer acl.c aescrypt.c aeskey.c aestab.c alaw.c app.c asterisk.c ast_expr2.c ast_expr2f.c astmm.c autoservice.c callerid.c cdr.c channel.c chanvars.c cli.c config.c config_old.c db.c devicestate.c dlfcn.c dns.c dnsmgr.c dsp.c enum.c file.c frame.c fskmodem.c image.c indications.c io.c jitterbuf.c loader.c logger.c manager.c md5.c muted.c netsock.c pbx.c plc.c poll.c privacy.c rtp.c say.c sched.c slinfactory.c srv.c strcompat.c tdd.c term.c translate.c ulaw.c utils.c ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com
[Asterisk-Users] [Announce] Web-MeetMe v1.3.3
Work intrudes again and I will not be able to get to modifying the db and gui to support per-conference flags as soon as I expected. So I have released an update with what I do have available. [Location] http://www.fitawi.com/Asterisk [Features] 1. Schedule new conferences a. Control start and end times b. Set conference pin # i. Generate one if the requester leaves it blank ii. Identify pin # conflicts (another conference with the same pin is scheduled at the same time) c. Set Admin and User passwords i. Generate a user password if an Admin pw is set but the User pw is blank 2. Email the details for a successfully scheduled conference 3. Separate views for Current, Past and Future conferences 4. Ability to modify the end time of a running conference a. Can also reschedule a past or future conference. 5. Monitor realtime conference activity a. Mute/Kick participants 6. Optional authentication a. Currently Active Directory or LDAP based b. Authentication is abstracted so unix/PAM/DB/RADIUS support could be easily added (but outside of my interest to do so (patches welcome)) 7. Users can only monitor, update or delete their conferences 8. Verified administrators can monitor, update or delete any conferences. 9. Updated to CVS-Head (a couple weeks ago, will target 1.2 soon) a. Changes to the Manager interface may have caused support for 1.0.X to slip, I cannot test that) There is one functional issue to be addressed, and that is that MeetMe tracks conference participants by channel. From a conference management perspective it makes more sense to track the participant by caller-id. I have a patch for 1.0.X on my site, but have not polished one for CVS-Head or the 1.2.0beta release. Thanks and enjoy, Dan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Compile error - x86_64
Asterisk has this error on compile: flex ast_expr2.fl ast_expr2.fl, line 50: unrecognized %option: reentrant ast_expr2.fl, line 51: unrecognized %option: bison-bridge ast_expr2.fl, line 52: unrecognized %option: bison-locations make: *** [ast_expr2f.c] Error 1 2.6.12-1.1447_FC4smp #1 SMP bison (GNU Bison) 2.0 Written by Robert Corbett and Richard Stallman. Copyright (C) 2004 Free Software Foundation, Inc. This is free software; see the source for copying conditions. There is NO warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: cvs update error?
Hello, I have attached my makefile. I don't know what I should be looking for in it but if it is somehow different to everyone elses make file, will someone please point that out? I never modified it in any way. How would I get a new copy of the Makefile from CVS? Many Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: 29 August 2005 17:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: cvs update error? On Mon, 2005-08-29 at 14:04 +0100, Aisling wrote: Hi, I am trying to update Asterisk from cvs as I think it might solve a secondary problem that I am experiencing (see below). In the /usr/src/asterisk directory I typed make upgrade. However I get an error: Makefile:16: *** missing separator. Stop. Make[2]L Leaving directory '/usr/src/asterisk' Make: *** [depend] Error 1 Has anyone come across this or does anyone know a way of solving this? Look at your Makefile it looks like there was a conflict during your make upgrade. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. Makefile.dat Description: Binary data ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and a Meridian Nortell Release 11
Hi there, We are using * with an Option 11C - we tried all of the various protocols and the only one we could get to work satisfactorily was 5ESS, with the * as CO and the Nortel as remote. The one drawback of this approach is getting name information for caller ID - because the Nortel sees the * as CO, it won't send the name information. /etc/zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #clear=1-24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf: [trunkgroups] [channels] context=incoming switchtype=5ess usecallingpres=yes echocancel=128 usecallerid=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 rxgain=-4.0 txgain=-6.0 group=1 callgroup=1 pickupgroup=1 signalling = pri_net channel = 1-23 musiconhold=default Any use? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 27-Aug-05, at 7:20 AM, Alvaro Parres wrote: Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel Release 11. I need to connect both of them. We are using MFC/R2 for this.. The Diagram: [ NORTEL ] ( AMI ) (DIGIUM) [ ASTERISK] we have green light at the digium card, and at asterisk we see all 31 channels as idle. But when i want to recive a call from the Nortel to the Asterisk i get at the Nortel only a empty sound, and after about 15 o 20 sec it's hangup. Any suggestion ? The log at Asterisk is: Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 - 0001 [1/ 1/Idle /Idle ] Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Detected Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Making a new call with CRN 32769 Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 1101 - [2/ 2/Idle /Idle ] Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:2865 handle_uc_event: Unicall/30 event Detected Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 - 0001 [1/ 1/Idle /Idle ] Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Detected Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Making a new call with CRN 32769 Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 1101 - [2/ 2/Idle /Idle ] Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:2865 handle_uc_event: Unicall/30 event Detected tel2*CLI Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 - 1001 [2/ 2/Seize ack /Seize ack] Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event: Unicall/30 event Far end disconnected Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:3198 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Call control(6) Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Drop call(cause=Normal Clearing [16]) Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event: Unicall/30 event Drop call Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Call control(7) Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Release call Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 1001 - [1/1000/Clear fwd /Seize ack ] Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 - 1001 [2/ 2/Seize ack /Seize ack ] Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event: Unicall/30 event Far end disconnected Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:3198 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Call control(6) Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Drop call(cause=Normal Clearing [16]) Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event: Unicall/30 event Drop call Aug 27
Re: [Asterisk-Users] Asterisk and a Meridian Nortell Release 11
On Mon, Aug 29, 2005 at 09:54:11AM -0700, Anthony Rodgers wrote: We are using * with an Option 11C - we tried all of the various protocols and the only one we could get to work satisfactorily was 5ESS, with the * as CO and the Nortel as remote. The one drawback of this approach is getting name information for caller ID - because the Nortel sees the * as CO, it won't send the name information. This is our experience as well: National ISDN-2 doesn't work (calls only go one way) because of limitations on the Meridian unit. 5ESS only works if the Meridian is CPE and Asterisk is NET. But you can only send CID names from Asterisk to Meridian, not vice versa. According to our phone system consultants (TAC Centre) there isn't any way to get the Meridian of this age to transmit CID names in this case. -- Karl A. Krueger [EMAIL PROTECTED] Network Security -- Linux/Unix Systems Support -- VoIP -- etc. Woods Hole Oceanographic Institution ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Noise on ZAP channel
I havea couple SIP phones on a PIII 1Ghz 256MB* server with a TDM01B connected to the PSTN. Calls between SIP phones are clear. Calls to the PSTN are quite noisy. The other person does not hear noise but I hear quite a bit. It is not an annoying sound but definitely much noisier than typical PSTN or even cell phone calls. I believe I have a TDM400P REV H card. I definitely don't have any IRQ issues. Everything not required is disabled in BIOS. Zaptel drivers have been compiled with defaults and with MMX and other enhancements. Have tried V1.0.9.1 and current 1.2 beta1 software. Nothing changes. Tried telco PSTN connection and VoIP provider connection via ATA which both sound clear when connected directly to an analog phone.Nothing changes. Tried adjusting RX/TX gain and echo cancellation in zaptel.conf. Nothing changes. Doesany one have any ideas? Could my FXO module be bad? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] teliax
Is there a problem at Teliax? I'm looking for a VoIP provider and when I call them they never answer the phone and the voice mail says it's full. Chris___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] teliax
\I concur. They seem to be always busy. Chris wrote: Is there a problem at Teliax? I'm looking for a VoIP provider and when I call them they never answer the phone and the voice mail says it's full. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- = Joshua Abbott, Support Technician http://www.successfulhosting.com/ Direct Line: PENDING Phone: (866) 494-5096 x1207 E-Fax: (419) 858-3241 Alt E-Fax: (801) 217-1123 [EMAIL PROTECTED] = The Success behind your web site! = ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] teliax
I like the plans they offer, but this doesn't give me much confidence in their ability.Can anyone recommend someone else? - Original Message - From: Joshua Abbott [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 29, 2005 12:18 PM Subject: Re: [Asterisk-Users] teliax \I concur. They seem to be always busy. Chris wrote: Is there a problem at Teliax? I'm looking for a VoIP provider and when I call them they never answer the phone and the voice mail says it's full. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- = Joshua Abbott, Support Technician http://www.successfulhosting.com/ Direct Line: PENDING Phone: (866) 494-5096 x1207 E-Fax: (419) 858-3241 Alt E-Fax: (801) 217-1123 [EMAIL PROTECTED] = The Success behind your web site! = ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] teliax
Joshua Abbott wrote: \I concur. They seem to be always busy. Chris wrote: Is there a problem at Teliax? I'm looking for a VoIP provider and when I call them they never answer the phone and the voice mail says it's full. Have you tried emailing them or using their online support? Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] teliax
Their online support says off line and goes to email.I have emailed them several times and still haven't got answers to my questions.Everytime I get a response from them I have to repeat my question and then I never hear the answer. Regards, Chris - Original Message - From: Darrick Hartman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 29, 2005 11:47 AM Subject: Re: [Asterisk-Users] teliax Joshua Abbott wrote: \I concur. They seem to be always busy. Chris wrote: Is there a problem at Teliax? I'm looking for a VoIP provider and when I call them they never answer the phone and the voice mail says it's full. Have you tried emailing them or using their online support? Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] teliax
Chris wrote: Their online support says off line and goes to email.I have emailed them several times and still haven't got answers to my questions.Everytime I get a response from them I have to repeat my question and then I never hear the answer. Regards, I'm glad I haven't shared the same experience. I had fairly quick replies to the questions I had. They've also been upgrading their equipment and network over the past several months. I had a few problems when I first signed up, but after working through the two simple issues, I haven't had a problem. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] teliax
Their plans look good, but it just feels like I am being ignored. Some guy named David emailed me off the Asterisk-biz list from Teliax with his direct number.I'll give that a try. Regards, Chris - Original Message - From: Darrick Hartman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 29, 2005 11:59 AM Subject: Re: [Asterisk-Users] teliax Chris wrote: Their online support says off line and goes to email.I have emailed them several times and still haven't got answers to my questions. Everytime I get a response from them I have to repeat my question and then I never hear the answer. Regards, I'm glad I haven't shared the same experience. I had fairly quick replies to the questions I had. They've also been upgrading their equipment and network over the past several months. I had a few problems when I first signed up, but after working through the two simple issues, I haven't had a problem. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Moving to New Zealand
Is there anyon here currently in New Zealand that use asterisk, I need to help getting voice and internet services. I will be moving in a week. Any help would be great. Please use the details below to get ahold of me. Thanks in advance. James Jones Signate, LLC [EMAIL PROTECTED] 415.442.4012 (office) 413.771.1402 (office) 413.977.6482 (mobile) 413.667.3105 (fax) 665 Third Street Suite 100 San Francisco, CA 94107-190 Asterisk Services and Training ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 ringing No voice
using ARTDIO clone IAX2 phone set connected on the same LAN as Asterisk server Ring... when off hook : - we can hear correctly the caller - but the caller continue to hear the ring tone Any idea ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream handytone 488 fxo
can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. i have been told that [EMAIL PROTECTED] has this built in to just a button hit, but i dont want to reinstall the box and would prefer to use asterisk directly Casey Boone ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] delay before dial on TDM04B
I am searching for a way to add a 2 second delay before calling out with Dial(). Sometimes I get the message you must first dial a 1 to place this call. I presume the phone company is missing the first digit pulsed out sometimes. How do I put a 2 second delay after coming offhook and before dialing the digits? Thanks, jerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New astGUIclient version released 1.1.6
Hello, We've released another update to our Asterisk GUI Client suite: 1.1.6 http://astguiclient.sf.net/ The client suite runs on Windows, UNIX and Mac, includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL client-side web app auto-dialer. This package is free as in GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this revision, we have finished the VICIDIAL web-client, added compatibility with the Asterisk 1.2 release tree, streamlined several server-side apps and added cpu percentages to our stats logging scripts. As of this release, all client apps and daily administration functions can be access through a web browser and we have tested our new AJAX-enabled(PHP, Javascript and XMLHTTPRequest) VICIDIAL client in production with great results. Let me know what you think. Thanks, MATT--- http://astguiclient.blogspot.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] delay before dial on TDM04B
Jerry Geis wrote: I am searching for a way to add a 2 second delay before calling out with Dial(). Sometimes I get the message you must first dial a 1 to place this call. I presume the phone company is missing the first digit pulsed out sometimes. How do I put a 2 second delay after coming offhook and before dialing the digits? Thanks, jerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can prepend a 'w' for a half-second wait which will resolve this problem. e.g., Zap/1/w19007529269 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 presence
Hi All, Just wondering if anyone has managed to get line presence working on the 7 indicator lights on a grandstream gxp-2000 with asterisk? If so, what is the trick? :) last week i asked the grandstream support for this, and got this short answer: This feature is not supported yet, it will be supported in the future. I have simple presence working with my polycom phones but cant seem to get it working with the gxp-2000 - is it available in the latest firmware or is it something that will be released later on? Or is there something tricky i need to do on teh * side? Cheers, Ben ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] delay before dial on TDM04B
Samy, Thanks for the suggestion - however I am confused on the wiki the 'w' stands for: *w*: Allow the /called/ user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.0.x) This is not a delay of any kind. Jerry [Asterisk-Users] delay before dial on TDM04B *Samy Kamkar* samy at fonality.com mailto:asterisk-users%40lists.digium.com?Subject=%5BAsterisk-Users%5D%20delay%20before%20dial%20on%20TDM04BIn-Reply-To=43134CDA.4020408%40pagestation.com /Mon Aug 29 13:07:05 CDT 2005/ * Previous message: [Asterisk-Users] delay before dial on TDM04B http://lists.digium.com/pipermail/asterisk-users/2005-August/123436.html * Next message: [Asterisk-Users] New astGUIclient version released 1.1.6 http://lists.digium.com/pipermail/asterisk-users/2005-August/123437.html * *Messages sorted by:* [ date ] http://lists.digium.com/pipermail/asterisk-users/2005-August/date.html#123438 [ thread ] http://lists.digium.com/pipermail/asterisk-users/2005-August/thread.html#123438 [ subject ] http://lists.digium.com/pipermail/asterisk-users/2005-August/subject.html#123438 [ author ] http://lists.digium.com/pipermail/asterisk-users/2005-August/author.html#123438 Jerry Geis wrote: / I am searching for a way to add a 2 second delay before calling out // with Dial(). // Sometimes I get the message you must first dial a 1 to place this call. // I presume the phone company is missing the first digit pulsed out // sometimes. // // How do I put a 2 second delay after coming offhook and before dialing // the digits? // // Thanks, // // jerry // // ___ // --Bandwidth and Colocation sponsored by Easynews.com -- // // Asterisk-Users mailing list // Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users // http://lists.digium.com/mailman/listinfo/asterisk-users // To UNSUBSCRIBE or update options visit: // http://lists.digium.com/mailman/listinfo/asterisk-users / You can prepend a 'w' for a half-second wait which will resolve this problem. e.g., Zap/1/w19007529269 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] teliax
Chris wrote: Is there a problem at Teliax? I'm looking for a VoIP provider and when I call them they never answer the phone and the voice mail says it's full. I don;t see any network problems, and I monitor Teliax and a few other providers. Teliax is my main provider and I have never had any problems worth worrying about. Send them an email, I find they always respond, at least if the question is reasonable. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream handytone 488 fxo
Casey Boone escreveu: can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. I have one but I too haven't been able to make it work. I've been looking at the config pages for the 488 and trying to make sense of the Route to PSTN configuration. Have you found any documentation for this? Keith Yoder ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MSG Waiting Off
I think Asterisk is sending some signal to my cordless phone that is causing it to constantly display message: MSG Waiting Off. The problem is that it is impossible to program anything into the phone or sometime dial a phone numbers as the when I try to program a number or dial a number and asterisk sends a signal that is causing it to display that massage and my dialing or programing is interrupt. What is causing it? -- #Joseph ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] delay before dial on TDM04B
Jerry Geis wrote: Samy, Thanks for the suggestion - however I am confused on the wiki the 'w' stands for: *w*: Allow the /called/ user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.0.x) This is not a delay of any kind. Jerry [Asterisk-Users] delay before dial on TDM04B *Samy Kamkar* samy at fonality.com mailto:asterisk-users%40lists.digium.com?Subject=%5BAsterisk-Users%5D%20delay%20before%20dial%20on%20TDM04BIn-Reply-To=43134CDA.4020408%40pagestation.com /Mon Aug 29 13:07:05 CDT 2005/ * Previous message: [Asterisk-Users] delay before dial on TDM04B http://lists.digium.com/pipermail/asterisk-users/2005-August/123436.html * Next message: [Asterisk-Users] New astGUIclient version released 1.1.6 http://lists.digium.com/pipermail/asterisk-users/2005-August/123437.html * *Messages sorted by:* [ date ] http://lists.digium.com/pipermail/asterisk-users/2005-August/date.html#123438 [ thread ] http://lists.digium.com/pipermail/asterisk-users/2005-August/thread.html#123438 [ subject ] http://lists.digium.com/pipermail/asterisk-users/2005-August/subject.html#123438 [ author ] http://lists.digium.com/pipermail/asterisk-users/2005-August/author.html#123438 Jerry Geis wrote: / I am searching for a way to add a 2 second delay before calling out // with Dial(). // Sometimes I get the message you must first dial a 1 to place this call. // I presume the phone company is missing the first digit pulsed out // sometimes. // // How do I put a 2 second delay after coming offhook and before dialing // the digits? // // Thanks, // // jerry // // ___ // --Bandwidth and Colocation sponsored by Easynews.com -- // // Asterisk-Users mailing list // Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users // http://lists.digium.com/mailman/listinfo/asterisk-users // To UNSUBSCRIBE or update options visit: // http://lists.digium.com/mailman/listinfo/asterisk-users / You can prepend a 'w' for a half-second wait which will resolve this problem. e.g., Zap/1/w19007529269 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Jerry, Check out: http://www.voip-info.org/tiki-index.php?page=Asterisk+Zap+channels Note this line on the page: /phonenumber/, if present, specifies which telephone number you wish to be connected with. Note that this makes sense only when you are dialing a telephone line (an FXO or PRI interface), not an internal extension. Within the phone number, you may use the special modifier *w* to indicate a half-second pause. You might want to use this to wait for a dialtone or for a pause while dialing digits. You may also use the special modifier *c* to allow for clear channel connections between PRI ports. w = half-a-second wait So, Zap/1/13105551212 would be a 2 second wait. However, I've dealt with a lot of phone providers and none have ever required more than a single half-a-second wait for them to begin detecting the DTMF tones, so you should be good with one 'w'. -samy ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] teliax
They are always there at the online chat.(during business hours) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Monday, August 29, 2005 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] teliax Chris wrote: Is there a problem at Teliax? I'm looking for a VoIP provider and when I call them they never answer the phone and the voice mail says it's full. I don;t see any network problems, and I monitor Teliax and a few other providers. Teliax is my main provider and I have never had any problems worth worrying about. Send them an email, I find they always respond, at least if the question is reasonable. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSG Waiting Off
On Mon, 2005-08-29 at 12:20 -0600, Joseph wrote: I think Asterisk is sending some signal to my cordless phone that is causing it to constantly display message: MSG Waiting Off. The problem is that it is impossible to program anything into the phone or sometime dial a phone numbers as the when I try to program a number or dial a number and asterisk sends a signal that is causing it to display that massage and my dialing or programing is interrupt. What is causing it? Or it could be my Sipura-3000 is sending some signal that is causing that message to appear. -- #Joseph ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and a Meridian Nortell Release 11
Where does the CNAM originate, is it sent to the Nortel from the PSTN and then passed on to *, or does it originate on the Nortel? There may be another way, but without more info I do not wan to peak out of context. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Monday, August 29, 2005 10:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and a Meridian Nortell Release 11 Hi there, We are using * with an Option 11C - we tried all of the various protocols and the only one we could get to work satisfactorily was 5ESS, with the * as CO and the Nortel as remote. The one drawback of this approach is getting name information for caller ID - because the Nortel sees the * as CO, it won't send the name information. /etc/zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #clear=1-24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf: [trunkgroups] [channels] context=incoming switchtype=5ess usecallingpres=yes echocancel=128 usecallerid=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 rxgain=-4.0 txgain=-6.0 group=1 callgroup=1 pickupgroup=1 signalling = pri_net channel = 1-23 musiconhold=default Any use? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 27-Aug-05, at 7:20 AM, Alvaro Parres wrote: Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel Release 11. I need to connect both of them. We are using MFC/R2 for this.. The Diagram: [ NORTEL ] ( AMI ) (DIGIUM) [ ASTERISK] we have green light at the digium card, and at asterisk we see all 31 channels as idle. But when i want to recive a call from the Nortel to the Asterisk i get at the Nortel only a empty sound, and after about 15 o 20 sec it's hangup. Any suggestion ? The log at Asterisk is: Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 - 0001 [1/ 1/Idle /Idle ] Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Detected Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Making a new call with CRN 32769 Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 1101 - [2/ 2/Idle /Idle ] Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:2865 handle_uc_event: Unicall/30 event Detected Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 - 0001 [1/ 1/Idle /Idle ] Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Detected Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Making a new call with CRN 32769 Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 1101 - [2/ 2/Idle /Idle ] Aug 27 09:16:20 WARNING[6496]: chan_unicall.c:2865 handle_uc_event: Unicall/30 event Detected tel2*CLI Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 - 1001 [2/ 2/Seize ack /Seize ack] Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event: Unicall/30 event Far end disconnected Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:3198 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Call control(6) Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Drop call(cause=Normal Clearing [16]) Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:2865 handle_uc_event: Unicall/30 event Drop call Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Call control(7) Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Release call Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 1001 - [1/1000/Clear fwd /Seize ack ] Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 - 1001 [2/ 2/Seize ack /Seize ack ] Aug 27 09:16:36 WARNING[6496]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/30 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Aug 27
Re: [Asterisk-Users] delay before dial on TDM04B
Samy Kamkar wrote: Jerry Geis wrote: Samy, Thanks for the suggestion - however I am confused on the wiki the 'w' stands for: *w*: Allow the /called/ user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.0.x) This is not a delay of any kind. Jerry [Asterisk-Users] delay before dial on TDM04B *Samy Kamkar* samy at fonality.com mailto:asterisk-users%40lists.digium.com?Subject=%5BAsterisk-Users%5D%20delay%20before%20dial%20on%20TDM04BIn-Reply-To=43134CDA.4020408%40pagestation.com /Mon Aug 29 13:07:05 CDT 2005/ * Previous message: [Asterisk-Users] delay before dial on TDM04B http://lists.digium.com/pipermail/asterisk-users/2005-August/123436.html * Next message: [Asterisk-Users] New astGUIclient version released 1.1.6 http://lists.digium.com/pipermail/asterisk-users/2005-August/123437.html * *Messages sorted by:* [ date ] http://lists.digium.com/pipermail/asterisk-users/2005-August/date.html#123438 [ thread ] http://lists.digium.com/pipermail/asterisk-users/2005-August/thread.html#123438 [ subject ] http://lists.digium.com/pipermail/asterisk-users/2005-August/subject.html#123438 [ author ] http://lists.digium.com/pipermail/asterisk-users/2005-August/author.html#123438 Jerry Geis wrote: / I am searching for a way to add a 2 second delay before calling out // with Dial(). // Sometimes I get the message you must first dial a 1 to place this call. // I presume the phone company is missing the first digit pulsed out // sometimes. // // How do I put a 2 second delay after coming offhook and before dialing // the digits? // // Thanks, // // jerry // // ___ // --Bandwidth and Colocation sponsored by Easynews.com -- // // Asterisk-Users mailing list // Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users // http://lists.digium.com/mailman/listinfo/asterisk-users // To UNSUBSCRIBE or update options visit: // http://lists.digium.com/mailman/listinfo/asterisk-users / You can prepend a 'w' for a half-second wait which will resolve this problem. e.g., Zap/1/w19007529269 Hi Jerry, Check out: http://www.voip-info.org/tiki-index.php?page=Asterisk+Zap+channels Note this line on the page: /phonenumber/, if present, specifies which telephone number you wish to be connected with. Note that this makes sense only when you are dialing a telephone line (an FXO or PRI interface), not an internal extension. Within the phone number, you may use the special modifier *w* to indicate a half-second pause. You might want to use this to wait for a dialtone or for a pause while dialing digits. You may also use the special modifier *c* to allow for clear channel connections between PRI ports. w = half-a-second wait So, Zap/1/13105551212 would be a 2 second wait. However, I've dealt with a lot of phone providers and none have ever required more than a single half-a-second wait for them to begin detecting the DTMF tones, so you should be good with one 'w'. -samy Also be advised that w ONLY seems to work with DTMF. For those who require pulse output, there is no way to delay the blind dialing. Any number of w's are ignored, so misdialing is probable if for any reason the Co isn't ready. John Novack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Internal Extensions Busy
Title: Message I have recently discovered a problem that I cannot dial internal extensions - I either get a busy tone or directed to voicemail depending on if the extension has voicemail. This was working fine, but not sure what has changed to stop this working. Today I did delete a load of extensions and setup and set of new ones. Now neither old or new can be dialled internally. Outgoing calls from each extension works fine. I'm getting the SIP registration for each phone - so they are definitely connected. The only other thing I can think of - I ran "yum -y update" which downloaded and installed a lot of stuff - I re-built the Zaptel and Network drivers after this. Your help is much appreciated Cheers Graham ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
