[asterisk-users] how to set the busy signal usign softphones
Hi to all. I'm using Asterisk 1.4 with Sjphone as softphone. My problem is that when a SIP user is busy, he still receive calls from asterisk. I've tried to setup the call-limit preference to 1, but with this kind of configuration the user can't transfer calls, as the system block the 2nd call generated to do the transfer. I've also tried to set the user as friend, limitonpeers = yes and call-limit =1. In that case the work-around works but only when the user is the receiver of the call that makes him busy. If the user is the caller, he still receive a second call. So, isn't there any method to limit the call available for a user to 1 but granting him the possibility to transfer a call? I know that there is the busy-level settings, but i'ts available only in 1.6. Thanks to all in advance. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set the busy signal usign softphones
nik600 a écrit : Hi to all. I'm using Asterisk 1.4 with Sjphone as softphone. My problem is that when a SIP user is busy, he still receive calls from asterisk. I've tried to setup the call-limit preference to 1, but with this kind of configuration the user can't transfer calls, as the system block the 2nd call generated to do the transfer. I've also tried to set the user as friend, limitonpeers = yes and call-limit =1. In that case the work-around works but only when the user is the receiver of the call that makes him busy. If the user is the caller, he still receive a second call. So, isn't there any method to limit the call available for a user to 1 but granting him the possibility to transfer a call? I know that there is the busy-level settings, but i'ts available only in 1.6. Thanks to all in advance. Have you tried to set the call-limit to 10 or 2 for example, i know it's what's needed for the queue system to detect busy on sip softphone ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [FreeBSD 6.3] Upgrading Zaptel messed up host
Hello Since the Ports collection showed that there were more recent versions of Asterisk and Zaptel, I tried to compile/install Zaptel, but it fails, even after stopping Zaptel cleanly, and even after stopping Asterisk itself, so I decided to just reboot. Now, when I type ztcfg -vv, I lose the SSH connection for a couple of minutes. === Dec 20 14:37:21 freebsd kernel: Zapata Telephony Interface Registered on major 196 Dec 20 14:37:21 freebsd kernel: Zaptel Version: zaptel-bsd-ng v0.0.1 Dec 20 14:37:21 freebsd kernel: Zaptel Echo Canceller: MG2 Dec 20 14:37:21 freebsd kernel: wctdm0 port 0xb400-0xb4ff mem 0xf500-0xf5000fff irq 9 at device 11.0 on pci2 Dec 20 14:37:21 freebsd kernel: wctdm0: [FAST] Dec 20 14:37:21 freebsd kernel: Freshmaker version: 71 Dec 20 14:37:21 freebsd kernel: Freshmaker passed register test Dec 20 14:37:21 freebsd kernel: Module 0: Installed -- AUTO FXO (FCC mode) Dec 20 14:37:21 freebsd kernel: Module 1: Not installed Dec 20 14:37:21 freebsd kernel: Module 2: Not installed Dec 20 14:37:21 freebsd kernel: Module 3: Not installed Dec 20 14:37:21 freebsd kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (1 modules) Dec 20 14:37:21 freebsd kernel: link_elf: symbol te11xp_init undefined === Any idea what's wrong? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set the busy signal usign softphones
On Sat, Dec 20, 2008 at 2:25 PM, Benoit maver...@maverick.eu.org wrote: Have you tried to set the call-limit to 10 or 2 for example, i know it's what's needed for the queue system to detect busy on sip softphone Yes, but if i set the call-limit to 2 the user receive more than 1 call (correctly...up to 2 calls), even when he is busy. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Needs more cpu usage
transcoding will eat you CPU 2008/12/20 Jason Kim asterja...@yahoo.com Hi, I am running * on centos5 using 4core cpu. When it is busy, * uses 99.9% of cpu max. How can I make * to use more cpu power? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to get /var/run/asteris/asterisk.ctl
2008/12/20 Scott Berry n7...@northlc.com Hello there everyone, Well I have set up Asteriks 6.0 and almost have Freepbx working too. However, freepbx is showing me that /var/run/asterisk/asterisk.ctl is not found. I confirmed that by going to the directory. How do I get /var/run/asterisk/asterisk.ctl put in correctly? I am using a Ubuntu 8.10 system. Thanks much. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users start it from /etc/init.d/asterisk start the script will create it... it only exist if asterisk is runing try to connect to asterisk doing asterisk -r maybe you will need to be root. if you cant connect if you normaly run asterisk as root run asterisk -c and it will show you why is not running. if you run asterisk as other user you MUST run asterisk whit that user or you will F**CK the instalation. su root password and then su asterisk_user and then asterisk -c regards David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set the busy signal usign softphones
I use the dev_state() function to find the status of the called phone. If it is BUSY then I call the busy() application to signal a busy tone. Firthermore, I also consult a MySQL table to see whether the user wants waiting calls or not and decide accordingly. __Yehavi: 2008/12/20 nik600 nik...@gmail.com On Sat, Dec 20, 2008 at 2:25 PM, Benoit maver...@maverick.eu.org wrote: Have you tried to set the call-limit to 10 or 2 for example, i know it's what's needed for the queue system to detect busy on sip softphone Yes, but if i set the call-limit to 2 the user receive more than 1 call (correctly...up to 2 calls), even when he is busy. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Needs more cpu usage
Your asterisk is using 99.9% of cpu and you want it to use more? Do you mean you want asterisk to use LESS cpu? CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Kim Sent: Friday, December 19, 2008 11:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Needs more cpu usage Hi, I am running * on centos5 using 4core cpu. When it is busy, * uses 99.9% of cpu max. How can I make * to use more cpu power? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Needs more cpu usage
On Fri, Dec 19, 2008 at 08:35:07PM -0800, Jason Kim wrote: Hi, I am running * on centos5 using 4core cpu. When it is busy, * uses 99.9% of cpu max. How can I make * to use more cpu power? If the work is done in a single thread, you can't spread it further. In top press 'H' (that is: shift-h) to get a separate list of threads. Do you see asingle thread taking most of the CPU time? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SMS text messaging capabilities
Hello! What kind of sms text messaging capabilities does Asterisk have? I do not know very much about about SMS technology, but I am looking for the following features: 1. mobile SIP devices can send and receive SMS messages 2. Asterisk server be able to accept and send SMS messages through PRI lines and Internet connections. I noticed that Asterisk has an SMS function, but I am not farmiliar enough with that technology to make it useful. Any help with this would be great! Regards, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 softphones keep ringing....
Running AsteriskNow, with FreePBX front end. Have two users who use softphones on notebooks in the field. Problem is that if the softphone receives a call, but the user is not available to pick it up, Asterisk will send the call to voice mail as normal, but the softphone will continue to ring.. I've tested the Virbiage and x-Lite softphones, same issue. Is there an IAX setting in Asterisk that applies here? Jerome ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS text messaging capabilities
On Sat, Dec 20, 2008 at 6:27 PM, Elliot Murdock murdo...@gmail.com wrote: What kind of sms text messaging capabilities does Asterisk have? I know nothing about PRI, but in Europe SMS can be used over fixed lines and there is a page of info on voip-info.org (should be easy to google the site) about SMS in various countries. I used SMS on our asterisk install for both send and receive and it worked beautifully. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS text messaging capabilities
2008/12/20 Elliot Murdock murdo...@gmail.com Hello! What kind of sms text messaging capabilities does Asterisk have? I do not know very much about about SMS technology, but I am looking for the following features: 1. mobile SIP devices can send and receive SMS messages 2. Asterisk server be able to accept and send SMS messages through PRI lines and Internet connections. I noticed that Asterisk has an SMS function, but I am not farmiliar enough with that technology to make it useful. Any help with this would be great! Regards, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users maybe this will be usefull http://www.voip-info.org/wiki/view/chan_mobile http://www.voip-info.org/wiki-Asterisk+cmd+Sms http://www.chan-mobile.org/?page_id=5 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS text messaging capabilities
Elliot Murdock wrote: Hello! What kind of sms text messaging capabilities does Asterisk have? Asterisk has the ability to use land lines to send SMS messages to a remote device that supports landline SMS. In Europe and much of the rest of the world SMS carriers provide a public PSTN gateway into the mobile phone networks. SMS carriers in the USA do not do this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS text messaging capabilities
But they do provide email-SMS gateways, which is enough for many applications of SMS. On Dec 20, 2008, at 3:12 PM, Eric \ManxPower\ Wieling e...@fnords.org wrote: Elliot Murdock wrote: Hello! What kind of sms text messaging capabilities does Asterisk have? Asterisk has the ability to use land lines to send SMS messages to a remote device that supports landline SMS. In Europe and much of the rest of the world SMS carriers provide a public PSTN gateway into the mobile phone networks. SMS carriers in the USA do not do this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4 with Lenny
Hello Tzafrir , On Fri, 19 Dec 2008, Tzafrir Cohen wrote: On Fri, Dec 19, 2008 at 09:32:57PM +0100, Loic Didelot wrote: Hi, I tried agx-addons with different version. I got it working till asterisk version 1.4.21 included on ubuntu with libtiff4. Starting from asterisk 1.4.22 it did not longer work. Just updated my backport. Originally intended to be in a Debian package but now I see that it won't make it. A patch vs. recent apps/app_fax.c (from 1.6.0) http://svn.debian.org/viewsvn/pkg-voip/asterisk-spandsp-plugins/trunk/debian/patches/app_fax_14?rev=6561view=log app_fax.c could be found oon the same area. Are these warning's OK ? Tia , JimL [CC] app_fax.c - app_fax.o app_fax.c:52: warning: no previous prototype for 'ast_tvdiff_sec' app_fax.c:63: warning: no previous prototype for 'ast_tvdiff_us' app_fax.c: In function 'phase_e_handler': app_fax.c:213: warning: implicit declaration of function 't30_get_tx_ident' app_fax.c:213: warning: assignment makes pointer from integer without a cast app_fax.c:214: warning: implicit declaration of function 't30_get_rx_ident' app_fax.c:214: warning: assignment makes pointer from integer without a cast app_fax.c: In function 'set_local_info': app_fax.c:272: warning: implicit declaration of function 't30_set_tx_ident' app_fax.c:276: warning: implicit declaration of function 't30_set_tx_page_header_info' app_fax.c: In function 'transmit_audio': app_fax.c:349: warning: unused variable 'fr' app_fax.c:347: warning: unused variable 'detect_tone' app_fax.c: At top level: app_fax.c:523: warning: 'transmit_t38' defined but not used -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4 with Lenny
On Sat, Dec 20, 2008 at 01:50:55PM -0900, Mr. James W. Laferriere wrote: Hello Tzafrir , On Fri, 19 Dec 2008, Tzafrir Cohen wrote: On Fri, Dec 19, 2008 at 09:32:57PM +0100, Loic Didelot wrote: Hi, I tried agx-addons with different version. I got it working till asterisk version 1.4.21 included on ubuntu with libtiff4. Starting from asterisk 1.4.22 it did not longer work. Just updated my backport. Originally intended to be in a Debian package but now I see that it won't make it. A patch vs. recent apps/app_fax.c (from 1.6.0) http://svn.debian.org/viewsvn/pkg-voip/asterisk-spandsp-plugins/trunk/debian/patches/app_fax_14?rev=6561view=log app_fax.c could be found oon the same area. Are these warning's OK ? Tia , JimL What version of spandsp do you use? [CC] app_fax.c - app_fax.o app_fax.c:52: warning: no previous prototype for 'ast_tvdiff_sec' app_fax.c:63: warning: no previous prototype for 'ast_tvdiff_us' app_fax.c: In function 'phase_e_handler': app_fax.c:213: warning: implicit declaration of function 't30_get_tx_ident' app_fax.c:213: warning: assignment makes pointer from integer without a cast app_fax.c:214: warning: implicit declaration of function 't30_get_rx_ident' app_fax.c:214: warning: assignment makes pointer from integer without a cast app_fax.c: In function 'set_local_info': app_fax.c:272: warning: implicit declaration of function 't30_set_tx_ident' app_fax.c:276: warning: implicit declaration of function 't30_set_tx_page_header_info' app_fax.c: In function 'transmit_audio': app_fax.c:349: warning: unused variable 'fr' app_fax.c:347: warning: unused variable 'detect_tone' app_fax.c: At top level: app_fax.c:523: warning: 'transmit_t38' defined but not used -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4 with Lenny
Hello Tzafrir , On Sun, 21 Dec 2008, Tzafrir Cohen wrote: On Sat, Dec 20, 2008 at 01:50:55PM -0900, Mr. James W. Laferriere wrote: On Fri, 19 Dec 2008, Tzafrir Cohen wrote: On Fri, Dec 19, 2008 at 09:32:57PM +0100, Loic Didelot wrote: Hi, I tried agx-addons with different version. I got it working till asterisk version 1.4.21 included on ubuntu with libtiff4. Starting from asterisk 1.4.22 it did not longer work. Just updated my backport. Originally intended to be in a Debian package but now I see that it won't make it. A patch vs. recent apps/app_fax.c (from 1.6.0) http://svn.debian.org/viewsvn/pkg-voip/asterisk-spandsp-plugins/trunk/debian/patches/app_fax_14?rev=6561view=log app_fax.c could be found oon the same area. Are these warning's OK ? Tia , JimL What version of spandsp do you use? spandsp-0.0.4pre16.tgz Which one is this patch compiling against successfully ? Tho later the make finally blew chuck at LD time ... make[2]: Leaving directory `/home/archive/asterisk/asterisk-1.4.22/main/db1-ast' [LD] abstract_jb.o acl.o aescrypt.o aeskey.o aestab.o alaw.o app.o ast_expr2.o ast_expr2f.o asterisk.o astmm.o astobj2.o audiohook.o autoservice.o callerid.o cdr.o channel.o chanvars.o cli.o config.o cryptostub.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o file.o fixedjitterbuf.o frame.o fskmodem.o global_datastores.o http.o image.o indications.o io.o jitterbuf.o loader.o logger.o manager.o md5.o netsock.o pbx.o plc.o privacy.o rtp.o say.o sched.o sha1.o slinfactory.o srv.o stdtime/localtime.o strcompat.o tdd.o term.o threadstorage.o translate.o udptl.o ulaw.o utils.o editline/libedit.a db1-ast/libdb1.a ../apps/modules.link ../cdr/modules.link ../channels/modules.link ../codecs/modules.link ../formats/modules.link ../funcs/modules.link ../pbx/modules.link ../res/modules.link - asterisk ../apps/app_fax.o: In function `fax_generator_generate': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:319: undefined reference to `fax_tx' ../apps/app_fax.o: In function `load_module': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:793: undefined reference to `span_set_message_handler' ../apps/app_fax.o: In function `phase_e_handler': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:196: undefined reference to `t30_get_transfer_statistics' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:204: undefined reference to `t30_completion_code_to_str' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:206: undefined reference to `t30_completion_code_to_str' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:213: undefined reference to `t30_get_tx_ident' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:214: undefined reference to `t30_get_rx_ident' ../apps/app_fax.o: In function `transmit_audio': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:375: undefined reference to `fax_init' ../apps/app_fax.o: In function `set_logging': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:260: undefined reference to `span_log_set_message_handler' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:261: undefined reference to `span_log_set_level' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:260: undefined reference to `span_log_set_message_handler' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:261: undefined reference to `span_log_set_level' ../apps/app_fax.o: In function `set_file': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:282: undefined reference to `t30_set_tx_file' ../apps/app_fax.o: In function `set_ecm': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:289: undefined reference to `t30_set_ecm_capability' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:290: undefined reference to `t30_set_supported_compressions' ../apps/app_fax.o: In function `transmit_audio': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:386: undefined reference to `fax_set_transmit_on_idle' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:388: undefined reference to `t30_set_phase_e_handler' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:501: undefined reference to `t30_set_phase_e_handler' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:504: undefined reference to `t30_terminate' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:505: undefined reference to `fax_release' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:450: undefined reference to `fax_rx' ../apps/app_fax.o: In function `set_file': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:284: undefined reference to `t30_set_rx_file' ../apps/app_fax.o: In function `set_local_info': /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:272: undefined reference to `t30_set_tx_ident' /home/archive/asterisk/asterisk-1.4.22/apps/app_fax.c:276: undefined reference to `t30_set_tx_page_header_info' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make[1]: Leaving
Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4 with Lenny
On Sat, Dec 20, 2008 at 02:21:28PM -0900, Mr. James W. Laferriere wrote: Hello Tzafrir , On Sun, 21 Dec 2008, Tzafrir Cohen wrote: What version of spandsp do you use? spandsp-0.0.4pre16.tgz Which one is this patch compiling against successfully ? 0.0.5pre4 . However with 0.0.4pre16 you should be able to build the agx-addons package mentioned above. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4 with Lenny
Hello Tzafrir , On Sun, 21 Dec 2008, Tzafrir Cohen wrote: On Sat, Dec 20, 2008 at 02:21:28PM -0900, Mr. James W. Laferriere wrote: On Sun, 21 Dec 2008, Tzafrir Cohen wrote: What version of spandsp do you use? spandsp-0.0.4pre16.tgz Which one is this patch compiling against successfully ? 0.0.5pre4 . However with 0.0.4pre16 you should be able to build the agx-addons package mentioned above. For some darned reason , the RxFax would not pickup an inbound fax . So I am trying other options . Your patched app_fax.c . Would 0.0.6pre3 be a problem you think ? Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 softphones keep ringing....
On Sat, Dec 20, 2008 at 5:54 PM, Jerome Deyle jde...@gmail.com wrote: Running AsteriskNow, with FreePBX front end. Have two users who use softphones on notebooks in the field. Problem is that if the softphone receives a call, but the user is not available to pick it up, Asterisk will send the call to voice mail as normal, but the softphone will continue to ring.. I've tested the Virbiage and x-Lite softphones, same issue. Is there an IAX setting in Asterisk that applies here? Jerome What version of asterisk are you running ? IIRC I experienced similar behaviours with 1.4.22... I'm now running 1.4.19 (as far as this version, because of other issues, fixed in 1.4.22 -- but IAX non-hangups broke 1.4.22 for me) -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
