RE: [Asterisk-Users] Language Problems
Thx Peter... I ooked all over thewiki for that tip but no luck... Thank you! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Sábado, 26 de Febrero de 2005 01:53 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Language Problems On Sat, 26 Feb 2005, Anton Krall wrote: and then Spanish as /var/log/asterisk/sounds/sp /var/log/asterisk/sounds/sp/phonetic /var/log/asterisk/sounds/sp/digit /var/log/asterisk/sounds/sp/letters Now, the normal voices ARE heard in spanish but all digit related voices are taken from the english dirs... why? Asterisk will add sp after the last directory name. You will need the following directories (or symlinks): /var/log/asterisk/sounds/sp /var/log/asterisk/sounds/phonetic/sp /var/log/asterisk/sounds/digit/sp /var/log/asterisk/sounds/letters/sp Peter ___ 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 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] Queue Auto fallthrough
I gave a queue setup like this, but I also have it setup so that if no
agents are online, the caller cannot get in but I discovered that if that's
the case, the call hangsup on the caller:
[soportetecnico]
;Soporte Tecnico
exten = 2,1,Playback(${SONIDOS}/transferringcall)
exten = 2,2,Queue(Soporte-Tecnico)
exten = 2-.,1,Playback(noagents)
I want to play a message tothecaller saying no agents are online but this
doesn't seem to be working... Any suggestions guys?
-- Executing Queue(SIP/intruder1-9e41, Soporte-Tecnico) in new stack
Feb 26 02:58:39 WARNING[29371]: app_queue.c:2429 queue_exec: Unable to join
queue 'Soporte-Tecnico'
== Auto fallthrough, channel 'SIP/intruder1-9e41' status is 'UNKNOWN'
__
Anton Krall
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[Asterisk-Users] Polycom SP300 problem solved
Hi, all Registration problem is solved now. I did not realise phones also have web interface. I used that to set up SIP server and authentication. Settings on the phone itself do not have all the options. Rudolf ___ 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] FRS *: an actual business use
I've noticed a growing number of stores using FRS radios. It would make sense to interface (via soundcard/console driver, with the nessacary electrical conversion) a VOX FRS radio to asterisk to allow someone in the office to page/talk with people on the floor or warehouse. You could throw that call (ie, all the radios) into a meetme conference. Then, you could have people in the office either dial that extension and/or have some of them always in that conference on a speaker phone (muted usually). Unless I'm not understanding your comments, the meetme conference isn't needed (assuming all radios are on the same channel). The radios become the meetme for all practical purposes. (When the base radio transmits, all remote radios listen.) While I haven't tried this (yet!), it does seem like it would be a useful feature. The restriction against PSTN interconnection would be met UNLESS your dialplan allowed outside (PSTN) callers into that conference. You /could/ allow remote softphone users into the conference. ___ 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 and SER
Hi Everyone, Just a curious question. Has anyone heard of any service provider who is using Asterisk and SER to provide their VOIP services? Thanks Walid ___ 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] open 723
yes, it works great even on a very high load machine with two tormenta2 quad E1/PRI cards. On Sat, 2005-02-26 at 03:45 +, Kanishka Somaratne wrote: has any one implemented open 723 at http://www.readytechnology.co.uk/open/g723.1 ___ 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 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] listening to gsm files
Hello list, I am having trouble listening to GSM files created by Asterisk using a browser. I am noticing that some of my users succeed in listening to them and some others don't. I guess it is something of a codec problem that does not seem to be installed on all machines (though they are all WinXP). Anybody knows what one should do to listen to GSM files? I send files through the browser as audio/GSM, this should be correct, right? Thnaks l. -- Assum est, versa et manduca. ___ 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] Queue Question
I have a quick question.. reading the wiki, I found this: Use the 'Local' channel construct to point to an appropriate dial-out extension in the dialplan if you'd like to add remote agents using AgentCallbackLogin() That's exactly what Im trying to do, so fasr I needed to make a new context that only has Dial cmds so that the call would not get routed to the voicemails... But, what is this local channel construct? __ Anton Krall ___ 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] Determine IP addres of a AIP/IAX user
Hello all!
Is there any possibility to determine the
IP address of a caller in my dialplan?
I would like to have a predefined channel variable
like ${CALLER_IP} but it seems it doesnt exist (http://www.voip-info.org/wiki-Asterisk+Variables)
.. is this list complete?
Are there any other possibility to store the
SIP/IAX callers IP address on every call?
Thanks
Niels
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RE: [Asterisk-Users] Determine IP addres of a AIP/IAX user
Niels
Are there any other possibility to store the
SIP/IAX callers IP address on every call?
Run the command database
show at the Asterisk command line (CLI). I believe this shows the information you
require. If it does, you can access the information within the dialplan using
DBGet()
Bill Seddon
Lyquidity Solutions Limited
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: February 26, 2005 10:32 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Determine IP addres of a AIP/IAX user
Hello all!
Is there any possibility
to determine the IP address of a caller in my dialplan?
I would like to have a
predefined channel variable like ${CALLER_IP} but it seems it doesnt
exist (http://www.voip-info.org/wiki-Asterisk+Variables)
.. is this list complete?
Are there any other
possibility to store the SIP/IAX callers IP address on every call?
Thanks
Niels
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[Asterisk-Users] ERROR: when compile app_addon_sql_mysql.c of asterisk_addon
Dear ALL: I got this error when try to compile asterisk_addon. Does anybody have solution about it ? cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given make: *** [app_addon_sql_mysql.o] Error 1 Best Regard Charles ___ 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: Getting PHP Config to work?
I had a look but $100 seems a bit steep to me at the minute! C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Herman Cremer Sent: 25 February 2005 14:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FW: Getting PHP Config to work? If you are using windows, have a look at Zend Studio that is used for PHP but can do wonders for other editing apps as well. -herman On Fri, 2005-02-25 at 15:52, C. Tomlinson wrote: Richard, I have been using WinSCP to transfer files across easily without messing with FTP accounts. I had not found that feature, many thanks for pointing it out :-D I will definitely use this from now on until I find a better solution. Do you have an easy way to reload asterisk after changing the files? Have putty open to do a reload? Or use the builtin terminal capabilities of WinSCP? This is a great fix as my main machine is currently Windows. However I would still like to get phpconfig working as it would be easier to use that across the internet etc. Thanks Again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Folwell Sent: 25 February 2005 13:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? C. Tomlinson wrote: I have been doing various testing with asterisk and its been going great. However I am a bit feedup of using vi for editing configs, and would rather do it from any machine on my LAN. Look at WinSCP: http://www.winscp.org/ which is a lovely program that initially purports to provide easier file transfer, but which has some very useful tricks up its sleeve - including editing remote files in place. It is (almost) worth installing Windows just to be able to use it. :-) If anyone knows of anything similar that runs under Linux please enlighten me! Richard ___ 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 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 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 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] Queue Question
On Sat, 26 Feb 2005, Anton Krall wrote: I have a quick question.. reading the wiki, I found this: Use the 'Local' channel construct to point to an appropriate dial-out extension in the dialplan if you'd like to add remote agents using AgentCallbackLogin() That's exactly what Im trying to do, so fasr I needed to make a new context that only has Dial cmds so that the call would not get routed to the voicemails... But, what is this local channel construct? Use google: asterisk local channel. Peter ___ 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: Getting PHP Config to work?
Hi Tzafrir, I do accept that there are many security issues with this setup. However I agree with the post in the previous thread: If the asterisk server is reachable from the outside over http or other unsecure protocols, it would be really dangerous. But in a trusty intranet-environment, where firewalls block every attempt to access the asterisk server from the outside, this solution should be save enough, even if nothing is really save enough ;-) . -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 25 February 2005 18:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? On Fri, Feb 25, 2005 at 04:43:50PM -, C. Tomlinson wrote: Hi, Thanks for the batchfile type, it's a handy one. I'm not editing over the internet, just local LAN for testing ATM. Protected via firewall. Would it not be fairly secure using a combination of the following: .htaccess file VPN? https access? Limit apache to only allow certain IP's? And the public keys thing. Secure agains what? What are the threats you consider? VPN and/or limit of IP addresses (in iptables or in apache's config) would serve to allow access only from certain addresses. But is this a relaistic limitation? I thout you wanted to be able to edit the configuration from various hosts. If this is only your setup, then an sftp-based setup is probably more convinient. Using a .htaccess file (or even better: an apache config snippet in /etc/apache/conf.d )you can force authentication to get to this directory. But then-again, you empower the apache user (www-data) to configure and control asterisk, and thus if anybody manages to make your web server execute an arbitrary script (e.g: can write to a .php file under the wwwroot) they can make asterisk execute arbitrary code as well. The chmod command makes Asterisk's configuration world-writable. So anybody with temporary write access to your system can again change asterisk's configuration. This breaks a general sanity assumption that only system users can write to the configuration. As a rule of thumb such a chmod should generally be replaced by adding a certain user to a certain group. You also change the permissions to the asterisk reload script to 777. Why does it need to be world-writable? This gives an attacker yet another place to inject executable code. In short: I still fail to see the atvantages of using phpconfig in your settings. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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 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: Getting PHP Config to work?
-Original Message- From: C. Tomlinson [mailto:[EMAIL PROTECTED] Sent: 26 February 2005 11:39 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FW: Getting PHP Config to work? Hi Tzafrir, I do accept that there are many security issues with this setup. I am fairly ignorant of the exact problems due to my lack of knowledge. However I agree with the post in the previous thread: If the asterisk server is reachable from the outside over http or other unsecure protocols, it would be really dangerous. But in a trusty intranet-environment, where firewalls block every attempt to access the asterisk server from the outside, this solution should be save enough, even if nothing is really save enough ;-) . Guido Hecken What exactly do you mean by an sftp based setup? Is this like the builtin editor in WinSCP? Phpconfig allows me to change the config by any pc on my LAN, using windows, mac, pocket pc(have to test this one) etc. This is handy for me for testing. I like the flexibility it gives me. The * box is behind a NAT firewall, the only ports open being those for IAX. If I setup a VPN in the future I will be able to access the phpconfig files securely (?) via that. It may not suite everyone. Maybe the 777 CHMOD could be done better, but this was the way it worked for me. I am fairly new to Linux and *, so my methods will not be the best. Thanks for all the informationif I get to a production box I will probably not use phpconfig! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 25 February 2005 18:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FW: Getting PHP Config to work? On Fri, Feb 25, 2005 at 04:43:50PM -, C. Tomlinson wrote: Hi, Thanks for the batchfile type, it's a handy one. I'm not editing over the internet, just local LAN for testing ATM. Protected via firewall. Would it not be fairly secure using a combination of the following: .htaccess file VPN? https access? Limit apache to only allow certain IP's? And the public keys thing. Secure agains what? What are the threats you consider? VPN and/or limit of IP addresses (in iptables or in apache's config) would serve to allow access only from certain addresses. But is this a relaistic limitation? I thout you wanted to be able to edit the configuration from various hosts. If this is only your setup, then an sftp-based setup is probably more convinient. Using a .htaccess file (or even better: an apache config snippet in /etc/apache/conf.d )you can force authentication to get to this directory. But then-again, you empower the apache user (www-data) to configure and control asterisk, and thus if anybody manages to make your web server execute an arbitrary script (e.g: can write to a .php file under the wwwroot) they can make asterisk execute arbitrary code as well. The chmod command makes Asterisk's configuration world-writable. So anybody with temporary write access to your system can again change asterisk's configuration. This breaks a general sanity assumption that only system users can write to the configuration. As a rule of thumb such a chmod should generally be replaced by adding a certain user to a certain group. You also change the permissions to the asterisk reload script to 777. Why does it need to be world-writable? This gives an attacker yet another place to inject executable code. In short: I still fail to see the atvantages of using phpconfig in your settings. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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 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: when compile app_addon_sql_mysql.c of asterisk_addon
Update both asterisk and asterisk-addons. This error is usually due to not updating include files in /usr/src/include/asterisk (this files are updated when a make install is done from asterisk soruce dir). Regards, Cristian Alonso On Sat, 26 Feb 2005 19:34:59 +0800, Charles Wang [EMAIL PROTECTED] wrote: Dear ALL: I got this error when try to compile asterisk_addon. Does anybody have solution about it ? cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given make: *** [app_addon_sql_mysql.o] Error 1 Best Regard Charles ___ 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 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] Wildcard failing to load on asterisk@home
I have a E110P which I can't seem to make function and i think it has something to do with interrupts, but I've never had to deal with them before and was wondering if someone could offer some guidance. a modprope zaptel loads fine. when i then do modprobe wcte11xp i get the following: /lib/modules/2.4.21-27.0.1.EL/misc/wcte11xp.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.1.EL/misc/wcte11xp.o: insmod /lib/modules/2.4.21-27.0.1.EL/misc/wcte11xp.o failed /lib/modules/2.4.21-27.0.1.EL/misc/wcte11xp.o: insmod wcte11xp failed I then went to # cat /proc/interrupts CPU0 0:5200010 XT-PIC timer 1: 3 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 50373 XT-PIC ide0 8: 1 XT-PIC rtc 10: 661881 XT-PIC usb-ohci, eth0 14: 0 XT-PIC ide2 NMI: 0 ERR: 0 and i don't see my wildcard, but... #dmesg [snip] Zapata Telephony Interface Registered on major 196 Specify address with base=0xN Registered Tormenta2 PCI usb.c: registered new driver wcusb Wildcard USB FXS Interface driver registered [snip] (should that be usb fxs??) and then a # lspci -bv [snip] 00:05.0 System peripheral: Compaq Computer Corporation Advanced System Management Controller Subsystem: Compaq Computer Corporation: Unknown device b0f3 Flags: medium devsel, IRQ 11 I/O ports at 1800 Memory at f5fe (32-bit, non-prefetchable) 00:08.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 795e:0001 Flags: bus master, medium devsel, latency 64, IRQ 11 I/O ports at 2800 Memory at f5fd (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 [snip] shows my card is there, but sharing an IRQ with something else. my zaptel.conf is span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk and # ztcfg -v Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 31 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) which i guess makes sense... Now, I'm a bit stuck as to the correct procedure for correcting this. Should I move the card to another slot, try going into the BIOS and allocating another IRQ no manually? is it even that that is causing the problem. (I haven't tried these yet as getting to the box is tricky, but if that will fix then i will obv make the trip) Could anyone advise me how to proceed further on this? This is my first zaptel/PRI build after numerous SIP and CAPI configs without problems... I'm hoping it is something simple, but i've never fiddled with IRQs before and just need someone to point me in the right direction. Thanks in advance r ___ 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] Digium BRI or quad BRI
Hi, I had similar questions. I've emailed few questions and got no response in 10 days from junghanns. So I decided to try Beronet cards (they will arrive shortly). I just cannot imagine to have support from someone that is not able to answer few simple technical questions about their cards in one week. HTH, regards, Rob. - Original Message - From: Brett, Gary [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, February 23, 2005 2:49 PM Subject: RE: [Asterisk-Users] Digium BRI or quad BRI Thanks Florian, that's great, is this card (junghanns QuadBRI) really stable with * ? Do you or anybody else have any experiences with this card and also is it ok to run multiple cards in one machine cheers -Original Message- From: Florian Overkamp [mailto:[EMAIL PROTECTED] Sent: 23 February 2005 12:58 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Digium BRI or quad BRI Hi, -Original Message- Hi there, quick question...do digium make any BRI cards (ISDN2) or even better a quad port BRI, maybe im going blind, but I cant see any on their website They don't. If you are in need of a european ISDN2 type, see if http://www.junghanns.net/asterisk/page17.html helps you out. Florian ___ 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 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 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] Queue Question
Found it! Interesting.. Local helps on not having to make 2 contexts, one with all extension and their dialouts and privs and one with just the local extensions and their voicemails without privs so callers wont be able to abuse... Am I right on this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Sábado, 26 de Febrero de 2005 05:38 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queue Question On Sat, 26 Feb 2005, Anton Krall wrote: I have a quick question.. reading the wiki, I found this: Use the 'Local' channel construct to point to an appropriate dial-out extension in the dialplan if you'd like to add remote agents using AgentCallbackLogin() That's exactly what Im trying to do, so fasr I needed to make a new context that only has Dial cmds so that the call would not get routed to the voicemails... But, what is this local channel construct? Use google: asterisk local channel. Peter ___ 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 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] HFC-S ISDN card on *@home
Hello, the easiest way ist to download BRI Stuff, untar it and to run the install script. The install script will automatically download Asterisk and patch it to work with ZAPHfc. On Mon, 21 Feb 2005 00:07:40 +0100, Erwin de Raad [EMAIL PROTECTED] wrote: It seems so simple, but I'm having no luck installing a HFC-s ISDN BRI card on [EMAIL PROTECTED] 0.5. I probably have to install BRI-stuff from Junghanns.net but that also downloads and installs another copy of * from Digium. I'm not sure if zaphfc has to be installed *before* Asterisk or if it's OK to do this afterwards. I've seen this question before, but: Anyone successfully installed a HFC-s card on [EMAIL PROTECTED] Please post the steps you had to take. I'm sure quite some list-members are interested! With kind regards Erwin ___ 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 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] Interface * with ATA from ATA FXS port?
Me again... I have service with a company that does not allow for a BYOD plan. They will not give out credential or server info either. Is it possible to run the FXS port of the ATA to an FXO port in *? The service I have is throug Broadvox Direct using the Mediatrix 2102. I have tried this using loop start and kewl start. The * box sees the incoming ring, picks up and starts my dial plan. But the Mediatrix never recognizes that the * box has picked up and continues to ring and then goes on to voicemail. I am beginning to think that this is not possible. Guess maybe I should have found another provider. I have a number port in progress right now and cannot stop it. Robert ___ 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] listening to gsm files
Anybody knows what one should do to listen to GSM files? I know QuickTime can play gsm files. Maybe your users that succeeded had it installed. hth ___ 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] ERROR: compile asterisk (from CVS HEAD)
Dear ALL: I got an error message lists below. Does anyone have the same problem? How to solve it? Best Regard Charles In file included from config.c:34: include/asterisk/app.h:62: array size missing in `options' make: *** [config.o] Error 1 ___ 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] Interface * with ATA from ATA FXS port?
Me again... I have service with a company that does not allow for a BYOD plan. They will not give out credential or server info either. Is it possible to run the FXS port of the ATA to an FXO port in *? The service I have is throug Broadvox Direct using the Mediatrix 2102. I have tried this using loop start and kewl start. The * box sees the incoming ring, picks up and starts my dial plan. But the Mediatrix never recognizes that the * box has picked up and continues to ring and then goes on to voicemail. I am beginning to think that this is not possible. Guess maybe I should have found another provider. I have a number port in progress right now and cannot stop it. Sounds like an ata or * config problem to me, but then without any copy/paste of what you've configured, debug data, etc, no one is going to get any closer then throwing a dart at the wall. Is that ata a Cisco or what? So, if you really want some help, try to collect and post some useful data in some reasonable manner. - config data from * for your ata box - config data from * for your extensions.conf - any debug that supports your boxes (eg, sip debug, zap debug, etc) - config data and/or debug data from the ata ___ 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] FRS *: an actual business use
I've noticed a growing number of stores using FRS radios. It would make sense to interface (via soundcard/console driver, with the nessacary electrical conversion) a VOX FRS radio to asterisk to allow someone in the office to page/talk with people on the floor or warehouse. You could throw that call (ie, all the radios) into a meetme conference. Then, you could have people in the office either dial that extension and/or have some of them always in that conference on a speaker phone (muted usually). Yah I started this thread with this basic requirement have been trying to track down if this is feasible. So I have been talking with local radio hams. They have brought to my attention some limitations with using a low end FRS/gmrs handset as the base station 1) battery , since this would be in used more than the individual handset but you could rig a direct power supply to it 2) talk time, these low end handset are designed to cut out after a configured period apparently 30-60 secs 3) the actual radio, under continuos use most experts here believe the radios would over heat and die a quick and certain death :) BUT the best suggestion I found was to use a real uhf radio tune it to the gmrs/FRS frequency and drop its power to the 2 watts like the GMFRS/FRS radios a Motorola cm200/cm300 http://www.motorola.com/cgiss/mobiles/cm300.shtml might work Then I believe you can use the zapata analog telephony adapter to interface to cm200/cm300 via the 9pin adpater and the cm200/300 mic jack and asterisk will interface via app_rpt... I would appreciate anybody more familiar with this technology to vet this config and raise any flag if they see any issues with this ...from a tech pov the legal issue has been raised I don't really want hear any more about that :( ___ 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] Interface * with ATA from ATA FXS port?
Robert Webb wrote: Me again... I have service with a company that does not allow for a BYOD plan. They will not give out credential or server info either. Is it possible to run the FXS port of the ATA to an FXO port in *? Can you answer the call from a POTS phone? Are you sure the cable is good connecting the two? Are you sure the FXO card is good? Output of the ATA should appear to Asterisk as a PSTN line. Perhaps your problem is really simple? John Novack ___ 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] ERROR: compile asterisk(from CVS HEAD) and got an error
Dear ALL: I got an error message lists below. Does anyone have the same problem? How to solve it? Best Regard Charles In file included from config.c:34: include/asterisk/app.h:62: array size missing in `options' make: *** [config.o] Error 1 ___ 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: compile asterisk(from CVS HEAD) and got anerror
Hi All Just made a post before but seem's not to appear in the list I replaced at line 62 of include/asterisk/app.h file struct ast_ivr_option options[]; /* All options */ With struct ast_ivr_option *options; /* All options */ It works but as i'm not very good a C don't know if it's ok Bets regards thierry -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Charles Wang Envoyé : samedi 26 février 2005 15:58 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] ERROR: compile asterisk(from CVS HEAD) and got anerror Dear ALL: I got an error message lists below. Does anyone have the same problem? How to solve it? Best Regard Charles In file included from config.c:34: include/asterisk/app.h:62: array size missing in `options' make: *** [config.o] Error 1 ___ 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 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] FRS *: an actual business use
I've noticed a growing number of stores using FRS radios. It would make sense to interface (via soundcard/console driver, with the nessacary electrical conversion) a VOX FRS radio to asterisk to allow someone in the office to page/talk with people on the floor or warehouse. You could throw that call (ie, all the radios) into a meetme conference. Then, you could have people in the office either dial that extension and/or have some of them always in that conference on a speaker phone (muted usually). Yah I started this thread with this basic requirement have been trying to track down if this is feasible. So I have been talking with local radio hams. They have brought to my attention some limitations with using a low end FRS/gmrs handset as the base station 1) battery , since this would be in used more than the individual handset but you could rig a direct power supply to it 2) talk time, these low end handset are designed to cut out after a configured period apparently 30-60 secs 3) the actual radio, under continuos use most experts here believe the radios would over heat and die a quick and certain death :) BUT the best suggestion I found was to use a real uhf radio tune it to the gmrs/FRS frequency and drop its power to the 2 watts like the GMFRS/FRS radios a Motorola cm200/cm300 http://www.motorola.com/cgiss/mobiles/cm300.shtml might work Then I believe you can use the zapata analog telephony adapter to interface to cm200/cm300 via the 9pin adpater and the cm200/300 mic jack and asterisk will interface via app_rpt... I would appreciate anybody more familiar with this technology to vet this config and raise any flag if they see any issues with this ...from a tech pov the legal issue has been raised I don't really want hear any more about that :( I've not played with those Motorolas, but the older ones required a special box to program the frequencies, etc. Doubtful that can be done from the front panel as service tech's would have a major problem chasing commercial users/employees who were playing around with the radio accidently changing frequencies and other programmable features. Also, Motorola has been rather careful with their designs in the past, not allowing a radio to be programmed in such a way as to allow it to be used in bands that would make it illegal. Best ensure exactly what frequencies it truly can be programmed to use; best guess is the FRS frequencies are restricted for obvious reasons. Ham radio phone patches are basically a 2-wire to 4-wire hybrid (not unlike pstn fxo interfaces), and require a fair amount of tweeking to minimize crosstalk/feedback, etc. They also are impedance matching devices intended to interconnect 600-ohm pstn lines with high-impendance radio inputs, etc. Therefore, proper impedance matching to whatever radio is selected will be required in any case. (Use of sound board mic and speaker jacks will still require impedance matching and level adjustments that should be well researched.) After all of the above is addressed, you'll still run into issues with the lack of control from *. E.g., who controls dropping the call? What happens when the call is not dropped and busy signal is constantly transmitted to all radios? etc, etc. ___ 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] Language Problems
On Sat, Feb 26, 2005 at 08:53:23AM +0100, Peter Svensson wrote: On Sat, 26 Feb 2005, Anton Krall wrote: Asterisk will add sp after the last directory name. You will need the following directories (or symlinks): /var/log/asterisk/sounds/sp /var/log/asterisk/sounds/phonetic/sp /var/log/asterisk/sounds/digit/sp ^^ this should be digits/sp (s) /var/log/asterisk/sounds/letters/sp Peter -- Tho/\/\as ___ 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] Error Message
Ever since I started using asterisk I see often this error message, can sombody tell me what it means? Feb 26 09:20:40 WARNING[29371]: app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! __ Anton Krall ___ 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: compile asterisk(from CVS HEAD) and got an error
See bug #3639 http://bugs.digium.com/bug_view_page.php?bug_id=0003639 Nothing's been committed yet though I think... On Sat, Feb 26, 2005 at 10:58:11PM +0800, Charles Wang wrote: Dear ALL: I got an error message lists below. Does anyone have the same problem? How to solve it? Best Regard Charles In file included from config.c:34: include/asterisk/app.h:62: array size missing in `options' make: *** [config.o] Error 1 ___ 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 -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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] NAT= setting for a public proxy
Hi, I'm chasing a bug in chan_sip.c where Asterisk is removing the rport parameter out of the via headers. Here's my scenario: UA - Snom NATf - Snom 4S Proxy - Asterisk Echo Test Function NATf, the proxy, and Asterisk are all on public IPs. So my question is: In chan_sip.c, copy_via_headers function, I see an if statement testing for (ast_test_flag(p, SIP_NAT) == SIP_NAT_ALWAYS) What in sip.conf do I do to toggle/change SIP_NAT to try to match this if statement? Following is my sip.conf for the proxy. Note I've tried nat=yes, nat=no, nat=always but the darn thing always takes the else instead of matching the if. #sip.conf ; Make asterisk register with a foreign proxy ; In this case: ; 9723048720 = account on foreign proxy ; passwd = password for foreign account ; abpusa.com = domain to register with ; 2000 = Asterisk extension to ring if somebody on the foreign proxy calls 9723048720 register = 9723048720:[EMAIL PROTECTED]/2000 ; Used in extension.conf [snomproxy] nat=no type=peer context=extensions host=abpusa.com ;disallow=all ;allow=ulaw Here is a sip debug: notice the via for 192.168 in the inbound INVITE, then notice the lack of it on the outbound 200 OK. Sip read: INVITE sip:[EMAIL PROTECTED]:5070 SIP/2.0 v: SIP/2.0/UDP 209.189.239.106:5060;branch=z9hG4bK-6f04a3acfdd3002c645fcb7605027073.1 v: SIP/2.0/UDP 209.189.239.106:5062;branch=z9hG4bK-dc2a770cf56399c4c0eb8ca964813dc9;nat=true v: SIP/2.0/UDP 192.168.5.102:5060;branch=z9hG4bK-d0t7a9g6nme4;rport=5060;received=4.13.144.17 Record-Route: sip:abpusa.com:5060;maddr=209.189.239.106;lr=1 Record-Route: sip:209.189.239.106:5062;transport=udp;dest=4.13.144.17-5060;to-tag=u2dogw3pvf;lr=1 f: sip:[EMAIL PROTECTED];tag=u2dogw3pvf t: sip:[EMAIL PROTECTED];user=phone i: [EMAIL PROTECTED] CSeq: 1 INVITE Max-Forwards: 68 m: sip:[EMAIL PROTECTED]:5060;line=70n796w7 P-Key-Flags: keys=3 User-Agent: snom190-3.56t Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 P-Asserted-Identity: Shanon Snom190 sip:[EMAIL PROTECTED];tag=u2dogw3pvf c: application/sdp l: 392 v=0 o=root 1476297067 1476297067 IN IP4 209.189.239.106 s=call c=IN IP4 209.189.239.106 t=0 0 m=audio 40126 RTP/AVP 18 4 9 3 0 8 101 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:9 g722/16000 a=rtpmap:3 gsm/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - a=setup:actpass 23 headers, 18 lines Using latest request as basis request Sending to 209.189.239.106 : 5060 (non-NAT) Found user '9723048721' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 9 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 209.189.239.106:40126 Found description format g729 Found description format g723 Found description format g722 Found description format gsm Found description format pcmu Found description format pcma Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 10 in extensions list_route: hop: sip:abpusa.com:5060;maddr=209.189.239.106;lr=1 list_route: hop: sip:209.189.239.106:5062;transport=udp;dest=4.13.144.17-5060;to-tag=u2dogw3pvf;lr=1 list_route: hop: sip:[EMAIL PROTECTED]:5060;line=70n796w7 Feb 26 09:10:51 NOTICE[10677]: chan_sip.c:3084 copy_via_headers: Shanon: Used Original Via # 0 Feb 26 09:10:51 NOTICE[10677]: chan_sip.c:3084 copy_via_headers: Shanon: Used Original Via # 1 Feb 26 09:10:51 NOTICE[10677]: chan_sip.c:3084 copy_via_headers: Shanon: Used Original Via # 2 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.189.239.106:5060;branch=z9hG4bK-6f04a3acfdd3002c645fcb7605027073.1 Via: SIP/2.0/UDP 209.189.239.106:5062;branch=z9hG4bK-dc2a770cf56399c4c0eb8ca964813dc9;nat=true Via: SIP/2.0/UDP 192.168.5.102:5060;branch=z9hG4bK-d0t7a9g6nme4;received=4.13.144.17 From: sip:[EMAIL PROTECTED];tag=u2dogw3pvf To: sip:[EMAIL PROTECTED];user=phone;tag=as067bdac9 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED]:5070 Content-Length: 0 to 209.189.239.106:5060 Feb 26 09:10:51 NOTICE[10677]: chan_sip.c:3084 copy_via_headers: Shanon: Used Original Via # 0 Feb 26 09:10:51 NOTICE[10677]: chan_sip.c:3084 copy_via_headers: Shanon: Used Original Via # 1 Feb 26 09:10:51 NOTICE[10677]: chan_sip.c:3084 copy_via_headers: Shanon: Used Original Via # 2 We're at 4.16.23.149 port 10798 Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via:
Re: [Asterisk-Users] T.38 fax summary
On Feb 25, 2005, at 11:41 AM, Steve Underwood wrote: Mark, In the time it took to write all that you could probably have read up enough about T.38 to realise you were talking complete rubbish :-) Gee thanks Steve. And your insight has been absolutely beneficial as well. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.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 phone speaker phone mic cutting out
Title: Message Hi i have an @home box with Uniden UIP200's and the speaker phone works great for recording my voicemail box and recording the digital receptionist menu but when i make calls out the PSTN the mic on the speaker phone acts like it only picks up when my voice is really high, make a noise and get closer to the mic all the sudden it starts working then if i back off it quits and visa-versa, but I know its not a problem with the mic because it works perfect between other phones in my office, just not when i make calls to the PSTN, if i use the handset it's fine too. Kurt Fankhauser WaveLincwww.wavelinc.com114 S. Walnut St.Bucyrus, OH 44820419-562-6405 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.5.0 - Release Date: 2/25/2005 ___ 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] Digium BRI or quad BRI
Hmm don't know about you but I rarely wait more than a couple of hours or answers from junghanns. We use junghanns cards in quite some setups and they work quite fine. On Sat, 26 Feb 2005 13:58:02 +0100, Robert Rozman [EMAIL PROTECTED] wrote: Hi, I had similar questions. I've emailed few questions and got no response in 10 days from junghanns. So I decided to try Beronet cards (they will arrive shortly). I just cannot imagine to have support from someone that is not able to answer few simple technical questions about their cards in one week. HTH, regards, Rob. - Original Message - From: Brett, Gary [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, February 23, 2005 2:49 PM Subject: RE: [Asterisk-Users] Digium BRI or quad BRI Thanks Florian, that's great, is this card (junghanns QuadBRI) really stable with * ? Do you or anybody else have any experiences with this card and also is it ok to run multiple cards in one machine cheers -Original Message- From: Florian Overkamp [mailto:[EMAIL PROTECTED] Sent: 23 February 2005 12:58 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Digium BRI or quad BRI Hi, -Original Message- Hi there, quick question...do digium make any BRI cards (ISDN2) or even better a quad port BRI, maybe im going blind, but I cant see any on their website They don't. If you are in need of a european ISDN2 type, see if http://www.junghanns.net/asterisk/page17.html helps you out. Florian ___ 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 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 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 -- Michal Bielicki http://www.asterisk.com.pl/ ___ 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 + SER
you do not need radius for ser and asterisk to speak to each other. if anything, i would suggest using SER for the endpoint and asterisk for the billing and accounting. -yair On Fri, 25 Feb 2005 23:32:42 -0500, Chris Ford [EMAIL PROTECTED] wrote: I just installed SER last night but if you want it ot talk to Asterisk I found that you should install FREERADIUS Server and RADIUS CLIENT. For it to function properly - Original Message - From: Nitesh Divecha [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, February 25, 2005 8:29 PM Subject: [Asterisk-Users] Asterisk + SER Hello All, Has anyone tried Asterisk with SER.? My main focus is billing and authentication of my endpoints. I want Asterisk to handle all my endpoints and SER to do billing/accounting stuff. Any help will be highly appreciated. Neel ___ 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 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 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: compile asterisk(from CVS HEAD) and got an error
Dear Martijn: Thank you for your tips. It is very important for me such an asterisk newbie. I have recompile the source of asterisk-1.0.5 and compile the asterisk-addon done. Anyway, thank you for your kind. Best Regard Charles On Sat, 26 Feb 2005 16:25:15 +0100, Martijn van Oosterhout [EMAIL PROTECTED] wrote: See bug #3639 http://bugs.digium.com/bug_view_page.php?bug_id=0003639 Nothing's been committed yet though I think... On Sat, Feb 26, 2005 at 10:58:11PM +0800, Charles Wang wrote: Dear ALL: I got an error message lists below. Does anyone have the same problem? How to solve it? Best Regard Charles In file included from config.c:34: include/asterisk/app.h:62: array size missing in `options' make: *** [config.o] Error 1 ___ 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 -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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: FRS over *
On the various technical issues raised here (OK, we posted the rules, we won't discuss the legality anymore) I think there is only one main obstacle to using FRS radios for extensions on *. They are simplex (push-to-talk, release-to-listen). The protocol for dealing with voice-activated-switching (VOX) has been used in ham and public safety simplex autopatches but it's really tricky to get right. It would be much easier if you used GMRS or business band radios programmed to transmit on one frequency and listen on another, that way the base can be set up with one pair of frequencies and the portables on the opposite split. Yes, you need two antennas or a duplexer, but this would be a better way to do it than trying VOX and carrier interruptions to make it work simplex. Some of the cheap FRS radios have surprisingly good RF performance, I wouldn't be too put off by hams telling you they will self-destruct. Yes, they are programmed to time out after a few seconds or a minute of talking, but you will need to make your transmissions brief anyway so you can hear what the other people on the channel are saying. They aren't as fragile as they look. FRS is also narrow band (3 kHz deviation, 11.25 kHz channel bandwidth). I don't think you would be able to get more than about 1200 bps of data over this channel. Garmin does transmit GPS data over GMRS at that rate. The modern (last 10 years or more) commercial Motorola radios don't need a special programmer, just an interface box and PC software (which can be expensive). But only a few of these have the narrow band mode needed for compatibility with FRS (and none are legal for FRS as they are not type-accepted as such). Nearly all are OK for GMRS though. These were very expensive radios though, and while you might get them on eBay for $10, the batteries will be much more than that. Only one brand of cheap GMRS radios that I've seen (Garmin) has the duplex mode that allows use with repeaters and duplex base stations. I think this is essential for successful integration with a phone system. My recommendation would be to use a duplex base station radio on a low power business band channel pair. Any of the low power UHF repeaters would be OK for this (the repeater logic is all handled by app_rpt). You can get a license for the itinerant frequencies that costs just a little more than a GMRS license, and be able to use real portable and mobile radios and real antennas. There are plenty of these available with dtmf pads so you could have full control of your * switch. ___ 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] listening to gsm files
The free utility WavePad for Win32 will play and edit GSM files as well: http://www.nch.com.au/wavepad/ To convert to/from GSM on Win32 you can use DBpowerAMP: http://www.dbpoweramp.com/dmc.htm And for Linux or Win32 you could use Sox of course: http://sox.sourceforge.net/ MATT--- -Original Message- From: Time Bandit [mailto:[EMAIL PROTECTED] Sent: Saturday, February 26, 2005 8:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] listening to gsm files Anybody knows what one should do to listen to GSM files? I know QuickTime can play gsm files. Maybe your users that succeeded had it installed. hth ___ 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 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] Wierd asterisk-perl compilation problem
I am running a fully updated Fedora Core 3 server, and installed a pretty thin system, and have just installed packages as needed. My problem is that I am trying to get asterisk-perl installed, but it keeps segmentation faulting on me. I know a little python but perl baffles me. # perl Makefile.PL Segmentation fault ==Strace ___ 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] Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite....
XLite does not support transfer... You have to buy their XPro -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mateo Meier Sent: Tuesday, February 22, 2005 3:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite Hey Guys Im trying to forward a call from the asterisk mainmenue to my second computer with X-Lite installed.. What I've done so far is this: Installed X-lite @my win PC.. X-Lite configuration: Menu | System Settings | SIP Proxy | default Display Name: mateo01 User Name Authorization User: mateo01 Password: Domain/Realm: 192.168.1.** SIP Proxy: 192.168.1.** 192.168.1.** = IP address of Asterisk and the sip.conf file looks like that: [mateo01] type=friend username=mateo01 callerid=mateo01 1234 host=dynamic secret= disallow=all allow=gsm allow=ulaw allow=alaw context=sip nat=no Now, Im unsure what to do ? whats next ? and what do I type in to extensions.conf instead of the following: exten=2,1,Dial(capi/720:078***) Thx for the help Mateo ___ 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 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] Wierd asterisk-perl compilation problem
On Sat, 2005-02-26 at 23:19 -0600, David Carroll wrote: I am running a fully updated Fedora Core 3 server, and installed a pretty thin system, and have just installed packages as needed. My problem is that I am trying to get asterisk-perl installed, but it keeps segmentation faulting on me. I know a little python but perl baffles me. # perl Makefile.PL Segmentation fault ==Strace Oops, accidentally hit send instead of attach :). I'm attaching the strace and the env outputs to see if that helps someone figure out what I have going on wrong. Asterisk is up and going great, I just can't seem to figure out what package I'm missing, or what is broken by the output. strace Description: Perl program HOSTNAME=www SHELL=/bin/bash TERM=xterm HISTSIZE=1000 OLDPWD=/var/lib/asterisk/agi-bin USER=root LS_COLORS=no=00:fi=00:di=00;34:ln=00;36:pi=40;33:so=00;35:bd=40;33;01:cd=40;33;01:or=01;05;37;41:mi=01;05;37;41:ex=00;32:*.cmd=00;32:*.exe=00;32:*.com=00;32:*.btm=00;32:*.bat=00;32:*.sh=00;32:*.csh=00;32:*.tar=00;31:*.tgz=00;31:*.arj=00;31:*.taz=00;31:*.lzh=00;31:*.zip=00;31:*.z=00;31:*.Z=00;31:*.gz=00;31:*.bz2=00;31:*.bz=00;31:*.tz=00;31:*.rpm=00;31:*.cpio=00;31:*.jpg=00;35:*.gif=00;35:*.bmp=00;35:*.xbm=00;35:*.xpm=00;35:*.png=00;35:*.tif=00;35: JAVA_PATH=/usr/share/java MAIL=/var/spool/mail/root PATH=/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/usr/X11R6/bin:/usr/share/java/bin:/usr/lib/jre/bin:/root/bin INPUTRC=/etc/inputrc PWD=/var/lib/asterisk/agi-bin/asterisk-perl-0.08 JAVA_HOME=/usr/share/java LANG=en_US.UTF-8 SSH_ASKPASS=/usr/libexec/openssh/gnome-ssh-askpass SHLVL=1 HOME=/root LOGNAME=root LESSOPEN=|/usr/bin/lesspipe.sh %s DISPLAY=:0.0 G_BROKEN_FILENAMES=1 XAUTHORITY=/root/.xauth8cO3WV _=/bin/env ___ 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] BRIstuff - synchronization with PSTN?
Hello, for a long time I've been having problems with analog modems and faxes communicationg over ISDN BRI lines. Now I began to suspect that this is due to * being out of sync with the PSTN. I have a quadBRI card with first two ports connected to PSTN, and defined as follows: span=1,1,0,ccs,ami span=2,2,0,ccs,ami This should mean that spans 1 and 2 are used as primary and secondary synchronization sources. However, when I check the spans with zttool it says: Sync source: Internally clocked Does this mean that ISDN interface is not synchronized with the PSTN? If so, why, and how can I correct it? Thanks ___ 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] BRIstuff - synchronization with PSTN?
it's a bug in zttool On Sat, 26 Feb 2005 18:40:19 +, Niksa Baldun [EMAIL PROTECTED] wrote: Hello, for a long time I've been having problems with analog modems and faxes communicationg over ISDN BRI lines. Now I began to suspect that this is due to * being out of sync with the PSTN. I have a quadBRI card with first two ports connected to PSTN, and defined as follows: span=1,1,0,ccs,ami span=2,2,0,ccs,ami This should mean that spans 1 and 2 are used as primary and secondary synchronization sources. However, when I check the spans with zttool it says: Sync source: Internally clocked Does this mean that ISDN interface is not synchronized with the PSTN? If so, why, and how can I correct it? Thanks ___ 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 -- Michal Bielicki http://www.asterisk.com.pl/ ___ 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] Wierd asterisk-perl compilation problem
Hello, A good rule of thumb for heavy perl users is to not use Fedora/RedHat. Or at least not use rpms or the preinstalled perl on the OS. RedHat has done a lot to screw up how perl works in the last several versions and there are a lot of angry perl developers that have just given up on the distro altogether. I suggest using another ditro(I know that's a little drastic, but you'll be better off in the long run) or at least install ActivePerl from ActiveState or download perl source and compile it on your system and use that. I use Slackware now and have no problems with perl or asterisk-perl on stock installs. MATT--- -Original Message- From: David Carroll [mailto:[EMAIL PROTECTED] Sent: Sunday, February 27, 2005 12:25 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Wierd asterisk-perl compilation problem On Sat, 2005-02-26 at 23:19 -0600, David Carroll wrote: I am running a fully updated Fedora Core 3 server, and installed a pretty thin system, and have just installed packages as needed. My problem is that I am trying to get asterisk-perl installed, but it keeps segmentation faulting on me. I know a little python but perl baffles me. # perl Makefile.PL Segmentation fault ==Strace Oops, accidentally hit send instead of attach :). I'm attaching the strace and the env outputs to see if that helps someone figure out what I have going on wrong. Asterisk is up and going great, I just can't seem to figure out what package I'm missing, or what is broken by the output. ___ 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] How to grab CallerId information
I am building a click to dial and CRM type web page and I'm having
trouble with something. I can make everything in the manager api work
as documented, but I can't seem to get a grip on how to tell what the
callerid is of an active call. Example: I know that on phone SIP/101
that there is an active call that originated from the outside. What's
the best way to get the callerid of that call?
I have attempted to put the callerid into the database with DBPut during
the initial call setup, but I don't really know that the call is
active. I can get the last busy and last unanswered callerid using
${DIALSTATUS}, but not the last or current answered. Anyone have any ideas?
Here's what I want to do (not using the Flash Operator Panel). If a
salesrep is on the phone, I want them to click a link on a webpage that
will open up a window with all of the customer information they would
need, based on the callerid of the active call. I already have a really
nice click to dial application and don't want a separate app. I also
don't want to monitor all of the time like the Flash Operator Panel
does. Anyone?
Thanks!
Mark
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Re: [Asterisk-Users] Wierd asterisk-perl compilation problem
On Feb 26, 2005, at 18:52, mattf wrote: A good rule of thumb for heavy perl users is to not use Fedora/RedHat. Or at least not use rpms or the preinstalled perl on the OS. RedHat has done a lot to screw up how perl works in the last several versions and there are a lot of angry perl developers that have just given up on the distro altogether. Funny thing is, this is true for Python as well. No one who cares about the things they run on it should *ever* tie themselves to the distribution's package. Compile your own is the standard recommendation and solution. jens ___ 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] Seting up for afirst time -- can not call
Okay, About the secret, comment out the line. You do have to set the secret in the phone. So when the INVITE is exchanged Asterisk will ask the phone for the secret, no secret, no connection. I don't have a polycom phone so that is about all I can help with. Oh yeah, you need a context [from-sip] [from-sip] exten = 101,1,Dial,(SIP/polycom_sp300_ext101) exten = 102,1,Dial,(SIP/polycom_sp300_ext102) As far as I know when the calls come into asterisk via SIP asterisk checks the [from-sip context] be default. Remember that Asterisk is first a PBX, then a VoIP/SIP Server. SIP is sort of step-child status in Asterisk. Race The Tyrant Vanderdecken. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf Ladyzhenskii Sent: Saturday, February 26, 2005 1:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Seting up for afirst time -- can not call Hi, all I am setting up Asterisk for the first time and have some problems. Setup is very simple -- Astersik box and two Polycom SP300 phones. I will add bells and whistles as I go, at the moment things are very simple. No TFTP servers, so phones run with their default configuration. I set up IP addresses, netmask and gateway IPs manually on the phones. Now, I have read of problems with polycom phones. Here is my sip.conf file: ; SIP configuration file [general] port=5060 bindaddr=0.0.0.0 context=default [polycom_sp300_ext101] type=user host=192.168.1.101 secret=101 context=default [polycom_sp300_ext101] type=peer secret=101 host=192.168.1.101 context=default callerid=Ext 101 [polycom_sp300_ext102] type=user host=192.168.1.102 secret=101 context=default [polycom_sp300_ext102] type=peer secret=102 host=192.168.1.102 context=default callerid=Ext 102 First question is about the secret. Should I set up something on teh phone? Is it phone password (default 456)? Now, I am trying to have some extensions. So I have edited the extensions.conf file and changed the [default] section: [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; ;include = demo exten = 101,1,Dial,(SIP/polycom_sp300_ext101) exten = 102,1,Dial,(SIP/polycom_sp300_ext102) The rest of the file is as is as it came with Asterisk. Now I run 'reload' command as CLI. Is ist all I have to do to be able to call between those two phones? If I try to call from one phone to another, after I enter first two digits '10', I get connecting on phone screen and instant busy tone. Any help is greatly appreciated. Thanks, Rudolf ___ 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 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] Dial out through Broadvoice
Hello all, When I call the Broadvoice number all is good. When I try to call out through DISA on my broadvoice line i get the following: Executing Dial(SIP/147.135.0.129-0815bc60, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 480 Temporarily Not Available back from 147.135.16.128 -- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy == Everyone is busy/congested at this time -- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new stack == Spawn extension (outgoing, 16037862111, 102) exited non-zero on 'SIP/147.135.0.129-0815bc60' Is this as simple as it seems? Broadvoice is circut busy? Can any one think of any other reason I might get this message? Or do I just need to call BroadVoice and complain? I have tried two different proxy's (ip's in /etc/hosts) and get the same error. in extensions.conf: [outgoing] exten = _1NXXNXX, 1, dial(SIP/[EMAIL PROTECTED],30) ; exten = _1NXXNXX, 2, congestion() ; No answer, nothing exten = _1NXXNXX, 102, busy() ; Busy in sip.conf: [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=192.168.123.100; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet register = [EMAIL PROTECTED]:XX:[EMAIL PROTECTED] [broadvoice1] type=friend username=603XXX fromuser=603XXX secret=XX host=proxy.bos.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes [bv-in-1] type=friend host=sip.broadvoice.com context=broadvoice dtmfmode=inband canreinvite=no nat=yes Thank you for any help. John Millican ___ 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] BRIstuff - synchronization with PSTN?
A bug in zttool?? OK, if that is the case, is there a way I can verify that the span is indeed synchronized with PSTN? Michael Bielicki wrote: it's a bug in zttool On Sat, 26 Feb 2005 18:40:19 +, Niksa Baldun [EMAIL PROTECTED] wrote: Hello, for a long time I've been having problems with analog modems and faxes communicationg over ISDN BRI lines. Now I began to suspect that this is due to * being out of sync with the PSTN. I have a quadBRI card with first two ports connected to PSTN, and defined as follows: span=1,1,0,ccs,ami span=2,2,0,ccs,ami This should mean that spans 1 and 2 are used as primary and secondary synchronization sources. However, when I check the spans with zttool it says: Sync source: Internally clocked Does this mean that ISDN interface is not synchronized with the PSTN? If so, why, and how can I correct it? Thanks ___ 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 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] Dial out through Broadvoice
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL PROTECTED]|30) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 Temporarily Not Available back from
147.135.16.128
-- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new stack
== Spawn extension (outgoing, 16037862111, 102) exited non-zero on
'SIP/147.135.0.129-0815bc60'
Is this as simple as it seems? Broadvoice is circut busy? Can any
one think
of any other reason I might get this message? Or do I just need to
call
BroadVoice and complain? I have tried two different proxy's (ip's in
/etc/hosts) and get the same error.
in extensions.conf:
[outgoing]
exten = _1NXXNXX, 1, dial(SIP/${EXTEN}
@proxy.bos.broadvoice.com,30) ;
exten = _1NXXNXX, 2, congestion() ; No answer, nothing
exten = _1NXXNXX, 102, busy() ; Busy
in sip.conf:
[general]
context=default ; Default context for incoming
calls
port=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=192.168.123.100 ; IP address to bind to
(0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
; Note: Asterisk only uses the first
host
; in SRV records
; Disabling DNS SRV lookups disables
the
; ability to place SIP calls based on
domain
; names to some other SIP users on the
Internet
register =
[EMAIL PROTECTED]:XX:[EMAIL PROTECTED]
[broadvoice1]
type=friend
username=603XXX
fromuser=603XXX
secret=XX
host=proxy.bos.broadvoice.com
fromdomain=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=yes
[bv-in-1]
type=friend
host=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
canreinvite=no
nat=yes
Try adding this line to sip:
insecure=very
see if that helps. if not, try a standard registration string instead
of the one broadvoice tells you to use.
Also - make sure you're using the password they sent you in an email -
not the one you used when you signed up on their website.
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[Asterisk-Users] Sangoma Cards
Hello list, I need a few words about the difference between sangoma quad E1 cards w/dsp vs. digium tormenta2 compatible cards. Does * really make use of the dsp's on these boards(sangoma)? How many % CPU do they each need (sangoma vs. digium)? Unfortunatelly i do not have the sangoma cards yet, they're on their way with DHL, but i'm very curious about them. Thanks for your time, Calin. ___ 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] Dial out through Broadvoice
I tried to call you number to see what I would get and you have a verizon
Voice messaging service.
Make sure you have your iax set up right in the Iax.conf and your outbaound
registering string going back out.
I have mine set up that I dial 6 to get out on my broadvoice line and 9 to
get out on my voice pulse line.
- Original Message -
From: Your Name [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com;
asterisk-users@lists.digium.com
Sent: Saturday, February 26, 2005 4:13 PM
Subject: Re: [Asterisk-Users] Dial out through Broadvoice
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL PROTECTED]|30) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 Temporarily Not Available back from
147.135.16.128
-- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new stack
== Spawn extension (outgoing, 16037862111, 102) exited non-zero on
'SIP/147.135.0.129-0815bc60'
Is this as simple as it seems? Broadvoice is circut busy? Can any
one think
of any other reason I might get this message? Or do I just need to
call
BroadVoice and complain? I have tried two different proxy's (ip's in
/etc/hosts) and get the same error.
in extensions.conf:
[outgoing]
exten = _1NXXNXX, 1, dial(SIP/${EXTEN}
@proxy.bos.broadvoice.com,30) ;
exten = _1NXXNXX, 2, congestion() ; No answer, nothing
exten = _1NXXNXX, 102, busy() ; Busy
in sip.conf:
[general]
context=default ; Default context for incoming
calls
port=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=192.168.123.100 ; IP address to bind to
(0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
; Note: Asterisk only uses the first
host
; in SRV records
; Disabling DNS SRV lookups disables
the
; ability to place SIP calls based on
domain
; names to some other SIP users on the
Internet
register =
[EMAIL PROTECTED]:XX:[EMAIL PROTECTED]
[broadvoice1]
type=friend
username=603XXX
fromuser=603XXX
secret=XX
host=proxy.bos.broadvoice.com
fromdomain=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=yes
[bv-in-1]
type=friend
host=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
canreinvite=no
nat=yes
Try adding this line to sip:
insecure=very
see if that helps. if not, try a standard registration string instead
of the one broadvoice tells you to use.
Also - make sure you're using the password they sent you in an email -
not the one you used when you signed up on their website.
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Re: [Asterisk-Users] Dial out through Broadvoice
On Saturday February 26 2005 4:30 pm, Chris Ford wrote:
I tried to call you number to see what I would get and you have a verizon
Voice messaging service.
if you called the 6037862111 that is a voicemail number tyhat i was calling to
test knowing it would not be busy and would not bother anyone.
Make sure you have your iax set up right in the Iax.conf and your outbaound
registering string going back out.
I have mine set up that I dial 6 to get out on my broadvoice line and 9 to
get out on my voice pulse line.
I am not using IAX at all. Did not think broadvoice supported it, am I wrong?
More Comments at BOTTOM
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL PROTECTED]|30) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 Temporarily Not Available back from
147.135.16.128
-- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new stack
== Spawn extension (outgoing, 16037862111, 102) exited non-zero on
'SIP/147.135.0.129-0815bc60'
Is this as simple as it seems? Broadvoice is circut busy? Can any
one think
of any other reason I might get this message? Or do I just need to
call
BroadVoice and complain? I have tried two different proxy's (ip's in
/etc/hosts) and get the same error.
in extensions.conf:
[outgoing]
exten = _1NXXNXX, 1, dial(SIP/${EXTEN}
@proxy.bos.broadvoice.com,30) ;
exten = _1NXXNXX, 2, congestion() ; No answer, nothing
exten = _1NXXNXX, 102, busy() ; Busy
in sip.conf:
[general]
context=default ; Default context for incoming
calls
port=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=192.168.123.100 ; IP address to bind to
(0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
; Note: Asterisk only uses the first
host
; in SRV records
; Disabling DNS SRV lookups disables
the
; ability to place SIP calls based on
domain
; names to some other SIP users on the
Internet
register =
[EMAIL PROTECTED]:XX:[EMAIL PROTECTED]
[broadvoice1]
type=friend
username=603XXX
fromuser=603XXX
secret=XX
host=proxy.bos.broadvoice.com
fromdomain=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=yes
[bv-in-1]
type=friend
host=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
canreinvite=no
nat=yes
Try adding this line to sip:
insecure=very
Added insecure=very and same message
see if that helps. if not, try a standard registration string instead
of the one broadvoice tells you to use.
Also - make sure you're using the password they sent you in an email -
not the one you used when you signed up on their website.
Registration seems to work and shows as registered when i run sip show
registry.
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Re: [Asterisk-Users] MusicOnHold
I'll give that a try. Do you make a symbolic link to Madplay as mpg123 or is there a way to configure * to use a different executable? An issue I have with MOH at all is that if I'm on a conference call, I don't want MOH to play. MARK. Ken Godee wrote: MF Hulber wrote: I'm looking for a simple way to disable MusicOnHold in my environment. I'm not really interested in having it and it causes too many problems with hanging mpg123 processes and memory management errors. The problem is, so many other modules seem to depend on it. I can't just cause a noload of MusicOnHold and be done. Does anyone have a simple solution? A solution that doesn't require a recompile is preferred but I'll appreciate and listen to any. After having the same issues you're having, we installed and now use Madplay. Been about 3 weeks and have not had a single issue with moh since. We where averaging several problems a week. http://www.underbit.com/products/mad/ musiconhold.conf [classes] default = custom:/var/lib/asterisk/mohmp3/,/usr/local/bin/madplay -Q --attenuate=-5 --mono -R 8000 --output=RAW:- rock1 = custom:/var/lib/asterisk/rock1/,/usr/local/bin/madplay -Q -z --attenuate=-5 --fade-in --mono -R 8000 --output=RAW:- ___ 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 Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: MF Hulber wrote: I'm looking for a simple way to disable MusicOnHold in my environment. I'm not really interested in having it and it causes too many problems with hanging mpg123 processes and memory management errors. The problem is, so many other modules seem to depend on it. I can't just cause a noload of MusicOnHold and be done. Does anyone have a simple solution? A solution that doesn't require a recompile is preferred but I'll appreciate and listen to any. [...] Content analysis details: (0.6 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO 0.5 URIBL_WS_SURBL Contains an URL listed in the WS SURBL blocklist [URIs: underbit.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] MusicOnHold
Sorry, in my haste I didn't read your musiconhold.conf that answers my question about setting up the executable. MARK. Ken Godee wrote: MF Hulber wrote: I'm looking for a simple way to disable MusicOnHold in my environment. I'm not really interested in having it and it causes too many problems with hanging mpg123 processes and memory management errors. The problem is, so many other modules seem to depend on it. I can't just cause a noload of MusicOnHold and be done. Does anyone have a simple solution? A solution that doesn't require a recompile is preferred but I'll appreciate and listen to any. After having the same issues you're having, we installed and now use Madplay. Been about 3 weeks and have not had a single issue with moh since. We where averaging several problems a week. http://www.underbit.com/products/mad/ musiconhold.conf [classes] default = custom:/var/lib/asterisk/mohmp3/,/usr/local/bin/madplay -Q --attenuate=-5 --mono -R 8000 --output=RAW:- rock1 = custom:/var/lib/asterisk/rock1/,/usr/local/bin/madplay -Q -z --attenuate=-5 --fade-in --mono -R 8000 --output=RAW:- ___ 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 Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: MF Hulber wrote: I'm looking for a simple way to disable MusicOnHold in my environment. I'm not really interested in having it and it causes too many problems with hanging mpg123 processes and memory management errors. The problem is, so many other modules seem to depend on it. I can't just cause a noload of MusicOnHold and be done. Does anyone have a simple solution? A solution that doesn't require a recompile is preferred but I'll appreciate and listen to any. [...] Content analysis details: (0.6 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO 0.5 URIBL_WS_SURBL Contains an URL listed in the WS SURBL blocklist [URIs: underbit.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] HELP NEEDED ASTERISK AND MEDIATRIX 1102
Do yourself a favor and get a Sipura SPA-2100 - much easier to configure and the quality is better than the Mediatrix unit. This was true with the earlier SIP and H323 software, however if you get their latest (11.70) it seems to be one of the better IADS out there. I do agree with you though configuration could go smoother. ___ 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' displays on 2nd line (CID Number Line) on Cisco 79x0 phones
I have found that I can make the phones display any one word on this
second line by adding a fromuser=word in sip.conf. This really isn't
good enough though. When you look at the received calls or missed calls
directory, each item has two lines, the first is the CID name, and the
2nd is supposed to be the CID number. However, if it is asterisk, or
some other word, when you hit the Dial softkey, it fails trying to
dial. If you hit the Edit softkey, the name is correct, and the Phone
field shows asterisk, or whatever it was changed to by the fromuser
setting.
I am just curious if this is something anyone has thought about. Is
there anyway there could be a specific command to change SIP header when
a Dial command is sent to a 79x0 phone to reflect the CID number in the
From: command. I don't know if the cisco phones are the only ones that
have this issue.
Also, I read a message on this list from September that asked this same
basic question. The answer given was to make sure the CID is correct.
I have tried many combinations of manually setting the CIDname, CIDnum,
and just CID right before my dial command, and nothing I have tried has
made any difference. I do have this simple logic which at least
displays the number if a name isn't there:
exten = s,1,answer
exten = s,2,LookupCIDName
exten = s,3,GotoIf($[${CALLERIDNAME} != ${CALLERIDNUM}]?5)
exten = s,4,SetCIDName(${CALLERIDNUM})
exten = s,5,Dial(SIP/{ARG1},18)
exten = s,6,Voicemail(su${ARG1})
Any help in this matter would be appreciated.
Albert Chaffman
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Re: [Asterisk-Users] Sangoma Cards
There are no DSP's on the sangoma cards, who gave you that idea ? The nice thing about those cards are: selective echo cancellation per span around 25% less interrupts created so less load auto select on 3.3V/5V and some other engineering details plus the warranty of 3 years. But no DSP's. cheers Michael On Sat, 26 Feb 2005 23:28:38 +0200, Calin Serbanescu [EMAIL PROTECTED] wrote: Hello list, I need a few words about the difference between sangoma quad E1 cards w/dsp vs. digium tormenta2 compatible cards. Does * really make use of the dsp's on these boards(sangoma)? How many % CPU do they each need (sangoma vs. digium)? Unfortunatelly i do not have the sangoma cards yet, they're on their way with DHL, but i'm very curious about them. Thanks for your time, Calin. ___ 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 -- Michal Bielicki http://www.asterisk.com.pl/ ___ 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] Sangoma Cards
From looking at the description, it seems that the Sangoma card (at least the quad version) *may* have a more robust hardware buffering mechanism than the TE4xxP series. If so, this might help solve some of the load-related issues that my customers have experienced in very large systems. Hope Digium takes note and makes their own improvements! I'm a loyal Digium customer and reseller and would like to stay that way... Cheers Scott Michael Bielicki wrote: There are no DSP's on the sangoma cards, who gave you that idea ? The nice thing about those cards are: selective echo cancellation per span around 25% less interrupts created so less load auto select on 3.3V/5V and some other engineering details plus the warranty of 3 years. But no DSP's. cheers Michael On Sat, 26 Feb 2005 23:28:38 +0200, Calin Serbanescu [EMAIL PROTECTED] wrote: Hello list, I need a few words about the difference between sangoma quad E1 cards w/dsp vs. digium tormenta2 compatible cards. Does * really make use of the dsp's on these boards(sangoma)? How many % CPU do they each need (sangoma vs. digium)? Unfortunatelly i do not have the sangoma cards yet, they're on their way with DHL, but i'm very curious about them. Thanks for your time, Calin. ___ 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 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] Wierd asterisk-perl compilation problem
On Sat, Feb 26, 2005 at 11:24:33PM -0600, David Carroll wrote: On Sat, 2005-02-26 at 23:19 -0600, David Carroll wrote: I am running a fully updated Fedora Core 3 server, and installed a pretty thin system, and have just installed packages as needed. My problem is that I am trying to get asterisk-perl installed, but it keeps segmentation faulting on me. I know a little python but perl baffles me. # perl Makefile.PL Segmentation fault ==Strace Oops, accidentally hit send instead of attach :). I'm attaching the strace and the env outputs to see if that helps someone figure out what I have going on wrong. Asterisk is up and going great, I just can't seem to figure out what package I'm missing, or what is broken by the output. USER=root PATH=/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/usr/X11R6/bin:/usr/share/java/bin:/usr/lib/jre/bin:/root/bin There are many directories on your PATH. Strangely, though, there is no matching LD_LIBRARY_PATH . What are the directories in /etc/ld.so.conf ? Anyway, I'd try to pack this package using cpanflute (is it still in /usr/lib/rpm ?) to get an rpm of that perl module. This is something that could be run as a normal user and does not require running that code as root. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ 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: Asterisk in front of Toshiba CTX
-Original Message-
Message: 8
Date: Fri, 25 Feb 2005 15:13:26 -0700
From: Daniel Burget [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk in front of Toshiba CTX
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
I have googled, and wiki'ed until blue. Is it possible to put
T1---*Toshiba CTX ? I have a TE405P, with one interface programmed
for the T1, I am not sure how to program the 2nd port to mimick the T1
to the Toshiba. The Zapata.conf
[channels]
switchtype=national
context=from-pstn
signalling=pri_cpe
usecallerid=asreceived
echocancel=yes
echocancelwhenbridged=no
echotraining=400
overlapdial=yes
immediate=no
group=0
channel = 1-23
-add the 2nd span to your config changing only the following parameters
-context=from-trashiba ;this is optional, you could use the same
context and sort things out in the dialplan but this makes it cleaner.
-signalling=pri_net
-group=1
-channel= 25-47
Zaptel.conf
bchan=1-23
dchan=24
span=1,0,0,esf,b8zs
-bchan=25-47
-dchan=48
-span=2,0,0,esf,b8zs
-your extensions.conf will have the context including the extensions that *
serves and at the bottom of that context use an include statement pointing
to another context like:
[trashiba_extensions]
Exten= _5XXX,Dial(Zap/G1/${EXTEN}) ;assuming 5XXX extensions
numbers on the trashiba
This works for the T1 into *, or Into Toshiba. I want the calls to go
into *, if they don't match exten.conf to go to the Toshiba. If the
Toshiba dials out, It goes into * and out via sip. So, the T1 will be
for internal DID only.
Is this possible, or am I chasing a dream?
-you are working with *. It is a dream. It is a good dream. And the
answer is always yes it is possible.
-try that, it should work.
Jason Kawakami
www.optellabs.com
Salt Lake City, UT
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RE: [Asterisk-Users] Wierd asterisk-perl compilation problem
I had read some folks complaining about perl on redhat so I suspected something like that. I'm a distro pragmatist so switching to debian/slackware/mandrake wouldn't be an issue, but I have this box set up pretty well so I'll just try and do perl from scratch and see if that fixes it. On Sat, 2005-02-26 at 12:52 -0500, mattf wrote: Hello, A good rule of thumb for heavy perl users is to not use Fedora/RedHat. Or at least not use rpms or the preinstalled perl on the OS. RedHat has done a lot to screw up how perl works in the last several versions and there are a lot of angry perl developers that have just given up on the distro altogether. I suggest using another ditro(I know that's a little drastic, but you'll be better off in the long run) or at least install ActivePerl from ActiveState or download perl source and compile it on your system and use that. I use Slackware now and have no problems with perl or asterisk-perl on stock installs. MATT--- -Original Message- From: David Carroll [mailto:[EMAIL PROTECTED] Sent: Sunday, February 27, 2005 12:25 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Wierd asterisk-perl compilation problem On Sat, 2005-02-26 at 23:19 -0600, David Carroll wrote: I am running a fully updated Fedora Core 3 server, and installed a pretty thin system, and have just installed packages as needed. My problem is that I am trying to get asterisk-perl installed, but it keeps segmentation faulting on me. I know a little python but perl baffles me. # perl Makefile.PL Segmentation fault ==Strace Oops, accidentally hit send instead of attach :). I'm attaching the strace and the env outputs to see if that helps someone figure out what I have going on wrong. Asterisk is up and going great, I just can't seem to figure out what package I'm missing, or what is broken by the output. ___ 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] FRS *: an actual business use
Rich Adamson wrote: I've noticed a growing number of stores using FRS radios. It would make sense to interface (via soundcard/console driver, with the nessacary electrical conversion) a VOX FRS radio to asterisk to allow someone in the office to page/talk with people on the floor or warehouse. You could throw that call (ie, all the radios) into a meetme conference. Then, you could have people in the office either dial that extension and/or have some of them always in that conference on a speaker phone (muted usually). Unless I'm not understanding your comments, the meetme conference isn't needed (assuming all radios are on the same channel). The radios become the meetme for all practical purposes. (When the base radio transmits, all remote radios listen.) I had intended for the meetme conf to allow more than one /phone/ user to communicate with the radio users at the same time. cheers, glenn ___ 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] call pickup with Sipura-3000
I can not make a call pickup to work with Sipura-3000. I have one SIP phone and one is connected to ATA Sipura-3000 I've in all sip.conf context callgroup=1 pickupgroup=1 in features.conf I've tired: pickupexten = *88 pickupexten = *8 Nothing works. What am I missing? -- #Joseph ___ 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] Wierd asterisk-perl compilation problem
On Sun, 2005-02-27 at 01:50 +0200, Tzafrir Cohen wrote: On Sat, Feb 26, 2005 at 11:24:33PM -0600, David Carroll wrote: On Sat, 2005-02-26 at 23:19 -0600, David Carroll wrote: I am running a fully updated Fedora Core 3 server, and installed a pretty thin system, and have just installed packages as needed. My problem is that I am trying to get asterisk-perl installed, but it keeps segmentation faulting on me. I know a little python but perl baffles me. # perl Makefile.PL Segmentation fault ==Strace Oops, accidentally hit send instead of attach :). I'm attaching the strace and the env outputs to see if that helps someone figure out what I have going on wrong. Asterisk is up and going great, I just can't seem to figure out what package I'm missing, or what is broken by the output. USER=root PATH=/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/usr/X11R6/bin:/usr/share/java/bin:/usr/lib/jre/bin:/root/bin There are many directories on your PATH. Strangely, though, there is no matching LD_LIBRARY_PATH . What are the directories in /etc/ld.so.conf ? Anyway, I'd try to pack this package using cpanflute (is it still in /usr/lib/rpm ?) to get an rpm of that perl module. This is something that could be run as a normal user and does not require running that code as root. /usr/X11R6/lib /usr/lib/mysql ___ 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] snom 190 funtion buttons
Hi All Does anyone know the corrrect procedure of how to monitor the pstn line status using the funtion buttons on snom 190 phones. I am using a tdm400p card with 4 fxo modules. I have tried using the hit command as documented in the wiki without success. Any assistance would be greatly appreciated Thanks Geoffrey Sachs ___ 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] Limit the call recording when pressing *1
I'm testing two options from dial command and can not make them to work.
L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left,
repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional.
The following special variables are optional for limit calls: (pasted
from app_dial.c)
w: Allow the called user to start recording after pressing *1 or what
defined in features.conf (Asterisk v1.0.x)
for example I've tired:
exten = 21,1,Dial(${phone1},20,r,w)
exten = 21,1,Dial(${phone1},20,r,L(5[:4][:1]))
but none is working.
I've *-1.0.5 but I can not find app_dial.c nor features.conf contains
any define recording options.
--
#Joseph
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Re: [Asterisk-Users] Limit the call recording when pressing *1
exten = 21,1,Dial(${phone1},20,r,L(5[:4][:1]))
I think should be:
exten = 21,1,Dial(${phone1},20,r,L(5:4:1))
The [] mean the parameter is optional, but you don't use them when
specifying the values.
--Luki
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Re: [Asterisk-Users] Limit the call recording when pressing *1
On Sat, 2005-02-26 at 22:14 -0800, Luki wrote:
exten = 21,1,Dial(${phone1},20,r,L(5[:4][:1]))
I think should be:
exten = 21,1,Dial(${phone1},20,r,L(5:4:1))
The [] mean the parameter is optional, but you don't use them when
specifying the values.
I've tried that too, doesn't work!
5min. has passed and the call wasn't disconnected nor I hear any warning
to message how many minutes are left.
--
#Joseph
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Re: [Asterisk-Users] call pickup with Sipura-3000
On Sat, 2005-02-26 at 18:22 -0700, Joseph wrote: I can not make a call pickup to work with Sipura-3000. I have one SIP phone and one is connected to ATA Sipura-3000 I've in all sip.conf context callgroup=1 pickupgroup=1 in features.conf I've tired: pickupexten = *88 pickupexten = *8 Nothing works. What am I missing? I found it! It can be solved by defining: pickupexten = 33 ;any unique number or in Line 1 dia plan (xx.|*xx) ;this permits passing *8 through Line1 -- #Joseph ___ 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] Limit the call recording when pressing *1
Joseph wrote:
On Sat, 2005-02-26 at 22:14 -0800, Luki wrote:
exten = 21,1,Dial(${phone1},20,r,L(5[:4][:1]))
I think should be:
exten = 21,1,Dial(${phone1},20,r,L(5:4:1))
The [] mean the parameter is optional, but you don't use them when
specifying the values.
I've tried that too, doesn't work!
5min. has passed and the call wasn't disconnected nor I hear any warning
to message how many minutes are left.
Options should be in a single section so that all options are
delimited by a single comma. Thus, try:
exten = 21,1,Dial(${PHONE1},20,rwL(5:4:1))
Kris
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tel;fax:+1 607 272 8634
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RE: [Asterisk-Users] SIP NOTIFY in stable branch?
I didn't realize that the stable branch was never added to... So it will NEVER have any more features than it currently has??? Clay -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Thursday, February 24, 2005 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP NOTIFY in stable branch? Clay Reiche wrote: Does anyone know when the SIP NOTIFY feature from the CLI will be part of the stable branch? Is there any way I can install just that HEAD feature? Yes, I know when. Never :-) Stable means stable, no new features will be added. It's not a difficult feature to backport, but if you haven't worked inside chan_sip before it could take you quite some time. ___ 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 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] Transfer not working
I have all my dial statements with tr at the end but everytime I try to use # as per defined on the features.conf file or any other function like *1 for monitor, nothing works... anybody had the same problem? Thx! ___ 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 + SER
Yes, I use this method too. On Sat, 26 Feb 2005 18:18:15 +0200, Yair Hakak [EMAIL PROTECTED] wrote: you do not need radius for ser and asterisk to speak to each other. if anything, i would suggest using SER for the endpoint and asterisk for the billing and accounting. -yair On Fri, 25 Feb 2005 23:32:42 -0500, Chris Ford [EMAIL PROTECTED] wrote: I just installed SER last night but if you want it ot talk to Asterisk I found that you should install FREERADIUS Server and RADIUS CLIENT. For it to function properly - Original Message - From: Nitesh Divecha [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, February 25, 2005 8:29 PM Subject: [Asterisk-Users] Asterisk + SER Hello All, Has anyone tried Asterisk with SER.? My main focus is billing and authentication of my endpoints. I want Asterisk to handle all my endpoints and SER to do billing/accounting stuff. Any help will be highly appreciated. Neel ___ 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 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 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 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
