RE: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 20:41 + schrieb David Bath:
Hi,
Thanks for quick response.
I changed it as you suggested, but it has the same effect:
In the console I get:
--Executing
Set(SIP/604625-b79140a8,CALLERID(number)=44${CALLERID(number)}) in
new stack
It's running the IF code correctly, but in the true it's just not
evaluating the variable...
Well, perhaps the IF hinders evaluation from happening?
It is by far not as elegant, but you could try
exten=123456,1,GotoIf($[${REGEX(^0..)} = 1]?2:3)
exten=123456,2,Set(CALLERID(num)=44${CALLERID(num):1})
exten=123456,3,ContinueYourDialplanHere
Btw. it should be CALLERID(num), not CALLERID(number), right?
BR
Anselm
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Re: [asterisk-users] zaptel compilation problems with linux 2.6.19
On Fri, Dec 01, 2006 at 10:55:24AM +0200, Roman Yeryomin wrote: On Thursday 30 November 2006 21:49, Tzafrir Cohen wrote:: On Thu, Nov 30, 2006 at 07:19:14PM +0200, Roman Yeryomin wrote: Hello! I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 -- all give the same error) with 2.6.19 kernel CC [M] /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o In file included from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:26, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xdefs.h:93: error: conflicting types for 'bool' include/linux/types.h:36: error: previous declaration of 'bool' was here In file included from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zconfig.h:9, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zaptel.h:34, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xproto.h:29, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:27, from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: include/linux/config.h:10:3: warning: no newline at end of file make[3]: *** [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o] Error 1 make[2]: *** [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp] Error 2 make[1]: *** [_module_/home/roman/install/asterisk/zaptel-1.4.0-beta2] Error 2 make[1]: Leaving directory `/home/roman/install/kernel/linux-2.6.19' make: *** [linux26] Error 2 seems that commenting out typedef int bool; in xpp/xdefs.h on line 93 works that out, but don't know if it's completely right thing to do Simply replacing that int with a _Bool will give several incompatible pointer type warnings. The following is from our internal working copy, with pathes removed for clarity: CC [M] xpp/card_fxo.o xpp/card_fxo.c: In function `__check_report_battery': xpp/card_fxo.c:38: warning: return from incompatible pointer type CC [M] xpp/card_fxs.o xpp/card_fxs.c: In function `__check_poll_digital_inputs': xpp/card_fxs.c:37: warning: return from incompatible pointer type CC [M] xpp/xbus-core.o CC [M] xpp/xpp_zap.o xpp/xpp_zap.c: In function `__check_zap_autoreg': xpp/xpp_zap.c:67: warning: return from incompatible pointer type xpp/xpp_zap.c: In function `__check_prefmaster': xpp/xpp_zap.c:68: warning: return from incompatible pointer type xpp/xpp_zap.c: In function `__check_xpp_ec': xpp/xpp_zap.c:70: warning: return from incompatible pointer type xpp/xpp_zap.c: In function `xpd_read_proc': xpp/xpp_zap.c:437: warning: unused variable `chans' xpp/xpp_zap.c: In function `proc_sync_write': xpp/xpp_zap.c:748: warning: int format, bool arg (arg 5) xpp/xpp_zap.c: In function `proc_xpd_ztregister_write': xpp/xpp_zap.c:816: warning: int format, bool arg (arg 3) Most of them seem to be related to the procfs interface. If you don't need xpp for yourself and can leave with those warnings, go ahead. I'll try to resolve them. hmm... make install also gives an error install -D -m 644 zaptel.h /usr/include/linux/zaptel.h install -D -m 644 torisa.h /usr/include/linux/torisa.h install -D -m 644 tonezone.h /usr/include/tonezone.h install -m 644 doc/ztcfg.8 /usr/share/man/man8 install -m 644 doc/zttool.8 /usr/share/man/man8 [ `id -u` = 0 ] /sbin/depmod -a 2.6.19 || : [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf build_tools/genmodconf linux26 tor2 torisa wcusb wcfxo wctdm wctdm24xxp ztdynamic ztd-eth wct1xxp wcte11xp pciradio ztd-loc ztdummy [: 66: ==: unexpected operator [: 66: ==: unexpected operator Unknown kernel build version requested... exiting. make: *** [install] Error 1 This is because of using non-standard '==' in the shell's test ([). Fixed in the SVN. As a quick-fix: sed -i -e 's/==/=/g' build_tools/genmodconf (replace every '==' with a single '=' in the genmod script) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Siemens Gigaset C450 IP vs S450 IP
Am Mittwoch, den 29.11.2006, 20:48 -0500 schrieb Andrew Joakimsen: Does anyone know where to source the Siemens Gigaset phones in North America? I called 1-800-SIEMENS and was told the Gigaset range is no longer marketed here since a few years ago. How far from being FCC compliant is the DECT standard? Probably ages, in DECT's European incarnation (there seem to be other DECTs defined, but Siemens AFAIK only produces DECTs for Europe). The reason is that DECT uses frequency ranges (~1900 MHz) that are also in use with the GSM1900 network now increasingly popular in the US. This frequency range was not in use before GSM, so with the introduction of T-Mobile(*) etc, using DECT in US had become a bad idea. Europe, and most parts of the rest of the world, use GSM900/GSM1800 which do not interfere with DECT. There is information about these frequencies etc in the wikipedia, for example at http://de.wikipedia.org/wiki/GSM#Verwendete_Frequenzen http://de.wikipedia.org/wiki/DECT#Funk.C3.BCbertragung (in German, the English site probably contains similar info) (*) The fact that T-Mobile is of German origin seems not to be well-known in US. I can actually see their HQ when I step out of my house, on the other side of the river Rhein; we have some T-M billing system database admin people in your local LUG :) BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel compilation problems with linux 2.6.19
Hi On Fri, Dec 01, 2006 at 01:43:20AM -0500, Matthew Rubenstein wrote: On Thu, 2006-11-30 at 17:56 -0700, [EMAIL PROTECTED] wrote: Message: 18 Date: Fri, 1 Dec 2006 00:56:10 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] zaptel compilation problems with linux 2.6.19 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Thu, Nov 30, 2006 at 02:44:03PM -0500, Matthew Rubenstein wrote: I'm having problems installing ztdummy on my CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP only to PSTN). I unpacked the kernel sources and headers in a directory, made (but not re/installed) the kernel, unpacked the zaptel-1.2.11 tarball, then went thru the make sequence. It seemed to proceed OK (without errors, just some warnings), but didn't seem to result in a loadable ztdummy kernel module. Complete (failed) install session transcript is attached to this message; details appended: - # cd path-to-zaptel-1.2.11-source # export KSRC=path-to-kernel-source-root-dir # make clean # make config [... series of shell script conditionals apparently executed OK ...] # make linux26 [... series of CC/LD reports, some warnings, no errors ...] # make install [... series of INSTALL messages, same warnings from (make linux26), no errors ...] # modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy # modprobe zaptel FATAL: Module zaptel not found. - (make linux26) generated some warnings about various usb_*_dev symbols undefined in [xpp,wcusb]/*.ko, but no actual errors. (make install) Those are harmless, IIRC. I'll try to recall their source. I suspected as such. But I don't think the server has full USB/UHCI support running, or fully installed: - # lsmod Module Size Used by binfmt_misc12168 1 dm_mod 59512 0 thermal13864 0 processor 25284 1 thermal fan 4772 0 floppy 63172 0 generic 4836 0 [permanent] ide_generic 1504 0 [permanent] # modprobe usb_uhci FATAL: Module uhci_hcd not found. # modprobe uhci FATAL: Module uhci_hcd not found. - repeated those warnings. (modprobe ztdummy) finished with Was depmod run? No, but trying it now (after the transcripted session) didn't seem to help: - # depmod # modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy - uname -r # uname -r 2.6.16-rc6-060427a so depmod, modprobe and such will look under /lib/modules/2.6.16-rc6-060427a , but the modules were installed elsewhere: ls -l /lib/modules/2.6*/misc/*.ko # ls -l /lib/modules/2.6*/misc/*.ko -rw-r--r-- 1 root root 198617 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/pciradio.ko -rw-r--r-- 1 root root 195365 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/tor2.ko -rw-r--r-- 1 root root 122139 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/torisa.ko -rw-r--r-- 1 root root 114623 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wcfxo.ko -rw-r--r-- 1 root root 164626 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wct1xxp.ko -rw-r--r-- 1 root root 340812 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wctdm24xxp.ko -rw-r--r-- 1 root root 215930 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wctdm.ko -rw-r--r-- 1 root root 204323 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wcte11xp.ko -rw-r--r-- 1 root root 155909 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wcusb.ko -rw-r--r-- 1 root root 343208 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/zaptel.ko -rw-r--r-- 1 root root 106184 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/ztd-eth.ko -rw-r--r-- 1 root root 92153 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/ztd-loc.ko -rw-r--r-- 1 root root 72401 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/ztdummy.ko -rw-r--r-- 1 root root 98511 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/ztdynamic.ko One way to fix this is to move the modules, or pass the kernel vesiosion explicitly to make with KVERS . However this raises the question: does the kernel source tree you used to build the module matches the running kernel version. Is it a kernel you have built? Is there a link /lib/modules/2.6.16-060427a/build ?
Re: [asterisk-users] Re: sip address in voicemail emails
Am Freitag, den 01.12.2006, 17:57 -0500 schrieb Mark Price: Hi, On 12/1/06, Mark Price [EMAIL PROTECTED] wrote: hi, I am using asterisk 1.2.10. I am trying to send sip links in asterisk voicemail, so that users can easily reply to emails. This does not seem to be straightforward. First, there seems to be no variable that prints out the domain name of the sip call, since I am including every variable mentioned on http://www.voip-info.org/wiki-Asterisk +config+voicemail.conf You might use your own notification script, s.t. you can send a Content-type: text/html for your E-Mail. To clarify to myself, I made a call from a different sip domain from a username that does not exist on the asterisk box, and found out that it is true: VM_CIDNUM contains the username, but not the domain name of the call. Therefore, as long as the username is a telephone number, we can work around that, but the message printed to describe a non-telephone-number phone call will be incorrect. Hi Mark, I guess there are incoming lines that send callerid rather reliably - PSTN will most probably send the correct number, if it sends one. OTOH, if you accept incoming SIP connections from unauthenticated hosts, they could basically send any from phone number they wish. Your dialplan should probably take care of that, possibly by prepending the callerid with some prefix (X_ or whatever). Your notification script could make use of that information to _not_ provide a callback link in that case. Several programs allow for special protocol handling; which mail client/internet browser you use will determine wether you can configure it to hand off sip: URIs to the proper program (X-Lite, a CTI prog...). BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4beta3 help
On Thu, Nov 30, 2006 at 09:10:59PM -0500, Doug Crompton wrote: I do a ./configure successfully but when I try doing a 'make' I get error 1 - menuselect What am I doing wrong? Please post a complete trace. The real error message should be a bit above the error message from make. BTW: you don't need to explicitly run 'make menuselect'. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Caller ID Rewrite
Hi All,
First, Edwin thanks for the suggestion in the previous email about
Regex. This unfortunately did not work... I believe it was correctly
evaluation the true condition (i.e. I got the same behaviour).
Anselm, thanks! This way does do it. I believe you must be correct -
the variables are not evaluated when they are the true or false part
of an IF function. I wonder if anyone knows if this is a known bug, or
whether it should be perhaps raised?
On the CALLERID(num) vs CALLERID(number) well. There seems to be
quite a lot of conflicting documentation. The upshot is I'm using
CALLERID(number) and CALLERID(name) and they both seem to work fine..
Thanks to all who made suggestions... my nice little rule is working now
:)
Cheers,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: 02 December 2006 09:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 20:41 + schrieb David Bath:
Hi,
Thanks for quick response.
I changed it as you suggested, but it has the same effect:
In the console I get:
--Executing
Set(SIP/604625-b79140a8,CALLERID(number)=44${CALLERID(number)}) in
new stack
It's running the IF code correctly, but in the true it's just not
evaluating the variable...
Well, perhaps the IF hinders evaluation from happening?
It is by far not as elegant, but you could try
exten=123456,1,GotoIf($[${REGEX(^0..)} = 1]?2:3)
exten=123456,2,Set(CALLERID(num)=44${CALLERID(num):1})
exten=123456,3,ContinueYourDialplanHere
Btw. it should be CALLERID(num), not CALLERID(number), right?
BR
Anselm
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[asterisk-users] Help with IAX Trunk
Hi all. I have an IAX trunk between 2 Asterisk servers. Everything is working correctly dialing between the servers as well as through the PSTN (a T1 connected to one of the servers). The second Asterisk server routes all calls to the PSTN via the first server. Calls to local 10-digit, and toll free calls are working properly. My long distance provider requires that a billing code be entered after dialing a long distance call. From the directly attached Asterisk server, these calls work when the user enters their PIN after dialing. From the second server (connected via an IAX trunk), I never get the tone to enter the long distance PIN..all I get is a steady ringtone. Has anyone encountered this or know how to fix it? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com http://www.autodatasolutions.com/ Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IAX Trunk
Dave Morrow wrote: My long distance provider requires that a billing code be entered after dialing a long distance call. From the directly attached Asterisk server, these calls work when the user enters their PIN after dialing. From the second server (connected via an IAX trunk), I never get the tone to enter the long distance PIN..all I get is a steady ringtone. Instead of having the user enter the billing code, maybe you could program it to be sent via the dial plan? Or, is the code different each time? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with IAX Trunk
Unfortunately, the codes are private for the individual. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Saturday, December 02, 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with IAX Trunk Dave Morrow wrote: My long distance provider requires that a billing code be entered after dialing a long distance call. From the directly attached Asterisk server, these calls work when the user enters their PIN after dialing. From the second server (connected via an IAX trunk), I never get the tone to enter the long distance PIN..all I get is a steady ringtone. Instead of having the user enter the billing code, maybe you could program it to be sent via the dial plan? Or, is the code different each time? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IAX Trunk
Dave Morrow wrote: Unfortunately, the codes are private for the individual. Then I would suggest that you prompt the user for that code, before the actual dial. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with IAX Trunk
H.interesting thought. Not sure how to do it though... I found this this morning. I think it might be the answer I seek http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=5303forum=2 David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Saturday, December 02, 2006 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with IAX Trunk Dave Morrow wrote: Unfortunately, the codes are private for the individual. Then I would suggest that you prompt the user for that code, before the actual dial. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: sip address in voicemail emails
Hi, Anselm On 12/2/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Freitag, den 01.12.2006, 17:57 -0500 schrieb Mark Price: Hi, On 12/1/06, Mark Price [EMAIL PROTECTED] wrote: hi, I am using asterisk 1.2.10. I am trying to send sip links in asterisk voicemail, so that users can easily reply to emails. This does not seem to be straightforward. First, there seems to be no variable that prints out the domain name of the sip call, since I am including every variable mentioned on http://www.voip-info.org/wiki-Asterisk +config+voicemail.conf You might use your own notification script, s.t. you can send a Content-type: text/html for your E-Mail. I don't understand how the notification script is useful for this purpose. The voicemail.conf page on voip-info.org that is referenced above says the following: The way it works is basically any time that somebody leaves a voicemail on the system (regardless of mailbox number), the command specified for externnotify is run with the arguments (in this order): context, extension, and number of voicemails in that mailbox In other words, the the documentation says that the externnotify command is not given any information at all regarding the source of the phone call. Thanks, Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RINGNOANSWER on 1.2
Hi, I've been trying to implement this [1] on 1.2.13 and whilst my twiddlings seem to work, I just wanted confirmation that I'm not doing something really stupid which could cause a segfault under certain conditions. My chan_queue.c addition is this one line: ast_queue_log(queue, qe-chan-uniqueid, outgoing-chan-name, RINGNOANSWER, %d, orig); The output in queue_log is of the format 1165076773|asterisk-21332-1165076763.17|ccuk|Local/[EMAIL PROTECTED],1| RINGNOANSWER|1 Is there a way I can just have 'Local/[EMAIL PROTECTED]' without the other stuff after it? [1] http://lists.digium.com/pipermail/asterisk-commits/2006-May/004096.html Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem in Poland
Hello All, I'm having problems connecting Asterisk to Telco in Poland (using E1). The telco guys are saying that the RING message is missing. How can I make Asterisk to send the RING message? Does anyone have any samples of zaptel and zapata for Poland? Best Regards, Alex Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IAX Trunk
On 09:48, Sat 02 Dec 06, Dave Morrow wrote: H.interesting thought. Not sure how to do it though... I found this this morning. I think it might be the answer I seek http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=5303forum=2 Probably yeah. The r option in the dial command will not pass early media but instead generates it's own. I find the r flag for dial and queue the wrong thing to do. In dial it will disable stuff like 'this call will cost you 300 euro a minute and that's something I really wanna hear. In queue() it will kill the periodic announcements. annoying as well. I removed them from everywhere in my extensions.conf and my system is much more usable. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detailed description of problem in Poland
Hi guys, Here is a bit more detailed information of my problem: If I connect Asterisk PBX to the Polish telco via E1, I don't get any red alarms or anything. The line seems to be fine and the inbound calls are also accepted by the Asterisk. However, whenever I try to make an outbound call, the call is either stuck (Asterisk just displays Called g1/482 and then nothing), or I get the following message: -- Called g2/ -- Channel 0/1, span 3 got hangup -- Zap/63-1 is circuit-busy -- Hungup 'Zap/63-1' == Everyone is busy/congested at this time (1:0/1/0) The telco guys say that my request to make an outbound call is missing a RING message. What must be set in the zapata.conf or zaptel.conf to make Asterisk send RING message? Does anyone have any sample zapata.conf or zaptel.conf for connection between Asterisk to Polish telco via E1? Any suggestions will be most appreciated, Alex Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 200+ analog phones connected to FXS modules
JS == Jon Schøpzinsky [EMAIL PROTECTED] writes: JS I would just guess that the PCI bus would be pretty busy, with 3 JS T1 cards. Couldn't that be a problem? Jon A T1 is less than 2Mbps. The PCI bus can just about handle 1Gbps ethernet. That's a LOT of T1's. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Caller ID Rewrite
AMH == Anselm Martin Hoffmeister [EMAIL PROTECTED] writes:
AMH Well, perhaps the IF hinders evaluation from happening? It is
AMH by far not as elegant, but you could try
AMH exten=123456,1,GotoIf($[${REGEX(^0..)} = 1]?2:3)
AMH exten=123456,2,Set(CALLERID(num)=44${CALLERID(num):1})
AMH exten=123456,3,ContinueYourDialplanHere
How about simply:
exten = 123456/0XX,1,Set(CALLERID(num)=44${CALLERID(num):1})
exten = 123456,1,NoOp()
exten = 123456,2,ContinueYourDialplanHere
/Benny
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Re: [asterisk-users] Problem in Poland
Hello All, I'm having problems connecting Asterisk to Telco in Poland (using E1). The telco guys are saying that the RING message is missing. How can I make Asterisk to send the RING message? Does anyone have any samples of zaptel and zapata for Poland? Best Regards, Alex which telco in Poland are you connected to? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem in Poland
If I'm not mistaken (that's how I was told), the inbound calls are managed by Telekomunikacija Polska, and outbound calls are managed by Profuturo. - Original Message From: Bartosz Jozwiak [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 2, 2006 7:06:20 PM Subject: Re: [asterisk-users] Problem in Poland Hello All, I'm having problems connecting Asterisk to Telco in Poland (using E1). The telco guys are saying that the RING message is missing. How can I make Asterisk to send the RING message? Does anyone have any samples of zaptel and zapata for Poland? Best Regards, Alex which telco in Poland are you connected to? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Caller ID Rewrite
Oh... that's an interesting idea Benny. I didn't realize you could use
TO/FROM type syntax in the dialplan...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benny
Amorsen
Sent: 02 December 2006 17:28
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Caller ID Rewrite
AMH == Anselm Martin Hoffmeister [EMAIL PROTECTED]
writes:
AMH Well, perhaps the IF hinders evaluation from happening? It is
AMH by far not as elegant, but you could try
AMH exten=123456,1,GotoIf($[${REGEX(^0..)} = 1]?2:3)
AMH exten=123456,2,Set(CALLERID(num)=44${CALLERID(num):1})
AMH exten=123456,3,ContinueYourDialplanHere
How about simply:
exten = 123456/0XX,1,Set(CALLERID(num)=44${CALLERID(num):1})
exten = 123456,1,NoOp()
exten = 123456,2,ContinueYourDialplanHere
/Benny
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Re: [asterisk-users] Problem in Poland
This is what PRI debug says on problematic call: Protocol Discriminator: Q.931 (8) len=42 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [28 07 75 6e 6b 6e 6f 77 6e] Display (len= 7) [ unknown ] [6c 0a 00 80 41 67 65 6e 74 30 39 39] Calling Number (len=12) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation permitted, user number not screened (0) 'Agent099' ] [70 04 80 34 38 32] Called Number (len= 6) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '482' ] -- Called g1/482 Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: STATUS (125) [08 03 81 e4 6c] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (6) ] Cause data 1: 6c (108) [14 01 06] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Present (6) -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: SETUP ACKNOWLEDGE (13) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Overlap sending, peerstate Overlap Receiving Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' == Spawn extension (from-sip, 1482, 1) exited non-zero on 'SIP/Agent099-0089f4f0' Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: RELEASE COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null - Original Message From: Bartosz Jozwiak [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 2, 2006 7:06:20 PM Subject: Re: [asterisk-users] Problem in Poland Hello All, I'm having problems connecting Asterisk to Telco in Poland (using E1). The telco guys are saying that the RING message is missing. How can I make Asterisk to send the RING message? Does anyone have any samples of zaptel and zapata for Poland? Best Regards, Alex which telco in Poland are you connected to? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Cheap talk? Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.com___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys PAP2t-NA and Asterisk
I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on the pap2's line on connect (holding the dial tone past the initial 9, dropping it with any other key)? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to 1.1.1.14
Hi Scott, I have direct contact with a support person from Grandstream. I will ask him about that and tell you what did he say as soon as possible. Please just wait. Regards Claudemir On 11/30/06, Scott Keagy [EMAIL PROTECTED] wrote: So I've got phones with ancient firmware, and the release notes for 1.1.1.14 say read the previous release notes and first upgrade to 1.1.0.16 The 1.1.0.16 firmware is not available for download from the grandstream website (at least I haven't found it). Any pointers on where to get this intermediate image? I already tried googling to no avail (didn't help that I was using a link with 2000 ms latency). Plus, any overall pointers for making this upgrade process a success would be appreciated. Regards, Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: 200+ analog phones connected to FXS modules
Saturday, December 2, 2006, 6:16:25 PM, Benny Amorsen wrote: JS == Jon Schopzinsky [EMAIL PROTECTED] writes: JS I would just guess that the PCI bus would be pretty busy, with 3 JS T1 cards. Couldn't that be a problem? Jon A T1 is less than 2Mbps. The PCI bus can just about handle 1Gbps ethernet. That's a LOT of T1's. Well, the data bandwidth is only one. The irq is the other, and that is the bottleneck. -- Best regards, Csibra Gergomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Low beep on voicemail
We've had a few people complain that the beep before leaving a voicemail is not loud enough and too short. Does anybody have a recorded beep that they can share, that is a little louder and a little longer? We've had this box in production for 2+ years, so I hate to mess with the gain on the PRI or anything like that because everything else works fine. I know nothing about recording sounds, and I am sure I could spend a few hours and come up with a suitable version, but I thought I'd ask around before I waste my time trying to figure it out. Thanks in advance. Peder ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rxfax or spandsp problems??
Hi! I am having problems with rxfax. When receiving a fax (on a Zap channel from a te110p), I see on the console: Dec 2 18:49:22 WARNING[31532]: channel.c:2341 set_format: Unable to find a codec translation path from unknown to unknown Dec 2 18:49:22 WARNING[31532]: app_rxfax.c:311 rxfax_exec: Unable to restore write format on 'Zap/8-1' extensions.conf is plain simple: fax,1,rxfax(filename.tif) fax,102,Goto(1) any ideas? -- lars pd: Using Asterisk 1.2.10 on debian, with libspandsp0 package. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel compilation problems with linux 2.6.19
On Sat, 2006-12-02 at 09:53 -0700, [EMAIL PROTECTED] wrote: Date: Sat, 2 Dec 2006 11:51:37 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] zaptel compilation problems with linux 2.6.19 To: Asterisk-Users asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi Hi, and thanks for the help :). On Fri, Dec 01, 2006 at 01:43:20AM -0500, Matthew Rubenstein wrote: On Thu, 2006-11-30 at 17:56 -0700, [EMAIL PROTECTED] wrote: Message: 18 Date: Fri, 1 Dec 2006 00:56:10 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] zaptel compilation problems with linux 2.6.19 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Thu, Nov 30, 2006 at 02:44:03PM -0500, Matthew Rubenstein wrote: I'm having problems installing ztdummy on my CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP only to PSTN). I unpacked the kernel sources and headers in a directory, made (but not re/installed) the kernel, unpacked the zaptel-1.2.11 tarball, then went thru the make sequence. It seemed to proceed OK (without errors, just some warnings), but didn't seem to result in a loadable ztdummy kernel module. Complete (failed) install session transcript is attached to this message; details appended: - # cd path-to-zaptel-1.2.11-source # export KSRC=path-to-kernel-source-root-dir # make clean # make config [... series of shell script conditionals apparently executed OK ...] # make linux26 [... series of CC/LD reports, some warnings, no errors ...] # make install [... series of INSTALL messages, same warnings from (make linux26), no errors ...] # modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy # modprobe zaptel FATAL: Module zaptel not found. - (make linux26) generated some warnings about various usb_*_dev symbols undefined in [xpp,wcusb]/*.ko, but no actual errors. (make install) Those are harmless, IIRC. I'll try to recall their source. I suspected as such. But I don't think the server has full USB/UHCI support running, or fully installed: - # lsmod Module Size Used by binfmt_misc12168 1 dm_mod 59512 0 thermal13864 0 processor 25284 1 thermal fan 4772 0 floppy 63172 0 generic 4836 0 [permanent] ide_generic 1504 0 [permanent] # modprobe usb_uhci FATAL: Module uhci_hcd not found. # modprobe uhci FATAL: Module uhci_hcd not found. - repeated those warnings. (modprobe ztdummy) finished with Was depmod run? No, but trying it now (after the transcripted session) didn't seem to help: - # depmod # modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy - uname -r # uname -r 2.6.16-rc6-060427a so depmod, modprobe and such will look under /lib/modules/2.6.16-rc6-060427a , but the modules were installed elsewhere: ls -l /lib/modules/2.6*/misc/*.ko # ls -l /lib/modules/2.6*/misc/*.ko -rw-r--r-- 1 root root 198617 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/pciradio.ko -rw-r--r-- 1 root root 195365 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/tor2.ko -rw-r--r-- 1 root root 122139 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/torisa.ko -rw-r--r-- 1 root root 114623 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wcfxo.ko -rw-r--r-- 1 root root 164626 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wct1xxp.ko -rw-r--r-- 1 root root 340812 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wctdm24xxp.ko -rw-r--r-- 1 root root 215930 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wctdm.ko -rw-r--r-- 1 root root 204323 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wcte11xp.ko -rw-r--r-- 1 root root 155909 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wcusb.ko -rw-r--r-- 1 root root 343208 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/zaptel.ko -rw-r--r-- 1 root root 106184 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/ztd-eth.ko -rw-r--r-- 1 root root 92153 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/ztd-loc.ko -rw-r--r-- 1 root root 72401 Nov 30 09:24
Re: [asterisk-users] Asterisk + Avaya S8700
On 12/1/06, Tomer Horn [EMAIL PROTECTED] wrote: Michel R Vaillancourt wrote: Tomer Horn wrote: Hello list, I am curious here if anybody here got an experience connecting Avaya to Asterisk using H323 / T1. I am completely lack of experience with Avaya and I wanna know if anybody here has connected Avaya to Asterisk using H323 and managed to stabilize it. Google provides mixed comments regarding the matter. The purpose of Asterisk on this matter is to provide outgoing calls from the Avaya through Asterisk, so features such as MWI and stuff are not necessary for me. Thanks, Tomer. I have done it with a Definity G3. It was actually pretty straight forward. Have you done it with H323 or T1/E1 ? I've done it both ways to a G3R and an 8700. The h.323 gateways from T1 on carrier interconnect side to an 8700 via h.323 signaling group are actually pre-1.2 Asterisk (still!) and the folks using it are very happy with it. The only gotcha I would warn you about would be packetization between the Avaya and Asterisk on the RTP side of things. If you don't get that right, you won't get good sound quality on the calls. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium through Octasic
On 11/30/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Thursday 23 November 2006 11:44, Heidi Mendoza wrote: We're looking at using 4 or 8 port T1 cards with echo cancellation and are evaluating brands to go with. We know that Sangoma has excellent solutions especially when it comes to echo. But we still have to hear about actual performance of a Digium card using the same Octasic DSP echo canceller. Excellent performance. I had an A104d which was giving some very odd audio artifacting, Sangoma replaced the card but did not test the original to ensure that the card was indeed defective or that the problem was solved with the replacement. I haven't put the replacement in service yet, as I had a TE407P on order and it arrived first. :-) After dealing with the crap that the TE406P was, the TE407P is *heaven*. Highly recommended. Ditto here as well. The TE412P and TE212P have been rock solid in deployments I've put them in to. Kudos to the Digium folks for getting it right here. They've got a great product that I wouldn't hesitate to recommend with this product line. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hold calling channel and ask called channel beforeconnect???
you can find an example on the wiki here: http://www.voip-info.org/wiki/view/Asterisk+cmd+dial On 12/1/06, Nigel J. Terry [EMAIL PROTECTED] wrote: I posted this a week ago and have had no response. Can someone tell me if I am asking a stupid question, i.e. is the answer either obvious or impossible? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nigel J. Terry Sent: Wednesday, November 22, 2006 10:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hold calling channel and ask called channel beforeconnect??? I am a newbie. Just got my Asterisk working and I love it. I want to do the following, believe it should be possible, but can't work out how: When I get an incoming call, I want to answer and just send ringing to the calling channel. Then I want to call the destination channel, send a message asking if they will accept the call, get a response (1 or 2) and then either connect the parties (1) or send the calling channel to voicemail (2). Any ideas, thanks Nigel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hold calling channel and ask called channel beforeconnect???
Hi Nigel, If I understand your question correctly, you can accomplish what you need in Trixbox/FreePBX by having your calls answered by a queue. When the caller is in this queue, he will hear music on hold until the call is answered by an agent. When the agent answers the call a recorded message can be played ahead of actually connecting the caller. With this feature I can be notified that the call is originating from a certain channel or line. This functionality could probably be modified to report the CLI of the incoming call. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada you can find an example on the wiki here: http://www.voip-info.org/wiki/view/Asterisk+cmd+dial On 12/1/06, Nigel J. Terry [EMAIL PROTECTED] wrote: I posted this a week ago and have had no response. Can someone tell me if I am asking a stupid question, i.e. is the answer either obvious or impossible? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nigel J. Terry Sent: Wednesday, November 22, 2006 10:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hold calling channel and ask called channel beforeconnect??? I am a newbie. Just got my Asterisk working and I love it. I want to do the following, believe it should be possible, but can't work out how: When I get an incoming call, I want to answer and just send ringing to the calling channel. Then I want to call the destination channel, send a message asking if they will accept the call, get a response (1 or 2) and then either connect the parties (1) or send the calling channel to voicemail (2). Any ideas, thanks Nigel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Linksys PAP2t-NA and Asterisk
I am doing this already. I assume you are using a 'batphone' dialplan on the pap2 that places calls on asterisk into the 's' extension. The asterisk feature you want is 'DISA' (Direct Inward System Access - I think). My sip.conf has the pap2 coming into context 'ata_in', so my asterisk dialplan looks like: [ata_in] exten = s,1,Answer exten = s,n,DISA(no-password|internal) [internal] ... my internal extensions here DISA gives the remote end dialtone, optionally after a password. Have a look on the wiki for all it's features. The security stuff mostly only applies if remote SIP connections can access. If anyone knows how to tell DISA to give a different dialtone then I'd love to know! HTH James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jason Michaelson Sent: Sunday, 3 December 2006 06:02 To: Asterisk Users Subject: [asterisk-users] Linksys PAP2t-NA and Asterisk I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on the pap2's line on connect (holding the dial tone past the initial 9, dropping it with any other key)? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Trouble with regexten
Well, I can't pretend to know how other people use it, but perhaps an example of how I use it would be helpful. Most of the sites that I maintain have a pair of boxes that are being loadbalanced (by UltraMonkey: www.ultramonkey.org), so I have no particular way of knowing who is registered to what box beforehand. Obviously, I need to know this. My solution is to use DUNDi and regexten. The DUNDi contexts are mapped into the context where the regextens take place (actually, it's the context where the 2 thru n priorities are, but the regcontext is included) and then I can just do a DUNDILOOKUP to found out the dialing information for any given device. It's simple, it works, and it's a good way to provide redundancy. I belive you may be expecting too much from regexten. It doesn't really do *that* much, but what it does do is useful. Regards, - Brad From: [EMAIL PROTECTED] on behalf of Andrew Joakimsen Sent: Thu 11/30/2006 10:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with regexten Because the REGEXTEN would be the phone number And the Device's userid would be the macaddress, settting regexten should create that association. There used to be an example on the voip-info wiki but its not there anymore. Would someone care to explain what regexten, in its current state, can do that the dialplan can't already do? The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys PAP2t-NA and Asterisk
James Harper wrote: I am doing this already. I assume you are using a 'batphone' dialplan on the pap2 that places calls on asterisk into the 's' extension. In the telephone industry, called a house phone If anyone knows how to tell DISA to give a different dialtone then I'd love to know! HTH James Set in indications.conf, though I believe that changes the Dialtone of an FXS port on the TDM400 as well. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Linksys PAP2t-NA and Asterisk
James Harper wrote: I am doing this already. I assume you are using a 'batphone' dialplan on the pap2 that places calls on asterisk into the 's' extension. In the telephone industry, called a house phone If anyone knows how to tell DISA to give a different dialtone then I'd love to know! Set in indications.conf, though I believe that changes the Dialtone of an FXS port on the TDM400 as well. Yes, sorry, I should have been more specific. Within the same dialplan I'd like to be able to present different dialtones with DISA. Internal dialton, and 'external' dialtone. Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys PAP2t-NA and Asterisk
James Harper wrote: Yes, sorry, I should have been more specific. Within the same dialplan I'd like to be able to present different dialtones with DISA. Internaldialton, and 'external' dialtone. Thanks James I suspect that would involve some digging into the source, given that there is only one dialtone specified. That wouldn't be a bad feature to have, but isn't sexy enough for anyone to want to work it out. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Low beep on voicemail
take a look on Audacity program is opensource and has the option Generate Beep, then just add some Gain as you want... On 12/2/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: We've had a few people complain that the beep before leaving a voicemail is not loud enough and too short. Does anybody have a recorded beep that they can share, that is a little louder and a little longer? We've had this box in production for 2+ years, so I hate to mess with the gain on the PRI or anything like that because everything else works fine. I know nothing about recording sounds, and I am sure I could spend a few hours and come up with a suitable version, but I thought I'd ask around before I waste my time trying to figure it out. Thanks in advance. Peder ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Answering Machine detection in Australia
Hello, Can anyone comment on the success of AMD/NVMachineDetect in an Australian setting? What kind of hit/miss ratio can we expect on a good quality g711 IAX tunk? Does the region even matter? I'm really not sure if these applications are tailored to a US/UK machines and VM services. Regards, Nick. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Linksys PAP2t-NA and Asterisk
Thanks for the help James! Not long after I sent the email I came across other instructions for using DISA exactly how you suggested. Works like a charm! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Harper Sent: Saturday, December 02, 2006 9:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Linksys PAP2t-NA and Asterisk I am doing this already. I assume you are using a 'batphone' dialplan on the pap2 that places calls on asterisk into the 's' extension. The asterisk feature you want is 'DISA' (Direct Inward System Access - I think). My sip.conf has the pap2 coming into context 'ata_in', so my asterisk dialplan looks like: [ata_in] exten = s,1,Answer exten = s,n,DISA(no-password|internal) [internal] ... my internal extensions here DISA gives the remote end dialtone, optionally after a password. Have a look on the wiki for all it's features. The security stuff mostly only applies if remote SIP connections can access. If anyone knows how to tell DISA to give a different dialtone then I'd love to know! HTH James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jason Michaelson Sent: Sunday, 3 December 2006 06:02 To: Asterisk Users Subject: [asterisk-users] Linksys PAP2t-NA and Asterisk I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on the pap2's line on connect (holding the dial tone past the initial 9, dropping it with any other key)? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
