Re: [asterisk-users] Asterisk on Pcengines Alix board
Is the PCI slot large enough for full height, half length PCI boards ? Yes. Has you heard of a PCI Express version ? No but the way chipsets are coming down in price, I would imagine someone will have it soon. John ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Copy or Make + Make Install
Tzafrir Cohen wrote: On Wed, Nov 28, 2007 at 10:47:44AM -0900, Mojo with Horan Company, LLC wrote: You might want the directory structure at /var/lib/asterisk as well, as it contains the current state of the voicemail boxes and any custom sound files that might have been added Voicemail boxes are actually under /var/spool/voicemail . /var/spool/asterisk/voicemail/ Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using existing extensions.conf macros, and co-habitation
Am Freitag, den 30.11.2007, 15:08 -0800 schrieb Philip Prindeville:
bump...
Philip Prindeville wrote:
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no default context) and two trunks coming
into separate contexts, i.e. one for residence and one for my at-home
business.
I noticed, however, that macro-stdexten depends on the default context:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well)
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring
the interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send
to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press
#, return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send
to voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press #,
return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1})
The issue is that I have, per virtual pbx (i.e. home or business), two
contexts
that these get used from. The internal-xyzzy and incoming-xyzzy
contexts (one
for each pbx, ie. xyzzy is home or else it's office).
I was wondering if there wasn't a more flexible solution to this issue, than
hard-coding a Goto(default,s,1) into them (I have no default context,
because it
would be meaningless).
Perhaps using Gosub and Return. Or do I need to hack the macro, and
pass in a
3rd argument (bletch)?
I am not a macro guy, but I see three possible ways of operation:
1. extend the macro to have a third parameter, which would be the
Context the macro is called from (and have
exten = s-BUSY,2,GOTO(${ARG3},s,1)
2. use a global variable for the same purpose
3. Check wether the ${CONTEXT} variable is still set to the calling
context in a Macro (no idea if it is, worth a NOOP, right?)
HTH
Anselm
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Re: [asterisk-users] REFER mesage extraction using SIP_HEADER
Thanks !! I am still confused, so are these functionalities SIPREFERREDBYHDR, SIPREFERTO are included in the 1.4 version ? - Arpit On Dec 1, 2007 1:31 AM, Norman W. Franke [EMAIL PROTECTED] wrote: On Dec 1, 2007, at 12:30 AM, [EMAIL PROTECTED] wrote: I would like to extract the information present in the SIP REFER message that comes to asterisk. Would SIP_HEADER() allow me to do that ? I have used SIP_HEADER() for extracting the to and from SIP headers previously. I wanted to do the exact same thing a while ago. However it is not possible as far as I can tell. I've tried it and verified the headers are being sent, but asterisk can't see them. It can read the headers from the original INVITE. This bug report: http://bugs.digium.com/view.php?id=4934 Complained of the same thing, but was ended as too much work and folks weren't sure it's even correct. Then this one: http://bugs.digium.com/print_bug_page.php?bug_id=8378 Talks about the Refered-By header in REFER messages, which seems to have been folded in to 1.4. It didn't solve the general case of other headers, however. I worked around it in my case, since the original invite actually had what I needed most of the time. Some odd cases just will remain broken. Norman Franke ASD, Inc. www.myasd.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality
Hi Veselin, You can verify SDP and RTP by running protocol analyzer such Ethereal, if you need instruction you can follow this link. http://www2.cs.uh.edu/~jsteach/cosc4377/2000fall/ethereal.html While TDM side, I think he is referring to the card, if the card is faulty or not. Best Regards, Joanna On Dec 1, 2007 9:27 AM, Veselin Kantsev [EMAIL PROTECTED] wrote: Thank you much for the prompt reply Salvatore. Would you have the time to explain further how should I go for verifying that SDP and RTP are OK. Also what is reffered to as the TDM site. Veselin On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote: Take a packet capture of your VoIP segment and verify that the SDP is correct and that the RTP is making it to the correct places. If all that looks good and this is a straight out quality problem, then you need to figure out if it's happening on the voip side or on the TDM side. You should make calls (with captures) VoIP to Voip passing the media through your asterisk and also try routing a tdm call in and back out. If you have the equipment, take a mos score of the TDM loop. Without any of the above, you will not be able to isolate the issue. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Veselin Kantsev Sent: Friday, November 30, 2007 2:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality Hello, I have an Asterisk running with a Sangoma A200 card with Hardware Echo cancelling connected to the UK PSTN. If a PSTN call comes in, voice both ways is OK, however if an outgoing call over the PSTN is made I can hear the other party OK but they can not, they can barely understand what I am saying, my voice is unclear fading and skipping. Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2 are OK too. I've tried gsm/ulaw/alaw codecs so far. Tried disabling the echo cancelling as well. Any suggestions will be greatly appreciated. Regards, Veselin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
Daryl G. Jurbala wrote: How recent? I tried switching from 1.2 to 1.4 about 4 months ago, and asterisk would stop accepting IAX connections in less than a day and would need to be restarted. Just look at the changelogs, there have been lots and lots and lots of commits to the iax channel driver as of late. (and the sip driver). ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Only call me once
Do you mean only once per day? On Dec 1, 2007 7:47 AM, Alex Balashov [EMAIL PROTECTED] wrote: Store a value indicating it has been called as a unique key in AstDB, and set your dial plan to check for it. On Fri, 30 Nov 2007, [EMAIL PROTECTED] wrote: Anyone have an idea how to implement a phone number that can only be called once? The first time it will process normally and any subsequent calls will be rejected. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Consulting/Integration Services Non-US US *u
to all, I am available for work either US or Non-US for * consulting, configuring, integration with other business applications. have been working with * for about three years on and off and would like to do this full time. am available for on-site or remote project work. have 20+ years UNIX/Linux (SuSE, redhat, debian, knoppix, lamppix, slackware, cdrouter, etc) system and application integration experience. have ISP experience, medical answering service experience, full life cycle software design and development in the gov't, financial, and private sectors. have worked with open source products apache, dns, inn, ntp, majordomo, sugarCRM, joomla, and countless others... please contact me off list for specifics and to discuss potential projects. I am in the philadelphia, US area. thanks, daveC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.15 Voicemail
Hi after having installed asterisk 1.4.15 my voicemail does not work anymore. Am I the only one? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Consulting/Integration Services Non-US US *u
On Sat, 1 Dec 2007, dave cantera wrote: [snip] You forgot i don't know what the shift key is and i don't understand what Non-Commercial Discussion means. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cdr_pgsql error in 1.4.15
In Asterisk 1.4.15 if I try to configure cdr_pgsql.conf , asterisk crash with this message asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_pgsql.so: undefined symbol: PQescapeStringConn Is this a knowed error? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Consulting/Integration Services Non-US US *u
steve, oops, you are right... sorry.. wrong list... daveC Steve Edwards wrote: On Sat, 1 Dec 2007, dave cantera wrote: [snip] You forgot "i don't know what the shift key is" and "i don't understand what Non-Commercial Discussion means." Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as non-root/best practices
Robert McNaught wrote: not in path [EMAIL PROTECTED] echo $PATH /usr/kerberos/bin:/usr/lib/courier-imap/bin:/usr/local/bin:/bin:/usr/bin:/usr/X11R6/bin:/home/admin/bin Is /sbin in your path? CP Robert McNaught wrote: my problem is that a non-privileged user, eg admin, cannot log in and connect to the console by issuing the following [EMAIL PROTECTED] asterisk -r bash: asterisk: command not found [EMAIL PROTECTED] whereis asterisk asterisk: /usr/sbin/asterisk /usr/lib/asterisk /usr/include/asterisk /usr/include/asterisk.h /usr/share/man/man8/asterisk.8 what is the best way to solve this problem? i have tried adding admin ALL=(ALL) ALL- I will prune back once I verify I can get this working into visudo, but even that returns asterisk:command not found Does anyone out there know the best way around this - I tried adding in a symbolic link in /usr/bin/asterisk to point to the /home/asterisk/asterisk-bin/sbin/asterisk file, which worked, but is a hack around the problem and don't believe this is the way It seems that non-privileged users cannot run commands in sbin, but can in bin directories Robert ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Then put /usr/sbin in your path, then you will be able to launch it. or you can use alias, which is particularly good if you need to use sudo to launch asterisk, my alias entry reads: alias asterisk='sudo /usr/sbin/asterisk -r' Then if I want to do the -rx thing I can just call asterisk -x because the -r is already present. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using existing extensions.conf macros, and co-habitation
Philip Prindeville wrote: Tilghman Lesher wrote: On Thursday 29 November 2007 13:29:17 Philip Prindeville wrote: [snip] The issue is that I have, per virtual pbx (i.e. home or business), two contexts that these get used from. The internal-xyzzy and incoming-xyzzy contexts (one for each pbx, ie. xyzzy is home or else it's office). I was wondering if there wasn't a more flexible solution to this issue, than hard-coding a Goto(default,s,1) into them (I have no default context, because it would be meaningless). Perhaps using Gosub and Return. Or do I need to hack the macro, and pass in a 3rd argument (bletch)? MacroExit or Gosub/Return would certainly be possibilities. The main thing to note is that this macro that you call standard is actually just an arbitrary example. It is by no means perfect, so feel free to adapt it to your own liking. Sure. I just figured that it would be nice if the canned macros worked out-of-the-box without modification, in the real world. I suppose I could file a bug, and then submit patches for the macro and documentation... -Philip ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The ability to modify the macros to your own needs is not a bug. Anyway try adding a few more args to your stdexten to handle the context name and the like so it doesn't need default. On another point, why would asterisk come with built in code example for a multi-tenant set-up? Please save your self some time and embarrassment by not submitting that particular bug. Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cross-compiling asterisk-1.4 for Debian on a slug
On Wednesday 28 November 2007 23:27:41 myself wrote: [..] dlfcn.c:1225: error: dereferencing pointer to incomplete type dlfcn.c:1225: error: dereferencing pointer to incomplete type dlfcn.c:1225: error: dereferencing pointer to incomplete type dlfcn.c:1225: error: dereferencing pointer to incomplete type dlfcn.c:1225: error: dereferencing pointer to incomplete type dlfcn.c:1225: error: dereferencing pointer to incomplete type dlfcn.c:1230: error: dereferencing pointer to incomplete type dlfcn.c:1231: error: dereferencing pointer to incomplete type dlfcn.c:1234: error: invalid application of `sizeof' to incomplete type `mach_header' dlfcn.c:1235: error: dereferencing pointer to incomplete type dlfcn.c:1235: error: dereferencing pointer to incomplete type dlfcn.c:1237: error: `LC_SYMTAB' undeclared (first use in this function) dlfcn.c:1237: error: dereferencing pointer to incomplete type dlfcn.c:1240: error: dereferencing pointer to incomplete type dlfcn.c:1241: error: dereferencing pointer to incomplete type dlfcn.c:1244: error: dereferencing pointer to incomplete type dlfcn.c:1249: error: dereferencing pointer to incomplete type dlfcn.c:1250: error: dereferencing pointer to incomplete type dlfcn.c:1250: error: `N_PEXT' undeclared (first use in this function) dlfcn.c:1251: error: dereferencing pointer to incomplete type dlfcn.c:1251: error: `N_EXT' undeclared (first use in this function) dlfcn.c:1254: error: increment of pointer to unknown structure dlfcn.c:1254: error: arithmetic on pointer to an incomplete type dlfcn.c:1257: error: dereferencing pointer to incomplete type dlfcn.c:1257: error: dereferencing pointer to incomplete type dlfcn.c:1259: error: dereferencing pointer to incomplete type dlfcn.c:1262: error: increment of pointer to unknown structure dlfcn.c:1262: error: arithmetic on pointer to an incomplete type dlfcn.c:1266: error: dereferencing pointer to incomplete type dlfcn.c:1267: error: dereferencing pointer to incomplete type dlfcn.c: In function `search_linked_libs': dlfcn.c:515: warning: value computed is not used dlfcn.c: In function `dlsymIntern': dlfcn.c:563: warning: unsupported arg to `__builtin_return_address' dlfcn.c: In function `dladdr': dlfcn.c:1198: warning: value computed is not used dlfcn.c:1221: warning: value computed is not used dlfcn.c:1235: warning: value computed is not used dlfcn.c: At top level: dlfcn.c:133: error: storage size of `mainStatus' isn't known dlfcn.c:83: warning: 'NSSymbol' declared `static' but never defined make[1]: *** [dlfcn.o] Error 1 make[1]: Leaving directory `/home/user/down/optware/cs04q3armel/builds/asterisk' make: *** [/home/user/down/optware/cs04q3armel/builds/asterisk/.built] Error 2 [...] gai_strerror.c:61: error: `EAI_IDN_ENCODE' undeclared here (not in a function) gai_strerror.c:61: error: initializer element is not constant gai_strerror.c:61: error: (near initialization for `values[16].code') gai_strerror.c:61: error: initializer element is not constant gai_strerror.c:61: error: (near initialization for `values[16]') make[6]: *** [gai_strerror.lo] Error 1 make[6]: Leaving directory `/home/user/optware/cs04q3armel/builds/gnutls/gl' make[5]: *** [all] Error 2 make[5]: Leaving directory `/home/user/down/optware/cs04q3armel/builds/gnutls/gl' make[4]: *** [all-recursive] Error 1 make[4]: Leaving directory `/home/user/down/optware/cs04q3armel/builds/gnutls' make[3]: *** [all] Error 2 make[3]: Leaving directory `/home/user/down/optware/cs04q3armel/builds/gnutls' make[2]: *** [/home/user/down/optware/cs04q3armel/builds/gnutls/.built] Error 2 make[2]: Leaving directory `/home/user/down/optware/cs04q3armel' make[1]: *** [/home/user/down/optware/cs04q3armel/builds/iksemel/.configured] Error 2 make[1]: Leaving directory `/home/user/down/optware/cs04q3armel' make: *** [/home/user/down/optware/cs04q3armel/builds/asterisk14/.configured] Error 2 Hello? Nobody any idea how to fix that or how to compile asterisk-1.4 for Debian/slug on an i686 machine? Or should i ask on the nslu2 mailing list? Greetings, Fabiano ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Only call me once
[EMAIL PROTECTED] wrote: Anyone have an idea how to implement a phone number that can only be called once? The first time it will process normally and any subsequent calls will be rejected. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yup store the number in DB if its new, if it's already in db reject. This is an incredibly simple thing to do, you can even just use the Asterisk internal DB for simplicity. Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Increasing the voice volume from the diguim cards
Hi List; Anyone knows a method (command) to increase the voice volume at diguim card level? Regards Bilal Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how. http://overview.mail.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime SIP BLF
On Nov 28, 2007, at 8:24 PM, [EMAIL PROTECTED] wrote: From memory - 'rtcachefriends=yes' should do the trick. PaulH Sorry for the late response, wanted to make sure everything else was still working. This did indeed solve the problem. The only side affect I have noticed is that changed I make to the realtime database don't get picked up immediately. Not sure what the cache timeout is but I am able to flush it manually so for the moment I don't care. :) Thanks, Daniel On Wed, 2007-11-28 at 16:56 -0800, Daniel Hazelbaker wrote: I am trying to get the presence/hints/BLF working along with Realtime SIP but I never get any busy notification. core show hints always shows the realtime sip user as idle. I have tried setting call-limit to various values, including 1 but nothing seems to help. I have tried limitonpeers both yes and no. Anybody got any other ideas? I do know the hinting is working as I can hint a Zap channel and it works fine. Daniel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do While loop
On Fri, 30 Nov 2007 10:54:47 -0900, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: Sorry it's in some pseudocode that doesn't really represent a language at all. BTW, how do most people write dialplans these days? Do they still use extensions.conf, or did they move to either AEL, AEL2, or even external scripts through AGI? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cross-compiling asterisk-1.4 for Debian on a slug
Fabiano Sidler wrote: Hello? Nobody any idea how to fix that or how to compile asterisk-1.4 for Debian/slug on an i686 machine? Or should i ask on the nslu2 mailing list? You did not make clear if you try to build on an i686 or on a slug (as your subject says) which is not x86 but Intel XScale. Anyhow: All I can say is that cross-compiling Asterisk is probably not an easy task although some improvements have been made recently. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Increasing the voice volume from the diguim cards
bilal ghayyad wrote: Hi List; Anyone knows a method (command) to increase the voice volume at diguim card level? Are you trying to do this at some other level than rxgain and txgain settings in zapata.conf? If so, for the analog cards there are some module parameters for doing so. For digital T1/E1 cards, the only way to do it is with the gain options in zapata.conf. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
On Friday 30 November 2007 04:17:36 Philipp Kempgen wrote: With SIP you can attach custom variables to calls (using X-... headers). IAX (Inter-Asterisk eXchange!) can't do that (yet). With IAX2 you can share variables too. I believe Tilghman had supplied a patch to do exactly that several months ago. -A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
Andrew Kohlsmith wrote: On Friday 30 November 2007 04:17:36 Philipp Kempgen wrote: With SIP you can attach custom variables to calls (using X-... headers). IAX (Inter-Asterisk eXchange!) can't do that (yet). With IAX2 you can share variables too. I believe Tilghman had supplied a patch to do exactly that several months ago. I believe this patch is in trunk but not in 1.4. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality
When you take your packet capture, you'll need to look at the sip messages with SDP attached to get the ip's and ports used for both media streams. Make sure that the ips are correct and that the port used can traverse between those ip's without being blocked by a packet filter or firewall. A lot of times, administrators will set a range of UDP ports that are allowed to pass their packet filter for media and your pbx or phones may be using a different range. This can cause audio loss. You'll need to eliminate that possibility. Sometimes checking your firewall/packet filters for blocks may also prove helpful in identifying problems. You should be aware that the logs from certain firewall products may not be comprehensive. For example, in the past I have seen packets dropped going through netscreens because of invalid headers and no entries appeared in the logs. If you ultimately believe a firewall may be blocking your traffic make sure you setup a capture port or a span on each side of the device and verify the traffic going to and leaving from the firewall using ethereal on a laptop or maybe a Nixon box if you are in a large distributed environment. Never trust a potentially broken device to report accurate information about it's function. TDM = Time Division Multiplexing TDM describes how channels are separated on T1's, etc. It's common to refer to those types of connections as TDM. http://en.wikipedia.org/wiki/Time-division_multiplexing -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Veselin Kantsev Sent: Friday, November 30, 2007 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality Thank you much for the prompt reply Salvatore. Would you have the time to explain further how should I go for verifying that SDP and RTP are OK. Also what is reffered to as the TDM site. Veselin On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote: Take a packet capture of your VoIP segment and verify that the SDP is correct and that the RTP is making it to the correct places. If all that looks good and this is a straight out quality problem, then you need to figure out if it's happening on the voip side or on the TDM side. You should make calls (with captures) VoIP to Voip passing the media through your asterisk and also try routing a tdm call in and back out. If you have the equipment, take a mos score of the TDM loop. Without any of the above, you will not be able to isolate the issue. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Veselin Kantsev Sent: Friday, November 30, 2007 2:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality Hello, I have an Asterisk running with a Sangoma A200 card with Hardware Echo cancelling connected to the UK PSTN. If a PSTN call comes in, voice both ways is OK, however if an outgoing call over the PSTN is made I can hear the other party OK but they can not, they can barely understand what I am saying, my voice is unclear fading and skipping. Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2 are OK too. I've tried gsm/ulaw/alaw codecs so far. Tried disabling the echo cancelling as well. Any suggestions will be greatly appreciated. Regards, Veselin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using existing extensions.conf macros, and co-habitation
Anthony Francis wrote: Philip Prindeville wrote: Tilghman Lesher wrote: On Thursday 29 November 2007 13:29:17 Philip Prindeville wrote: [snip] The issue is that I have, per virtual pbx (i.e. home or business), two contexts that these get used from. The internal-xyzzy and incoming-xyzzy contexts (one for each pbx, ie. xyzzy is home or else it's office). I was wondering if there wasn't a more flexible solution to this issue, than hard-coding a Goto(default,s,1) into them (I have no default context, because it would be meaningless). Perhaps using Gosub and Return. Or do I need to hack the macro, and pass in a 3rd argument (bletch)? MacroExit or Gosub/Return would certainly be possibilities. The main thing to note is that this macro that you call standard is actually just an arbitrary example. It is by no means perfect, so feel free to adapt it to your own liking. Sure. I just figured that it would be nice if the canned macros worked out-of-the-box without modification, in the real world. I suppose I could file a bug, and then submit patches for the macro and documentation... -Philip The ability to modify the macros to your own needs is not a bug. Anyway try adding a few more args to your stdexten to handle the context name and the like so it doesn't need default. On another point, why would asterisk come with built in code example for a multi-tenant set-up? Please save your self some time and embarrassment by not submitting that particular bug. Anthony Using the default context is a bad idea, as is pointed out in several places (including the SECURITY document, the O'Reilly book, and several good online tutorials). Besides, what's the point of having all the flexibility that you have in Asterisk if you're going to shoot yourself in the foot by having canned macros that limit that flexibility? Let's say I file the bug, and someone closes it. What's the harm? Someone doing a search of the database will at least later have the suggested patch as a possible resolution to their trying to address the same or a similar requirement. Good thing I'm not easily embarrassed, or I'd find your attitude stifling. And to address your question: it wouldn't be code *for* multi-tenant. It would be code that *didn't preclude* multi-tenant. Anything worth doing is worth doing right. Examples provided with Asterisk should showcase its power and flexibility. Not limit/ignore it. -Philip ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Only call me once
[EMAIL PROTECTED] wrote: Anyone have an idea how to implement a phone number that can only be called once? The first time it will process normally and any subsequent calls will be rejected. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yup store the number in DB if its new, if it's already in db reject. This is an incredibly simple thing to do, you can even just use the Asterisk internal DB for simplicity. Anthony Yeah fairly easy, but why would you want to? Is this part of a verification process like throwaway URL's that get emailed to me when I sign up at a web site? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.15 crash without generating core file
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 equis software wrote: Hi, I'm testing Asterisk 1.4.15 with the -g option. When it crash didn´t generate core file in the /tmp folder. What is happening?? Check the directory you were in when you ran Asterisk. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHUfcaDQNt8rg0Kp4RApnEAKC7BjvKEhMcQlNwNzOurWw1pLdz+ACfYzpA P37Q505SDDIQmbi+2CQ+Prc= =O62/ -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration state: Failed
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Probably you have deny=something instead of disallow=all. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHUfd5DQNt8rg0Kp4RAnysAJwJPfhiaHTkVl1/vxgrqEpl2nIKpACdHQbP f27510ioiqIoVf6TTrWYK4s= =oU1d -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Requiring a login to a phone
Hi List, We have a remote asterisk SIP phone at the cottage. I'd like it to have minimal privileges when it first registers with Asterisk. Ideally it should be in a restricted context. Dialing any number would intercept the call and tell the person to log on. This way, if the phone was stolen or someone got into the cottage, we wouldn't have a bunch of surprise charges on our phone bill... :-) Once the phone has been authenticated, it should go into a context with normal privileges. After a couple of days of non-use, it should auto-logout to the restricted context. How can I change the sip context of a phone on the fly, based on authentication login? Any ideas? Thanks, Steve sip.conf: --- ; phone at the cottage [155] context=restricted-155 ... extensions.conf [restricted-155] exten _X.,1,NoOp(All Calls filter through this if not logged in on 155] exten _X.,n,Answer exten _X.,n,Wait(1) exten _X.,n,Playback(You must log in to use this phone) exten _X.,n,Authenticate(65535) // if the person authenticates sucessfully, change the context of ext 155 // from restricted-155 to sip-phones.(HOW???) [sip-phones] ; normal sip phone outgoing context ... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: Avaya
Avaya makes 52% of it's revenue from professional services. In enterprises, you generally have 3 budgets: Captial, expense, professional services Avaya figured out that they could make more money tapping into professional services portion of the budget with charge by the hour union consultants than by selling equipment. Avaya is also the most pervasive vendor in the space when it come to calling dev products GA, so they can get their customers to pay them to beta test. Avaya's newest ploy is to get customers hooked on their systems and after 6 - 12 months of shear hell supporting the products, they kindly offer to outsource your voice infrastructure support using a system called SIG. SIG requires you to place a collector box on your network with an IPSEC VPN nailed up to Avaya corporate. This gives them full unchecked access to your network. Exciting huh? Introducing Avaya into a corporate network is about as smart as introducing syphalis into a high school. Sure, it was all fun and games at first, but eventually it catches up to you. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesse Molina Sent: Saturday, December 01, 2007 1:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Off-Topic: Avaya Salvatore Giudice wrote: They are cheap. You only have to pay for the box and the maintenance percentage. That is indeed the Avaya way. First you buy it, then you rent it. Stop paying their maintenance fees and their dial into your PBX and cripple the OS by removing customer maintenance command permissions. Hell, Avaya won't even give you root on any of their servers. You cant audit the box and you can't poll them unless you pay them money to join their partner program and get their SDK. If you already have Avaya, you should just buy Message Networking or a Mitel voicemail server if you want seamless voicemail with Avaya. However, you should know that using Avaya is probably a bad idea to begin with. Until February 07, the majority Avaya's soft switch products were running on Redhat 9, which was unsupported since 2003. Avaya was only managing a dozen packages and they've always left it up to the customer to know when they need an update, requiring the customer to request a field load. It has to be the worst update model in the industry when it comes to infrastructure monitoring and patching. By using Avaya, you are blindly trusting them to properly maintain a Linux appliance. This is something they are not capable of and you can't even audit them. Avaya is what happens to organizations when they have ignorant telecom infrastructure engineers deciding what products to buy. Avaya focuses sales on those engineers because they k now their products won't pass certification by network, systems, or security engineers. Telecom engineers only look for features and usually get their asses handed to them after they put Avaya VoIP into their infrastructure. Bravo. A well-deserved lambasting of this awful vendor. -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requiring a login to a phone
Steve, You might be able to swing it using the configuration updater that's part of the manager API as of 1.4.0: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+UpdateConfig -- Alex On Sat, 1 Dec 2007, Steve Johnson wrote: Hi List, We have a remote asterisk SIP phone at the cottage. I'd like it to have minimal privileges when it first registers with Asterisk. Ideally it should be in a restricted context. Dialing any number would intercept the call and tell the person to log on. This way, if the phone was stolen or someone got into the cottage, we wouldn't have a bunch of surprise charges on our phone bill... :-) Once the phone has been authenticated, it should go into a context with normal privileges. After a couple of days of non-use, it should auto-logout to the restricted context. How can I change the sip context of a phone on the fly, based on authentication login? Any ideas? Thanks, Steve sip.conf: --- ; phone at the cottage [155] context=restricted-155 ... extensions.conf [restricted-155] exten _X.,1,NoOp(All Calls filter through this if not logged in on 155] exten _X.,n,Answer exten _X.,n,Wait(1) exten _X.,n,Playback(You must log in to use this phone) exten _X.,n,Authenticate(65535) // if the person authenticates sucessfully, change the context of ext 155 // from restricted-155 to sip-phones.(HOW???) [sip-phones] ; normal sip phone outgoing context ... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cdr_pgsql error in 1.4.15
On Saturday 01 December 2007 09:43:41 equis software wrote: In Asterisk 1.4.15 if I try to configure cdr_pgsql.conf , asterisk crash with this message asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_pgsql.so: undefined symbol: PQescapeStringConn Is this a knowed error? This sounds like version skew -- like you have headers from a later version of Postgres, but an earlier version of the libraries. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Pcengines Alix board
On Thu, 29 Nov 2007 23:55:38 -0600, John Faubion [EMAIL PROTECTED] wrote: The newer CF cards are making this nearly a mute point. Seems like I provide updated software often enough that I never have CF cards wear out. I guess /tmp can live in RAM, but what about eg. recording ten-twenty WAV files to /var a day, and logs into /var/log? Do I have to worry about the card wearing out in six months? I'm not sure I understand the need for the PCI card to be perpendicular to the board. So I can use a flatter box. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Requiring a login to a phone
On Saturday 01 December 2007 18:09:27 Steve Johnson wrote:
Hi List,
We have a remote asterisk SIP phone at the cottage.
I'd like it to have minimal privileges when it first registers with
Asterisk. Ideally it should be in a restricted context. Dialing any
number would intercept the call and tell the person to log on. This
way, if the phone was stolen or someone got into the cottage, we
wouldn't have a bunch of surprise charges on our phone bill... :-)
Once the phone has been authenticated, it should go into a context
with normal privileges. After a couple of days of non-use, it should
auto-logout to the restricted context.
How can I change the sip context of a phone on the fly, based on
authentication login?
I wouldn't. I'd do authentication on the fly, using a database of some kind.
extensions.conf:
[sip-phones]
exten = _X.,1,Set(lastlogin=${ODBC_LOGIN(${CUT(CHANNEL,-,1)})})
; Logins expire after 86400 sec = 24 hours
exten = _X.,n,GosubIf($[0${lastlogin} + 86400 ${EPOCH}]?restricted,s,1)
exten = _X.,n,Dial(Zap/g1/${EXTEN})
[restricted]
; VMAuthenticate terminates the call if authentication fails.
exten = s,1,VMAuthenticate
exten = s,n,Set(ODBC_LOGIN(${CUT(CHANNEL,-,1)})=${EPOCH})
exten = s,n,Set(lastlogin=${EPOCH})
exten = s,n,Return
func_odbc.conf:
[LOGIN]
dsn=asterisk
read=SELECT lastlogin FROM logins WHERE channel='${ARG1}'
write=UPDATE logins SET lastlogin=${VAL1} WHERE channel='${ARG1}'
logins.sql:
CREATE TABLE logins (
channel CHAR(50) PRIMARY KEY,
lastlogin INTEGER,
);
INSERT INTO logins VALUES ('SIP/100',0);
INSERT INTO logins VALUES ('SIP/101', 0);
INSERT INTO logins VALUES ('SIP/102', 0);
--
Tilghman
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Re: [asterisk-users] Requiring a login to a phone
For such a simple application I'd use AstDB to avoid having to hassle with
an external database (and also means this sort of dialplan will work even on
embedded/slimmed Asterisk boxes that may not have db modules
loaded/available). In any case, what Tilghman said is what I'd suggest as
well.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+db
http://www.the-asterisk-book.com/unstable/funktionen-db.html
Also consider allowing emergency number dialing to bypass authentication, if
applicable.
-jr
On Dec 1, 2007 5:32 PM, Tilghman Lesher [EMAIL PROTECTED]
wrote:
On Saturday 01 December 2007 18:09:27 Steve Johnson wrote:
Hi List,
We have a remote asterisk SIP phone at the cottage.
I'd like it to have minimal privileges when it first registers with
Asterisk. Ideally it should be in a restricted context. Dialing any
number would intercept the call and tell the person to log on. This
way, if the phone was stolen or someone got into the cottage, we
wouldn't have a bunch of surprise charges on our phone bill... :-)
Once the phone has been authenticated, it should go into a context
with normal privileges. After a couple of days of non-use, it should
auto-logout to the restricted context.
How can I change the sip context of a phone on the fly, based on
authentication login?
I wouldn't. I'd do authentication on the fly, using a database of some
kind.
extensions.conf:
[sip-phones]
exten = _X.,1,Set(lastlogin=${ODBC_LOGIN(${CUT(CHANNEL,-,1)})})
; Logins expire after 86400 sec = 24 hours
exten = _X.,n,GosubIf($[0${lastlogin} + 86400 ${EPOCH}]?restricted,s,1)
exten = _X.,n,Dial(Zap/g1/${EXTEN})
[restricted]
; VMAuthenticate terminates the call if authentication fails.
exten = s,1,VMAuthenticate
exten = s,n,Set(ODBC_LOGIN(${CUT(CHANNEL,-,1)})=${EPOCH})
exten = s,n,Set(lastlogin=${EPOCH})
exten = s,n,Return
func_odbc.conf:
[LOGIN]
dsn=asterisk
read=SELECT lastlogin FROM logins WHERE channel='${ARG1}'
write=UPDATE logins SET lastlogin=${VAL1} WHERE channel='${ARG1}'
logins.sql:
CREATE TABLE logins (
channel CHAR(50) PRIMARY KEY,
lastlogin INTEGER,
);
INSERT INTO logins VALUES ('SIP/100',0);
INSERT INTO logins VALUES ('SIP/101', 0);
INSERT INTO logins VALUES ('SIP/102', 0);
--
Grover Beach, California, USA
http://blog.joshrichards.org[EMAIL PROTECTED]+1 (805)
471-6923
http://www.linkedin.com/in/joshrichards
Supporting these causes: Kiva.org RoomToRead.org
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[asterisk-users] Answer Machine/Fax/modem detection
Has anyone sucessfully implimented a fax or modem detection dial plan? I'm originating calls from asterisk using a list of numbers and dropping the destination into an IVR menu but need to do something different if a modem or fax answers. I tried to use the NVBackgroundDetect() application but i think that is for receiving faxes only. Any help would be appreciated. Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
