Re: [Asterisk-Users] Does PCI Developer Kit work with kernel 2.6
I just got it working with a 2.6 kernel without any problems, although I'm having a little bit of trouble setting the callerid on it Keith On Jul 1, 2005, at 11:00 PM, Michael Jia wrote: Hi, In digium website. http://store.yahoo.com/asteriskpbx/newitastdmde.html It is said Dev Kit PCI card works with 2.4 kernel. I am wondering if it is also working with 2.6 kernel? Anyone knows? Thanks Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E3 card
Eric Wieling aka ManxPower wrote: Kevin P. Fleming wrote: Tamas J wrote: Does anybody know what will be the around price for the announced E3 card from Digium? When is it planned to be ready? Pricing and release date have not been announced at this time. The product has also been removed from Digium's web site, as far as I can tell. Perhaps they decided the market is not that big, and reduced the priority of the card. If so, they are not the only ones to do this. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pattern matching based on callerid, not working
On 07/02/05 02:15 Matthew Boehm said the following: according to the wiki, I should be able to do this: exten = _9./3003,1,Set(CALLERID(number)=281443) exten = _9./3004,n,Set(CALLERID(number)=281444) exten = _9./3005,n,Set(CALLERID(number)=281445) exten = _9./3006,n,Set(CALLERID(number)=281446) i believe these four should be on the same priority. asterisk dialplans will bork if the next priority doesnt match. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pattern matching based on callerid, not working
On 2 Jul 2005, at 08:48, Dinesh Nair wrote:
On 07/02/05 02:15 Matthew Boehm said the following:
according to the wiki, I should be able to do this:
exten = _9./3003,1,Set(CALLERID(number)=281443)
exten = _9./3004,n,Set(CALLERID(number)=281444)
exten = _9./3005,n,Set(CALLERID(number)=281445)
exten = _9./3006,n,Set(CALLERID(number)=281446)
i believe these four should be on the same priority. asterisk
dialplans will bork if the next priority doesnt match.
Or better yet, fold them into a single line - like:
exten = _9./300[3456],1,Set(CallerID(number)=28144${CALLERIDNUM}:1)
Tim.
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Re: [Asterisk-Users] MOH - request to schdule in the past
I have googled this to death, and all I get are reference to the MoH needing a Zaptel timing source, and then people saying no they don't any more. -- Set Response Timeout to 2 -- Executing BackGround(SIP/211-57ba, my-greeting) in new stack -- Playing 'my-greeting' (language 'en') == CDR updated on SIP/211-57ba -- Executing MusicOnHold(SIP/211-57ba, ) in new stack -- Started music on hold, class 'default', on SIP/211-57ba -- Stopped music on hold on SIP/211-57ba == Spawn extension (webtree, 7, 1) exited non-zero on 'SIP/211-57ba' Jul 1 20:19:32 NOTICE[1376]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! It seems to work for a little while after a reboot then dies a death. The extension just gets dead air, and the error message is on the CLI, it always comes up after a call, and seemingly randomly inbetween. Any suggestions where to look? extensions.conf has exten = _[123],3,Dial(SIP/200SIP/202SIP/210SIP/211SIP/212|40|m) and exten = 7,1,musiconhold() both do the same thing. musiconhold.conf has [classes] default = mp3:/var/lib/asterisk/mohmp3,-z ;loud = mp3:/var/lib/asterisk/mohmp3 ;random = quietmp3:/var/lib/asterisk/mohmp3,-z From your system command line (not asterisk), type 'mpg123' and tell us what version of mpg123 you're running. If its not v0.59r or v0.59q, then get one of those installed. (Lots of notes say v0.59r only, however I've been using v0.59q on RHv9 and Fedora 3 boxes with no problems.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Operators Panel for Asterisk
IPSwitchBoard Version 0.121 - 02 July 2005 * Extensions can be added to speed dial number. This can be used to dial speed dial numbers from any phone connected to your asterisk system. This requires that you configure your dial plan to take advantage of this feature. See sample Dial Plan in the IPS Manual. Download: http://ipswitchboard.thorben.dk _ IPSwitchBoard is a FREE Windows.Net application that will allow you to: Unattended/attended transfers. Park calls and retrieve/forward them again. Organize all your SIP, IAX and Zap extensions (automatically retrieved from Asterisk). Hotel/Call shop Billing module Monitor all extensions, queues, agents and Parked Calls. Dynamically log extensions in and out of queues. Integration with CRM software on the web. Browse Call Records and make Charts. Record calls and transfer wav files to the PC automatically. Set Do Not Disturb on Extensions and give a reason. Speed Dialing. Share speed dial number between users of IPS ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] make error for zaptel
On Friday 01 Jul 2005 16:43, Zoltan Szecsei wrote: Bob Goddard wrote: On Friday 01 Jul 2005 15:14, Zoltan Szecsei wrote: Hi Bob, Thanks - I'll run with the README idea of yours. Your comment regarding re-boot however is not valid. I also thought that was the case and (as I said on the first line of my message) I specifically rebooted the box. Have to confess I am really flumuxed why the symbolinc link differs from the uname -r name. I cannot see what the problem is with the output of 'uname -r'! I'm saying that I though that if uname -r returns: 2.6.11.4-20a-smp then I would expect that /usr/src/linux would link to linux-2.6.11.4-20a-smp and it does not, it links to linux-2.6.11.4-21.7 see: gl0:/usr/src # ls -la [...] First, learn to trim your posts. I have already read the signature and I don't need to read it any more. You will have updated the kernel via Yast. Yast will remove all previous versions. You have 2 kernels at least partially installed. Your system is broken. When you updated the kernel, the linux link was recreated, but Yast did not manage to remove the previous version. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Errors Question
Today we have been having some problems with the dchannel of out T1's. I was wondering if there is a way for asterisk to send out an email or page whenan error occurs. Not I know errors happen quite offen for many reasons, but I would like an email sent when there is a TI problem, or specefic other problems. I am using the safe_asterisk to run asterisk and it sends an email when asterisk crashes, which is not very often. Take a look at using a combination of logging and swatch. Goggle for 'swatch syslog'. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] editing time to say astcc-noanswer
i dont know how to edit the time 3ms for ringing in astcc when it says there is no body to answer.i want to change this time to 4ms but i dont know how.please help please. __ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PortaOne's Radius client for Asterisk
hello i m trying to use radius with asterisk http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth how to fix this patch 8. Make sure that your Asterisk includes all related bug fixes and patches, namely: - SIPGetHeaders for chan_sip (derived from chan_sip2 ) i m using asterisk-1.0.9 extensions.conf exten = _X.,1,SIPGetHeader(SIP_Authorization=Proxy-Authorization) Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does PCI Developer Kit work with kernel 2.6
On Saturday 02 July 2005 02:04, Keith Caldwell wrote: I just got it working with a 2.6 kernel without any problems, although I'm having a little bit of trouble setting the callerid on it You can't set outgoing callerID on analog PSTN lines. This is a telephone network limitation, not a Digium hardware limitation. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does PCI Developer Kit work with kernel 2.6
On Saturday 02 July 2005 02:04, Keith Caldwell wrote: I just got it working with a 2.6 kernel without any problems, although I'm having a little bit of trouble setting the callerid on it You can't set outgoing callerID on analog PSTN lines. This is a telephone network limitation, not a Digium hardware limitation. Think he's trying to set it on a TDM-fxs module (not fxo). Or did I miss something. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does PCI Developer Kit work with kernel 2.6
On Saturday 02 July 2005 07:57, Rich Adamson wrote: Think he's trying to set it on a TDM-fxs module (not fxo). Or did I miss something. Nope I am probably the one who is missing it, it's early here. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom - Asterisk - Vegastream
Looking for some advice from vegastream users if possible? I am having a nightmare trying to find the best settings for G729 and G723.1 codecs. My users are using Snom phones. Any recommendations as to the best codec settings would be very appreciated as trial and error is proving long and unproductive! Thanks, Neil ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Epia C3 Linux
On Fri, Jul 01, 2005 at 09:53:33AM -0700, Wiley Siler wrote: Anyone know a good distro for an Epia Mobo with the C3 chip? Debian, as for any hardware :-p I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Debian i386 packages should work on VIA Epia-s as-is. Specifically, see http://bugs.debian.org/294884 Does anyone know a good install for this processor/mobo combo? I have has some success stories of Rapid with it. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems loading asterisk .
Bharat M. Sarvan wrote:
Hello everybody,
I have made a application of my own. (I.e. Def ( )). I am able to
compile the application successfully. And the .so file is created as
well. But when I load asterisk I get the following error.
[Def.so]Jul 1 19:20:06 WARNING[15664]: loader.c:295
ast_load_resource: No key routine in module
/usr/lib/asterisk/modules/Def.so
You need to return the Asterisk GPL Key, have a look at app_skel.c (if I
remember correctly).
I.E.:
char *key()
{
return ASTERISK_GPL_KEY;
}
And from module.h:
#define ASTERISK_GPL_KEY \
This paragraph is Copyright (C) 2000, Linux Support Services,
Inc. \
In order for your module to load, it must return this key via a function \
called \key\. Any code which includes this paragraph must be licensed
under \
the GNU General Public License version 2 or later (at your option).
Linux \
Support Services, Inc. reserves the right to allow other parties to
license \
this paragraph under other terms as well.
(Sorry about the line wrapping)
--
Cheers,
Matt Riddell
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[Asterisk-Users] Audio delay w/ call forwarding
I have experienced a * problem with all forwarded calls where the inbound caller cannot hear any audio for 2-4 seconds after the forwarded call is answered, causing the caller--who cannot hear anything--to think there is no connection and thus hangs up. If the caller waits a couple of seconds, audio is restored and everything is OK. The problem didn't seem to be there when I first set up the server about 2 months ago. After eliminating potential hardware problems, etc., I figured out that when I change jitterbuffer=yes to jitterbuffer=no in iax.conf the audio delay problem is solved, but the call quality degrades. My prior settings were: [general] bindport = 4569 bindaddr = 0.0.0.0 delayreject=yes disallow=all allow=ulaw allow=alaw mailboxdetail=yes register = [EMAIL PROTECTED] dropcount=3 jitterbuffer=yes maxjitterbuffer=500 maxexcessbuffer=300 canreinvite=no Any ideas on tweaking iax.conf to optimize call quality, but avoid the audio delay with forwarded calls? Regards, Mike Michael Hillerbrand ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC
Problem resolved with Astcc, certified fully working.Juan Luis Moyano [EMAIL PROTECTED] wrote: Ade Agbero wrote: Finally, We have lift off, a shaky one though. I deleted my Astcc.gi and replaced it with Darren's copy posted on his website and I have finally been able to get something recorded as BILLCOST. I got it working too here with Darren's astcc.agi. And billing as expected so finally It's working. It would be nice if someone could update the cvs with Darren's astcc.agi, because the current one doesn't work, even patched.. it gets worse. Thanks for your attention Darren!-- Juan Luis Moyano[EMAIL PROTECTED]___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger NEW - crystal clear PC to PC calling worldwide with voicemail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error with app_addon_sql_mysql.c
Hi People! Having interesting issues with app_addon_sql_mysql.c: [EMAIL PROTECTED]:/usr/src/asterisk-addons# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 This is a installation of Slackware 10.1 with Mysql 4.1.12 (source). Any ideas? Regards, Sahil Gupta VoiceValley On Fri, 1 Jul 2005, Brian West wrote: You could have just done ln -s asterisk-1.0.9 asterisk and it would have fixed that. It should by default do -I../asterisk /b --- Anakin: “You’re either with me, or you’re my enemy.” Obi-Wan: “Only a Sith could be an absolutist.” On Jun 30, 2005, at 1:13 AM, Chris Mason (Lists) wrote: Marcel van Kaam, Fonetica wrote: I had the same problem with installing addons. I checked out in the file cdr_addons_mysql.c what the location of the asterisk.h must be and changed the cdr_addons_mysql.c to the location of the asterisk.h file. After this it worked. Also to be sure do: locate asterisk.h to check or you have the file on your system. Marcel Yes, that worked. For the record, it had to be #include ../asterisk-1.0.9/asterisk.h -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Provider Survey
List Receiver wrote: Having used Broadvoice for a while with marginal service, I want to move on to another provider. So my question to the List is who is good? I know now one service is perfect but somebody out there has to be decent. Who have you guys had the best luck with? I suggest, if your installation is mission critial at all, you use a dialplan setup that has failover between two providers with failover to PSTN, and that you consider Teliax and NuFone as Providers. I have found these are serious players, they are not lemonade stands, and they offer failover to pstn for incoming calls also. I have never had an instance where both are down, but my experiene is relatively short. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk showing more than once on ps
The system startup script /etc/init.d/asterisk calls the script /usr/sbin/safe_asterisk In safe_asterisk, the program is started with -c by default (console on TTY9). That explains why it is starting with a console, but not why it's running so many times! Here is what my system (FC3) shows: [EMAIL PROTECTED] sbin]# ps ax | grep asterisk 3371 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk 3417 ?S 0:00 asterisk -vvvg -c 6846 ?S 0:00 asterisk -vvvg -c 6848 ?S 0:00 asterisk -vvvg -c 6849 ?S 0:00 asterisk -vvvg -c 6850 ?S 0:00 asterisk -vvvg -c 6853 ?S 0:01 asterisk -vvvg -c 6854 ?S 0:00 asterisk -vvvg -c 8479 pts/1S+ 0:00 grep asterisk Can anyone explain why asterisk is being launched 7 times? Thanks, OCG -Original Message- From: Luki [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk showing more than once on ps Do not know why, but have noticed redhat = 1, and debian = many Not quite. RedHat Enterprise also = many at times, depending on number of concurrent calls; usually one when idle. Maybe it has something to do with kernel 2.4 vs 2.6 and how threads show up in ps. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] passing through MWI info from SBC
Woah woah woah.. why not just disable SBC voicemail and have asterisk handle it? I don't understand why you would go to such great lengths when you can just have Asterisk deal with it. On 7/1/05, andrew matthews [EMAIL PROTECTED] wrote: Is there alternative access to voicemail? Like web access? If there was then you can write a program to check the site every 5 or 10 minutes maybe every 30, and parse out the message listing and then send MWI to the phones if there is a VM waiting. Or You could get one of those cheep 20 dollar devices and hook it up to the serial port on the asterisk box and then you can use sty to monitor the port for changes. Then send the message that way. I'd have to research it a little more but it very possible. Just a thought :) On 7/1/05, Chris Gamble [EMAIL PROTECTED] wrote: Does this just sound worse than it is? With SBC you are out of luck, since Asterisk doesn't detect dialtone ( it dials blind, sometimes too quickly for the CO to catch the first digit, resulting in wrong numbers )) or stutter dialtone either, and reportedly has had any indication of the DC status of a POTS line removed due to problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Novack Sent: Friday, July 01, 2005 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] passing through MWI info from SBC Mike Myers wrote: Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all exchange voicemails with each other on the SBC voicemail system. They can leave messages for each other without having the phones ring, etc... We have a 2 yr old at home, and her sister has some small kids too, so that's how they manage to send voicemails when they are unsure if the kids are sleeping, etc... Anyway, preserving this capability of using the SBC VM and being notified when a message is waiting is critical for good WAF. The vonage line and voipjet line can be intergrated into the Asterisk VM. My Nortel venture phones light the MWI if any line has VM on it, and the display tells you which lines have VM waiting. I would love to be able to duplicate this function on the Polycom's and hopefully the Sipura's as well. I've looked for answers on this, but haven't found one, hence the post. My apologies if I have missed something. Thanks much, Mike You haven't missed much. With SBC you are out of luck, since Asterisk doesn't detect dialtone ( it dials blind, sometimes too quickly for the CO to catch the first digit, resulting in wrong numbers )) or stutter dialtone either, and reportedly has had any indication of the DC status of a POTS line removed due to problems. Only choice would to port the number to a VOIP provider and provide the VM in Asterisk. Similar problem with Vonage VM. John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk showing more than once on ps
Hello All , On Sat, 2 Jul 2005, Michael Stahl wrote: The system startup script /etc/init.d/asterisk calls the script /usr/sbin/safe_asterisk In safe_asterisk, the program is started with -c by default (console on TTY9). That explains why it is starting with a console, but not why it's running so many times! Here is what my system (FC3) shows: [EMAIL PROTECTED] sbin]# ps ax | grep asterisk 3371 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk 3417 ?S 0:00 asterisk -vvvg -c 6846 ?S 0:00 asterisk -vvvg -c 6848 ?S 0:00 asterisk -vvvg -c 6849 ?S 0:00 asterisk -vvvg -c 6850 ?S 0:00 asterisk -vvvg -c 6853 ?S 0:01 asterisk -vvvg -c 6854 ?S 0:00 asterisk -vvvg -c 8479 pts/1S+ 0:00 grep asterisk Can anyone explain why asterisk is being launched 7 times? Thanks, OCG -Original Message- From: Luki [mailto:[EMAIL PROTECTED] Sent: Friday, July 01, 2005 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk showing more than once on ps Do not know why, but have noticed redhat = 1, and debian = many Not quite. RedHat Enterprise also = many at times, depending on number of concurrent calls; usually one when idle. Maybe it has something to do with kernel 2.4 vs 2.6 and how threads show up in ps. --Luki Below ps is from a * server on slackware 10.0 using the command to start(**) . So I am not sure the '-c' is what is creating the multiple threads . linux-2.6 issue maybe ? Hth , JimL # ps -auxww | grep aster root 115 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 123 0.0 1.1 11916 5944 ?SJun30 0:01 /usr/sbin/asterisk -d -v -v -v root 125 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 130 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 131 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 132 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 139 0.2 1.1 11916 5944 ?SJun30 6:08 /usr/sbin/asterisk -d -v -v -v root 155 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 156 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 157 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v root 158 0.0 1.1 11916 5944 ?SJun30 0:00 /usr/sbin/asterisk -d -v -v -v (**) # after expansion of variables . /usr/sbin/asterisk -d -v -v -v /var/log/asterisk/debug # sudo asterisk -V Asterisk CVS-HEAD-05/01/05-14:10:09 # uname -a Linux asterisk-1 2.6.11.8 #1 Sun May 1 12:04:14 MDT 2005 i686 unknown unknown GNU/Linu -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E3 card
Just get one of these. The PCI 921-CDS is a low-cost channelized DS3 WAN adapter that can be used in ImageStream's Industrial Series routers or OEM products running Linux. The PCI 921-CDS can individually address all of the DS0s and T1s in a DS3, and it can be used in a wide range of applications including WAN and Internet routing, Voice-over-IP (VoIP), medical imaging, and digital video on demand. http://imagestream.com/PCI_921-CDS.html On 7/2/05, Steve Underwood [EMAIL PROTECTED] wrote: Eric Wieling aka ManxPower wrote: Kevin P. Fleming wrote: Tamas J wrote: Does anybody know what will be the around price for the announced E3 card from Digium? When is it planned to be ready? Pricing and release date have not been announced at this time. The product has also been removed from Digium's web site, as far as I can tell. Perhaps they decided the market is not that big, and reduced the priority of the card. If so, they are not the only ones to do this. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] passing through MWI info from SBC
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jon Radon Sent: Saturday, July 02, 2005 10:49 AM To: andrew matthews; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] passing through MWI info from SBC Woah woah woah.. why not just disable SBC voicemail and have asterisk handle it? I don't understand why you would go to such great lengths when you can just have Asterisk deal with it. Because in most cases, the POTS provider will not disable voicemail on a per number basis as it is a part of the 'package'. So there really isn't an option to do that. The other issue is that with call waiting, if you do not answer the call there is no way to have Asterisk handle the voicemail. So here is where you would still need the pass through in order for the subscriber to know there was a message. For me, I just got a $5 per month DID and forwarded all my POTS call to it. I get up to two simultaneous incoming calls that Asterisk handles completely. It also includes a voicemail system where if my connection goes down or I exceed those two calls, then if someone leave a message, it gets emailed to me. The other benefit is that I still have my POTS line and use it for all my local outgoing calls and use a 1.3 cent per minute provider for all LD calls and I still save money by not spending that extra $20 a month for unlimited LD on my POTS line. Has worked great so far. Only down side is if my DID connection drops, I have no way to call into the house. But, I just use the cell to call the wife in that case. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AMP/A@H (asterisk at home) custom incoming routing
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Friday, July 01, 2005 11:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AMP/[EMAIL PROTECTED] (asterisk at home) custom incoming routing A few things to followup on my earlier post: 1.) Definitely put the [tdm-in] context in the file /etc/asterisk/ extensions_custom.conf. That way your changes will not get overwritten. 2.) I am still unable to make call waiting on the incoming ZAP line work, b/c I have not thought up a good way to make this happen. hasn't anyone done this before? 3.) When setting up the ZAP trunk, I found it usefull to put w in the dial prefix field to force the system to wait for the dial tone. If I didn't do that, I could not call out on the ZAP Channel. Tom Tom, Actually, it is the extensions_custom.conf that DOES get overwritten. Unles things have completely changed since version 1.0. You should be using the extensions.conf to place all your permamnant changes. The _custom was there to tell you that those are the custom settings for [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What to use h323 or oh323 ???
I m new to asterisk n i've got an IP phone that supports h323 protocol but i dont know how to configure asterisk to use it... i m comfortable in using sip iax softphones butthere is no h323.conf in /etc/asterisk/ i read that i've to compile some files but i m confused regarding h323 oh323 .. which one should i use.. plz tell me or atleast give some helpful link__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DTMF Problem...
IAX doesn't use INBAND DTMF. Denis Galvão. On 01 de jul de 2005, at 03:23, Mark Edwards wrote: Hi. Probably been asked before, but my IAX provider assures me its not their problem I have a IAX connection to a peer providing a DID. I am dialing up my number, seeing the DTMF tones come down the line, and the * IVR is just ignoring them. IAX debug output is: Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: DTMF Subclass: 1 Timestamp: 02608ms SCall: 00016 DCall: 3 [ 210.80.176.12:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 02608ms SCall: 3 DCall: 00016 [210.80.176.12:4569] for a press of 1 I am assuming this is the DTMF inband problem, but I appear unable to convince my provider. Can I work around this on * or do I have to go back to SIP? Mark -- regards, Mark P. Edwards TEL:+61 408 601 107 SKYPE: mark.p.edwards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC
Sorry I was not available yesterday. It was Canada Day and we got to celebrate Alberta's centenial. What did you wind up doing to get it working? Darren Wiebe [EMAIL PROTECTED] Ade Agbero wrote: Problem resolved with Astcc, certified fully working. */Juan Luis Moyano [EMAIL PROTECTED]/* wrote: Ade Agbero wrote: Finally, We have lift off, a shaky one though. I deleted my Astcc.gi and replaced it with Darren's copy posted on his website and I have finally been able to get something recorded as BILLCOST. I got it working too here with Darren's astcc.agi. And billing as expected so finally It's working. It would be nice if someone could update the cvs with Darren's astcc.agi, because the current one doesn't work, even patched.. it gets worse. Thanks for your attention Darren! -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger http://us.rd.yahoo.com/mail/uk/taglines/default/messenger/*http://uk.messenger.yahoo.com NEW - crystal clear PC to PC calling worldwide with voicemail http://us.rd.yahoo.com/mail/uk/taglines/default/messenger/*http://uk.beta.messenger.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Make Webvmail Error
Manjit Riat wrote: I did a make webvmail and I get the following error on redhat 9.0 No HTTP directory make : *** [webvmail] Error 1 I have the perl-suidperl rpm installed and apache installed Thanx . The webvmail make script isn't terribly intelligent about finding your web directory, which varies alot from one linux distrib to the next. I'm looking into making a patch for it to parameterize a little better. If you go into the make (Makefile) and search for $HTTPDIR you can sort of see where it is expecting to install it and change it by hand for now if you want. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (Simple?) ENUM Question
I've been doing some reading on ENUM and am almost ready to start testing with it. However, I have a question. As I understand things the following ENUM entry would return info for all telephone numbers of any length beginning with 00393. The Asterisk pattern would be _00393. (notice the dot) *.3.9.3.0.0 IN NAPTR 100 100 u SIP+E2U !^+*00393(.*)!sip:[EMAIL PROTECTED] . I would like to know if it's possible to have an ENUM entry that would be the equiv of the Asterisk pattern _21XX (notice no dot). -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is it possible to setup group voicemail in Asterisk?
Hello there, I'm a new Asterisk user and I wonder if it is possible to associate a voicemail box with a group of users, i.e., a single recorded message is sent to everyone in that group. If so, where can I find more information about that? Thanks in advance, Leo Burd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is it possible to setup group voicemail inAsterisk?
Hi Leo, here's a suggestion: in your dialplan (extensions.conf) send multiple users to the same mailbox (e.g. 999) if they do not pick up within 30 seconds: ; SIP Phone 100, Tom exten = 100,1,Dial(SIP/100,30) exten = 100,2,VoiceMail(999) ; SIP Phone 200, Eric exten = 200,1,Dial(SIP/200,30) exten = 200,2,VoiceMail(999) and in your voicemail.conf you do not specify a single user's email address ([EMAIL PROTECTED] or [EMAIL PROTECTED]), but a email group created in your email system then the voicemail file is sent to all the email-addresses in this group. this is not an asterisk issue, you should take a look at the manual of your email system on how to create groups. regards, roland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Burd Sent: Saturday, July 02, 2005 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Is it possible to setup group voicemail inAsterisk? Hello there, I'm a new Asterisk user and I wonder if it is possible to associate a voicemail box with a group of users, i.e., a single recorded message is sent to everyone in that group. If so, where can I find more information about that? Thanks in advance, Leo Burd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] play message to callee before connect to incoming call
Hello, i try to do the following: 1) call comes in on extension 999 2) caller should hear music (NOT MoH!!!) 3) a call should be initiated to SIP Phone 100 4) when SIP Phone 100 is answered, a sound file should be played to the user at SIP Phone 100 5) the incoming call (at extension 999) should be connected to SIP Phone 100 any suggestions on how to implement this in an easy way? Thanks in advance, Roland Zagler ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What to use h323 or oh323 ???
Hi Adeel http://www.inaccessnetworks.com/projects/asterisk-oh323 Please visit there, you will find your way. Bashir - Original Message - From: Adeel -31 To: asterisk-users@lists.digium.com Sent: Saturday, July 02, 2005 9:13 AM Subject: [Asterisk-Users] What to use h323 or oh323 ??? I m new to asterisk n i've got an IP phone that supports h323 protocol but i dont know how to configure asterisk to use it... i m comfortable in using sip iax softphones butthere is no h323.conf in /etc/asterisk/ i read that i've to compile some files but i m confused regarding h323 oh323 .. which one should i use.. plz tell me or atleast give some helpful link __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is it possible to setup group voicemail in Asterisk?
exten = 1234,5,Voicemail(u,1234234534564567) As you can see the same voicemail will go to all the users. On 7/2/05, Leo Burd [EMAIL PROTECTED] wrote: Hello there, I'm a new Asterisk user and I wonder if it is possible to associate a voicemail box with a group of users, i.e., a single recorded message is sent to everyone in that group. If so, where can I find more information about that? Thanks in advance, Leo Burd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] play message to callee before connect to incoming call
On Jul 2, 2005, at 10:23 AM, Roland Zagler wrote: Hello, i try to do the following: 1) call comes in on extension 999 2) caller should hear music (NOT MoH!!!) 3) a call should be initiated to SIP Phone 100 4) when SIP Phone 100 is answered, a sound file should be played to the user at SIP Phone 100 5) the incoming call (at extension 999) should be connected to SIP Phone 100 Rough pseudo-code follows, experiment and report your results to the list: 999 dial(SIP/100|20|m(soundfile)A(announcementfile)) OR 999 background(soundfile) 999 dial(SIP/100|20|A(announcementfile)) -- Robert Goodyear Brand Up LLC http://www.brand-up.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] play message to callee before connect to incomingcall
try this one exten = 999,1,Answer() exten = 999,2,playback(~.mp3) exten = 999,3,dial (sip/100) exten = 999,4,playbackground(~.mp3) exten = 999,h,Hangup() not sure abt playbackground should be before the dial command or after From: [EMAIL PROTECTED] on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] play message to callee before connect to incomingcall Hello, i try to do the following: 1) call comes in on extension 999 2) caller should hear music (NOT MoH!!!) 3) a call should be initiated to SIP Phone 100 4) when SIP Phone 100 is answered, a sound file should be played to the user at SIP Phone 100 5) the incoming call (at extension 999) should be connected to SIP Phone 100 any suggestions on how to implement this in an easy way? Thanks in advance, Roland Zagler ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call forwarding, most basic case
hello all,
i need some help and after trying the wiki i'm even more confused than i was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is disconnected.
as you can see, i don't want any *21 or #21, and then the number, i
dont even want the caller to be able to pick the number to forward to,
the simplest case possible, and a different extension (155) to turn
the forwarding off (for now, then i'll put them in a menu together or
something.)
so, i know i need an extension like this:
exten =154,1, Answer
exten = 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
exten =153,3, Hangup
but line 2 is giving me fits, and the documentation is a bit thin. i'm
confused about the families in the database - do i have to create
them, or are they aready there?
of course, if i'm barking up the wrong tree and there's a much simpler
way to do this please tell me.
thanks,
yair
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Re: [Asterisk-Users] play message to callee before connect to incomingcall
On Jul 2, 2005, at 12:55 PM, Mahmoud Badran wrote: try this one exten = 999,1,Answer() exten = 999,2,playback(~.mp3) exten = 999,3,dial (sip/100) exten = 999,4,playbackground(~.mp3) exten = 999,h,Hangup() not sure abt playbackground should be before the dial command or after Mahmoud: you don't pass file extension to the playback app, and there's no such app called playbackground. Plus the OP wanted the announcement to hit the callee solely. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] play message to callee before connect toincoming call
Thank you, Robert! The announcementfile plays well, but at Dial-option m i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. Before connecting to SIP Phone 100 the caller should hear a soundfile... wiki says nothing about an Dial-option to play a soundfile to the caller ;-( Roland Zagler -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Saturday, July 02, 2005 8:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] play message to callee before connect toincoming call On Jul 2, 2005, at 10:23 AM, Roland Zagler wrote: Hello, i try to do the following: 1) call comes in on extension 999 2) caller should hear music (NOT MoH!!!) 3) a call should be initiated to SIP Phone 100 4) when SIP Phone 100 is answered, a sound file should be played to the user at SIP Phone 100 5) the incoming call (at extension 999) should be connected to SIP Phone 100 Rough pseudo-code follows, experiment and report your results to the list: 999 dial(SIP/100|20|m(soundfile)A(announcementfile)) OR 999 background(soundfile) 999 dial(SIP/100|20|A(announcementfile)) -- Robert Goodyear Brand Up LLC http://www.brand-up.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Resolving groupcalls
You can also set anything you wish into the CDR variables. We came
up with the whole CDR variable thing for this exact purpose. Check
cdr_custom to log it like you want.
ie Set(CDR(GROUP)=${GROUPCALL})
/b
PS don't for get to come to cluecon!
On Jun 30, 2005, at 4:15 AM, Chris Coulthurst wrote:
Oops, sent that last one prematurely!
How about the accountcode setting? You could get user information
from
that, right?
Maybe you could send:
Asterisk -rx 'show channels'
..and when you get the data, you'd know which channels are up and
alive
(full names).
You could then re-run the command with the channel information:
Asterisk -rx 'show channel SIP/201-ec69'
..you'd get a dump, with the end looking something like this:
CDR Variables:
level 1: clid=Chris Office 201
level 1: src=201
level 1: dst=18009427433
level 1: dcontext=unlimited
level 1: channel=SIP/201-ec69
level 1: dstchannel=IAX2/provider-7
level 1: lastapp=Dial
level 1: lastdata=iax2/[EMAIL PROTECTED]/2047622726
level 1: start=2005-06-30 02:10:35
level 1: answer=2005-06-30 02:10:38
level 1: end=2005-06-30 02:10:38
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: accountcode=019284718233 --account code
unique to the user
level 1: uniqueid=1120122635.400
Anyway, maybe something like that...
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Martin Czarnowski
|Sent: Thursday, June 30, 2005 12:58 AM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Resolving groupcalls
|
|
|Hi,
|
|I'm trying to write a tool, which shows me the state of the current
|calls. For this purpose I'm reading from Pipe the Asterisk output and
|parse it... asterisk -vr | mytool
|
|However, the problem ist how to get the information about who got
this
|call in the group. The Zap channels are assigned dynamical.
|Only thing I
|can see which channel is connect to the caller but not who is
|using the
|channel.
|
|I know there is the CDR output in Master.csv. But it shows me
|the same.
|The other problem with CDR is, that it shows me the Info only
|after the
|call is finished. That's why I'm trying to parse the asterisk output.
|
|My extensions.conf looks like this..
|GROUPCALL = Zap/g2/1200021Zap/g2/1200022Zap/g2/1200023
|.
|.
|exten = s,1,Dial(${GROUPCALL})
|
|
|___
|Asterisk-Users mailing list
|Asterisk-Users@lists.digium.com
|http://lists.digium.com/mailman/listinfo/asteri|sk-users
|To
|UNSUBSCRIBE or update options visit:
| http://lists.digium.com/mailman/listinfo/asterisk-users
|
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Re: [Asterisk-Users] play message to callee before connect toincoming call
On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote: Thank you, Robert! The announcementfile plays well, but at Dial-option m i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Noted, which is why I offered option two. Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. But it doesn't REQUIRE input. Background completes when then sound file ends. Are you saying you want to move on to announcing the call to the callee as soon as it comes in while the caller is listening to the soundfile? I was following your sequential steps in your post, but if you intend to fork the process and be doing two things at once, then it's more complex. Before connecting to SIP Phone 100 the caller should hear a soundfile... wiki says nothing about an Dial-option to play a soundfile to the caller ;-( Sure it does... BACKGROUND. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux Firewall Question
On Fri, Jul 01, 2005 at 12:15:06PM -0400, Michael Stahl wrote: You should be able to do a good job with IPTABLES which is included in FC3. You can limit source destp IP and protocol, etc. Type man iptables | more for more details... Which will not get you anywhere. There are a number of relevant HOWTOs on this subject. One reasonable starting point is iptables (actually: netfilter)'s homepage: http://netfilter.org/documentation/index.html#documentation-howto Also worth mentioning is the command with the confusing name 'iptables-save', which dumps the current iptables rules and iptables-estore which restores from those rules. iptables-save is handy as a simple rules browser. But you should generate those rules in some automated way, because you'll need to somehow change them from a remote location when you'll least expect that. This is why I'd avoid most of the graphical apps such as firestarter. Shorewall mentioned earlier is also very handy. One of its design goals is to save you from mistakes and make it much less probable that you lock yourself out. However I chose to avoid it beccause it creates relatively complex and expensive rules: with Asterisk you'd want ti minimize the amounts of tests each good voip packet is subject to before it is allowed in. Shorewall's rules will subject the packet to many unnecessary rules. It may also be confusing for a simple one-interface server setup. That's why I have decided to write one myself. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DTMF Problem...
Denis Galvão - iSolve wrote: IAX doesn't use INBAND DTMF. Denis Galvão. Denis, A clarification, I hope, just to make Mark aware of the small difference. IAX sends DTMF in the signaling 'stream', that happens to follow the same path as the media. But, in IAX DTMF is not sent as voice payload (as the typical inband when is being used in SIP as one example), just happen to be sent in the same flow from source host/port to destination host/port. In SIP, when we use inband, in fact, the DTMF tone will be treated as a voice 'piece', and will be sent in RTP packets without any special considerations :-) From what I understand, that is one of the reasons with SIP inband doesn't mix well with any codec other than G.711. (compression could distort the tone). Mark, you may want to ask your provider to check the same kind of trace you did in your side. (after all, it seems much like you ARE sending the DTMF to them, is not like will get distorted :-) On 01 de jul de 2005, at 03:23, Mark Edwards wrote: Hi. Probably been asked before, but my IAX provider assures me its not their problem I have a IAX connection to a peer providing a DID. I am dialing up my number, seeing the DTMF tones come down the line, and the * IVR is just ignoring them. IAX debug output is: Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: DTMF Subclass: 1 Timestamp: 02608ms SCall: 00016 DCall: 3 [ 210.80.176.12:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 02608ms SCall: 3 DCall: 00016 [210.80.176.12:4569] for a press of 1 I am assuming this is the DTMF inband problem, but I appear unable to convince my provider. Can I work around this on * or do I have to go back to SIP? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What to use h323 or oh323 ???
On Sat, Jul 02, 2005 at 09:13:47AM -0700, Adeel -31 wrote: I m new to asterisk n i've got an IP phone that supports h323 protocol but i dont know how to configure asterisk to use it... There are currently a number of options for h323 support: - The original chan_h323 . Works only with HEAD. Requires some specific version of openh323 and pwlib - chan_oh323 . There is a certain version that works with Stable (and it should be availble in Debian debs). Another one in HEAD. Requires it's own specific version of the monster (openh323 and pwlib, IIRC) - chan_ooh323c . A new h323 stack. Does not require the monster. However it is probably the least tested of all options - chan_woomra: an attempt to hide the monster. Still requires openh323 and pwlib and something called opal on top of it, but all asterisk sees is a simple wrapper that talkes to that server through a pipe/socket. Should work with both HEAD and stable. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] play message to callee before connect toincomingcall
sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. the behaviour is just like MoH, but the problem is, that the caller has to hear a soundfile from the beginning, and MoH does not do this as the sound files inside the directory specified in musiconhold.conf are started when asterisk is started. an idea i had was to write an agi script which is called in dialplan before i issue a playback command to the caller, which is producing a callfile to call sip phone 100, but how can i connect these 2 calls afterwards? is there a possibility to connect the answered sip phone 100 to the incoming call (that is still listening to the playback command)? roland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Saturday, July 02, 2005 9:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] play message to callee before connect toincomingcall On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote: Thank you, Robert! The announcementfile plays well, but at Dial-option m i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Noted, which is why I offered option two. Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. But it doesn't REQUIRE input. Background completes when then sound file ends. Are you saying you want to move on to announcing the call to the callee as soon as it comes in while the caller is listening to the soundfile? I was following your sequential steps in your post, but if you intend to fork the process and be doing two things at once, then it's more complex. Before connecting to SIP Phone 100 the caller should hear a soundfile... wiki says nothing about an Dial-option to play a soundfile to the caller ;-( Sure it does... BACKGROUND. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Telephoning Announcements -- Suggestions?
In the subdivision where I live, we have a well that time to time has problems. Currently, our trustees call me, I take a message, and then call the people on our phone calling tree and give them the message. They, in turn, pass the message on to the rest of the residents. We have a few problems with the way this works -- people don't bother to relay the message, or, when they do relay the message, they get the message wrong. I was thinking about using an Asterisk server -- kind of a hot line for our subdivision. A trustee can call in and record an announcement, and the server would phone each resident and play the announcement to them. At other times we can use the server to play up and coming events and so on. Obviously, I need to create the menus and authenticate the trustees, record the message, and have that trigger an agi to create outbound dial files -- and some sort of cron or agi job to get statistics of who has acknowledged the message and who hasn't so that the trustee can phone the resident themselves. I'm thinking about a small PC located at one of the trustees' house (or perhaps my own) with one phone line interface. Alas, no high speed internet here so I can't use VOIP instead of a POTS phone line. Does anyone have suggestions/tips/cautions for me? Are there any applications that might help me? Should I create a program that makes the calls via the manager interface, etc. Thanks, Scott Nelson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Telephoning Announcements -- Suggestions?
Hi Scott, everything thing you are looking to do is possible. You might need to offer a bounty for someone if you don't feel comfortable to do it yourself but checkout the call agi scripts. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Scott Nelson Sent: Saturday, 2 July 2005 4:05 PM To: Asterisk Users Mailing List - Asterisk Users Subject: [Asterisk-Users] Telephoning Announcements -- Suggestions? In the subdivision where I live, we have a well that time to time has problems. Currently, our trustees call me, I take a message, and then call the people on our phone calling tree and give them the message. They, in turn, pass the message on to the rest of the residents. We have a few problems with the way this works -- people don't bother to relay the message, or, when they do relay the message, they get the message wrong. I was thinking about using an Asterisk server -- kind of a hot line for our subdivision. A trustee can call in and record an announcement, and the server would phone each resident and play the announcement to them. At other times we can use the server to play up and coming events and so on. Obviously, I need to create the menus and authenticate the trustees, record the message, and have that trigger an agi to create outbound dial files -- and some sort of cron or agi job to get statistics of who has acknowledged the message and who hasn't so that the trustee can phone the resident themselves. I'm thinking about a small PC located at one of the trustees' house (or perhaps my own) with one phone line interface. Alas, no high speed internet here so I can't use VOIP instead of a POTS phone line. Does anyone have suggestions/tips/cautions for me? Are there any applications that might help me? Should I create a program that makes the calls via the manager interface, etc. Thanks, Scott Nelson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA2000 behind NAT
Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___ HOME ___ OFFICE SPA2000 --- Linux Box -- Asterisk Box 192.168.0.253192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention I can the audio when the destination is ringing, but I can hear the voice of the person when it reponds. The person in the other side can hear me, but I can not hear anything from him. I can not hear the voice prompts for the voicemail (*98) or the operator voice, but can leave voice messages to other SIP devices and they can hear my messages. This is my sip.conf [105] username=105 type=friend secret=105 qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Guilllermo Salas HOME 105 My ext on line 1 of the Sipura is 105, and is registred with the * box: -- Registered SIP '105' at 200.93.220.27 port 5060 expires 3600 asterisk*CLI sip show peer 105 asterisk*CLI * Name : 105 Secret : Set MD5Secret: Not set Context : from-internal Language : es FromUser : FromDomain : Callgroup: (0) Pickupgroup : (0) Mailbox : [EMAIL PROTECTED] LastMsgsSent : 2 Dynamic : Yes Expire : 4 Expiry : 900 Insecure : No Nat : Always ACL : No CanReinvite : No PromiscRedir : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 200.93.xxx.xb Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Username : 105 Codecs : 0xc011f (g723|gsm|ulaw|alaw|g726|g729|h261|h263) Codec Order : (g729|g723|gsm|g726|ulaw|alaw|h261|h263) Status : UNKNOWN Useragent: Full Contact : sip:[EMAIL PROTECTED]:5060 And this is the output of sip debug peer 105 when I call to *98 (for voice messages): asterisk*CLI sip debug peer 105 SIP Debugging Enabled for IP: 200.93.xxx.xb:5060 Sip read: NOTIFY sip:sip.mydomain.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-67ea7370 From: Guillermo Salas M sip:[EMAIL PROTECTED];tag=4f2df183b116b70c To: sip:sip.mydomain.net Call-ID: [EMAIL PROTECTED] CSeq: 4 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Sipura/SPA2000-2.0.2 Content-Length: 0 10 headers, 0 lines Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-67ea7370 From: Guillermo Salas M sip:[EMAIL PROTECTED];tag=4f2df183b116b70c To: sip:sip.mydomain.net;tag=as038653dd Call-ID: [EMAIL PROTECTED] CSeq: 4 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 200.93.xxx.xb:5060 Destroying call '[EMAIL PROTECTED]' asterisk*CLI Sip read: NOTIFY sip:sip.mydomain.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-d386a279 From: Guillermo Salas M sip:[EMAIL PROTECTED];tag=4f2df183b116b70c To: sip:sip.mydomain.net Call-ID: [EMAIL PROTECTED] CSeq: 6 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Sipura/SPA2000-2.0.2 Content-Length: 0 10 headers, 0 lines Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-d386a279 From: Guillermo Salas M sip:[EMAIL PROTECTED];tag=4f2df183b116b70c To: sip:sip.mydomain.net;tag=as5099fa8f Call-ID: [EMAIL PROTECTED] CSeq: 6 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 200.93.xxx.xb:5060 Destroying call '[EMAIL PROTECTED]' asterisk*CLI I dial *98 to get into the voice message system: asterisk*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-600583f3 From: Guillermo Salas M sip:[EMAIL PROTECTED];tag=4f2df183b116b70c To: sip:[EMAIL PROTECTED];tag=as65eec750 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Max-Forwards: 70 Contact: Guillermo Salas M sip:[EMAIL PROTECTED] User-Agent: Sipura/SPA2000-2.0.2 Content-Length: 0 10 headers, 0 lines asterisk*CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-ec22067b From: Guillermo Salas M sip:[EMAIL PROTECTED];tag=4f2df183b116b70c To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username=105,realm=asterisk,nonce=47a68adb,uri=sip:[EMAIL PROTECTED],algorithm=MD5,response=8e60f592df094f9b852a59544b9da384 Contact: Guillermo Salas M sip:[EMAIL PROTECTED] Expires: 240 User-Agent: Sipura/SPA2000-2.0.2 Content-Length: 422 Content-Type: application/sdp v=0 o=- 12384 12384 IN IP4 192.168.0.253 s=- c=IN IP4 192.168.0.253 t=0 0 m=audio 16468 RTP/AVP 4 0 2 8 18 96 97 98 100 101 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101
Re: [Asterisk-Users] Telephoning Announcements -- Suggestions?
If you do add a bounty, I'll add a little bit $50US or so to it. I know there are others that have written or would like a script like this as it was discussed a few months ago on this list I believe. Darren Wiebe [EMAIL PROTECTED] Dean Collins wrote: Hi Scott, everything thing you are looking to do is possible. You might need to offer a bounty for someone if you don't feel comfortable to do it yourself but checkout the call agi scripts. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Scott Nelson Sent: Saturday, 2 July 2005 4:05 PM To: Asterisk Users Mailing List - Asterisk Users Subject: [Asterisk-Users] Telephoning Announcements -- Suggestions? In the subdivision where I live, we have a well that time to time has problems. Currently, our trustees call me, I take a message, and then call the people on our phone calling tree and give them the message. They, in turn, pass the message on to the rest of the residents. We have a few problems with the way this works -- people don't bother to relay the message, or, when they do relay the message, they get the message wrong. I was thinking about using an Asterisk server -- kind of a hot line for our subdivision. A trustee can call in and record an announcement, and the server would phone each resident and play the announcement to them. At other times we can use the server to play up and coming events and so on. Obviously, I need to create the menus and authenticate the trustees, record the message, and have that trigger an agi to create outbound dial files -- and some sort of cron or agi job to get statistics of who has acknowledged the message and who hasn't so that the trustee can phone the resident themselves. I'm thinking about a small PC located at one of the trustees' house (or perhaps my own) with one phone line interface. Alas, no high speed internet here so I can't use VOIP instead of a POTS phone line. Does anyone have suggestions/tips/cautions for me? Are there any applications that might help me? Should I create a program that makes the calls via the manager interface, etc. Thanks, Scott Nelson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Telephoning Announcements -- Suggestions?
Scott Nelson wrote: In the subdivision where I live, we have a well that time to time has problems. How about just fix the well :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC
I used the astcc.agi you posted on your website,and changed the"BRANDS" INCvalue from 6 to 60. So, I guess all you need to do now is incorporate your astcc.agi file into the CVS version. Thanks goes to you and those who contributed and responded to my HELP HELP HELP plea. From a satisfied ASTCC user, Ade.Darren Wiebe [EMAIL PROTECTED] wrote: Sorry I was not available yesterday. It was Canada Day and we got to celebrate Alberta's centenial. What did you wind up doing to get it working?Darren Wiebe[EMAIL PROTECTED]Ade Agbero wrote: Problem resolved with Astcc, certified fully working. */Juan Luis Moyano <[EMAIL PROTECTED]>/* wrote: Ade Agbero wrote: Finally, We have lift off, a shaky one though. I deleted my Astcc.gi and replaced it with Darren's copy posted on his website and I have finally been able to get something recorded as BILLCOST. I got it working too here with Darren's astcc.agi. And billing as expected so finally It's working. It would be nice if someone could update the cvs with Darren's astcc.agi, because the current one doesn't work, even patched.. it gets worse. Thanks for your attention Darren! -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger NEW - crystal clear PC to PC calling worldwide with voicemail ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Colored asterisk -R?
Hi folks, when I start asterisk directly, I get a colored CLI. When connect to a already running asterisk with asterisk -R, it's never colored, despite I'm running both from the same console (tty). Is there a way to force asterisk -R into color mode? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Enum or DUNDi
I've been reading a bit about Enum and DUNDi and still have something not very clear to me. This is a HYPOTHETICAL scenario: I have 4 asterisk servers. All of them are handling registrations of both SIP and IAX2 UAs. SIP agents are being load balanced by something like SER. I have another server in charge of load balancing IAX2 UAs registration (some sort of dynamic firewall telling the remote host what IP address to connect to) (may be there is already a IAX2 load balancer out there). The idea behind this is that any agent (whether it's SIP or IAX2) can register to any of the asterisk servers at any given time, given the load on the servers. Therefore, I would assume that all asterisks servers have a replica of the sip.conf, iax.conf, and some intelligent dial plan that would be able to handle all these. I THINK the questions I have are: 1) Since all asterisk servers can, at any time, accept registration from any SIP or IAX2 client, can they all have the same sip.conf and iax2.conf? 2) If the answer to question 1 is yes, then if I use something like ENUM or DUNDi so that all asterisk servers can learn where the actual clients are, will the asterisk servers announce their knowledge once the client registers on the asterisk server, or will the fact that their configuration is hard-coded in the sip.conf and iax2.conf assumes that all the asterisks servers will simply say, I can route call to this address, even if that client has not actually registered on a particular server? I hope my questions were clear. Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call forwarding, most basic case
Try this
http://www.voip-info.org/tiki-index.php?page=Asterisk+Tips+follow+me
I used and it works well. Rather than segregate calls based on caller ID, it
carries the caller's ID through to the forwarded phone (cell phone, or
other?), but inserts a 0 before the number, that way you know it is an *
related call. If you don't answer (don't like the caller) or can't answer,
the call goes to voice mail.
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Yair Hakak
Sent: Saturday, July 02, 2005 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] call forwarding, most basic case
hello all,
i need some help and after trying the wiki i'm even more confused than i
was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is disconnected.
as you can see, i don't want any *21 or #21, and then the number, i
dont even want the caller to be able to pick the number to forward to,
the simplest case possible, and a different extension (155) to turn
the forwarding off (for now, then i'll put them in a menu together or
something.)
so, i know i need an extension like this:
exten =154,1, Answer
exten = 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
exten =153,3, Hangup
but line 2 is giving me fits, and the documentation is a bit thin. i'm
confused about the families in the database - do i have to create
them, or are they aready there?
of course, if i'm barking up the wrong tree and there's a much simpler
way to do this please tell me.
thanks,
yair
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RE : [Asterisk-Users] Colored asterisk -R?
Asterisk -gc Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Stefan Gofferje Envoyé : samedi 2 juillet 2005 22:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] Colored asterisk -R? Hi folks, when I start asterisk directly, I get a colored CLI. When connect to a already running asterisk with asterisk -R, it's never colored, despite I'm running both from the same console (tty). Is there a way to force asterisk -R into color mode? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura SPA2000 behind NAT
Carlos, Thank you for your fast response :) , this is the output of iptables -nL on my linux box: [EMAIL PROTECTED]:/home/guillermo # iptables -nL Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination ACCEPT all -- 192.168.0.0/24 0.0.0.0/0 ACCEPT all -- 0.0.0.0/0192.168.0.0/24 Chain OUTPUT (policy ACCEPT) target prot opt source destination [EMAIL PROTECTED]:/home/guillermo # iptables -nL -t nat Chain PREROUTING (policy ACCEPT) target prot opt source destination Chain POSTROUTING (policy ACCEPT) target prot opt source destination MASQUERADE all -- 192.168.0.0/24 0.0.0.0/0 Chain OUTPUT (policy ACCEPT) target prot opt source destination This is my very-small and simple firewall script: [EMAIL PROTECTED]:/home/guillermo # cat /etc/init.d/firewall # Cargar Modulos modprobe ip_tables modprobe ip_nat_ftp modprobe ip_conntrack_ftp modprobe ip_nat_irc modprobe ip_conntrack_irc # Habilitar el forward echo 1 /proc/sys/net/ipv4/ip_forward # Flush iptables -X iptables -F iptables -X -t nat iptables -F -t nat # Habilitar nat para 192.168.0.0/24 iptables -t nat -A POSTROUTING -o eth0 -s 192.168.0.0/24 -j MASQUERADE # Permitir el forward para 192.168.0.0/24 iptables -A FORWARD -s 192.168.0.0/24 -j ACCEPT iptables -A FORWARD -d 192.168.0.0/24 -j ACCEPT # EOF On Sat, 2005-07-02 at 16:39 -0400, Carlos Alperin wrote: Guillermo, This is an issue with your router. Do you have open the ports 5060 for SIP? Also, RTP needs to be open from 16384 to 32767. Saludos, Carlos Alperin Senior System Engineer Seneca Communications, LLC [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M Sent: Saturday, July 02, 2005 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sipura SPA2000 behind NAT Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___ HOME ___ OFFICE SPA2000 --- Linux Box -- Asterisk Box 192.168.0.253192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention I can the audio when the destination is ringing, but I can hear the voice of the person when it reponds. The person in the other side can hear me, but I can not hear anything from him. I can not hear the voice prompts for the voicemail (*98) or the operator voice, but can leave voice messages to other SIP devices and they can hear my messages. This is my sip.conf [105] username=105 type=friend secret=105 qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Guilllermo Salas HOME 105 My ext on line 1 of the Sipura is 105, and is registred with the * box: -- Registered SIP '105' at 200.93.220.27 port 5060 expires 3600 asterisk*CLI sip show peer 105 asterisk*CLI * Name : 105 Secret : Set MD5Secret: Not set Context : from-internal Language : es FromUser : FromDomain : Callgroup: (0) Pickupgroup : (0) Mailbox : [EMAIL PROTECTED] LastMsgsSent : 2 Dynamic : Yes Expire : 4 Expiry : 900 Insecure : No Nat : Always ACL : No CanReinvite : No PromiscRedir : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 200.93.xxx.xb Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Username : 105 Codecs : 0xc011f (g723|gsm|ulaw|alaw|g726|g729|h261|h263) Codec Order : (g729|g723|gsm|g726|ulaw|alaw|h261|h263) Status : UNKNOWN Useragent: Full Contact : sip:[EMAIL PROTECTED]:5060 And this is the output of sip debug peer 105 when I call to *98 (for voice messages): asterisk*CLI sip debug peer 105 SIP Debugging Enabled for IP: 200.93.xxx.xb:5060 Sip read: NOTIFY sip:sip.mydomain.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-67ea7370 From: Guillermo Salas M sip:[EMAIL PROTECTED];tag=4f2df183b116b70c To: sip:sip.mydomain.net Call-ID: [EMAIL PROTECTED] CSeq: 4 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Sipura/SPA2000-2.0.2 Content-Length: 0 10 headers, 0 lines Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-67ea7370 From: Guillermo Salas M sip:[EMAIL PROTECTED];tag=4f2df183b116b70c To: sip:sip.mydomain.net;tag=as038653dd Call-ID: [EMAIL PROTECTED] CSeq: 4 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 200.93.xxx.xb:5060 Destroying call '[EMAIL PROTECTED]'
RE: [Asterisk-Users] Sipura SPA2000 behind NAT
Guillermo, I'm not very expert with iptables, but this is the issue: I don't see the forwarding from the ip of the sipura box ( that should be the only one to receive both UDP RTP traffic on the 5060 16384 to 32767 ports. On the other hand, the Asterisk box is also in an fix ip, so the traffic on UDP and the RTP only needs to be open between this two IP's. If you don't get audio, it is because UDP packets are drop when they come back to you from Asterisk, or from the other side of the call. What I see is that your firewall looks to be open on all IP's, but doesn't Mean that is in the right ports. Sorry if my advice is difuse, but as I said before, I'm not an expert on iptables. You can do an easy test, if you have an cheap Router as a Linksys or D-Link You only need to forward the ports to the ip address of the Sipura box. Is all that you need to make this work. Lamento no poder ser mas especifico. Saludos Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M Sent: Saturday, July 02, 2005 4:56 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Sipura SPA2000 behind NAT Carlos, Thank you for your fast response :) , this is the output of iptables -nL on my linux box: [EMAIL PROTECTED]:/home/guillermo # iptables -nL Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination ACCEPT all -- 192.168.0.0/24 0.0.0.0/0 ACCEPT all -- 0.0.0.0/0192.168.0.0/24 Chain OUTPUT (policy ACCEPT) target prot opt source destination [EMAIL PROTECTED]:/home/guillermo # iptables -nL -t nat Chain PREROUTING (policy ACCEPT) target prot opt source destination Chain POSTROUTING (policy ACCEPT) target prot opt source destination MASQUERADE all -- 192.168.0.0/24 0.0.0.0/0 Chain OUTPUT (policy ACCEPT) target prot opt source destination This is my very-small and simple firewall script: [EMAIL PROTECTED]:/home/guillermo # cat /etc/init.d/firewall # Cargar Modulos modprobe ip_tables modprobe ip_nat_ftp modprobe ip_conntrack_ftp modprobe ip_nat_irc modprobe ip_conntrack_irc # Habilitar el forward echo 1 /proc/sys/net/ipv4/ip_forward # Flush iptables -X iptables -F iptables -X -t nat iptables -F -t nat # Habilitar nat para 192.168.0.0/24 iptables -t nat -A POSTROUTING -o eth0 -s 192.168.0.0/24 -j MASQUERADE # Permitir el forward para 192.168.0.0/24 iptables -A FORWARD -s 192.168.0.0/24 -j ACCEPT iptables -A FORWARD -d 192.168.0.0/24 -j ACCEPT # EOF On Sat, 2005-07-02 at 16:39 -0400, Carlos Alperin wrote: Guillermo, This is an issue with your router. Do you have open the ports 5060 for SIP? Also, RTP needs to be open from 16384 to 32767. Saludos, Carlos Alperin Senior System Engineer Seneca Communications, LLC [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M Sent: Saturday, July 02, 2005 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sipura SPA2000 behind NAT Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___ HOME ___ OFFICE SPA2000 --- Linux Box -- Asterisk Box 192.168.0.253192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention I can the audio when the destination is ringing, but I can hear the voice of the person when it reponds. The person in the other side can hear me, but I can not hear anything from him. I can not hear the voice prompts for the voicemail (*98) or the operator voice, but can leave voice messages to other SIP devices and they can hear my messages. This is my sip.conf [105] username=105 type=friend secret=105 qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Guilllermo Salas HOME 105 My ext on line 1 of the Sipura is 105, and is registred with the * box: -- Registered SIP '105' at 200.93.220.27 port 5060 expires 3600 asterisk*CLI sip show peer 105 asterisk*CLI * Name : 105 Secret : Set MD5Secret: Not set Context : from-internal Language : es FromUser : FromDomain : Callgroup: (0) Pickupgroup : (0) Mailbox : [EMAIL PROTECTED] LastMsgsSent : 2 Dynamic : Yes Expire : 4 Expiry : 900 Insecure : No Nat : Always ACL : No CanReinvite : No PromiscRedir : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 200.93.xxx.xb Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Username
[Asterisk-Users] Re: passing through MWI info from SBC
Jon Radon wrote: Woah woah woah.. why not just disable SBC voicemail and have asterisk handle it? I don't understand why you would go to such great lengths when you can just have Asterisk deal with it. Jon, etc..., the issue here is her family all uses special features of SBC voicemail. E.g. Her mom leaves a VM for my wife at 10:30 PM after the baby is asleep without fear of ringing the phones and possibly waking up the baby. They use it like email. They sign in and create messages, and even send a message to multiple recipients, all without ever ringing the phone. When a VM is recieved the destination party's MWI lights and whenever this is detected, they dail the SBC VM and listen to the message, reply, etc... Since SBC VM doesn't network with Asterisk or anyone else, I need to find a way of passing the MWI info from the SBC line to the handsets in the house, else my wife won't know the she has VM from her family, and then the whole system breaks down. As I said before, everyone in her family has SBC VM. Anytime one of the husbands moved someone off of SBC, they were back in less than a week after realizing it broke how the family used voicemail. I can't get my wife off of SBC VM and onto Asterisk without moving everyone else off, which is very hard, and then I'd have to do training and answer any complaints, etc... So that's why I need to be able to have asterisk detect a message waiting on the SBC VM system and display it. Unless I can be backward compatible with the current analog phone system we have, the whole asterisk project will be nixed, and I'll have to go buy some more Venture handsets to replace the broken ones. :-( Thanks, Mike __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Colored asterisk -R?
[EMAIL PROTECTED] schrieb: Asterisk -gc I don't see a -R in that... Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Colored asterisk -R?
I'm running both from the same console (tty). Is there a way to force asterisk -R into color mode? It works fine for me to run astersik -rc Hatton ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] play message to callee before connect toinco mingcall
You can send both paties to a meetme conference with Manager Redirect. Or if you are feeling more adventurous you could load the Manager Bridge patch that I posted to the bugtracker two months ago. It allows bridging of any two existing channels together through a manager action: http://bugs.digium.com/view.php?id=4297 MATT--- -Original Message- From: Roland Zagler [mailto:[EMAIL PROTECTED] Sent: Saturday, July 02, 2005 4:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] play message to callee before connect toincomingcall sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. the behaviour is just like MoH, but the problem is, that the caller has to hear a soundfile from the beginning, and MoH does not do this as the sound files inside the directory specified in musiconhold.conf are started when asterisk is started. an idea i had was to write an agi script which is called in dialplan before i issue a playback command to the caller, which is producing a callfile to call sip phone 100, but how can i connect these 2 calls afterwards? is there a possibility to connect the answered sip phone 100 to the incoming call (that is still listening to the playback command)? roland -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Saturday, July 02, 2005 9:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] play message to callee before connect toincomingcall On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote: Thank you, Robert! The announcementfile plays well, but at Dial-option m i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Noted, which is why I offered option two. Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. But it doesn't REQUIRE input. Background completes when then sound file ends. Are you saying you want to move on to announcing the call to the callee as soon as it comes in while the caller is listening to the soundfile? I was following your sequential steps in your post, but if you intend to fork the process and be doing two things at once, then it's more complex. Before connecting to SIP Phone 100 the caller should hear a soundfile... wiki says nothing about an Dial-option to play a soundfile to the caller ;-( Sure it does... BACKGROUND. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura SPA2000 behind NAT
Hello This iptables setup won't work You need specific rules for the incoming UDP packets with status ESTABLISHED and RELATED like these simple ones Remember it's a statefull firewall. In the nat section -A POSTROUTING -p udp -m udp -m state --state RELATED -j MASQUERADE -A POSTROUTING -p udp -m udp -m state --state ESTABLISHED -j MASQUERADE And in the filter section -A FORWARD -p udp -m udp -m state --state RELATED -j ACCEPT -A FORWARD -p udp -m udp -m state --state ESTABLISHED -j ACCEPT Best regards Thierry -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Guillermo Salas M Envoyé : samedi 2 juillet 2005 22:56 À : asterisk-users@lists.digium.com Objet : RE: [Asterisk-Users] Sipura SPA2000 behind NAT Carlos, Thank you for your fast response :) , this is the output of iptables -nL on my linux box: [EMAIL PROTECTED]:/home/guillermo # iptables -nL Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination ACCEPT all -- 192.168.0.0/24 0.0.0.0/0 ACCEPT all -- 0.0.0.0/0192.168.0.0/24 Chain OUTPUT (policy ACCEPT) target prot opt source destination [EMAIL PROTECTED]:/home/guillermo # iptables -nL -t nat Chain PREROUTING (policy ACCEPT) target prot opt source destination Chain POSTROUTING (policy ACCEPT) target prot opt source destination MASQUERADE all -- 192.168.0.0/24 0.0.0.0/0 Chain OUTPUT (policy ACCEPT) target prot opt source destination This is my very-small and simple firewall script: [EMAIL PROTECTED]:/home/guillermo # cat /etc/init.d/firewall # Cargar Modulos modprobe ip_tables modprobe ip_nat_ftp modprobe ip_conntrack_ftp modprobe ip_nat_irc modprobe ip_conntrack_irc # Habilitar el forward echo 1 /proc/sys/net/ipv4/ip_forward # Flush iptables -X iptables -F iptables -X -t nat iptables -F -t nat # Habilitar nat para 192.168.0.0/24 iptables -t nat -A POSTROUTING -o eth0 -s 192.168.0.0/24 -j MASQUERADE # Permitir el forward para 192.168.0.0/24 iptables -A FORWARD -s 192.168.0.0/24 -j ACCEPT iptables -A FORWARD -d 192.168.0.0/24 -j ACCEPT # EOF On Sat, 2005-07-02 at 16:39 -0400, Carlos Alperin wrote: Guillermo, This is an issue with your router. Do you have open the ports 5060 for SIP? Also, RTP needs to be open from 16384 to 32767. Saludos, Carlos Alperin Senior System Engineer Seneca Communications, LLC [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M Sent: Saturday, July 02, 2005 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Sipura SPA2000 behind NAT Hi, I've one Sipura SPA2000 at home behind a linuxbox with two network adapters (eth0 for WAN and eth1 for LAN) doing NAT/DHCP: ___ HOME ___ OFFICE SPA2000 --- Linux Box -- Asterisk Box 192.168.0.253192.168.0.1 eth1 200.93.xxx.a 200.93.xxx.b eth0 My problem is when I try to call to any trunk or extention I can the audio when the destination is ringing, but I can hear the voice of the person when it reponds. The person in the other side can hear me, but I can not hear anything from him. I can not hear the voice prompts for the voicemail (*98) or the operator voice, but can leave voice messages to other SIP devices and they can hear my messages. This is my sip.conf [105] username=105 type=friend secret=105 qualify=no port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Guilllermo Salas HOME 105 My ext on line 1 of the Sipura is 105, and is registred with the * box: -- Registered SIP '105' at 200.93.220.27 port 5060 expires 3600 asterisk*CLI sip show peer 105 asterisk*CLI * Name : 105 Secret : Set MD5Secret: Not set Context : from-internal Language : es FromUser : FromDomain : Callgroup: (0) Pickupgroup : (0) Mailbox : [EMAIL PROTECTED] LastMsgsSent : 2 Dynamic : Yes Expire : 4 Expiry : 900 Insecure : No Nat : Always ACL : No CanReinvite : No PromiscRedir : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 200.93.xxx.xb Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Username : 105 Codecs : 0xc011f (g723|gsm|ulaw|alaw|g726|g729|h261|h263) Codec Order : (g729|g723|gsm|g726|ulaw|alaw|h261|h263) Status : UNKNOWN Useragent: Full Contact : sip:[EMAIL PROTECTED]:5060 And this
RE: [Asterisk-Users] call forwarding, most basic case
By user do you mean the caller (initiator of the call) or the recipient? If
you mean that user is the call recipient, it is very easy. The caller's call
comes to you with its Caller ID--if you want the call to go to VM, then
don't answer the call. I use this for forwarding to other PSTN lines (cell,
remote offices, etc..), although I would guess the same thing applies to SIP
phones. The dial plan variables are only necessary if you want to pass
caller ID from the originating caller through to the forwarded number. If
you don't use the variable then the caller ID you would see would be that
from the Asterisk configuration and not from the actual caller.
The 0 inserted into the number is helpful if you have calls forwarded
simultaneously to your cell phone (or other) so that you can see by the zero
that it is a forwarded call rather than a direct call to your PSTN number (I
guess you could also use this with internal calls to distinguish calls that
are forwarded from different extension numbers). If it is a forwarded call
then by not answering it, it would go to Asterisk VM. If a direct call, it
would go to whatever aswering funtion is set up on your cell phone (or other
PSTN phone). [Please reply through the mailing list]. Mike.
-Original Message-
From: Yair Hakak [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 02, 2005 5:05 PM
To: Mike Hillerbrand
Subject: Re: [Asterisk-Users] call forwarding, most basic case
hi,
thanks for your answer, but i'm not sure i understand. this dialplan says
1. call the extension
2. set a variable with the callerIDNum
3. dial out to the follow me number with a 0 prepended to the callerID
4. switch the callerID back to the original
5. go to voicemail
how does the user turn this on and off? that's what i'm trying to do
in my case. i want the user to be able to switch between asterisk
calling his extension and asterisk sending the call directly to
voicemail.
-yair
On 7/2/05, Mike Hillerbrand [EMAIL PROTECTED] wrote:
Try this
http://www.voip-info.org/tiki-index.php?page=Asterisk+Tips+follow+me
I used and it works well. Rather than segregate calls based on caller ID,
it
carries the caller's ID through to the forwarded phone (cell phone, or
other?), but inserts a 0 before the number, that way you know it is an *
related call. If you don't answer (don't like the caller) or can't answer,
the call goes to voice mail.
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Yair Hakak
Sent: Saturday, July 02, 2005 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] call forwarding, most basic case
hello all,
i need some help and after trying the wiki i'm even more confused than i
was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is disconnected.
as you can see, i don't want any *21 or #21, and then the number, i
dont even want the caller to be able to pick the number to forward to,
the simplest case possible, and a different extension (155) to turn
the forwarding off (for now, then i'll put them in a menu together or
something.)
so, i know i need an extension like this:
exten =154,1, Answer
exten = 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
exten =153,3, Hangup
but line 2 is giving me fits, and the documentation is a bit thin. i'm
confused about the families in the database - do i have to create
them, or are they aready there?
of course, if i'm barking up the wrong tree and there's a much simpler
way to do this please tell me.
thanks,
yair
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Re: [Asterisk-Users] call forwarding, most basic case
Yair,
One option is like this:
1) User dials ext. 154 to activate call forward (to voicemail)
2) User dials ext. 155 to de-activate call forward
3) Macro to check incoming calls for database entry's
4) The local extention must use that macro (or other way of screening)
1)
exten = 154,1,Answer
exten = 154,2,Set(DB(CFIM/${CALLERIDNUM})=${CALLERIDNUM})
exten = 154,3,Hangup
2)
exten = 155,1,Answer
exten = 155,2,DBdel(CFIM/${CALLERIDNUM})
exten = 155,3,Hangup
3)
[macro-test]
exten = s,1,Set(CFIM=${DB(CFIM/${ARG1})})
exten = s,2,GotoIf(${CFIM} = CFIM/${ARG1}?1|1)
exten = s,3,Dial(${ARG2}|${ARG3}|${ARG5})
exten = 1,1,VoiceMail(u${CFIM})
4)
exten = 202,1,Macro(test|${EXTEN}|SIP/202|15||tr)
Regards,
/* Ferdy */
http://asterisk.nsec.nl
info(AT)nsec(DOT)nl
Yair Hakak wrote:
hello all,
i need some help and after trying the wiki i'm even more confused than i was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is disconnected.
as you can see, i don't want any *21 or #21, and then the number, i
dont even want the caller to be able to pick the number to forward to,
the simplest case possible, and a different extension (155) to turn
the forwarding off (for now, then i'll put them in a menu together or
something.)
so, i know i need an extension like this:
exten =154,1, Answer
exten = 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
exten =153,3, Hangup
but line 2 is giving me fits, and the documentation is a bit thin. i'm
confused about the families in the database - do i have to create
them, or are they aready there?
of course, if i'm barking up the wrong tree and there's a much simpler
way to do this please tell me.
thanks,
yair
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Re: [Asterisk-Users] Problem with DTFM and complex international setup
Right... that's the one. My mistake. On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote: I don't find this option in the Makefile. I find RADIO_RELAX which is something to do with radios and DTMF. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohit Muthanna Sent: 01 July 2005 23:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem with DTFM and complex international setup Try compiling Asterisk with RELAX_DTMF (See Makefile). Mohit. On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote: We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed Asterisk so that it is possible, using a PIN code, to dial out from Asterisk onto the local PSTN. This also works occasionally. Looking at the message from the Asterisk box it is clear that sometimes numbers are missed or repeated in the dial string. This I suspect is because Asterisk is listening to the DTMF tones but the signal is dropped; sometimes the drop means that a whole digit is dropped and sometimes is means that a digit is repeated. Does anyone know how I can fix this to make it more reliable (out-of-band DTMF?) or a better way to achieve a reliable setup? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] There are 10 types of people. Those who understand binary, and those who don't. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] There are 10 types of people. Those who understand binary, and those who don't. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DTMF Problem...
From what I understand, that is one of the reasons with SIP inband doesn't mix well with any codec other than G.711. I believe it's just the ulaw/alaw PCM codecs that allow inband DTMF for SIP. Anything else will just chew it up. Mohit. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What to use h323 or oh323 ???
It's a little bit hard to compile but Try oh323 first. Although, There will be some few situations that H323 will work better than oh323. So, have both. Isamar On Sat, 2 Jul 2005, Adeel -31 wrote: I m new to asterisk n i've got an IP phone that supports h323 protocol but i dont know how to configure asterisk to use it... i m comfortable in using sip iax softphones but there is no h323.conf in /etc/asterisk/ i read that i've to compile some files but i m confused regarding h323 oh323 .. which one should i use.. plz tell me or atleast give some helpful link __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LDAP search application for Asterisk
I'm sending an Asterisk module I've written to see if anybody finds it useful or wants to provide some feedback. The name of the module is app_ldap and the application it provides is named LDAPSearch. LDAPSearch allows any kind of searches on an LDAP directory from the Asterisk dialplan. It returns its results using channel variables and it was specifically designed to be able to dial by name (users, contacts from an addressbook, etc.). One of its most interesting features is the extension it adds to the LDAP filter syntax (#= operator) to be able to match LDAP entries against a series of digits using the standard touch tone phone key mapping to make its comparisons. The syntax used by the application is the following: LDAPSearch(filter,[attr_1[:attr_2[:...]]][,max entries[,sort attr[,base DN[,scope) The documentation explaining its configuration and usage is included in doc/html/app_ldap.html. The only strange thing needed to build it is SCons (http://www.scons.org), which btw is an excellent build tool. You'll also need an LDAP client library and an LDAP server to test it. Everything has been tested using the OpenLDAP client library against both the OpenLDAP server and Microsoft Active Directory 2000. You can download it from: http://comellas.com.ar/app_ldap/app_ldap-0.1.0.tar.bz2 You'll need some knowledge of LDAP to be able to use this application successfully. Please report any problems you may have with it. -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DTMF Problem...
Thanks guys - appreciate the comments. I understand that IAX does not support inband dtmf, but I still can't fathom why 9 times out of 10 my * box is ignoring DTMF's even though they are showing up in the IAX2 protocol debug output. The really annoying thing is that I can't consistently reproduce the issue - every now and again it just works. I have logged a bug 004631 and will see what happens from here. cheers, Mark On 7/3/05, Mohit Muthanna [EMAIL PROTECTED] wrote: From what I understand, that is one of the reasons with SIP inband doesn't mix well with any codec other than G.711.I believe it's just the ulaw/alaw PCM codecs that allow inband DTMFfor SIP. Anything else will just chew it up.Mohit.___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users-- regards, Mark P. EdwardsFWD: 667917 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HW Capacity plan - How many Digium is recomended per server
Hello all, I'm evaluating a VRU project which has huge requirements. I'm looking for metrics but I haven't found anything that cover my requirements Initial estimation: Erlang 61.450 BTH 25.980 T1 req. 88 Digium HW support 4 T1 per card, assuming that each server host 3 Wildcard then 7 clustered servers will be required to support this load, but, How many Digium cards is recommended per server?, servers may be 2 CPU Xeon 2.4GHz to 3GHz Thanks, MS ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with DTFM and complex international setup
Do you think this might have an impact on http://bugs.digium.com/view.php?id=4631? Mark On 7/3/05, Mohit Muthanna [EMAIL PROTECTED] wrote: Right... that's the one. My mistake.On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote: I don't find this option in the Makefile. I find RADIO_RELAX which is something to do with radios and DTMF. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Mohit Muthanna Sent: 01 July 2005 23:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem with DTFM and complex international setup Try compiling Asterisk with RELAX_DTMF (See Makefile). Mohit. On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote: We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed Asterisk so that it is possible, using a PIN code, to dial out from Asterisk onto the local PSTN. This also works occasionally. Looking at the message from the Asterisk box it is clear that sometimes numbers are missed or repeated in the dial string. This I suspect is because Asterisk is listening to the DTMF tones but the signal is dropped; sometimes the drop means that a whole digit is dropped and sometimes is means that a digit is repeated. Does anyone know how I can fix this to make it more reliable (out-of-band DTMF?) or a better way to achieve a reliable setup? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] There are 10 types of people. Those who understand binary, and those who don't. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users--Mohit Muthanna [mohit (at) muthanna (uhuh) com] There are 10 types of people. Those who understand binary, and thosewho don't.___Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- regards,Mark P. EdwardsFWD: 667917 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audiocodes MP-108 FXO to Asterisk HELP
Does anyone have configs on the MP-108 FXO to asterisk setup? I'm doing my best with the limited docs, but having very little success. Thanks, -Darren ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting * to a Ericsson BP250
Hi List, This is somewhat off-topic since the problem itself isnt asterisk but the Ericsson BP250 I want to connect to. But since there have been a couple of posts relating in part to that system I am hoping someone can help me out. What we want to do: PRI --- BP250 --- Asterisk Currently the BP250 handles extension 000 - 899. We now would like to forward a chunk of 100 extensions to the asterisk system. We would like to use asterisk to deploy voice-mail fax services and voip/SIP connections to home-office users. Also we would like to connect to different location using IAX. The Problem: The BP250 was installed and is serviced by an Ericsson vendor. Since we have a service contract I called them and explained what I needed - a link to my internal asterisk system forwarding those 100 extensions The tech guy then wanted to know what I wanted it for. SO I explained that we wanted to deploy voice-mail and connect home users using SIP as well as IAX peerings to other locations. He said It's not possible But that he would talk to a colleague to find a solution and that he would get back to me. The following week I received and offer for 1. A voice-mail add-on für our BP250 2. A Voip interface for the BP250 alternatively they would be happy to discuss a new system based on VOIP entirely. Since I couldn't believe that there was no solution to my Problem I asked for the BP250 / RASC Documentation. According to our vendor such User Manuals do not exist and that knowledge of the system is only available through Ericsson training courses. So I am somewhat stuck in my efforts. I googled for the past week but couldn't find anything that would really help me. I still don't believe that the BP250 is incapable of forwarding those 100 extensions to an internal PRI. Does anyone know if it is possible?? And maybe even how ? If I knew how I could exert some pressure on my vendor to implement it. Otherwise I would have to take his word and my company would have to purchase the addons for the BP250 Thanks in advance to all of you. Best regards Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: TDM11B Dev Kit PCI + Asterisk CVS Head
Ok, after hours of research I finally found the problem. I found a
document from digium at
http://www.digium.com/asterisk_handbook/zapata.conf.html
which states that everything above the channel=x statement applies
to that interface which seems a little backwards to me. After
reconfiguring I have
context=internal
signalling=fxo_ks
callerid=Keith 100
channel=1
context=pstn-in
signalling=fxs_ks
callerid=asrecieved
channel=4
Just in case anyone else has the same problem.
Keith
On Jul 1, 2005, at 10:35 PM, Keith Caldwell wrote:
I just downloaded the latest asterisk and zaptel from cvs and I'm
having trouble setting the callerid on the fxs interface. From what
I've read you should be able to set it in the zapata.conf but I
can't get it to work. I had a cheap X100P clone but recently bought
the Dev Kit PCI . I've built a seperate server and have everything
up and running except everytime I go to check my voicemail I have
to enter my extension each time. Here is the exten for voicemail
that I have set up.
[vmail-main]
exten = 580,1,Wait,1
exten = 580,2,Answer
exten = 580,3,VoiceMailMain([EMAIL PROTECTED])
exten = 580,4,Wait,2
exten = 580,5,hangup
I've set up a test extension that will say cid
exten = 222,1,Wait,2
exten = 222,2,SayDigits(${CALLERIDNUM})
exten = 222,3,Hangup
It works with a sip soft phone I've tested it with.
Here is the zapata.conf
[channels]
busydetect=no
musiconhold=default
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
immediate=no
context=internal
signalling=fxo_ks
channel=1
context=pstn-in
callerid=Keith 100
signalling=fxs_ks
channel=4
callerid=asrecieved
Any Idea what I'm doing wrong?
Keith
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Re: [Asterisk-Users] IAX DTMF Problem...
On Saturday 02 July 2005 19:56, Mark Edwards wrote: Thanks guys - appreciate the comments. I understand that IAX does not support inband dtmf, but I still can't fathom why 9 times out of 10 my * box is ignoring DTMF's even though they are showing up in the IAX2 protocol debug output. The really annoying thing is that I can't consistently reproduce the issue - every now and again it just works. I have logged a bug 004631 and will see what happens from here. You aren't by any chance trying to pick up DTMF while Playback() is executing are you? Use Background() for that. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hidecallerid on analog line
chawki hammoud wrote: In the ISDN case, setcallerid or hidecallerid can be configured and I am aware that Asterisk doesn't support that on analog line. My question is whethere there is something like add-on script or hardware that will do the job. The teleco company provide the callerid service, but no private number service. If I understand the signalling architecture correctly, if your telco does not provide the service, you are out of luck. A quick explenation: ISDN (PRI for the purpose of this example) uses out of band signalling to provide what is essentially a trunk line plus a signalling channel to your PBX. Basically one of the DS0 channels on the T1 is used for signalling and called the D channel (while your voice DS0's are called B channels). Analog lines are completely different. In general, very little if any signalling is expected to be carried over an analog line. Therefore, the advanced signalling capabilities that ISDN offer are unavailable on analog simply because the telco architecture doesn't support it. Basically, in this case, *all* caller-id data originates from the telco's switch. The only way to suppress this information is to tell the switch to suppress it. Unless they offer such a service (prefixing the number with something, for example, to activate it), you have no access to the system. In this case there is nothing you can add to make this work. Best Wishes, Chris Travers Metatron Technology Consulting --- Robert Webb [EMAIL PROTECTED] wrote: On Wed, 29 Jun 2005 13:56:00 -0700 (PDT) chawki hammoud [EMAIL PROTECTED] wrote: Is there a way to hide the callerid on analog line on outgoing calls. Any ideas whether it could be done through configuration, a script or hardware. Thanks; It would have to be done through who ever provides your POTS service. They provide the caller ID to who you are calling. Some have the option to block it. Asterisk cannot be configured to do this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Chris Travers n:Travers;Chris email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival long starting time
I have installed festival and it works, but it takes a long time, till it starts. People hang up before they can listen the info. Q: 1. How can I shorten this time? 2. If two people call this extension, the second one must wait till the first one is finished. Is there a way to serve multiple poeple? 3. I have setup festival as explained in the first example. What are the advantages / disadvantages of the three methodes? 4. How long can a info be? 5. Besides weather.agi, what other application exist or what could be useful to use festival within asterisk bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sometimes yes - sometimes no (dialplan)
Robert Goodyear wrote: On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX DTMF Problem...
I hear you. background is in definitely in use in my extensions.conf here. Hopefully this partially accounts for the 10% of times when it _does_ work! ;-)Mark On 7/3/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Saturday 02 July 2005 19:56, Mark Edwards wrote: Thanks guys - appreciate the comments. I understand that IAX does not support inband dtmf, but I still can't fathom why 9 times out of 10 my * box is ignoring DTMF's even though they are showing up in the IAX2 protocol debug output. The really annoying thing is that I can't consistently reproduce the issue - every now and again it just works. I have logged a bug 004631 and will see what happens from here.You aren't by any chance trying to pick up DTMF while Playback() is executing are you?Use Background() for that.-A.___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- regards,Mark P. EdwardsFWD: 667917 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem registering Asterisk Dual Server
Here is my configuration everything would seems be straight forward, but I can not register both asterisk with each other. Both asterisks have Static IP but they are behind firewall/router, so it means I have to use Register statement. I'm a bit confused about the register statement. How can they can register with each other when both firewalls are blocking port 4569? Do I have to open ports 4569 on both servers in order to register them with each other? SERVER 1: register = Ast-1-in:[EMAIL PROTECTED] [Ast-2-in] type=user secret=password2 context=incoming disallow=all allow=ulaw ;trunk=yes [Ast-2-out] type=peer disallow=all allow=ulaw host=dynamic == SERVER 2: register = Ast-2-in:[EMAIL PROTECTED] [Ast-1-in] type=user secret=password1 context=incoming disallow=all allow=ulaw ;trunk=yes [Ast-1-out] type=peer disallow=all allow=ulaw host=dynamic -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call forwarding, most basic case
hello Mike,
we are talking about very different things here. please look at my
original mail again. I want the call recipient to be able to toggle on
and off do not disturb. I don't want the phone to ring at all.
thanks,
yair
On 7/3/05, Mike Hillerbrand [EMAIL PROTECTED] wrote:
By user do you mean the caller (initiator of the call) or the recipient? If
you mean that user is the call recipient, it is very easy. The caller's call
comes to you with its Caller ID--if you want the call to go to VM, then
don't answer the call. I use this for forwarding to other PSTN lines (cell,
remote offices, etc..), although I would guess the same thing applies to SIP
phones. The dial plan variables are only necessary if you want to pass
caller ID from the originating caller through to the forwarded number. If
you don't use the variable then the caller ID you would see would be that
from the Asterisk configuration and not from the actual caller.
The 0 inserted into the number is helpful if you have calls forwarded
simultaneously to your cell phone (or other) so that you can see by the zero
that it is a forwarded call rather than a direct call to your PSTN number (I
guess you could also use this with internal calls to distinguish calls that
are forwarded from different extension numbers). If it is a forwarded call
then by not answering it, it would go to Asterisk VM. If a direct call, it
would go to whatever aswering funtion is set up on your cell phone (or other
PSTN phone). [Please reply through the mailing list]. Mike.
-Original Message-
From: Yair Hakak [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 02, 2005 5:05 PM
To: Mike Hillerbrand
Subject: Re: [Asterisk-Users] call forwarding, most basic case
hi,
thanks for your answer, but i'm not sure i understand. this dialplan says
1. call the extension
2. set a variable with the callerIDNum
3. dial out to the follow me number with a 0 prepended to the callerID
4. switch the callerID back to the original
5. go to voicemail
how does the user turn this on and off? that's what i'm trying to do
in my case. i want the user to be able to switch between asterisk
calling his extension and asterisk sending the call directly to
voicemail.
-yair
On 7/2/05, Mike Hillerbrand [EMAIL PROTECTED] wrote:
Try this
http://www.voip-info.org/tiki-index.php?page=Asterisk+Tips+follow+me
I used and it works well. Rather than segregate calls based on caller ID,
it
carries the caller's ID through to the forwarded phone (cell phone, or
other?), but inserts a 0 before the number, that way you know it is an *
related call. If you don't answer (don't like the caller) or can't answer,
the call goes to voice mail.
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Yair Hakak
Sent: Saturday, July 02, 2005 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] call forwarding, most basic case
hello all,
i need some help and after trying the wiki i'm even more confused than i
was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is disconnected.
as you can see, i don't want any *21 or #21, and then the number, i
dont even want the caller to be able to pick the number to forward to,
the simplest case possible, and a different extension (155) to turn
the forwarding off (for now, then i'll put them in a menu together or
something.)
so, i know i need an extension like this:
exten =154,1, Answer
exten = 154,2,DBput(CFIM/${CALLERIDNUM}=Voicemail(u{CALLERIDNUM})
exten =153,3, Hangup
but line 2 is giving me fits, and the documentation is a bit thin. i'm
confused about the families in the database - do i have to create
them, or are they aready there?
of course, if i'm barking up the wrong tree and there's a much simpler
way to do this please tell me.
thanks,
yair
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RE: [Asterisk-Users] Telephoning Announcements -- Suggestions?
That's all doable. How many residents are you talking about? -- could take quite a while to call them all. Considering you have outlay in hardware, phone-cost, utilities (a 100W computer draws $5-$10/month), consider fixing that well as someone suggested. -Original Message- From: Scott Nelson [mailto:[EMAIL PROTECTED] Sent: Saturday, July 02, 2005 3:05 PM To: Asterisk Users Mailing List - Asterisk Users Subject: [Asterisk-Users] Telephoning Announcements -- Suggestions? In the subdivision where I live, we have a well that time to time has problems. Currently, our trustees call me, I take a message, and then call the people on our phone calling tree and give them the message. They, in turn, pass the message on to the rest of the residents. We have a few problems with the way this works -- people don't bother to relay the message, or, when they do relay the message, they get the message wrong. I was thinking about using an Asterisk server -- kind of a hot line for our subdivision. A trustee can call in and record an announcement, and the server would phone each resident and play the announcement to them. At other times we can use the server to play up and coming events and so on. Obviously, I need to create the menus and authenticate the trustees, record the message, and have that trigger an agi to create outbound dial files -- and some sort of cron or agi job to get statistics of who has acknowledged the message and who hasn't so that the trustee can phone the resident themselves. I'm thinking about a small PC located at one of the trustees' house (or perhaps my own) with one phone line interface. Alas, no high speed internet here so I can't use VOIP instead of a POTS phone line. Does anyone have suggestions/tips/cautions for me? Are there any applications that might help me? Should I create a program that makes the calls via the manager interface, etc. Thanks, Scott Nelson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE : [Asterisk-Users] Colored asterisk -R?
asterisk -nr n - no colors -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Saturday, July 02, 2005 3:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE : [Asterisk-Users] Colored asterisk -R? Asterisk -gc Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Stefan Gofferje Envoyé : samedi 2 juillet 2005 22:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] Colored asterisk -R? Hi folks, when I start asterisk directly, I get a colored CLI. When connect to a already running asterisk with asterisk -R, it's never colored, despite I'm running both from the same console (tty). Is there a way to force asterisk -R into color mode? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | Network Security Specialist V_/_ Heckler Koch - the original point and click interface ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
