Re: [asterisk-users] New generic sounds
Eric Wieling wrote: The word Dialing... and Calling... As in Dialing 911, please wait... and as in Calling 911, please wait... oooh boy wouldn't I be frustrated if I heard that instead of a ring when I dialed 911? what else is it gonna tell me? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New generic sounds
Philipp Kempgen wrote: Mojo with Horan Company, LLC schrieb: Eric Wieling wrote: The word Dialing... and Calling... As in Dialing 911, please wait... and as in Calling 911, please wait... oooh boy wouldn't I be frustrated if I heard that instead of a ring when I dialed 911? what else is it gonna tell me? Thank you for calling 911. All of our representatives are currently busy. Your estimated hold time is 2 hours and 15 minutes. Thank you for your patience. ... MOH Regards, Philipp Kempgen LOL haha that's what I was thinking... but Eric and Doug's comments are very true. I had 112 dialed on my cell the other day (I'm in the US, though) and accidentally hit the dial button instead of the clear button (they're very close). The instant that happened, the phone said it was dialing 'Emergency Number' so I hit the abort button immediately. Like no more than a second after I hit the dial button. It aborted immediately. 911 Emergency -- we just received a hangup from this number came calling back within 15 seconds Yes, a slight pause there would have helped me avoid that! Thanks guys! -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] noisy analog lines
To help you adjust your rxgain and txgain appropriately, you can ask your telco for the phone number for a milliwatt test line. 10 is a pretty high number for the gain, although it DOES depend on your distance from the telco and the line quality. My rxgain ranges between 2.375 and 2.945 depending on the specific channel. I'm a half mile from the center. Ian wrote: Hi all I have a small problem here. We are running Asterisk 1.4.18 and libpri 1.4.3 (how can I check the zaptel version again?) on an Intel core 2 duo 1.6Ghz, 2GB ram under Ubuntu server. We have 4 analog line coming into the box via a TDM 800 wildcard with echo cancel module and quad fxo modules. The server has been running smoothly with almost no problems for awhile now. Recently I started picking up problem with the voice clarity on our end, it sounds like a mobile going through a low signal patch. I asked the person on the other end and they can hear me loud and clear. I bumped the txgain up a notch a while back, can it be because of this? I ran a top and saw that the server only have about 16Mb free ram, can this be a possible cause? My zapata.conf and zaptel.conf are below. Thanks in advance Ian # less /etc/zaptel.conf # Autogenerated by /usr/sbin/zapconf on Fri Feb 29 16:12:07 2008 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: WCTDM/0 Wildcard TDM800P Board 1 (MASTER) fxsks=1 fxsks=2 fxsks=3 fxsks=4 # channel 5, WCTDM/0/4, no module. # channel 6, WCTDM/0/5, no module. # channel 7, WCTDM/0/6, no module. # channel 8, WCTDM/0/7, no module. # Global data loadzone= za defaultzone = za # less /etc/asterisk/zapata.conf [trunkgroups] ; define any trunk groups [channels] ;hardware channels ;default ;groep nommers en rede ; 1 = Landlyn ; 2 = Selfoon ; Span 1: WCTDM/0 Wildcard TDM800P Board 1 (MASTER) ;;; line=1 WCTDM/0/0 signalling=fxs_ks callerid=asreceived context=incoming_calls group=2 busydetect=yes usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes pulsedial=no callprogress=yes busycount=5 subscribecontext=GXP_BLF overlapdial=no toneduration=200 txgain=10.0 rxgain=10.0 channel = 1 ;;; line=2 WCTDM/0/1 FXSLS signalling=fxs_ks callerid=asreceived context=incoming_calls group=1,2 busydetect=yes usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes pulsedial=no callprogress=yes busycount=5 subscribecontext=GXP_BLF txgain=10.0 rxgain=10.0 overlapdial=yes channel = 2 ;;; line=3 WCTDM/0/2 signalling=fxs_ks callerid=asreceived context=incoming_calls group=1 busydetect=yes usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes pulsedial=no callprogress=yes busycount=5 subscribecontext=GXP_BLF txgain=10.0 rxgain=10.0 overlapdial=yes channel = 3 ;;; line=4 WCTDM/0/3 signalling=fxs_ks callerid=asreceived context=incoming_calls group=1 busydetect=yes usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes pulsedial=no callprogress=yes busycount=5 subscribecontext=GXP_BLF txgain=20.0 rxgain=10.0 overlapdial=yes channel = 4 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 license count...
Raúl Gómez C. wrote: Another silly question, In the first Digium link posted before there is a line that said *The G.729 codec works with all Digium cards*, but this license will work with a Sangoma Remora Card??? Or do I need to buy it from Sangoma??? (I don't know if the are selling G729 licenses) The codec in use for a specific channel doesn't even care if that channel exists over zapata analog or digital cards, sip channels, iax[2] channels, smoke signals, etc. If you care to use ping pong balls and the atlantic ocean as your medium, you should be able to interface with the g729 codec if you still needed to :D Although I wouldn't expect there to be much error correction inherent in the Atlantic. The codecs are modules for *asterisk* and not for the cards themselves. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 license count...
Raúl Gómez C. wrote: LOL!!! Thanks Mojo! On Sat, Apr 19, 2008 at 12:07 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: The codec in use for a specific channel doesn't even care if that channel exists over zapata analog or digital cards, sip channels, iax[2] channels, smoke signals, etc. If you care to use ping pong balls and the atlantic ocean as your medium, you should be able to interface with the g729 codec if you still needed to :D Although I wouldn't expect there to be much error correction inherent in the Atlantic. The codecs are modules for *asterisk* and not for the cards themselves. Moj Hehe you got it =D ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 license count...
Atis Lezdins wrote: The codec in use for a specific channel doesn't even care if that channel exists over zapata analog or digital cards, sip channels, iax[2] channels, smoke signals, etc. If you care to use ping pong balls and the atlantic ocean as your medium, you should be able to interface with the g729 codec if you still needed to :D Although I wouldn't expect there to be much error correction inherent in the Atlantic. I would not risk sending my data trough new cutting edge transports You mentioned. Instead I prefer to use proven technologies, and preferably documented in RFC - for example RFC 2549 IP over Avian Carriers with Quality of Service. There are even some modifications to this by using flash cards instead of paper, and that beats speed of ADSL. However that still doesn't seems best for my VoIP traffic because of latency. The codecs are modules for *asterisk* and not for the cards themselves. That's true. Regards, Atis We at Atlantic ColdStreak are pleased to offer SLA, Sea Lion Augmentation, to even our most basic transatlantic voip packages. The ping pong balls have a 99.99% 'up'time, guaranteed to float until eaten. -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor fdiles
robert boardman wrote: Hi, I have a load of files recorded with MixMonitor that are out of sync ie one leg of the call is 2-3 seconds behind the other, is this a bug in Asterisk 1.4.18, or am I possibly doing something wrong Is it possible to edit the file and re sync the a/b leg? Thanks for your help Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No, because if I recall correctly, the audio streams are not distinct in any way, i.e. left and right sides of a stereo stream. Note to anybody in particular -- If the conversations WERE mixed in this way, caller on left speaker, callee on right speaker, that would be very cool :) On playback, it would seem like a conversation was going on between two people in the room in distinct locations :D -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan extension priorities
Tilghman Lesher wrote: On Friday 18 April 2008 13:48:04 Roderick A. Anderson wrote: Second questions. Well possibly three questions. Can I create in a context a priority that skips a chunk. The example in Paul Mahler's book indicates so but I'd like to confirm, without/before testing, my code. This is desired so I can add/remove/augment dialplans/contexts that have a common jump to point. exten = 701,1, ... exten = 701,2, ... exten = 701,n, ... exten = 701,n, GotoIf( ... , 701,33) exten = 701,n, ... exten = 701,33, ... So I can add and and remove lines both before and after the GotoIf line. Yes. The priority n simply means take the last priority that was used in the dialplan and add 1 to it. So second part/question. Is there a Manual' for * 1.4.x? And the third par/question. Is the any books out or nearly so that cover * 1.4. I really hate typing in a bunch of stuff only to find it doesn't work. :-) Try O'Reilly for Asterisk: The Future of Telephony, second edition. Full disclosure: I assisted in the technical review of the book, and a few sections of the appendices are almost completely my contribution. Not to mention some of the modules. In the example, I believe the OP was using n algebraically to show the areas between 2 and 33 without realizing that n was in fact a valid priority number. Roderick, yes, you can use Goto or GotoIf to skip entire sections of dialplan logic. You either need to number your priorities fully or use labels with your n's if you choose to use n priorities. exten = s,1, exten = s,2,Goto(s,4) exten = s,3, exten = s,4, OR exten = s,1 exten = s,n,Goto(s,superjump) exten = s,n exten = s,n(superjump), I believe that's how it works, but it's from memory, so might not be quite right. Moj -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR and transfers! :(
Raúl Gómez C. wrote:
Hi list,
snip
I think this is a very common scenario so, how are you doing to handle this
situation???
What if you were to set an account code to the extension that is
requesting the long-distance call?
So person at extension 111 requests a long distance call to
808-555-1212. Lets say the receptionist dials, then, *111*8085551212...
The PBX does something like:
exten = _*XXX*NXXNXX,1,SetAccountCode(${EXTEN:1:3})
exten = _*XXX*NXXNXX,n,Dial(Zap/G1/${EXTEN:5})
then, the receptionist transfers the call to extension 111, which again
sets the account code to111.
Seems the account code would help the CDRs to make more sense? Maybe I
overlooked something :)
When anybody in the office dials 111, however, the accountcode will
still be set in my scenario. This might lead to the user at 111 being
charged for inter-office calls! So:
exten = _XXX,1,Dial(SIP/${EXTEN})
exten = _#XXX,1,SetAccountCode(${EXTEN:1})
exten = _#XXX,2,Goto(${EXTEN:1},1)
And the receptionist transfers long distance calls to #111
Moj
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Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
J. Oquendo wrote: Its fine and dandy, but the problem is you're still getting 5 packets. You're still saturated period. No QoS in the world outside of your provider and more bandwidth can alleviate that. Your provider is not going to care what you do once its passed to the CPE. So look at it logically again. QoS on a home router... Useless COMING IN. Going out... Means little but helps MINIMALLY. I think the road to success, when talking about upstream at least, is partially paved by trying to keep maximum traffic at 4 packets instead of 5, if 5 is going to saturate the link. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QOS for outgoing SIP ... Who needs QoS anyway!
J. Oquendo wrote: it does, when someone can realistically point this out please let me know so I can switch from a DS3 to T1 and save money. Use the T1 for voice and get a DSL modem for your data use? :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Nestor A. Diaz wrote: 1. I use a queue with just on sip device, one call at a time, however and without reason just after some couple of hours the sip device show in use and then no calls are transfered from the queue to the sip device, i do a sip show inuse and this is the result:asterisk -rx sip show inuse * User name In use Limit 200 0 3 * Peer name In use Limit 200 1/0 3 Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf, recreate 200 extensions and reload sip.conf Does a simple sip reload work, or do you really need to go to all the trouble of removing the peer definition? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialed number notify at invalid dial situation
Anonymous wrote:
Originally posted by: mailto:
Hi all
Now I'm making IVR sequance that is customised [mainmanu].
I wish to notify invaid command like a following
exten = i,1,playback('your command is ...')
exten = i,2,playback(${EXTEN}) ; Say 'i' oops! ;-(
exten = i,3,playback(' is incorrect! please again ')
# This exten lines are figure for instruction.
# I know to use with gsm filename.
but ${EXTEN} meaning 'i' that isn't dialed number.
Does anyone have good idea?
please help
---
Masakazu Nakano.
Dairiten.com - an open source VoIP and Ubiquitus Portal site in Japan.
http://www.dairiten.com:81/modules/news/
powered by xoops at http://www.xoops.org
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If you were to use Read to in your IVR instead of Background or
WaitExten, you could then reuse later the variable you read. I haven't
tested this to see if Goto *sends* you to the i extension when you try
to go to a non-existent extension... but *you* could :)
[mainmanu]
exten = s,1,Answer()
exten = s,n,Playback(Press 1, 2, or 3)
exten = s,n,Read(pressedbutton|Press one,two,or three|1)
exten = s,n,Goto(mainmanu,${pressedbutton},1)
exten = 1,1,blah
exten = 2,1,blah
exten = 3,1,blah
exten = i,1,NoOP(${pressedbutton})
--
*Mojo Wentworth*
HORAN COMPANY, LLC
403 Lincoln Street, Suite 210
Sitka, AK 99835
(907) 747-
(907) 747-7417 - Fax
[EMAIL PROTECTED]
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Re: [asterisk-users] dialed number notify at invalid dial situation
Mojo with Horan Company, LLC wrote:
[mainmanu]
exten = s,1,Answer()
exten = s,n,Playback(Press 1, 2, or 3)
exten = s,n,Read(pressedbutton|Press one,two,or three|1)
exten = s,n,Goto(mainmanu,${pressedbutton},1)
Oops,
shouldn't have that second priority in there. Because Read is playing
the prompt, Playback is unnecessary.
Moj
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Re: [asterisk-users] Zap Codec
In the sip peer definition, disallow=all allow=g729 allow=ulaw SHOULD work. Asterisk can't transcode g729, so it should fall on ulaw for the ZAP calls. But, when your polycoms talk with each other, as long as all necessary REINVITEs happen, they should use the 729 codec I believe. Remember however, that many options to the Dial application, like t,w,m,k (or so) REQURE asterisk to remain in the media path. moj Jeremy Mann wrote: Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail: afternoon audio file is missing
Alejandro Cabrera Obed wrote: Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I edit /etc/asterisk/voicemail.conf with envelope=yes and after that I left a message in a given mailbox near 11:00 AM. When a dial the voicemail number in order to hear the message, the Astreisk server close the cal and I get this error from te CLI: [Apr 10 14:09:08] WARNING[12955]: file.c:563 ast_openstream_full: File digits/afternoon does not exist in any format [Apr 10 14:09:08] WARNING[12955]: file.c:866 ast_streamfile: Unable to open digits/afternoon (format 0x2 (gsm)): No such file or directory [Apr 10 14:09:08] WARNING[12955]: say.c:409 wait_file: Unable to play message digits/afternoon I use language=es and I use the AsteriskSounds_ES.tar.gz (unpacked in /var/lib/asterisk/sound/es) package in order to get Spanish audio files. What can I do to correct the afternoon file error ??? That's odd, my afternoon is not in the 'digits' folder. Maybe yours isn't either; try moving afternoon.* into the digits folder...? [EMAIL PROTECTED] ~]$ locate afternoon /var/lib/asterisk/sounds/afternoon.ulaw /var/lib/asterisk/sounds/afternoon.wav /var/lib/asterisk/sounds/afternoon.ul Moj -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors
IIRC, using the utility 'screen' might work for you? Moj Mike wrote: Ah, not bad. When I start asterisk with /usr/sbin/asterisk -c I get the colors, but if I start it without -c and then connect to the console using /usr/sbin/asterisk -r I get no color. Since I want this to be running in the background, how do I fix this so I get to have my cake and eat it too? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mik Cheez Sent: Wednesday, April 09, 2008 19:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors Correct me if I'm wrong, but if you run asterisk as a service this happens. There is/was some dispute as to the fallacy of using 'safe_asterisk' anyway. Start it at the command line to see the pretty colors. Mike wrote: Hi, I`ve just made a leap from * 1.2.7 to 1.4.19. It took a while to fix all the deprecated stuff, but everything seems to be working fine now, except for a little tiny thing. I lost all color in my CLI, which makes it harder to debug. Is there something that needs doing? I didn't explicitely disable colorization from the command line, and I did try using nocolor=no in the config files. No luck. Regards, Mike -- -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help with Cisco 7960
That's what he's doing, he's asking someone with better sight to help him out and tell him what buttons to press! :) I've dialed in the dark enough times to know you don't need braille on the buttons to find the 3x4 array and use it properly without eyes. Sorry, Steve, but I had a twinge of 'what if *I* was blind' when you said that. Moj Steve Totaro wrote: In that case, I guess I would ask somone with better sight to help me out, uless they have braille on the buttons. Thanks, Steve Totaro On Sun, Apr 6, 2008 at 5:09 PM, Christian [EMAIL PROTECTED] wrote: Hello, I know how to unlock the phone and what the password is. I am asking this kind of question because i am visually impaired and cannot see the screen. many thanks, Christian On 2008-04-06 at 17:05 Steve Totaro wrote: You probably have to unlock it first. Google or voip-info.org is your friend. On Sun, Apr 6, 2008 at 5:02 PM, Christian [EMAIL PROTECTED] wrote: Hello all, I need some help with my Cisco 7960 enabling TFTP. Does anyone know what numbers to press in the menu? Or can I enable this through telnet? Many thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is this possible..
Yeah, Asterisk I think would be more than capable of doing that. It'll need some work to glue it all together. A lot of this would be written as an AGI script, and PHP or so for the webpage part of it. Sounds fun! blackwater dev wrote: We currently have an application used by the trucking industry to find freight to move. Now, the trucker does a search around Boston (for example) and gets 100 loads returned. They start at the first and call the company who has the freight, the company may say, sorry, someone just booked that so they go to the next number and call. It might take them 3-4 calls to find one that's still available. What I want to do is allow the trucker to click a check box by several loads and just click a button...it calls the first person and the system asks if its available, if yes it asks if he wants to hear the truckers credentials, if yes, it reads them and then asks if he wants to talk to the trucker and if so calls the trucker to connect the two. If the load is take then the system asks if he wants it removed from our system. If it's taken or he doesn't want to talk to the trucker, it just goes to the next number to call. In theory the trucker can click x rows and just sit back back and know when he gets a call the system has found an available load that he is approved by the broker to take. thanks, Eddie On Fri, Mar 7, 2008 at 12:53 AM, Adam Moffett [EMAIL PROTECTED] wrote: I think he's talking about an automated system. It's definitely possible with asterisk, whether or not it's a good idea. I really see this is useless since we alreadu got pricegrabbers buy.com and froogle they all list the itme in stock on the site there is really no need for a $30k a year operator to read it for the person. just my $0.02 On 3/6/08, blackwater dev [EMAIL PROTECTED] wrote: I'm head of RD for a dot com company and we are looking to create a prototype using asterisk. Basically we people who visit our site and search for goods listed by other people. Once something is found, a phone number is listed in the results and person A calls person B to see if the item is available, cost, etc. I'd like for the person searching to be able to click on 10 items they are interested in then click another button which would have asterisk start at the first, call person B, ask if the item is available, if yes, then call person A and connect the two, if not, it says thanks, and calls the next person on the list. Is this possible with Asterisk? Second, anyone looking for some contract work to help get this prototype running? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Web page to show online extensions?
faraz wrote: FOP is quite clunky! Also the flash is almost un-usable with a large number of extensions Would love to see something in PHP/Ajax which could be lightweight and fast. Last version of FOP I downloaded had a DHTML client in addition to the fat Flash client, I'm pretty happy with that. I embed it into our windows boxen's desktops, works great! Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending audio to a channel
On 3/25 Justin Newman wrote a message to the list mentioning his
SystemAnnounce application that broadcasts audio to all active channels,
I suspect his code would be easy to modify to broadcast to a single
channel...
Moj
John Hass wrote:
I have a voicemail application that users can listen to messages and
leave messages. I am looking for a way to play a beep tone to a user
when a new message is received when they are on the phone.
Here is what I have come up with:
in extensions.conf:
[beepvoicemail]
exten = 1000,1,answer()
exten = 1000,2,NoCDR()
exten = 1000,3,wait(2)
exten = 1000,4,Set(TIMEOUT(absolute)=5)
exten = 1000,5,playback(voicemail/beeps)
exten = 1000,7,SendDTMF(9)
exten = 1000,8,hangup()
exten = 2000,1,Set(TIMEOUT(absolute)=5)
exten = 2000,2,NoCDR()
exten = 2000,3,extenspy(,g(${mailbox})WqX)
exten = 2000,4,hangup()
Here is what I run:
Action: Originate
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
Context: beepvoicemail
Exten: 1000
Priority: 1
Callerid: Pager 1000
Variable: mailbox=$mailbox_user
I am using perl to originate so lets say mailbox 80085 left a message
for 8675309 $mailbox_user would contain 8675309 everyone that is logged
onto the system is part of there own spygroup the spygroup is always the
mailbox number.
This works when it doesn't crash Asterisk or the application does not
get stuck on extenspy for hours and hours.
Is there anyway to have an application that can just send audio to a
channel without having to use extenspy (it's sort of overkill for what I
need)
Thanks For the help.
--John
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Re: [asterisk-users] TDM410E card, 1 FXO module - how to dial Out
sean darcy wrote:
Kevin P. Fleming wrote:
Mojo with Horan Company, LLC wrote:
P.S. If you can't dial seven digit numbers in your area, but you miss
it, you can restore that behavior if you feel like selecting a default
area code:
exten = _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
Here, if I dial a seven digit number, asterisk dials 907 followed by my
seven digits out the phone line.
Well, sort of. This will also trigger if you dial the first 7 digits of
a 10-digit number from a device that doesn't dial 'en bloc', since there
is no longer any way to distinguish 7-vs-10 digit numbers by the number
pattern. In other words, this will work fine if you are dialing from a
SIP phone, but not if you are dialing from an analog phone.
With some trepidation, I can say my home system doesn't seem to work
that way. Using an analog phone, I can deal 3, 7, 10 or 11 numbers and
all goes as I expect.
After seeing this post, I wondered why :). It seems * waits about 4 secs
to see if all the numbers are dialed. Or is it some fortuitous order of
the includes ( vaguely remembering posts about how extensions were
searched)?
extensions.conf:
[internal]
include = outbound-local
include = outbound-long-distance
include = office-extensions
[outbound-local]
exten = _NXX,1,Answer()
exten = _NXX,n,Dial(${faxline}/${EXTEN})
[outbound-long-distance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,Dial(iax2/office/${EXTEN})
exten =_NXXNXX,1,Answer()
exten =_NXXNXX,n,Dial(iax2/office/${EXTEN})
[office-extensions]
exten =_1XX,1,Answer()
exten =_1XX,n,Dial(iax2/office/${EXTEN})
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I was led to believe that it WOULD wait a few seconds, unless the '!'
match character was on there in the dialplan. Kevin led me to believe
otherwise though. Any further input, anyone?
Moj
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Re: [asterisk-users] TDM410E card, 1 FXO module - how to dial Out
Kevin P. Fleming wrote:
Mojo with Horan Company, LLC wrote:
P.S. If you can't dial seven digit numbers in your area, but you miss
it, you can restore that behavior if you feel like selecting a default
area code:
exten = _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
Here, if I dial a seven digit number, asterisk dials 907 followed by my
seven digits out the phone line.
Well, sort of. This will also trigger if you dial the first 7 digits of
a 10-digit number from a device that doesn't dial 'en bloc', since there
is no longer any way to distinguish 7-vs-10 digit numbers by the number
pattern. In other words, this will work fine if you are dialing from a
SIP phone, but not if you are dialing from an analog phone.
I know you're not the person I should be asking Are you sure? but it
did seem like when I had an analog phone plugged into an FXS in a TDM
card that asterisk paused a bit to make sure I wasn't entering any more
digits, because I didn't use the wildcard '!' maybe? Just getting
confused, I guess -- It must have been when my IAXy was installed!
Thanks for the correction :)
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Re: [asterisk-users] voicemail custom greeting
That might not be where your voicemail files live, but if that IS, maybe asterisk currently goes 'the person at extension XYZ is [on the phone,unavailable] rather than playing greetings out of there. Do you have an Old folder in there? an INBOX folder? Then it's probably the right spot. I'd try dumping your wav file in there :) unavail, greet, and busy. Moj Mark Quitoriano wrote: On Sat, Mar 29, 2008 at 7:26 AM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: You could save it to your asterisk voicemail directory, which is often something like: /var/spool/asterisk/voicemail/your_context/your_voicemailbox_number The files used are unavail.*, busy.*, and greet.* -- Asterisk will choose the easiest-to-deal-with sound format when playing the files, so that's why there's threeish of each (WAV, wav, and gsm on my box). In my experience, I just delete the two extra ones and asterisk just makes-do with what it's got :) i can't see any unavail.* or busy.* wav or gsm files. can i just create one and put it there as unavail. and busy. ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel support removed from Asterisk
Olivier wrote: And what about SIP support ? Should it be removed in 1.6 or 1.8 ? Where have you been? SIP's been deprecated since 1.2. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Control of RTP open ports
Alejandro Cabrera Obed wrote: Can Asterisk control the RTP open ports the voip clients use ??? Or the RTP open ports depend on the voip clients ??? It depends on the VoIP clients. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK FXO hangup detection with a twist
Steve Davies wrote: Could you point me at some reference material for how this differs from KS, and what compatibility issues this might cause with other equipment? Has anyone tried this in the UK? Would BT even understand the request for ground-start signalling? KS (Kewl Start) simply lets asterisk/zaptel autodetect whether LoopStart or GroundStart is in use, so you don't have to muck with your configs as much. It's *not* something provided by the telco. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding iaxy's (iaxies?)
Steve Edwards wrote: 4) How do YOU find an Iaxy on your network? I was most easily able to find them by watching my DHCP server logs. You're right about the -b switch to ping, that's required. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM410E card, 1 FXO module - how to dial Out
Paul Whitby wrote:
Hello
Newbie question here: I have a box running Ubuntu Linux 7.10 gutsy
gibbon, and have a single Digium TDM410E card, with 1 FXO module
fitted and connected to my landline. I have it answering the landline,
directing to SIP phones, diverting to voicemail etc - and it works
great. What I can't work out is how to dial Out from this single card.
It is possible? if so, is it possible to handle both Incoming and
Outgoing calls, in the same configuration (obviously not at the same
time)? Thanks for any assistance.
Add some lines to the context your phones are in:
exten = _1NXXNXX,1,Dial(Zap/1/${EXTEN},,TWK)
exten = _0NXXNXX,1,Dial(Zap/1/${EXTEN},,TWK)
exten = _NXXNXX,1,Dial(Zap/1/1${EXTEN},,TWK)
exten = _NXX,1,Dial(Zap/1/${EXTEN},,TWK)
The fourth one only applies if you can dial seven digit numbers in your
local area, it seems phone companies are requiring ten digit dialing
more and more.
Moj
P.S. If you can't dial seven digit numbers in your area, but you miss
it, you can restore that behavior if you feel like selecting a default
area code:
exten = _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
Here, if I dial a seven digit number, asterisk dials 907 followed by my
seven digits out the phone line.
Moj
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Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume
Doug Lytle wrote: John Meksavan wrote: level high and still, the same problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it had no affect You'd want to fiddle with the txgain(Transmit) Doug He might actually want to deal with rxgain, because it could be perceived as a low volume coming into the box from the PSTN, hence being 'received' into asterisk...? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files
Sync the clocks on your asterisk boxen using NTP or whatever, and then 'touch' the call files into the future so each asterisk waits before processing it...? Might get them closer. Another option is get all three boxes into the same meetme room, waiting a few seconds for them to be ready if you want, and play the sound file to the meetme room. Moj Jerry Geis wrote: I am trying to use call files that dial and play a wave file on 3 asterisk boxes console dsp. This is working. The 3 boxes are noticeably out of sync. From using 3 different call files (time to process) I'm sure is the time delay. Is there a way to get these audios more in sync? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail custom greeting
Mark Quitoriano wrote: Hi, I have a wav file recording that i want to use on my voicemail, how can i set this up? thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You could save it to your asterisk voicemail directory, which is often something like: /var/spool/asterisk/voicemail/your_context/your_voicemailbox_number The files used are unavail.*, busy.*, and greet.* -- Asterisk will choose the easiest-to-deal-with sound format when playing the files, so that's why there's threeish of each (WAV, wav, and gsm on my box). In my experience, I just delete the two extra ones and asterisk just makes-do with what it's got :) Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two phones fail to agree on codec, asterisk at fault?
martin f krafft wrote: What's going on here? From all I can tell, the clients do the right thing, each selecting the first codec offered by asterisk (which they support), but asterisk is going a bit lala here, isn't it I think Brent's on to it there -- as he suggested, get your allow= and disallow= statements in each [peer], rather than in [global] ;) Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXy device
Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ADPCM codec and IAXy device
bilal ghayyad wrote: Hi All; I need to buy one IAXy device, but I discovered that it supports only g711 and ADPCM codec, so I was wonder that it does not support g729 or GSM?! Anyway, what is that ADPCM and how much it consumes bandwitdh? Also, asterisk support such codec? What its name in the configuration? Any advise? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I've heard, and I think Eric Wieling just confirmed it on the mailing list today, The IAXy does not support highly compressed codecs... I seem to recall that there's a space for ADPCM in the IAXy provisioning file, but I also seem to remember that this codec was not implemented in the IAXy's firmware. I've never tested it, so I don't know for sure. And if this was true, of course it could have changed by now :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXy device
I guess I've never run asterisk without ANY echo cans :) It's just that the echo was minor enough that MG2 et. al did a fine job. Thanks! Moj Eric Wieling wrote: You will never get latency on a network low enough for echo to be perceived as sidetone (like on analog). If you want to get rid of echo you must cancel echo. Mojo with Horan Company, LLC wrote: Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling users to the external domain using Asterisk
Aadilkhan Maniyar wrote: Hi All, I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17 and using it to make SIP calls. I have a configuration of Asterisk which serves the users in a particular domain, say internal.com I would like to make a SIP call from [EMAIL PROTECTED] to [EMAIL PROTECTED] I have added the following lines in extensions.conf exten = charles,1,Dial(SIP/[EMAIL PROTECTED]) exten = charles,2,Hangup Asterisk does a DNS SRV lookup and resolves the external.com to its proper IP and calls are established. But the problem with the above configuration is that I have manually added users that are in the external domain. Is there any way wherein I can call the users in external.com without adding them in the extensions.conf? Any help would be appreciated. Thanks, Aadil ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I could be wrong about this, but isn't that what a switch statement is for? So you might check to see if the dialed number is local to internal.com, then you might do a switch statement to external.com's dialplan if it wasn't local? moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk parking hold and transferdigittimeo ut
Guido Hecken wrote:
-Ursprüngliche Nachricht-
Von: Mojo with Horan Company, LLC [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 25. März 2008 23:23
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Asterisk parking hold and
transferdigittimeout
It seems that the dialplan comes into play. If your parking
lot is 700,
and you have any extension patterns that COULD begin with that, then
asterisk will wait to make SURE you're not typing 700:
Let's say that 700 is my parking lot extension.
exten = _NXXNXX,1,blahblahblah
This could match 7005551212, so asterisk waits around to make
sure I'm
not trying to find any more buttons before it accepts that I
meant 700.
As an example, if your parking lot extension was **, then
asterisk could
be pretty darn sure that that won't match anything else, and
will accept
it directly as a number to transfer too.
SOLUTION ###
Thanks for the tip, it was really the dialplan. In our * installations we
have an
outgoing context, named capi-out starting with this:
[capi-out]
exten = _XXX.,1,DoSomethingReallyImpressive()
...
After I changed it to:
[capi-out]
include = notfall ; special context for 3-digit emergency numbers
exten = _.,1,DoSomethingReallyImpressive()
...
[notfall]
exten = _11X,1,Dial(CAPI/ISDN3/${EXTEN}/b,60,tT)
...
BTW these includes are really magic, cause sometimes they don't do what you
(especially I) expext.
Please take a look at this:
EXAMPLE ###
;DIALPLAN
...
[capi-in]
include = capi-in-sub
exten = _955623XX,1,DoSomethingReallyImpressive()
...
[capi-in-sub]
exten = 9556230,1,DoSomethingReallyImpressive()
exten = 95562315,1,DoSomethingAnybodyWouldExpect()
...
Now, what happens:
Call for 9556230 reaches capi-in, is redirected through include statement to
capi-in-sub and executed.
So far so fine, expected behaviour.
Call for 95562315 reaches capi-in and is executed direct, the include
directive isn't executed at all!
Why?
Through the include statement, asterisk has to look first in capi-in-sub,
there it should
find this extension:
exten = 95562315,1,DoSomethingAnybodyWouldExpect()
...
and follow the dialplan under capi-in-sub since a valid extension was found.
What's wrong, any ideas?
Regards,
Guido Hecken
gwsNetTech
Guido Hecken
Quirrenbacher Str. 36
53639 Königswinter
Germany
fon +49(2244) 870663
fax +49(2244) 870664
mobil +49(179) 1267353
web http://www.gwsnettech.de
mailto:[EMAIL PROTECTED]
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According to this post:
http://lists.digium.com/pipermail/asterisk-dev/2007-April/027281.html
Includes are tacked on to the end of the dialplan they are mentioned
in, not where they stand.
So, since your exten = _955623XX,1,DoSomethingReallyImpressive()
matches, asterisk doesn't need to even bother checking the included context.
Moj
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Re: [asterisk-users] Broadcast/Announce app
Steve Edwards wrote: On Tue, 25 Mar 2008, Justin Newman wrote: Does anyone have use for a broadcast/annouce app? I wrote SystemAnnounce which will play a specified file to all active channels (in an UP or bridged state). This was originally to tell users to get off the system, but there are several other uses... I also wrote a new CallPickup and CallPark app, both of which work more as expected (supply actual extension numbers, etc). Let me know if there is any interest and I'll post the code. Silly question :) Yes, please post URL's. Justin, do you need hosting space? I'm excited to play with what you've got :) Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk parking hold and transferdigittimeout
Guido Hecken wrote: Hi, anyone out there with the same problems and a possible solution to the following? The functions callparking and hold use the same transferdigittimeout in features.conf. While I think 3 to 5 seconds are enough to let the user find their keys on the phone, the double ammount of time ( 2 x 5 secs) you have to wait before a call is parked and the parkposition is announced, is really too long. Did I miss something in the documentation? We are using SVN-branch-1.4-r96449. Regards, Guido Hecken gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It seems that the dialplan comes into play. If your parking lot is 700, and you have any extension patterns that COULD begin with that, then asterisk will wait to make SURE you're not typing 700: Let's say that 700 is my parking lot extension. exten = _NXXNXX,1,blahblahblah This could match 7005551212, so asterisk waits around to make sure I'm not trying to find any more buttons before it accepts that I meant 700. As an example, if your parking lot extension was **, then asterisk could be pretty darn sure that that won't match anything else, and will accept it directly as a number to transfer too. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling extension from CLI?
I think what you want is: originate LOCAL_CHANNEL application dial REMOTE_CHANNEL some examples: originate SIP/112 application dial Local/[EMAIL PROTECTED] originate SIP/112 application dial Local/[EMAIL PROTECTED] ;Echo Chamber exten originate SIP/112 application dial ZAP/g1/18005551212 Moj Vincent wrote: Hello For testing purposes, is it possible to call an extension from the command-line interface, just so I can trigger calls to AGI scripts from a test extension? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to capture destination number when receive call through ZAP
Distinctive Ringing might be available from your telecom provider.
mark morreny wrote:
Hi all,
I am using Digium PCI board to receive PSTN call through regular phone
line. It is no problem for me to receive calls, but I am not able to
capture the destination number through the ZAP channel
exten = s, n, Verbose(1|destination to ${EXTEN} )
${EXTEN} returns 's' instead of the actual destination number. Since
I have multiple phone numbers, I want to be able to route different
calls to different places.
Is this possible to do with Asterisk?
Thanks,
Mark
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Re: [asterisk-users] Access rights between AGI and Web server?
Vincent wrote: Hello I run AGI scripts from extensions.conf to save data into an SQLite database file, but this file must also be accessible in read-write mode by PHP scripts served by Lighttpd. As far as I can tell, Asterisk runs by default as root:wheel. I don't know if AGI scripts also run as root:wheel. Lighttpd runs as www:www, and if I create a new SQLite database through PHP scripts, they're created as www:wheel. What do you recommend I do so both AGI scripts and PHP scripts can work with a common SQLite file? Should I run Asterisk as www:www, www:wheel? Something else? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Don't forget that in PHP you have access to chown(), chgrp(), and chmod() -- You can change the files' permissions or uid/guid just after you create them. If the AGIs do run as root:wheel, then there should be no problem, because they should be able to access the db files? ?php $u = posix_getpwuid(posix_getuid()); $g = posix_getgrgid(posix_getgid()); echo This script is running as .$u['name'].:.$g['name']; ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie IVR: How to read() before playback() is finished?
Lee, John (Sydney) wrote: I am working on a menu to accept input from a caller like as follows: Exten = 100,1,Answer() Exten = 100,n,Playback(LONG-MESSAGE) Exten = 100,n,Read(OPTION,,2) ... When I tested it, I noticed if I start pressing a key before the Playback() is finished, the input is not buffered (simply ignored) and I have to listen to the whole message before I could re-enter again. Is there a way that I could press a key and it will be Read() before the Playback is finished? It seems like a lot of IVR system in the market can doing that and I am wondering if I have missed something in Asterisk. Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Besides the Background() app mentioned, you might like the WaitExten() app Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Access rights between AGI and Web server?
Vincent wrote:
On Mon, 24 Mar 2008 11:05:32 -0800, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
?php
$u = posix_getpwuid(posix_getuid());
$g = posix_getgrgid(posix_getgid());
echo This script is running as .$u['name'].:.$g['name'];
?
1. Here's the output:
echo exec('id') . hr;
$u = posix_getpwuid(posix_getuid());
$g = posix_getgrgid(posix_getgid());
echo This script is running as .$u['name'].:.$g['name'];
=
uid=80(www) gid=80(www) groups=80(www)
This script is running as www:www
Now, that was run under a webserver. right? not under asterisk as an
AGI? I thought we were expecting to see root:wheel :)
I understand that it shouldn't matter WHERE you run it from...
Does -w perms on a dir mean you can't modify files within the dir?
Means you can't CREATE new files in the dir, but you can modify existing
files, right? I guess what I'm wondering is if sqlite does something
like this, to keep the transaction atomic:
1. load test.sqlite to memory
2. add the record
3. dump it to disk in a tmp file, test.sqlite.asdfasdf
4. rm test.sqlite mv test.sqlite.asdfasdf test.sqlite
So I'm wondering if step 3 is breaking because go-w (and group is wheel)
on agi-bin dir?
Did you follow me?
Moj
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Re: [asterisk-users] Access rights between AGI and Web server?
Glad you got it! Moj P.S. This is not typical, right? If I do NOT have write access to a directory, I can still write to files that already exist in that directory, as long as I have write access to said files, I think... Maybe I'm just talking out loud, but it seems like if you had write access to temp.sqlite, you could do what you need to do, /unless/ sqlite tries to create a temporary file and mv it over the top of temp.sqlite, as this would require write access in the directory. Vincent wrote: On Mon, 24 Mar 2008 12:09:00 -0800, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: Now, that was run under a webserver. right? not under asterisk as an AGI? I thought we were expecting to see root:wheel :) Yup, sorry about: I forgot to say that I use a single SQLite database to share data between Asterisk and some PHP scripts. Found what it was: Even if a file is set to 664 and owned by the right user, the _directory_ in which the file lives has precedence. In this case, I just chowned it to root:www, and chmoded it to 664: [/usr/local/share/asterisk/agi-bin]# ll drwxrwxr-x 3 root www 512 Mar 24 22:05 . drwxr-xr-x 9 root wheel512 Mar 14 08:05 .. -rw-rw-r-- 1 www www 3072 Mar 24 22:05 test.sqlite Learned something new today. Thanks for the help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to capture destination number when receive call through ZAP
Rob Hillis wrote:
Distinctive ring is still not going to provide the line that was
called in the ${EXTEN} variable, so you're still stuck with dialplan
trickery to figure out which number was rung.
Mojo with Horan Company, LLC wrote:
Distinctive Ringing might be available from your telecom provider.
Of course, but I didn't think OP was stuck on using ${EXTEN}, so was
assuming they'd be great with separate contexts.
Anybody say distinctive ring detection has worked out well for them? I
haven't tried it, myself, but would appreciate the potential if it did
work reliably.
Moj
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Re: [asterisk-users] Calls to sip extensions not defined
Ricardo B. wrote:
Hi all, new to the list and this is probably a basic question and
couldn't find anything clear googling around but I don't know how to
handle calls to sip extensions not defined on sip.conf while using
pattern matching. On my example I have sip extensions 10, 11, 12, and
13 on sip.conf. On a basic extension.conf I set up a pattern starting
with 1 and a second digit should dial the sip extension entered by
the user and if the user don't pick up or is unavailable the call
goes to the user voicemail and then hangup. This basic setup can be
seen next:
[default]
exten = _1X,1,Dial(SIP/${EXTEN},10)
exten = _1X,2,VoiceMail([EMAIL PROTECTED],u)
exten = _1X,3,HangUp()
Now, what happens if the user dials 15? Then the pattern is applied
and the asterisk tries to dial that sip extension that doesn't exist,
the next step that is the voicemail also fails as 15 is not defined on
voicemail.conf and finally reaches the last step where it hang ups.
This can be seen on the cli output copied below:
astbox*CLI
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/10-0820d8e0, SIP/15|10)
in new stack
[Mar 21 19:57:48] WARNING[14321]: chan_sip.c:2860 create_addr: No such
host: 15
[Mar 21 19:57:48] WARNING[14321]: app_dial.c: dial_exec_full:
Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/10-0820d8e0,
[EMAIL PROTECTED]|u) in new stack
[Mar 21 19:57:48] WARNING[14321]: app_voicemail.c:2808
leave_voicemail: No entry in voicemail config file for '15'
-- Executing [EMAIL PROTECTED]:3] Hangup(SIP/10-0820d8e0, ) in new
stack
== Spawn extension (default, 15, 3) exited non-zero on 'SIP/10-0820d8e0'
astbox*CLI
What I am looking for is to play Playback(pbx-invalid) if a user
enters a sip extension not created. I've been testing a few options
using DIALSTATUS, AVAILSTATUS and their values but without luck as if
the sip phone 11 is not registered the pbx-invalid message.
Thansk for reading and any suggestion will be welcome.
Richard
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Not sure the scale of this job exactly, so this could be overkill - but
for small setups with too-hefty-servers, I tend to grep the
voicemail/sip config files with -c switch to test for presence of stuff
like that
^\[15\]$
with a properly constructed expression one could determine if a peername
like such is defined and not commented out
You could of course grep the cli output of sip show peer 15 to see if
the peer is reachable, if you use qualify... or if it even exists :)
just some ideas. These would probably kill a busy production box :)
Moj
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Re: [asterisk-users] Want to know Frequency and lenght of Frame
Zoa wrote: Mojo with Horan Company, LLC wrote: Aren't all the frames in asterisk 20ms long, no exceptions? Isn't ilbc the exception ? Even though the ilbc codec likes multiples of 50 for its frame size (Is this right?), I was under the impression that asterisk broke everything down to 20ms slin samples internally, unless it was just directly bridging two similarly-codeced channels. I would imagine that Sanjay meant zaptel hardware anyway, as the SIT is an in-band pattern meant for our ears. (I think that SIP would simply return a cause code out-of-band describing WHY the call failed, but would not pass any RTP audio.) Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
No, I meant if I leave this office, what to do when the cpu fan or power supply breaks on our current * box :) They might just be so worried that they'd *want* something like the 3Com V3000 :) Steve Totaro wrote: Call your dealer as I am sure you would have a support contract. Haven't really seen one break yet though. VxWorks is what runs satellites and junk ;-) Thanks, Steve Totaro On Wed, Mar 19, 2008 at 7:18 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: Steve Totaro wrote: Anyways, as to the four FXO system, I would not think twice to steer that customer to the 3Com V3000. Interesting :) When I (the tech guy) leave this office, they just *could* be asking me what to do when it breaks? lol :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to configure Voice mail for multi users.
Mian M Asif wrote:
Hi eric,
can you please tell me how can i save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1),
exten = s,n,Set(OLD_EXTEN=${EXTEN})
Then later, just use ${OLD_EXTEN}
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Re: [asterisk-users] Want to know Frequency and lenght of Frame
[EMAIL PROTECTED] wrote: I am planning to write a module to find if a Special Information was detected or not. Can anyone please help me to figure out the below fields? 1. The Frequency of a frame 2. Length of frame in milliseconds Aren't all the frames in asterisk 20ms long, no exceptions? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
An off-the-shelf 5+ year old MSI MS-6378X-L motherboard, 1.6GHz AMD, 512 RAM, 10 extensions, no more than three concurrent calls: [EMAIL PROTECTED] ~]$ uptime 11:31:45 up 103 days, 1:00, 2 users, load average: 0.00, 0.00, 0.00 But: [EMAIL PROTECTED] ~]$ sudo asterisk -rx 'core show uptime' System uptime: 9 hours, 32 minutes, 25 seconds I reboot every evening :) Drew, what's the uptime on your asterisk process on that box that's been up for 193 days? Drew Gibson wrote: Bill Andersen wrote: This is not a troll. I've used my real email because I want this taken seriously. I'm not trying to make anyone mad, I just want some real discussion on this issue. Please bare with me... I'm a USER of Asterisk. We purchased 3 commercially available Asterisk Based PBXs a little over a year ago. (I won't mention which one at this point - I don't want to bad mouth them - yet!) Two of the systems are very small (5 SIP lines/6 Polycom phones). The third is on a PRI with 30 Polycom phones. My smaller sites work pretty good. I've only had to restart Asterisk every month or so. However, my 30 station system is a continuous headache. I average a restart at least once a week. Sometimes a couple of times in the week. I'm always being called to fix something that just stopped working. I DON'T WANT TO GET INTO A Well, don't just complain, tell us your setup and we can help you get it working. This list HAS helped me figure out some of the issues. THANK YOU! But the purpose of this post is more of a fact finding mission. 1) Was choosing Asterisk for our company the wrong decision... a) IF... I expect a phone system to just work. Once it is configured, a phone system should just work with very little attention. My previous system was a Comdial with external voice mail on a DOS based PC. I LITERALLY WENT OVER 4 YEARS WITHOUT HAVING TO REMOVE POWER TO THE COMDIAL CONTROL OR RE-BOOT THE VOICE MAIL PC. b) IF... I really only need a phone system that allows an operator to answer each call and transfer them to the appropriate person. I need voice mail, but very little auto attendant features (mostly after hours). All the bells and whistles that Asterisk offers are cool, but don't bring that much to the table for our purpose. c) IF... Stability is more of an issue than high end features? 2) Are there any users out there that really DO have an Asterisk system that just works like clockwork? I'm saying, once setup, run for a year (or more) without any issues? 3) If SO, Should I simply consider a different vendor? 4) If NOT, and if my expectations are that a system SHOULD just run and run without any problems. Is Asterisk simply not my solution. Is Asterisk not REALLY ready for production. Because in my mind (as a user of phone services), dealing with the phone system, even on a MONTHLY basis, means that the system is NOT really production ready... Before we installed an Asterisk based PBX, I spent maybe 4 hours per YEAR with phone issues (setting up a new station?). Since we moved to an Asterisk based PBX, I spend 4 hours (or more) every WEEK! Am I expecting too much? Bill I don't think you are expecting too much. We have:- 130 physical extensions including 24x7 inbound call centre Debian on Dell server [EMAIL PROTECTED]:~# uptime 13:15:31 up 192 days, 23:49, 2 users, load average: 0.00, 0.01, 0.00 (Power was removed to switch to new UPS) asterisk*CLI show version Asterisk 1.2.24 built by root @ asterisk on a i686 running Linux on 2007-09-08 17:17:07 UTC asterisk*CLI show uptime System uptime: 63 days, 4 hours, 26 minutes, 40 seconds (Asterisk was restarted after queue config changes) We had a single power supply and single drive fail in one incident in Feb 2007 (one drive of RAID 1). System stayed up but was taken down for 15 minutes to swap the drive. PS was hot-swapped when it arrived later. regards, Drew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
I'm just a user :) we do real estate appraisals, and I found the time to roll my own (so to speak) pbx. We're on 1.4.4, TDM card with four FXOs. Honestly, you'll find it's easy to toss some zaptel and asterisk tarballs onto a system and compile them. You'll probably learn a lot along the way, but I won't liken it to the deep end of a swimming pool -- only halfway down! Moj Bill Andersen wrote: Thank you to everyone that replied to my post. I started to reply to most of them, but it is getting a little out of hand. Again, thank you. It actually makes me think the problem is not so much with Asterisk as it is with implementation. (My Vendor) Although this is a users list, I think it is more of a list for Asterisk resellers. I'd be interested in how many of you are simply using Asterisk as your phone system and NOT selling your services or an Asterisk based solution? Anyone? Just a user? That being said. As just a user of Asterisk, it is clear that if I want to continue with Asterisk, it looks like I really need to learn the ins-and-outs of Asterisk and ditch my pre-packaged solution. Off to Amazon for to find TFOT (I want the hard copy :) Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
Steve Totaro wrote: Anyways, as to the four FXO system, I would not think twice to steer that customer to the 3Com V3000. Interesting :) When I (the tech guy) leave this office, they just *could* be asking me what to do when it breaks? lol :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Asterisk ready for Prime-Time?
He could mean SIP or IAX Al Baker wrote: Quote This code is pre-Asterisk 1.0... It processes quite a few calls daily, I have about 1,800 DID numbers pointed at it, Are you SURE on that figure. Since you cold have at MOST 4 T1's coming into that box, 1,800 DIDs pointing to it sems like one hell of a congestion problem and a Dialplan thicker than War and Peace RE Kushner List Account wrote: Drew Gibson wrote: The box has been up since we upgraded the UPS, time before was for the disk failure in Feb 2007. Asterisk has now been up for 5 hours, 44 minutes (yes, by Murphy's Law, I'm troubleshooting a problem butrestart when convenient does not impact real uptime) but yesterday it had been up for 63+ days (last restart was for queue config changes) This is stock code on stock OS on stock hardware. We don't tweak it, poke at it, fiddle with it, update it unless necessary. We do OS and Asterisk updates on planned maintenance days infrequently) KISS and don't fsck with it! I have an Asterisk box running CVS-HEAD-08/21/04 with a T400P that currently has 17 weeks, 11 hours, 27 minutes, 51 seconds of uptime on a server that hasn't been rebooted in nearly a year. This code is pre-Asterisk 1.0... It processes quite a few calls daily, I have about 1,800 DID numbers pointed at it, there are several thousand wrong number calls a day besides the traffic I send through it. -Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Telemarketer Torture....
It must accept attachments, or we wouldn't get all these HTML messages, right? I think that's how HTML messages get through is attached :) Definitely not sure though. On another note, I heard a rumor a while back that messages over 40k might be held for moderation? Moj Drew Gibson wrote: James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Anyone have the telemarketer torture prompts? I would seriously like to revive this. - -- James Finstrom -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH3I8qdloC7YyaIOoRAlAjAJ9Hp+3SS2Z8179HecWIETp4RVDzWQCeMizp fW2JPZdYl/uxG1ziUwYnHGo= =QPbv -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi James, I have a copy of the prompts. Will the list accept attachments? regards, Drew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Desperately need help with Asterisk setup
I agree, seems odd you didn't have a [peername] section for your softphone in your sip.conf. aren't 404 errors a likely symptom of this? :) Mojo Steve Totaro wrote: Pete, You are connecting via a SIP softphone correct? Where is that in your sip.conf? On Mon, Mar 17, 2008 at 11:42 AM, Pete Kay [EMAIL PROTECTED] wrote: Hi, My sip.conf has the allow=gsm as shown in the following: [general] port = 5060 bindaddr = 0.0.0.0 context = others register =outraspace:[EMAIL PROTECTED]/outraspace nat=yes externip=58.251.75.251 localnet=192.168.1.0/255.255.255.0 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm qualify=yes All the sound files are in /var/lib/asterisk/sounds instead. Is it correct? I have tried both Wengo and xlite, but same result. I can't figure out what caused the 404 error. Any idea? Thank you so much for your help. Pete On Mon, Mar 17, 2008 at 10:34 PM, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Montag, den 17.03.2008, 21:38 +0800 schrieb Pete Kay: Hi, Here is the SIP debug output for the playback test. Thank you so much for your help. Hi Pete, [Mar 18 05:33:08] -- Executing [EMAIL PROTECTED]:1] Answer(SIP/2000-081e0738, ) in new stack [Mar 18 05:33:08] Audio is at 192.168.1.101 port 10028 [Mar 18 05:33:08] Adding codec 0x4 (ulaw) to SDP [Mar 18 05:33:08] Adding codec 0x8 (alaw) to SDP [Mar 18 05:33:08] Adding non-codec 0x1 (telephone-event) to SDP I do not see gsm here. Any reason not to allow that codec? Or did I miss something? You wrote you enabled it, so it should be here IMO. --- Transmitting (NAT) to 192.168.1.102:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK793126083;received=192.168.1.102;rport=5060 From: 2001 sip:[EMAIL PROTECTED];tag=2612560371 To: sip:[EMAIL PROTECTED];tag=as0ca1ddb0 Call-ID: [EMAIL PROTECTED] CSeq: 20 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 404 does not sound good. Please, look which sound files exist on your system (e.g. what does find /usr/share/asterisk -file vm-goodbye* say?) Another point: Which client do you use, is it Wengo or is it Xlite? Or both? In that case: Any differences? BR Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pre-pending certain digits (like 9) to an outbound call number
Like this?
exten = _XNPANXX,1,Dial(Zap/g1/9${EXTEN}|20)
Notice it matches 18005551212 and it dials 918005551212. (The 9 before the
${EXTEN})
Moj
Joshua Kinard wrote:
Hey all,
Working slowly on getting the myriad number of parts to my fax system plan
together, and one of the pieces I want to nail is how to go about, for the
outbound context (fax-out) pre-pending a digit onto a number? I.e., for all
my testing right now, I've been dialing '91XX', as the asterisk
server doing faxing junctions into my old Rolm CBX switch, and so I need the
'9' digit to dial outside numbers. However, for deployment, I'd like to save
the users confusion and have the server automatically append that leading '9'
digit.
That possible by chance? I assume it is, but off the top of my head, it
didn't seem intuitive. Below is the exten lines for my [fax-out] context,
followed by some test exten lines that wound up failing:
exten = _X.,1,Dial(Zap/g1/${EXTEN}|20)
exten = _X.,n,Busy
exten = _X.,n,Hangup
; Test appending 9?
;;exten = _9XNPANXX,1,Dial(Zap/g1/${EXTEN}|20)
;;exten = _9XNPANXX,n,Busy
;;exten = _9XNPANXX,n,Hangup
I was trying to do some basic matching to the NANP formula to catch when
someone accidentally mistypes a number, but that didn't match up and asterisk
was complaining that no exten lines in the [fax-out] context were matching.
Also, is it possible offhand to block the dialing of certain numbers in the
same context? I.e., just as a check, to block faxes to 900 numbers? I
believe my Rolm CBX will do this for me, as it's got a pretty extensive list
of area codes and exchanges that are known to be sinister in nature
pre-loaded (probably needs updating, though...), but I figured that if I
could block it in asterisk, to do so. Save the Rolm a wee bit of processing
and all (it is old, and probably senile...)
That, and I'd like to filter accidental '9911...' dials using this technique
(which would dial 911 emergency, and that wouldn't be good, since I doubt
faxes are a good method of calling in an emergency (unless they have a color
fax and can discern that the red ink really isn't red ink...)).
Thoughts anyone? Thanks!,
--Josh
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Re: [asterisk-users] DID number
http://vitelity.net has 800# DIDs for $0.50/month plus usage (which is like $0.02/min I think)This price has been very bearable for me to just experiment with -- I can ask anyone I want to call me to test my services and they don't have to worry about toll charges Moj Mike wrote: hey Folks, Just curious if anyone has suggestions on how one can get a near FREE(I hope) DID number. I am experimenting with asterisk, for home use. thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem transferring calls some of the times
Are you using buttons on your phone to effect the transfer, or are you using codes defined in features.conf? Moj Ian wrote: Hi, Mojo with Horan Company, LLC said the following on 20-Feb-08 09:31 PM: Is it AFTER you have parked a call? Meaning, for example, you transfer an incoming call to 700. No problem. Later, when it's picked up from 701, can it NOT be transferred again? Moj No I don't park the call. The call comes in, and gets redirected to our receptionists phone, from there it gets transferred to another extension (the bosses secratary) and then gets transferred (to the boss). now the problem, sometimes that transfer fails, other times the call dont even want to leave the receptionists phone. The big thing about this problem is that it comes and goes, like yesterday we didn't have a problem, and I did not change a thing. Ian Ian wrote: Hi All Sorry to be a bother again but seems like I just cant get away from the problems. This time my problem is that *sometimes* a user cant transfer a call from one extension to another, I have narrowed down the problem to it only happening to calls from outside the internal system. The wierd thing about the problem is that it comes and goes one moment the user can transfer, and the next call he can't. I am running: * Asterisk 1.4.17 * Zaptel 1.4.7.1 * Libpri 1.4.3 Using the following phones and firmware * Grandstream GXP2000 (with ext pad) : 1.1.4.14 * Grandstream BT200 : 1.1.4.18 I have set up the phones to log debug logs to a syslog server, I am still trying to figure out what exactly the log says. Is it an * problem, or Grandstream problem Does anyone know if I am able to see the keysequence the user types into the phone (just in case it might even be a user made problem), I have tried scanning though the logs of a failed call, but could not see any lines that can be a keypress, or maybe I am looking in the incorrect spot? Your help will be greatly appreciated. Let me know if, in any way, I can shed some more light on the subject. Thanks in advance Ian -- www.vddi.co.za http://www.vddi.co.za/ I Coetzee IT Tegnikus Telefoon: 012 664 2300 Selfoon : 079 522 6519 Faks: 012 644 2902 E-pos : [EMAIL PROTECTED] Skype : vddb_igcoetzee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem transferring calls some of the times
Sorry, I jut got your other message stating the steps your boss' secretary uses to transfer calls, so this question's time is past. I'm curious if the 'flash' button is the only way those phones can do a transfer. Do they have any other transfer keys, or could you try the featuremap codes? Our polycom transfer buttons have always just worked, but my users, for some reason, all felt more comfortable using DTMF keypresses... dunno why :) So we all press ## to do a blind transfer now, or ** to auto-park to first parking space. Moj Mojo with Horan Company, LLC wrote: Are you using buttons on your phone to effect the transfer, or are you using codes defined in features.conf? Moj Ian wrote: Hi, Mojo with Horan Company, LLC said the following on 20-Feb-08 09:31 PM: Is it AFTER you have parked a call? Meaning, for example, you transfer an incoming call to 700. No problem. Later, when it's picked up from 701, can it NOT be transferred again? Moj No I don't park the call. The call comes in, and gets redirected to our receptionists phone, from there it gets transferred to another extension (the bosses secratary) and then gets transferred (to the boss). now the problem, sometimes that transfer fails, other times the call dont even want to leave the receptionists phone. The big thing about this problem is that it comes and goes, like yesterday we didn't have a problem, and I did not change a thing. Ian Ian wrote: Hi All Sorry to be a bother again but seems like I just cant get away from the problems. This time my problem is that *sometimes* a user cant transfer a call from one extension to another, I have narrowed down the problem to it only happening to calls from outside the internal system. The wierd thing about the problem is that it comes and goes one moment the user can transfer, and the next call he can't. I am running: * Asterisk 1.4.17 * Zaptel 1.4.7.1 * Libpri 1.4.3 Using the following phones and firmware * Grandstream GXP2000 (with ext pad) : 1.1.4.14 * Grandstream BT200 : 1.1.4.18 I have set up the phones to log debug logs to a syslog server, I am still trying to figure out what exactly the log says. Is it an * problem, or Grandstream problem Does anyone know if I am able to see the keysequence the user types into the phone (just in case it might even be a user made problem), I have tried scanning though the logs of a failed call, but could not see any lines that can be a keypress, or maybe I am looking in the incorrect spot? Your help will be greatly appreciated. Let me know if, in any way, I can shed some more light on the subject. Thanks in advance Ian -- www.vddi.co.za http://www.vddi.co.za/ I Coetzee IT Tegnikus Telefoon : 012 664 2300 Selfoon: 079 522 6519 Faks : 012 644 2902 E-pos : [EMAIL PROTECTED] Skype : vddb_igcoetzee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 301/501 Keymapping
That can be found in the monstrous admin guide for the phone, seemly in Section 3.1.7 in my ancient version 1.5.0 document. It shows me that on the 501, that button is 9 instead of 23. http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip301.html There's a link to the administrator's guide down there under Setup Maintenance Documents. Moj Rob Schall wrote: I know how to remap a key on a polycom 301 and 501 But does anyone know of a list of mapping keys? For example, the Do Not Disturb on a 301 is #23. I got that one by just guessing though. Thanks, Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get a clean, basic configuration?
Delete extensions.ael too, unless you're using AEL instead of the dialplan Mindaugas Kezys wrote: We do: in modules.conf: noload = pbx_ael.so noload = pbx_dundi.so noload = res_config_pgsql.so noload = res_smdi.so in extensions.conf delete every context [default], [demo], whatever in sip.conf, iax.conf delete all peer/users if any Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vincent Sent: Thursday, February 21, 2008 4:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to get a clean, basic configuration? Hello I'm using a standard Asterisk install with default settings, and when I run reload, I see that Asterisk fetches configuration information from a lot more sources than just my extensions.conf and sip.conf. For instance: -- Registered indication country 've' -- Registered indication country 'za' -- Setting default indication country to 'us' == Parsing '/etc/asterisk/features.conf': Found == Parsing '/etc/asterisk/adsi.conf': Found == Parsing '/etc/asterisk/dundi.conf': Found == Parsing '/etc/asterisk/extensions.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4094 pbx_load_module: Starting AEL load process. [Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4101 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. etc. How can I go and trim things down? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get a clean, basic configuration?
Tzafrir Cohen wrote: Delete extensions.ael too, unless you're using AEL instead of the dialplan extensions.ael is harmless on its own. It seemed that the default extensions.ael created some demo contexts and extensions that might befuddle a new user, I could be wrong ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem transferring calls some of the times
Is it AFTER you have parked a call? Meaning, for example, you transfer an incoming call to 700. No problem. Later, when it's picked up from 701, can it NOT be transferred again? Moj Ian wrote: Hi All Sorry to be a bother again but seems like I just cant get away from the problems. This time my problem is that *sometimes* a user cant transfer a call from one extension to another, I have narrowed down the problem to it only happening to calls from outside the internal system. The wierd thing about the problem is that it comes and goes one moment the user can transfer, and the next call he can't. I am running: * Asterisk 1.4.17 * Zaptel 1.4.7.1 * Libpri 1.4.3 Using the following phones and firmware * Grandstream GXP2000 (with ext pad) : 1.1.4.14 * Grandstream BT200 : 1.1.4.18 I have set up the phones to log debug logs to a syslog server, I am still trying to figure out what exactly the log says. Is it an * problem, or Grandstream problem Does anyone know if I am able to see the keysequence the user types into the phone (just in case it might even be a user made problem), I have tried scanning though the logs of a failed call, but could not see any lines that can be a keypress, or maybe I am looking in the incorrect spot? Your help will be greatly appreciated. Let me know if, in any way, I can shed some more light on the subject. Thanks in advance Ian -- www.vddi.co.za http://www.vddi.co.za/ I Coetzee IT Tegnikus Telefoon : 012 664 2300 Selfoon : 079 522 6519 Faks : 012 644 2902 E-pos : [EMAIL PROTECTED] Skype : vddb_igcoetzee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need to Connect offices in Dubai and Pakistan
How about a computer with a copy of asterisk at each end? You'd need good network connectivity between them. A recent post by Gordon Henderson states that GSM calls can take up to 32K/sec with IP overhead, less probably if they are trunked into an IAX connection. For landline quality, Gordon states you'd need 80K/sec per call. Assuming you can meet these requirements (consider the number of concurrent calls) then go for it. To use existing analog phones at your offices you'd need some FXS ports or a channel bank. Or you could upgrade to IP phones. Mojo Kashif Naeem wrote: Hello All We need to connect our client's offices located in Dubai and Pakistan. Suggest us some economical solution. -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com http://www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] MSN: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk config file online editor
No problem, hope it gets you where you need to be :)
Moj
Anton Krall wrote:
This is a good start, thx Moj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: martes, 19 de febrero de 2008 01:35 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk config file online editor
Like 15 lines of php and html?
?php
$fn = /etc/asterisk/extensions.conf;
if ($_REQUEST['action'] == write $_REQUEST['contents'] != )
{
rename($fn, $fn...date(U));
$fp = fopen($fn, wt);
fwrite($fp, $_REQUEST['contents']);
fclose($fp);
}
?
form
h1?=$fn?/h1
textarea name=contents?php include $fn ?/textarea
input type=hidden name=action value=write
input type=submit value=Save File input type=reset value=Reset
/form
Security holes galore! clean it up a bit :) And check on permissions
issues, that your httpd can write to the file.
Moj
Anton Krall wrote:
Guys, Im looking for a good text file editor for asterisk config files
that can be embedded on a web page for online editing (on an
interface),
any recommendations?
Anton Krall
Direccion General
Intruder Consulting
A Division of IntruderEnterprises S.A. de C.V.
www.Intruder.com.mx
www.IntruderStore.com.mx
Tel. 3872-2200 ext. 201
Tel. 01-800-INTRUDER (01-800-468-7833)
Email: [EMAIL PROTECTED]
Como lo estoy haciendo? Contacte a mi Director:
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Re: [asterisk-users] SiP call generator
Sure, run 10 concurrently and see how it sounds. Scale up by a factor of 10 until it sounds crappy then start scaling down. shrug At least I think that's what Atis meant. Moj Tzafrir Cohen wrote: On Wed, Feb 20, 2008 at 11:51:55PM +0200, Atis Lezdins wrote: Test of audio quality is something I'm not really sure how to do. Run tests, and ChanSpy() them? See at which point decrease of quality becomes hearable. Manually??? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk config file online editor
Like 15 lines of php and html?
?php
$fn = /etc/asterisk/extensions.conf;
if ($_REQUEST['action'] == write $_REQUEST['contents'] != )
{
rename($fn, $fn...date(U));
$fp = fopen($fn, wt);
fwrite($fp, $_REQUEST['contents']);
fclose($fp);
}
?
form
h1?=$fn?/h1
textarea name=contents?php include $fn ?/textarea
input type=hidden name=action value=write
input type=submit value=Save File input type=reset value=Reset
/form
Security holes galore! clean it up a bit :) And check on permissions
issues, that your httpd can write to the file.
Moj
Anton Krall wrote:
Guys, Im looking for a good text file editor for asterisk config files
that can be embedded on a web page for online editing (on an interface),
any recommendations?
Anton Krall
Direccion General
Intruder Consulting
A Division of IntruderEnterprises S.A. de C.V.
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Re: [asterisk-users] DialPlan help with Analog Fax Machine
Jim Duda wrote: == Spawn extension (incoming-dial, fax, 0) exited non-zero on 'Zap/4-1' Yes, I DO think that's a little odd. It should be priority 1, shouldn't it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium stopped TDM400P production: alternatives??
There are some tdm400 cards on ebay, http://search.ebay.com/tdm400 Moj Giorgio Incantalupo wrote: Hi, Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Thank you! Giorgio. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Touch monitor file name format
Will Set(MONITOR_FILENAME=/blahblah/filename) work for you?
Moj
Jaap Winius wrote:
Hi list,
The default file name format for touch monitor (automon) recordings is:
auto-${EPOCH}-caller-calee
It's possible to use the ${TOUCH_MONITOR} variable to change the
'caller-calee' part, but what about the 'auto-${EPOCH}-' part?
I've been trying to use ${MONITOR_EXEC_ARGS} to add some more commands
after the somix sequence for mp3 conversion. This should work, but
I've so far failed to produce any mp3 files because I'm not able to
predict the above epoch number. If I could alter 'auto-${EPOCH}-', or
if it was stored in a variable I could use, then I'm sure my plan will
succeed.
Any ideas?
Thanks,
Jaap
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Re: [asterisk-users] Telephone line signaling configuration in Egypt for FXO ports
bilal ghayyad wrote: [channels] rxgain=15.0 txgain=15.0 Wow! Is this necessary? Is this something you took from a sample config somewhere, or numbers that you arrived at through trial and error? They seem a bit high in my experience, *but* I've never been to Egypt before, and I sure wouldn't be surprised if this was necessary -- just wanted your confirmation ;) Mojo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending a message from inside voicemailmain.
Tilghman Lesher wrote: On Monday 11 February 2008 11:55, Mojo with Horan Company, LLC wrote: William F. Acker WB2FLW +1-303-722-7209 wrote: Thanks for mentioning contexts. All of us are in the default context. So I started playing around with the options pertaining to contexts. I found that if I uncommented searchcontexts=yes, I could send from inside. The explanation says that if the parameter is set to no, only the default context will be searched, which should have worked for me. By setting it to yes, I now have lots of happy users. Thanks again. Instead of searchcontexts=yes, can you put your context name on the end of the voicemail box number? [EMAIL PROTECTED] and see if that works as well :) Maybe its quicker than searchcontexts, I don't know :) How exactly do you suggest typing the @ symbol and letter characters using a DTMF touchpad? This was a bug, plain and simple, and it's now fixed in SVN 1.4, and it will be in 1.4.19, whenever that is released. And now I realize that Directory REQUIRES the vm-context parameter. /me leaves his foot where it is for a while so he just can't say anything ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending a message from inside voicemailmain.
Tilghman Lesher wrote: On Monday 11 February 2008 11:55, Mojo with Horan Company, LLC wrote: William F. Acker WB2FLW +1-303-722-7209 wrote: Thanks for mentioning contexts. All of us are in the default context. So I started playing around with the options pertaining to contexts. I found that if I uncommented searchcontexts=yes, I could send from inside. The explanation says that if the parameter is set to no, only the default context will be searched, which should have worked for me. By setting it to yes, I now have lots of happy users. Thanks again. Instead of searchcontexts=yes, can you put your context name on the end of the voicemail box number? [EMAIL PROTECTED] and see if that works as well :) Maybe its quicker than searchcontexts, I don't know :) How exactly do you suggest typing the @ symbol and letter characters using a DTMF touchpad? This was a bug, plain and simple, and it's now fixed in SVN 1.4, and it will be in 1.4.19, whenever that is released. haha :) Yes I'll admit that one came out a bit wrong. What I was meaning to say was to pass the context to the Directory() application in the dialplan, but I was thinking of Voicemail([EMAIL PROTECTED]) terminology when I typed it, not Directory(), which takes just the context. Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending a message from inside voicemailmain.
William F. Acker WB2FLW +1-303-722-7209 wrote: Thanks for mentioning contexts. All of us are in the default context. So I started playing around with the options pertaining to contexts. I found that if I uncommented searchcontexts=yes, I could send from inside. The explanation says that if the parameter is set to no, only the default context will be searched, which should have worked for me. By setting it to yes, I now have lots of happy users. Thanks again. Instead of searchcontexts=yes, can you put your context name on the end of the voicemail box number? [EMAIL PROTECTED] and see if that works as well :) Maybe its quicker than searchcontexts, I don't know :) Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pulling my hair out over voicemail
Don't forget to 1000,1,Answer the call Moj John Von Essen wrote: Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes to voicemail, and message is stored on server. I created an extension to retrieve the messages: exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain And that worked. Granted, everything is still defaults, so when I dial 1000, I get the Comedian Mail greeting, then it prompts for mailbox and password, then I get the menu. Now, here is how it gets weird. Today I go and setup a new second SIP phone, and proceed to set it up for voicemail. Inbound calls go to voicemail properly when nobody answers, but I cant retrieve the messages. When I dial extension 1000, its rings for 2 seconds, then just goes silent. No greeting, no mailbox prompts, nothing. Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. -john ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Asterisk using C
Soumya Kat wrote: Hi, I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8 system. Asterisk works fine for me and I can log into Asterisk-GUI and monitor asterisk. What I would like to know is how to get information such as SIP users, number of SIP connections and traffic associated with those from asterisk using a C Code. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Although the code is pretty messy, you can see how I got this sort of information from asterisk for my monitor project AstSee. Source is available at http://www.astsee.com/ Mojo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 300 Echo
After Andrew's suggestion, if that isn't the problem, spend some more time on OSLEC to be darn sure it's operating properly -- that thing works like a champ for my crappy lines! Moj Brent Davidson wrote: We're deploying an asterisk-based phone system at all of our branch offices in an effort to eliminate long-distance costs incurred from the constant branch to branch calls. We're using the Snom 300's at all offices for the desk phones and X100P cards to interface to 2 analog lines. I'm having a problem tuning all the echo out of the system. So far two branches are using the new system and they are both reporting echo on both incoming and outgoing calls. The echo seems to be confined to the Snom 300 phones and is not heard by the person on the zap line. The echo is only the voice of the person using the Snom phone. There doesn't seem to be any echo of the analog line audio. I have tried adjusting the gain of the lines, turning on echo cancellation, Turning on echo training and nothing seems to work. At one of the branches, I re-compiled asterisk and zaptel using the OSLEC drivers and that doesn't seem to have had any effect either. What am I missing? Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need good voicemail documentation
Jaap Winius wrote: * Why can't I delete any voicemail messages? (Response: Message undeleted.) * Why can't I listen to the messages in the Old folder? * Why can't I use the advanced options? (Response: I'm sorry, I did not understand your response.) * How come if I put [EMAIL PROTECTED] in my phone's context of sip.conf, do I get an error? (CLI: ...Remote host can't match request NOTIFY to call...) I don't think you will find any of these in an asterisk voicemail documentation project. You need to examine the CLI with sufficient verbosity, and ask for our help if you don't understand what's in there. These are all problems that would be VERY atypical to have with asterisk's voicemail. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk G722
rachid wrote: Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. It's known as 'wideband' audio, to provide you with a keyword you can use to track asterisk's implementation of it. From voip-info[1]: ... Speex - which supports 8, 16 and 32 kHz sample rates and is open source freeware. So if you are looking for wideband VoIP, look at /Speex/. and A caveat for Asterisk hacks: The internal guts of Asterisk are still substantially geared for 8 kHz sampling, so arriving wideband signals will end up downsampled. I understand this is pervasive enough in the core code that it is not likely to evolve past 8 kHz for some time to come. So just tell your client to not /ask/ for 8kHz audio. I wonder, in a SIP reinvite situation, the UAs would re-choose codecs, wouldn't they? So even if asterisk didn't support 16kHz for, say, IVRs, two UAs, once REINVTEd, probably *would* choose 16kHz if they agreed on it.Am I right on REINVITEs providing opportunity for UAs to battle out codecs again? Moj [1] http://www.voip-info.org/wiki/view/Wideband+VoIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk G722
Mojo with Horan Company, LLC wrote: So just tell your client to not /ask/ for 8kHz audio. As Kevin just pointed out, apparently you do NOT have to tell your client to ask for 8kHz audio. May I ask what client you are using? Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P phone won't ring
Have you swapped the phones between the FXS ports to see if the phone rings? Moj Shane Wegner wrote: Hello all, I have two handsets connected to FXS ports on a TDM400P, both GE models but one rings and the other does not. The phone models are not identical. The phone which doesn't ring on the TDM does ring when connected to a regular POTS line and I tried connecting another phone to the port and it rings fine. So, I'm presuming the TDM is ringing the handsets somehow differently than the telco in a way which most phones like but this particular one doesn't deal with. On the wctdm module I've tried ringboost=1 and fastringer=1 but neither made a difference, not that fastringer=1 should as I'm in Canada where we use 20HZ I believe. Just wondering if there are any other settings in the Zaptel modules or Asterisk to change the ring properties. Best, Shane ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Console/dsp, makes me sound like a Dalek
Thomas Kenyon wrote: The server that I will need to get this running on has an 82801EB/ER (ICH5/ICH5R) AC'97 sound controller (and no expansion space left to put another card in). Just a suggestion, don't forget there are USB audio devices available that work with linux, you may have an extra usb port ;) Not suggesting it because I've tried this for asterisk, just thinking outside the box :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT POlycom question
randulo wrote: On Feb 4, 2008 9:34 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: In my recollection, [EMAIL PROTECTED] worked when I tried it, without sip or a colon. xxx could be anything at all. I noted this behavior back in 2006: http://lists.digium.com/pipermail/asterisk-users/2006-March/146393.html Note, that was with asterisk 1.2 I am running asterisk 1.2 although it shouldn't matter because I do not want to go thru asterisk (hence the OT) the number I put in the directory or dial in manually is of the style [EMAIL PROTECTED] (no colon or sip) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users For me, that worked fine back in 2006 exactly as you have it. I have url-dialing turned off right now so can't double-check. Sorry it's not working for you. There are quite a few places that could break IMO. On second thought, I tried another angle: I pointed the phone's microbrowser at a page containing the following: a href=tel://[EMAIL PROTECTED]Joe Smith/abr a href=tel://[EMAIL PROTECTED]John Smith/a And it worked like a charm. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT POlycom question
randulo wrote: I have an IP 500 and I have tried everything I can think of to call a SIP number like this :[EMAIL PROTECTED] without the call trying to go through the registered servers. I even added an emergency server and number in the sip.cfg. Dialing the number manually or in the directory appears to try the call but then immediately shows Number so, no such luck. Is anyone doing this and if so, how do I do it? thx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In my recollection, [EMAIL PROTECTED] worked when I tried it, without sip or a colon. xxx could be anything at all. I noted this behavior back in 2006: http://lists.digium.com/pipermail/asterisk-users/2006-March/146393.html Note, that was with asterisk 1.2 Mojo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-gui installation hangs
A COMPLETE shot in the dark, but: Tomasz Zieleniewski wrote: [Feb 4 09:33:09] == Parsing '/home/asterisk/asterisk/1.4/pbx/etc/asterisk/manager.conf': [Feb 4 09:33:09] Found If this is where you've got everything installed, i.e. with a base of /home/asterisk/asterisk/1.4/pbx/, maybe: [Feb 4 09:33:15] WARNING[3304]: app_system.c:107 system_exec_helper: Unable to execute '/sbin/zapscan.bin' Should be /home/asterisk/asterisk/1.4/pbx/sbin/zapscan.bin ? Just a thought. A soft link may help you, or find what is trying to spawn that zapscan program and fix it. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Losing CALLERID{dnid}
what about astdb? is that too much of a global variable?
moj
Arjan Kroon | Mobillion wrote:
Hi,
I’m using videocalling on asterisk 1.4.10.
When I setup the videocall with exten =
n,1,h324m_gw([EMAIL PROTECTED]), I loose the variable DNID
(${CALLERID(dnid)})
Before the videocall is set up, this variable is filled and after this
videocall this variable is empty.
Also all local variables are empty.
If al look at the A-number (${CALLERID(num)} this variable is not
empty after the videocall is set up.
Does anybody know how to ‘remember’ the variable ${CALLERID(dnid)} ?
A global variable is not an option.
Kind Regards.
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Re: [asterisk-users] CallerID shows wrong values in manager interface
Trust me, I don't WANT you to look at my code, it's butt-ugly! lol, lust kidding... -- but at http://www.astsee.com/ you can download the source code to my AstSee project -- it may provide some insight into what needs to be (or CAN be) gleaned from asterisk. I struggled with all this a year or more ago and manged to get it to work fairly as expected. A problem you might notice is that I believe I mistakenly update my internal arrays before asterisk's manager interface has sent a complete packet... argh But you can see me dealing with NewState, NewExten, NewChannel, etc etc and what I do with them :) Moj Devraj Mukherjee wrote: Thanks all :) Appreciate it. On Feb 1, 2008 12:04 PM, Ex Vito [EMAIL PROTECTED] wrote: I've struggled with this recently. In short: - Observed behaviour is expected as of asterisk 1.2 and later, as previously described by Mojo - If you want to get the caller id for the channel calling (dialling) into that channel for that specific Newstate: Ringing event, you can use the 'o' flag to the Dial command; in this case you'll get old asterisk 1.0 behaviour -- do you really want to depend on such an old behaviour ? well I decided I didn't... - Otherwise, you'll need to track other events (IIRC, at least, Dial, AgentCalled, Newstate, etc) in the AMI so as to know who is calling who at a given instant - BEWARE: if memory serves me right (search the list archives in the Nov/Dec timeframe), the behaviour is not 100% homogeneous for different channel types SIP, ZAP, mISDN, IAX, etc. What this means for a simple Dial() from one channel to the other is that a) at times you get the Dial event first then the Newstate: Ringing event; and that b) with other/different orig/dest channel types you'll get the events in the reverse order... Nothing much but: i) you'll have to track them either way and ii) it reveals that the AMI events aren't 100% clean!!! :/ -- exvito On Feb 1, 2008 12:08 AM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: The snippet is asterisk telling you I'm just letting you know that the correct caller id for Channel: SIP/103-098500d8 is CallerID: 103 This is absolutely correct, it's just not a piece of information you expected to be receiving at that point. You probably also received a packet like that with the following: Channel: SIP/101- CallerID: 101 telling you, again, the caller id for only that channel. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID shows wrong values in manager interface
The snippet is asterisk telling you I'm just letting you know that the correct caller id for Channel: SIP/103-098500d8 is CallerID: 103 This is absolutely correct, it's just not a piece of information you expected to be receiving at that point. You probably also received a packet like that with the following: Channel: SIP/101- CallerID: 101 telling you, again, the caller id for only that channel. Moj Devraj Mukherjee wrote: CallerIDName: unknown State: Ringing Event: Newstate Privilege: call,all Uniqueid: 1201748091.843 Channel: SIP/103-098500d8 CallerID: 103 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: dtmf mode
My polycoms all have dtmfmode=rfc2833 and they work fine on both asterisk's IVRs and external ones brought to me from the PSTN: [120] type=friend context=internalaugmented secret=a_secret host=dynamic *dtmfmode=rfc2833* Moj Jarga Jallow wrote: Hi, I am having trouble making a selection when I call a number and need to make a selection to go to an extension with my polycom phones 301. Anybody have an idea how to fix this problem? Thanks in advance. Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 www.2mcctv.com http://www.2mcctv.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
Steve Edwards wrote: Or, as a quick dirty... DATE=$(date +%F-%H-%M-%S) COUNT=$(sudo /usr/sbin/asterisk -r -x sip show channels | wc -l) echo $DATE $COUNT /tmp/channel-counts in a shell script executed every second in cron. every *second* from cron? how the heck would I you do that? sub-minute accuracy from cron is something I don't know how to do. Maybe it's a different version of cron...? The only way I would achieve that would be to run something every minute that self-perpetuated for the rest of that minute... for x in `seq 1 58`; do ( DATE=$(date +%F-%H-%M-%S) COUNT=$(sudo /usr/sbin/asterisk -r -x sip show channels | wc -l) echo $DATE $COUNT /tmp/channel-counts ) sleep 1s done which is honestly very messy. I promise I'm not being sarcastic. I actually *am* curious if there are versions of cron that will go sub-minute. Moj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peak number of calls?
Steve Edwards wrote: in a shell script executed every second in cron. every *second* from cron? how the heck would I you do that? sub-minute accuracy from cron is something I don't know how to do. Sheese -- that's what I get by trying to type without putting down the crack pipe :) You're right -- the * in the first column of your crontab means minutes, not seconds. Ok, I'm NOT on the crack pipe then ;) I was wondering. Sticking to the slimy hack i described! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Console app
What does 'make menuselect' let you choose? Under #3, Channel Driveers, does chan_alsa have XXX through it so you can't select it? does chan_oss have XXX? This would indicate to you that the pieces of alsa or oss asterisk would need are not installed properly. Moj Gilberto Nunes wrote: Hi all I build an Asterisk, with asterisk 1.4.16.1 source. I have notice, that the console app don't appear on CLI... Is theres some options to turn on, when I compile asterisk? Thanks... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attended transfers manager or phone
Some phones have the auto-answer ability. So your phone could have two extensions, one for normal use and one for auto-answer use. Redirect or Originate, as you were, to the auto-answer extension on the phone. So the phone would already put itself offhook, and asterisk would continue and build up the other end of the bridge. Polycom soundpoint phones, for example, but many others have this ability. an example extension setup might be exten = 110,1,Dial(SIP/110) exten = #110,1,SipAddHeader(...whatever your phone needs to make it autoanswer) exten = #110,2,Dial(SIP/110) Don't know about phones that allow ip control of their state, though. Moj Christian Ejlertsen wrote: Well I'm sure this issue has been bean up a few time since it's one of the only ones I can't find a real simple answer to. I'm trying to find away to do attended transfers through the manager interface, for a pc switchboard / Agent client solution, but so far coming up short. The action Originate is part of the solution, but what really I want is the phone being taken off-hook and then being able to dial the number without having to answer the dial-back first. 1. One solution, though an ugly one, would be using Originate, but use a phone that has some sort tcp/ip interface that allows for taking the phone off-hook. 2. A Better solution would be using a phone that allows dialling and taking the phone off-hook on-hook etc. via some tcp/ip interface. 3. Yet another solution, though I do not favour this one since I really don't want to maintain the sip phone code, would be programming a soft sip phone with all the bells and whistles and adding the switchboard functionality to that (name searching, status email so on and so forth. In the end all I need is just a software or hardware phone, sip/iax, which can be told via tcp/ip to go off-hook, on-hook, dial, transfer and perhaps status requests. If such a phone exists that would do the trick, the rest is manageable via the Asterisk Manager console. I'm guessing some people have messed with this problem before so I hope that someone has some information about this kind of thing :) Thank you in advance Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Directories Used by Asterisk
It is when you type 'make install' that these directories get created. 'make linux26' IS obsolete as another poster mentioned. broadband Voice wrote: I successfully obtained the Asterisk code and extracted them into /usr/src. When I make and install asterisk, zaptel, libpri etc. Are they supposed to move automatically into their respective directories? I cannot find: /etc/asterisk/ /usr/lib/asterisk/modules/ /var/lib/asterisk Do I have to manually create them or this is failed install? Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Digit Map
Jerry Jones wrote: Yours should work if you wait long enough for t to timeout. I think your digit map needs a T on the end of it if you want to allow timeouts for that match. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_echo.c
I would GUESS that if this line is removed, asterisk is settling on slin codec for the channel and does not try to negotiate anything better? Hence it will work without it. Mojo Bhrugu Mehta wrote: hi, all I have test echo application for just fun. I can'nt understand why this is used below in .c file, format = ast_best_codec(chan-nativeformats); ast_set_write_format(chan, format); ast_set_read_format(chan, format); without this this application work fine. then why this is used. any suggestion?? Bhrugu mehta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Digit Map
Doug Lytle wrote: Michael Munger wrote: only connects me to a dial tone and says Enter More Digits. It actually says this? I would say then it's not the phone, but your phone system's programming. The Polycoms don't verbally say anything, at least not the ones I deal with. Doug No it doesn't SAY it -- the polycoms put on the screen Enter more digits. I think it's when what you've dialed doesn't match an entry in your digit map, or possibly when asterisk says that extension does not match anything So try: 011XXT in your digit map, meaning 011 plus at least six digits, consider it good because you can't know how long the string will be in advance. You want to allow for the smallest possible, which I suspect would be a three digit country code, like in Tonga (676) -- and you want to allow for the longest possible, to account for stuff like in Tajikistan: 992 37962 is BEFORE the local number, so you'd want 011+ at least 9 Xs following it 011XX -- Tricky! ** Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
