Re: [asterisk-users] extensions.conf vs. AEL
In my opinion the dialplan isn't where that logic belongs. /b On Oct 3, 2007, at 12:32 AM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? Thanks! __Yehavi: ___ --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] Having problems posting to the list
but they do, apparently On 10/2/07, robert boardman [EMAIL PROTECTED] wrote: Hi All I'm having problems posting to this list, no bounces the mails just dont show any advice how to get the postings through is there filtering? 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 ___ --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 slow dial out - TDM400P
On 10/2/07, Ken Williams [EMAIL PROTECTED] wrote: Any suggestions would be greatly appreciated. Try removing all the echo cancel stuff just to see if that makes any difference at all. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple iax users on the same host
Hi i'm setting up a hylafax server, using iaxmodem to talk with asterisk (asterisk and hylafax are both on the same lan). Can i setup on the same host (Hylafax) multiple iax accounts ? (each account is used by a iaxmodem instance). The account can be on the same port or should i change the port for each iax account? Thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --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] extensions.conf vs. AEL
Where would you suggest all the logic goes Brian? Garth Garth van Sittert BSc (Physics Computer Science) - Main: 08600 BITCO Phone: +27 (0)11 875 6900 Fax:+27 (0)11 875 6901 Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] MSN:[EMAIL PROTECTED] Web:www.bitco.co.za Brian West wrote: In my opinion the dialplan isn't where that logic belongs. /b On Oct 3, 2007, at 12:32 AM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? Thanks! __Yehavi: ___ --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] meetme conference using g729?
On 10/3/07, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 02 October 2007 16:55:52 Brian West wrote: On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote: anyway still if there's a hack for meetme to work with g729 codec this won't be an issue. So is there a hack or patch that i can use any codec for meetme? tnx You still do not understand. It doesn't matter if the call coming in is g729 you must transcode it to signed linear, mix the frames and then code it back into g729 you end up with quality loss doing that. Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. yeah i still don't understand. this is what i want to do. I want asterisk not to compress and decompress codecs. so either i can use SLIN as my codec for my SIP or IAX. or i can remove SLIN codec in meetme and change it to g729a so there's is no compression and decompression. do you get what i want to do? 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] Secondary Dialtone and selecting a specific line from Zap/g
I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone, not a dialtone provided by Asterisk. Al lists wrote: Correction, on FXO port not FXS, second, read his email first: Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP Phone or Broadtel IP Phone, so if user select that button then he will be sure that his outside call will be via that specific line. Just assign a key on your phone to dial that extension, and you will have dial tone on selected line, then as a traditional PBX you can send any digits to your provider. On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: ignorepat continues dialtone after a leading digit has been dialed on FXS ports. How does ignorepat help this guy? Al lists wrote: ignorpat is your friend On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote: Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Simply don't use groups. Use channels directly. To dial via the specific Zaptel channel NN, use Zap/NN Am I missing anything? Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ --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] Queue members, URI.
On Tuesday 02 October 2007 19:30:44 Thomas Kenyon wrote: Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/number@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the party. (and if need be, there is the ability in queues to run a script on connection iirc). I'm migrating to Local/number@context right now (from Agent/ channels), and it seems to me that Local channels doesn't show (busy) in show queues. This will probably require for me to do some overhead work for correctly displaying agent status in monitoring software, but i think i will be able to do it by combining core show channels with show queues. I'm not sure is it related to Agent channels that could accept only one call or SIP channel status. I would expect queue to show even Local channel as busy if there is active call trough it. I think this really can't be accomplished by dialplan logics, as dialplan is not executed upon show queues Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --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] Having problems posting to the list
Mee too, a lot of the messages I'm sending seem to disappear. l. In data Tue, 02 Oct 2007 22:38:26 +0200, robert boardman [EMAIL PROTECTED] ha scritto: Hi All I'm having problems posting to this list, no bounces the mails just dont show any advice how to get the postings through is there filtering? robb _ -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue members, URI.
I believe that using the Local/[EMAIL PROTECTED] format will give you a bit more flexibility in the dialplan design, as there is an added degree of indirection. In the end I think this is only marginally costier than the raw channel format (unless you use the /n option) and should provide for a better laid-out dialplan. Just my $0.02, l. In data Tue, 02 Oct 2007 18:30:44 +0200, Thomas Kenyon [EMAIL PROTECTED] ha scritto: Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/number@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the party. (and if need be, there is the ability in queues to run a script on connection iirc). TIA for any clarification. -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference using g729?
Mark, Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. yeah i still don't understand. this is what i want to do. I want asterisk not to compress and decompress codecs. so either i can use SLIN as my codec for my SIP or IAX. or i can remove SLIN codec in meetme and change it to g729a so there's is no compression and decompression. do you get what i want to do? Thanks! Tilghman wrote it out: You can not mix two compressed audio streams together. You first have to uncompress them. Even if both audio streams use the same codec, they are compressed thus have to be uncompressed for the mixing of the audio to happen. Better? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] meetme conference using g729?
In article [EMAIL PROTECTED], Mark Quitoriano [EMAIL PROTECTED] wrote: yeah i still don't understand. this is what i want to do. I want asterisk not to compress and decompress codecs. so either i can use SLIN as my codec for my SIP or IAX. or i can remove SLIN codec in meetme and change it to g729a so there's is no compression and decompression. do you get what i want to do? Thanks! Yes, but it can't be done. In order to allow each conference participant to hear all the others at once, it is necessary to mix the audio by adding the contents of each channel. It is impossible to mix G.729 compressed because there is not a simple mathematical relationship between the output data and multiple input data. The mathematical way to do it would be what you are trying to avoid: convert each incoming stream to signed linear samples, then perform the mixing by adding those samples together, and then convert the outgoing mixed stream back to G.729 or whatever. This is what Asterisk does with any kind of codec that talks to Meetme, whether it be uLaw, ALaw, GSM, G.729, ILBC, and it doesn't need all participants to be using the same codec. Why were you so set on mixing G.729 without decoding/encoding? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] meetme conference using g729?
Tilghman Lesher wrote: Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. Ummm, why?? Unless you can explain some technical reason for this, looks like about 11 lines to change, +3 for correct log messages, +1 for a define, +~3 to add it as a nice config option in meetme.conf. So, in all about... 18 lines worth of code to get it running on any available codec, configurable from meetme.conf, which IMHO would make a lot of sense for single-codec systems... especially for G.729 due to better use of licenses, but for others too, due to load reduction and improved audio quality... Of course, I could be missing something obvious, please correct me if that's the case. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_read prematurely bridges channels
Hi list, Running Asterisk 1.4.10: When using the M() option for Dial to execute a macro, then executing a Read within the macro, once streaming of the audio file specified in Read has completed, and the channel attempts to read input from the destination channel where the macro is executed, the source channel stops ringing/moh, and audio from the source is bridged into the destination. I have tried various options to the Read application, but none have altered the results. An example of the behaviour: Calling party on channel SIP/YYY dials XXX, hits dialplan: exten = XXX,1,Dial(SIP/ZZZ,,mM(mymacro)) Macro looks something like: [macro-mymacro] exten = s,1,Playback(somefile) ;This plays fine on channel SIP/ZZZ exten = s,n,Read(somevar,audioprompt) ;audioprompt plays fine, then immediately after playing the prompt, channel SIP/ZZZ starts hearing audio data from SIP/YYY, moh stops on SIP/YYY, however no audio from SIP/ZZZ is sent to SIP/YYY until the macro exits exten = s,n,Playback(someotherfile) ;This and subsequent audio from the macro is not heard on either channel Once the macro finishes, audio is passed between the two channels as expected, however there is obviously something very wrong happening in app_read for the channel to be what appears to be partially bridged before the macro completes, though I can't see what it is. I thought it might be answering the chan, but the 'n' option to Read should skip around this, and has no effect. I've searched the bugtracker to no avail, full debug gives no useful data that I can see - is this a known bug, and does anyone have a workaround? Regards, Peter Fern ___ --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] Having problems posting to the list
Same here lenz wrote: Mee too, a lot of the messages I'm sending seem to disappear. l. In data Tue, 02 Oct 2007 22:38:26 +0200, robert boardman [EMAIL PROTECTED] ha scritto: Hi All I'm having problems posting to this list, no bounces the mails just dont show any advice how to get the postings through is there filtering? 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
Re: [asterisk-users] meetme conference using g729?
Peter Fern wrote: Tilghman Lesher wrote: Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. Ummm, why?? Unless you can explain some technical reason for this, looks like about 11 lines to change, +3 for correct log messages, +1 for a define, +~3 to add it as a nice config option in meetme.conf. So, in all about... 18 lines worth of code to get it running on any available codec, configurable from meetme.conf, which IMHO would make a lot of sense for single-codec systems... especially for G.729 due to better use of licenses, but for others too, due to load reduction and improved audio quality... lol. +2 lines of comments. Could you post the patch? ;) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk doesn't answer to incoming call
Hi: I installed A102d sangoma's card successfully but Asterisk doesn't answer to incoming call from pstn and console doesn't show any message of incoming call in the other word when I diall the number of E1 I can't connect to asterisk and dial the number of extension. I'd apreciateany idea. - Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compiling new version libpri
If I upgrade libpri 1.4.0 to 1.4.1, do I then need to recompile asterisk even though I'm not upgrading asterisk? -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [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] meetme conference using g729?
On 3 Oct 2007, at 10:16, Mark Quitoriano wrote: On 10/3/07, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 02 October 2007 16:55:52 Brian West wrote: On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote: anyway still if there's a hack for meetme to work with g729 codec this won't be an issue. So is there a hack or patch that i can use any codec for meetme? tnx You still do not understand. It doesn't matter if the call coming in is g729 you must transcode it to signed linear, mix the frames and then code it back into g729 you end up with quality loss doing that. Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. yeah i still don't understand. this is what i want to do. I want asterisk not to compress and decompress codecs. so either i can use SLIN as my codec for my SIP or IAX. or i can remove SLIN codec in meetme and change it to g729a so there's is no compression and decompression. do you get what i want to do? Thanks! Not exactly. Here are the facts: meetme mixes in SLIN. Any data arriving in anything other than slin will get transcoded twice, once on the way in and again on the way out. Now some opinions: The more efficient the compression of the codec, the less well it copes with decoding and re-encoding. Ulaw and Alaw are simple and not that efficient, but you don't lose any more by re-encoding than you did by decoding in the first place. Tighter codecs like 729 and GSM you will definitely hear the difference. Theory: If you have a conference where there is only _ever_ one speaker at a time, you could (in theory) optimize meetme to do without mixing, and if all the participants were using the same codec, you could get away with not re-encoding by sending out the appropriate incomming packet to all (other) members. I'm guessing that isn't the case for you. Advice: use Ulaw - it's a decent tradeoff for this sort of thing. Tim. ___ --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] Compiling new version libpri
On Wed, Oct 03, 2007 at 07:16:17AM -0500, Chris Stinson wrote: If I upgrade libpri 1.4.0 to 1.4.1, do I then need to recompile asterisk even though I'm not upgrading asterisk? To the best of my knowledge: no. Unless you have some non-standard patches to one of the versions (and not to the other) that have changed the interface. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Hey
For me this is very common. As soon as you define a problem in words (Email or otherwise) instead of concepts in your head, boom, the answer jumps out. That's why I really like whiteboards. I can draw the concept and then put it into words, side by side. Almost always figure out my issue this way. Thanks, Steve Totaro Ken Williams wrote: Just a quick thanks for all being here. I started to type up a message and realized my problem, so instead I'm saying thanks for all the good information you all pass through my mailbox every day and giving me a place to realize my error before I even ask the question. ___ --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 slow dial out - TDM400P
Ken Williams wrote:
Below is a copy of my log, zapata.conf extensions.conf that relate to
the ZAP lines. Basically when we dial out it takes on 10-12 seconds
before the ZAP line actaully picks up. I'm hoping to find out what the
cause is for this as it's causing user grief with extremely long connect
times, and I believe it may be causing issues of cross lines (an
outgoing call gets mixed with an incoming call, both ending up on the
same line). Incoming calls are processed fairly quickly, about 3
seconds which is perfectly acceptable.
[Oct 2 10:30:27] DEBUG[22199] chan_zap.c: Dialing 'xxx'
[Oct 2 10:30:27] DEBUG[22199] chan_zap.c: Deferring dialing...
[Oct 2 10:30:27] VERBOSE[22199] logger.c: -- Called 4/xxx
[Oct 2 10:30:35] DEBUG[22199] chan_zap.c: Engaged echo training on
channel 4
[Oct 2 10:30:38] DEBUG[22199] chan_zap.c: Echo cancellation already on
[Oct 2 10:30:38] VERBOSE[22199] logger.c: -- Zap/4-1 answered
SIP/717-08c387d0
ZAPATA.CONF
[channels]
language=en
echocancel=256
echocancelwhenbridged=256
echotraining=800
rxgain=6.0
txgain=0.0
faxdetect=no
signalling=fxs_ks
context=from-zaptel
group=0
channel = 2
signalling=fxs_ks
context=from-zaptel
group=0
channel = 3
---
EXTENSIONS.CONF
TRUNK_OPTIONS=rTt ;r here because of the 10-12 second delay
exten = _1NXXNXX,1,Dial(ZAP/2/${EXTEN},120,${TRUNK_OPTIONS})
exten = _1NXXNXX,n,Dial(ZAP/3/${EXTEN},120,${TRUNK_OPTIONS})
exten = _1NXXNXX,n,Dial(ZAP/4/${EXTEN},120,${TRUNK_OPTIONS})
exten = _1NXXNXX,n,Hangup()
---
Note that my extensions.conf used to have a single line exten =
_1NXXNXX,1,Dial(ZAP/g0/${EXTEN},120,${TRUNK_OPTIONS}) but I changed
it to see if this way sped things up at all, it doesn't.
Any suggestions would be greatly appreciated.
What version of Asterisk is this and what is your hardware? I am
assuming it is POTS.
Thanks,
Steve
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Re: [asterisk-users] Asterisk 1.4.12 and Asterisk-addons 1.4.3 released
Don Pobanz wrote: the Asterisk release contains a large number of bug fixes for all parts of Asterisk. I am thankful to see the amount of fixes that have gone into this release. However, seeing this many fixes does not give me a warm fuzzy feeling that we won't see a lot more fixes in the near future. So are bug fixes good or bad? ;-) And more importantly, will any of the remaining bugs bite me? Branch 1.4 has one important to us feature that 1.2 does not and that is the queue autofill option. Because of this one feature, I have been wanting to switch to the 1.4 branch for some time. We have a backup system that I will be using for testing. If all goes well, we will move to the 1.4 branch. I hope many others are doing the same so the stability of 1.4 can be improved to the point where no one is concerned. Thanks to all the developers for improving an already great product! Don Pobanz Another reason to call for a 1.2 spoon or fork! Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ping
must be blacklisted, i have posted like 4 messages and none are showing up. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Resolving digit strings using pound/hash.
Hi all, The thing that has bugged me about Asterisk since I first started playing with it, is the fact that the pound sign/hash/octothorp doesn't resolve digit conflicts or cancel timing on a variable length string such as a tie line code or when you call numbers in a country whose length can be different between numbers in the same plan. In North America, we see this when calling places such as Germany. Thanks to Atis, this now seems to work properly in DISA fixed via bug 10754. Can we please have this effect expanded to cover all cases where Asterisk collects digits such as dialing into an IVR, zap FXS channel, and everywhere else. Thanks. -- Bill in Denver ___ --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] Resolving digit strings using pound/hash.
On Wednesday 03 October 2007 15:41:08 William F. Acker WB2FLW +1-303-722-7209 wrote: Hi all, The thing that has bugged me about Asterisk since I first started playing with it, is the fact that the pound sign/hash/octothorp doesn't resolve digit conflicts or cancel timing on a variable length string such as a tie line code or when you call numbers in a country whose length can be different between numbers in the same plan. In North America, we see this when calling places such as Germany. Thanks to Atis, this now seems to work properly in DISA fixed via bug 10754. Can we please have this effect expanded to cover all cases where Asterisk collects digits such as dialing into an IVR, zap FXS channel, and everywhere else. Hi, Can you pinpoint (with examples) where it is not that way? From my experience this is already working nearly everywhere. At least it's for Read's and incoming calls. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --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 doesn't answer to incoming call
fateme fatah wrote: Hi: I installed A102d sangoma's card successfully but Asterisk doesn't answer to incoming call from pstn and console doesn't show any message of incoming call in the other word when I diall the number of E1 I can't connect to asterisk and dial the number of extension. Without seeing your configs for the E1 setup or your dial statement, nobody will be able to help you. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by 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] Having problems posting to the list
On Wed, 2007-10-03 at 14:14 +0300, Zoa wrote: Same here Yes, I'm aware that some people are having problems posting to the mailing list, and I'm working with Digium's IT staff to try to correct the problems. It seems to be related to our inbound spam filtering. (The weird thing is that new messages seem to get lost, but replies to existing messages seem to come through just fine.) I'll let you know as soon as I have an update, but I know Digium's IT team is currently swamped in helping out with the move to Digium's new offices. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf vs. AEL
You have various scripting languages things like that can go in! /b On Oct 3, 2007, at 4:12 AM, Garth van Sittert wrote: Where would you suggest all the logic goes Brian? Garth Garth van Sittert BSc (Physics Computer Science) - Main: 08600 BITCO Phone: +27 (0)11 875 6900 Fax: +27 (0)11 875 6901 Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Web:www.bitco.co.za Brian West wrote: In my opinion the dialplan isn't where that logic belongs. /b On Oct 3, 2007, at 12:32 AM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? Thanks! __Yehavi: ___ --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] meetme conference using g729?
On Wednesday 03 October 2007 06:09:01 Peter Fern wrote: Tilghman Lesher wrote: Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. Ummm, why?? Unless you can explain some technical reason for this, looks like about 11 lines to change, +3 for correct log messages, +1 for a define, +~3 to add it as a nice config option in meetme.conf. So, in all about... 18 lines worth of code to get it running on any available codec, configurable from meetme.conf, which IMHO would make a lot of sense for single-codec systems... especially for G.729 due to better use of licenses, but for others too, due to load reduction and improved audio quality... Of course, I could be missing something obvious, please correct me if that's the case. I invite you to try it. You could make a lot of really smart people look like fools if you're able to mix compressed audio together without decompressing, or you might make yourself look like a fool, because you get back garbage for attempting to mix compressed data. While I won't go so far as to say mixing compressed audio is impossible without decompressing first, it is not *simple* by any means whatsoever. In fact, I would go so far as to say that not only are you likely to degrade the audio even further, but the CPU time it would take is an order of magnitude higher than the current methodology. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf vs. AEL
Yehavi Bourvine +972-8-9489444 wrote: Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? I just think its the default so probably many new people to Asterisk start there and then possibly move over to AEL or AGI scripts later on as needs become more complex... For those that have been in the Asterisk community for a longer period of time, the traditional flat line script was all that was available until AEL came along as far as I know. I wrote an automated dialplan generator so much of *our* systems had the traditional flat script because its much easier to produce that traditional asterisk script from a GUI that generates script for you. I prefer pascal syntax personally, so we use a pascal based AGI/FastAGI engine that I wrote for much of our more advanced logic. In the end, it probably comes down to preference and need, I would think. Nice to be proficient in writing it all; flat scripts, AEL, AGI/FastAGI/Manager API (using your programming/script language of prefernce)this way we can have more tools to solve more problems for our customers or company. --- Warm Regards, Lee ___ --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 upgrade trouble (1.2.10 - 1.2.20.1)
On Tue, Oct 02, 2007 at 06:20:54PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 12:47:55PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote:
Hello!
I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am using
asterisk 1.2.10 with one TDM2400P (all 6 module in use) and one
TE405P. When I upgrade to 1.2.20.1 the order of cards mess up and
therefore zaptel.conf is unusable and gives error. Why is it happen
and what do I need to change in zaptel.conf?
now zaptel.conf is:
loadzone=hu
defaultzone=hu
# GSM
fxsks=1-4
# FAX
fxoks=5-8
# EXT
fxoks=9-24
# PRI
span=2,1,0,ccs,hdb3,crc4
bchan=25-39
dchan=40
bchan=41-55
zttool shows that my TDM is the first device and T4XXP is from second
to fifth. Nice. After zaptel upgrade TDM is on the fifth position. And
channels in zaptel.conf were messed up of course because 1-24 is not
for TDM.
Was there also a change in the kernel ?
In the value of MODULES in /etc/{sysconfig,default}/zaptel ?
Thanks for your answer. Same kernel (2.6.11) and no change in zaptel:
MODULES=$MODULES wct4xxp # TE405P - Quad Span T1/E1 Card (5v
version)
MODULES=$MODULES wctdm24xxp # TDM2400P - Modular FXS/FXO interface
(1-24 ports)
Might MODULES order count? If so why that wasn't count with zaptel 1.2.10?
It counts if the modules weren't loaded earlier on boot time, and if you
use the proper init.d script.
I see. I've been using the init.d script from release which I think
proper. Is it means there is logical difference between 1.2.10 and
1.2.20.1 on init.d script level? Because 1.2.10 is working flawless
with this MODULES setup.
Are you sure you use zaptel.init from 1.2.20.1 ?
I really don't think you can guarantee that the MODULES list will
triumph any automalic load the system has on startup (which is why
Astribank drivers won't auto-register by default).
I can't think of any change that should have affected the modules
loading order by default. Unless this is some timing issue - a race
between automatic loading and manual loading.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] extensions.conf vs. AEL
Lee Jenkins wrote: Yehavi Bourvine +972-8-9489444 wrote: Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? I just think its the default so probably many new people to Asterisk start there and then possibly move over to AEL or AGI scripts later on as needs become more complex... For those that have been in the Asterisk community for a longer period of time, the traditional flat line script was all that was available until AEL came along as far as I know. I wrote an automated dialplan generator so much of *our* systems had the traditional flat script because its much easier to produce that traditional asterisk script from a GUI that generates script for you. I prefer pascal syntax personally, so we use a pascal based AGI/FastAGI engine that I wrote for much of our more advanced logic. In the end, it probably comes down to preference and need, I would think. Nice to be proficient in writing it all; flat scripts, AEL, AGI/FastAGI/Manager API (using your programming/script language of prefernce)this way we can have more tools to solve more problems for our customers or company. --- Warm Regards, Lee Let us not forget that AEL cannot be stored in a database therefore rendering you unable to utilize realtime. -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [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] Cisco 7940G licensing with asterisk
But I believe Cisco is the only manufacturer producing a phone with a gigabit port for connecting a desktop pc. Anyone know of any other? Glenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Monday, October 01, 2007 7:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk Just buy the Linksys SPA962's they work better than the cisco phones in a NAT env. /b On Oct 1, 2007, at 6:13 PM, Andrew Joakimsen wrote: My understanding is: Smartnet: service contract basically allows you to download the newest sw release. Besides that you can buy phones without a license. Presumably as spares But you must buy a SIP license to technically be allowed to use that software that can be obtained from Smartnet. I know there was some changes a year or two back, but wasn't that just pricing? ___ --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] extensions.conf vs. AEL
Wouldnt that take a very large portion of datapower, to startup the parsers and such, instead of having the whole dialplan natively in Asterisk. We always try to do as much as possible in dialplan, so that we are not reliant on external scripts. Kind Regards Jon Leren Schøpzinsky -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: 3. oktober 2007 15:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] extensions.conf vs. AEL You have various scripting languages things like that can go in! /b On Oct 3, 2007, at 4:12 AM, Garth van Sittert wrote: Where would you suggest all the logic goes Brian? Garth Garth van Sittert BSc (Physics Computer Science) - Main: 08600 BITCO Phone: +27 (0)11 875 6900 Fax: +27 (0)11 875 6901 Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Web:www.bitco.co.za Brian West wrote: In my opinion the dialplan isn't where that logic belongs. /b On Oct 3, 2007, at 12:32 AM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something? Thanks! __Yehavi: ___ --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] ping
Pong On 10/2/07, Steve Totaro [EMAIL PROTECTED] wrote: must be blacklisted, i have posted like 4 messages and none are showing up. ___ --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] extensions.conf vs. AEL
Let us not forget that AEL cannot be stored in a database therefore rendering you unable to utilize realtime. AEL converted into standard extensions.conf syntax in the dialplan. ___ --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 slow dial out - TDM400P
So, I updated to 1.4.12 last night and it appears my problem is mostly
gone now. Not sure what the difference was, but it now takes about 3
seconds before the ZAP line picks it up. I was on 1.4.10.1 before that,
and yes POTS. Removing the echo cancellation at this point makes no
difference, not sure if it would've pre-.12.
I'm leaving the 'r' in the dial statement as 3 seconds is kind of an
awkward amount of time for a dialtone after you hit dial, but I could
remove it and it wouldn't be the end of the world.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
Williams
Sent: Tuesday, October 02, 2007 10:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Zaptel slow dial out - TDM400P
Below is a copy of my log, zapata.conf extensions.conf that relate to
the ZAP lines. Basically when we dial out it takes on 10-12 seconds
before the ZAP line actaully picks up. I'm hoping to find out what the
cause is for this as it's causing user grief with extremely long connect
times, and I believe it may be causing issues of cross lines (an
outgoing call gets mixed with an incoming call, both ending up on the
same line). Incoming calls are processed fairly quickly, about 3
seconds which is perfectly acceptable.
[Oct 2 10:30:27] DEBUG[22199] chan_zap.c: Dialing 'xxx'
[Oct 2 10:30:27] DEBUG[22199] chan_zap.c: Deferring dialing...
[Oct 2 10:30:27] VERBOSE[22199] logger.c: -- Called 4/xxx
[Oct 2 10:30:35] DEBUG[22199] chan_zap.c: Engaged echo training on
channel 4
[Oct 2 10:30:38] DEBUG[22199] chan_zap.c: Echo cancellation already on
[Oct 2 10:30:38] VERBOSE[22199] logger.c: -- Zap/4-1 answered
SIP/717-08c387d0
ZAPATA.CONF
[channels]
language=en
echocancel=256
echocancelwhenbridged=256
echotraining=800
rxgain=6.0
txgain=0.0
faxdetect=no
signalling=fxs_ks
context=from-zaptel
group=0
channel = 2
signalling=fxs_ks
context=from-zaptel
group=0
channel = 3
---
EXTENSIONS.CONF
TRUNK_OPTIONS=rTt ;r here because of the 10-12 second delay
exten =
_1NXXNXX,1,Dial(ZAP/2/${EXTEN},120,${TRUNK_OPTIONS})
exten =
_1NXXNXX,n,Dial(ZAP/3/${EXTEN},120,${TRUNK_OPTIONS})
exten =
_1NXXNXX,n,Dial(ZAP/4/${EXTEN},120,${TRUNK_OPTIONS})
exten = _1NXXNXX,n,Hangup()
---
Note that my extensions.conf used to have a single line exten =
_1NXXNXX,1,Dial(ZAP/g0/${EXTEN},120,${TRUNK_OPTIONS}) but I changed
it to see if this way sped things up at all, it doesn't.
Any suggestions would be greatly appreciated.
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[asterisk-users] Configuration files inside SQLite3
Hello all, Is it possible to store, read and write configuration files in an SQLite3 database instead of using the configuration files inside the /etc/asterisk/ directory? If it is then can you point me to the right documentation on how to do this or probably hints on how to do this? Thank you in advance. GNUbie ___ --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] Secondary Dialtone and selecting a specific line from Zap/g
I can't help you with that. I only wanted to point out that ignoreopat is not what you need. On Polycom SIP phones you continue dialtone by placing a , in the phone's dialplan. SIP phones have their own internal dialplan that is not part of Asterisk's dialplan. You would have to check the docs for your phone. Not all SIP phones can continue dialtone. bilal ghayyad wrote: I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone, not a dialtone provided by Asterisk. Al lists wrote: Correction, on FXO port not FXS, second, read his email first: Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP Phone or Broadtel IP Phone, so if user select that button then he will be sure that his outside call will be via that specific line. Just assign a key on your phone to dial that extension, and you will have dial tone on selected line, then as a traditional PBX you can send any digits to your provider. On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: ignorepat continues dialtone after a leading digit has been dialed on FXS ports. How does ignorepat help this guy? Al lists wrote: ignorpat is your friend On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote: Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Simply don't use groups. Use channels directly. To dial via the specific Zaptel channel NN, use Zap/NN Am I missing 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] extensions.conf vs. AEL
On Oct 3, 2007, at 9:39 AM, Jon Schøpzinsky wrote: Wouldnt that take a very large portion of datapower, to startup the parsers and such, instead of having the whole dialplan natively in Asterisk. We always try to do as much as possible in dialplan, so that we are not reliant on external scripts. Kind Regards Jon Leren Schøpzinsky Stepping thru the dialplan line by line is one of the most inefficient things in Asterisk... Every priority it checks and rechecks the dialplan and priorty at the very least 5 times per priority. I think this is one thing being addressed in 1.4 and later. Dialplan logic isn't a language in my opinion. /b ___ --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 upgrade trouble (1.2.10 - 1.2.20.1)
On 10/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 06:20:54PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 12:47:55PM +0200, Artifex Maximus wrote:
On 10/2/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote:
Hello!
I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am
using
asterisk 1.2.10 with one TDM2400P (all 6 module in use) and one
TE405P. When I upgrade to 1.2.20.1 the order of cards mess up and
therefore zaptel.conf is unusable and gives error. Why is it happen
and what do I need to change in zaptel.conf?
now zaptel.conf is:
loadzone=hu
defaultzone=hu
# GSM
fxsks=1-4
# FAX
fxoks=5-8
# EXT
fxoks=9-24
# PRI
span=2,1,0,ccs,hdb3,crc4
bchan=25-39
dchan=40
bchan=41-55
zttool shows that my TDM is the first device and T4XXP is from
second
to fifth. Nice. After zaptel upgrade TDM is on the fifth position.
And
channels in zaptel.conf were messed up of course because 1-24 is not
for TDM.
Was there also a change in the kernel ?
In the value of MODULES in /etc/{sysconfig,default}/zaptel ?
Thanks for your answer. Same kernel (2.6.11) and no change in zaptel:
MODULES=$MODULES wct4xxp # TE405P - Quad Span T1/E1 Card (5v
version)
MODULES=$MODULES wctdm24xxp # TDM2400P - Modular FXS/FXO interface
(1-24 ports)
Might MODULES order count? If so why that wasn't count with zaptel
1.2.10?
It counts if the modules weren't loaded earlier on boot time, and if you
use the proper init.d script.
I see. I've been using the init.d script from release which I think
proper. Is it means there is logical difference between 1.2.10 and
1.2.20.1 on init.d script level? Because 1.2.10 is working flawless
with this MODULES setup.
Are you sure you use zaptel.init from 1.2.20.1 ?
I am absolutely sure because I had to copy with my own hands.
I really don't think you can guarantee that the MODULES list will
triumph any automalic load the system has on startup (which is why
Astribank drivers won't auto-register by default).
I can't think of any change that should have affected the modules
loading order by default. Unless this is some timing issue - a race
between automatic loading and manual loading.
I will try with MODULES in right order and report the result. I think
there must be difference between two init.d script because of
different effect.
bye,
a
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[asterisk-users] Asterisk Keep Loosing Registration
Hello All, For some odd reasons my Asterisk is keep on loosing registration of my SIP devices. On the SIP device it shows I am RESISTED but when I do sip show peers it shows my sip endpoints are UNREACHABLE. And it keeps on flapping Peer '903456' is now UNREACHABLE! and Peer '903456' is now REACHABLE!... I changed my maxexpiry and defaultexpiry to 3600 in sip.conf but still it didn't help. I am using Asterisk 1.2.18 with Real-Time config. Any help will be appreciated... Cheers, Nitesh ___ --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] meetme conference using g729?
I have been following this discussion. You do have a point. However, the way * works right now. If a channel does not require trans-coding to get into a conference, coder usage is counted. So I really do not know what difference putting the transcoding in meetme is going to make. I mean, how could this better contribute to better use of G729 licenses. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Fern Sent: Wednesday, October 03, 2007 7:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] meetme conference using g729? Tilghman Lesher wrote: Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. Ummm, why?? Unless you can explain some technical reason for this, looks like about 11 lines to change, +3 for correct log messages, +1 for a define, +~3 to add it as a nice config option in meetme.conf. So, in all about... 18 lines worth of code to get it running on any available codec, configurable from meetme.conf, which IMHO would make a lot of sense for single-codec systems... especially for G.729 due to better use of licenses, but for others too, due to load reduction and improved audio quality... Of course, I could be missing something obvious, please correct me if that's the case. ___ --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] meetme conference using g729?
But his preference of G729 is to save bandwidth. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Wednesday, October 03, 2007 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] meetme conference using g729? Not exactly. Here are the facts: meetme mixes in SLIN. Any data arriving in anything other than slin will get transcoded twice, once on the way in and again on the way out. Now some opinions: The more efficient the compression of the codec, the less well it copes with decoding and re-encoding. Ulaw and Alaw are simple and not that efficient, but you don't lose any more by re-encoding than you did by decoding in the first place. Tighter codecs like 729 and GSM you will definitely hear the difference. Theory: If you have a conference where there is only _ever_ one speaker at a time, you could (in theory) optimize meetme to do without mixing, and if all the participants were using the same codec, you could get away with not re-encoding by sending out the appropriate incomming packet to all (other) members. I'm guessing that isn't the case for you. Advice: use Ulaw - it's a decent tradeoff for this sort of thing. Tim. ___ --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-bugs] Constant LAGGGED extensions
Doug, Look at the list. It seems you and Nitesh Divecha may be having the same problem. Maybe you guys can confirm that you have the same issue and figure out what is in common, such as Asterisk version or whatever. Thanks, Steve Doug Reid wrote: Hi Steve I have tried a constant ping and get no problem on that. I assume that the registration would use UPD and ping would use TCP so would this give any indication? Should I be looking at UPD or TCP? Thanks Doug Reid -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: 03 October 2007 03:42 PM To: Doug Reid Subject: Re: [asterisk-bugs] Constant LAGGGED extensions Doug Reid wrote: Hi All I have a problem that is affecting 3 of our Asterisk sites and have tried all possible to rectify this if anyone can shed some light on this? We have constant LAGGED, SIP extensions. The extensions will lagg in groups of 2 - 20 phones at a time and will recover very shortly after. We have tried so far: * Setting all phones (Snom and Polycom) to forced full 100M full duplex and forced all ports on the switches (HP and Cisco) to 100M full duplex. * Setting up QOS on all UPD ports and traffic. * Tried a number of Asterisk releases (1.2.20 -1.2.24) as I read about a bug on the UDP port of SIP. This does seem like a network issue but we have tried all possible solutions on the network side and I must now look at Asterisk. Below is a typical output that will show on our CLI and log files: Oct 2 18:57:49 NOTICE[9751] chan_sip.c: Peer '2177' is now TOO LAGGED! (2037ms / 2000ms) Oct 2 18:57:49 NOTICE[9751] chan_sip.c: Peer '2184' is now TOO LAGGED! (2041ms / 2000ms) Can you ping the phones from your Asterisk box? Keep pinging and when you get the lagged message and see if it indeed a network isssue. Thanks, Steve Totaro ___ --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] meetme conference using g729?
If bandwidth were not an issue, I would think everyone would opt for ulaw or alaw. Why compress and use CPU cycles and G729 licenses if there were no bandwidth issues? Thanks, Steve totaro Wai Wu wrote: But his preference of G729 is to save bandwidth. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Wednesday, October 03, 2007 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] meetme conference using g729? Not exactly. Here are the facts: meetme mixes in SLIN. Any data arriving in anything other than slin will get transcoded twice, once on the way in and again on the way out. Now some opinions: The more efficient the compression of the codec, the less well it copes with decoding and re-encoding. Ulaw and Alaw are simple and not that efficient, but you don't lose any more by re-encoding than you did by decoding in the first place. Tighter codecs like 729 and GSM you will definitely hear the difference. Theory: If you have a conference where there is only _ever_ one speaker at a time, you could (in theory) optimize meetme to do without mixing, and if all the participants were using the same codec, you could get away with not re-encoding by sending out the appropriate incomming packet to all (other) members. I'm guessing that isn't the case for you. Advice: use Ulaw - it's a decent tradeoff for this sort of thing. Tim. ___ --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] extensions.conf vs. AEL
Lee Jenkins wrote: Why most people don't use it? Am I missing something? I think it looks too much like C. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by 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 Keep Loosing Registration
Hi Nitesh, The reachable/unreachable determination is not connected to registration expiry parameters in any way. There is a qualify= parameter (that has a default value, and I think it may be on by default) that is associated with all SIP peers. It is basically a way to say that the SIP peer should be pinged periodically (with a blank SIP OPTIONS message that returns some sort of response from the other end) and that if the round-trip latency on that transaction exceeds whatever number of milliseconds, the host should be deemed 'unreachable' until the next ping is attempted and the RTL moves back into qualified territory. This is the parameter to qualify, e.g. qualify=2000 means declare the host unreachable if its round-trip latency through SIP ping exceeds 2000 ms. I generally find the behaviour of this facility to be rather spurious with end-user phones and prone to false alarms, for whatever reason. So, I either give it a really high value, or say qualify=no. Best of luck, -- Alex On Wed, 3 Oct 2007, Nitesh Divecha wrote: Hello All, For some odd reasons my Asterisk is keep on loosing registration of my SIP devices. On the SIP device it shows I am RESISTED but when I do sip show peers it shows my sip endpoints are UNREACHABLE. And it keeps on flapping Peer '903456' is now UNREACHABLE! and Peer '903456' is now REACHABLE!... I changed my maxexpiry and defaultexpiry to 3600 in sip.conf but still it didn't help. I am using Asterisk 1.2.18 with Real-Time config. Any help will be appreciated... Cheers, Nitesh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Secondary Dialtone and selecting a specific line from Zap/g
For another tone frequency for the outside dialtone, try putting this value [EMAIL PROTECTED],[EMAIL PROTECTED],[EMAIL PROTECTED];*(.4/0/1),10(*/0/2+3) in the Outside Dialtone field. It will give you a slight pause followed by a different dialtone frequency. On a Linksys/Siprua 941, that would be at the top of the Regional page. However, you won't hear any secondary dialtone unless you put a comma after EVERY initial '9' in the dialplan string for each line in use. On a 941, that would be at the bottom of the Ext 1 and Ext 2 pages of the web interface. I suggest the dialplan string of: (*xx|[1-7]xx|9,[3469]11|98|99|9,[2-9]xx|9,11|9,[2-9]xx|9,1[2-9]xx[2-9]xx|9,011xxx.) - Walt Joyce Eric ManxPower Wieling wrote: I can't help you with that. I only wanted to point out that ignoreopat is not what you need. On Polycom SIP phones you continue dialtone by placing a , in the phone's dialplan. SIP phones have their own internal dialplan that is not part of Asterisk's dialplan. You would have to check the docs for your phone. Not all SIP phones can continue dialtone. bilal ghayyad wrote: I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone, not a dialtone provided by Asterisk. Al lists wrote: Correction, on FXO port not FXS, second, read his email first: Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP Phone or Broadtel IP Phone, so if user select that button then he will be sure that his outside call will be via that specific line. Just assign a key on your phone to dial that extension, and you will have dial tone on selected line, then as a traditional PBX you can send any digits to your provider. On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: ignorepat continues dialtone after a leading digit has been dialed on FXS ports. How does ignorepat help this guy? Al lists wrote: ignorpat is your friend On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote: Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Simply don't use groups. Use channels directly. To dial via the specific Zaptel channel NN, use Zap/NN Am I missing 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 ___ --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 slow dial out - TDM400P
Looks like a bug they have fixed with the latest 1.4.x release.
Please, can we have a 1.2.x spoon? Instead of just security fixes, the
spoon should also include bug fixes and backports or new functionality
in later Asterisk versions.
Thanks,
Steve Totaro
Ken Williams wrote:
So, I updated to 1.4.12 last night and it appears my problem is mostly
gone now. Not sure what the difference was, but it now takes about 3
seconds before the ZAP line picks it up. I was on 1.4.10.1 before
that, and yes POTS. Removing the echo cancellation at this point
makes no difference, not sure if it would've pre-.12.
I'm leaving the 'r' in the dial statement as 3 seconds is kind of an
awkward amount of time for a dialtone after you hit dial, but I could
remove it and it wouldn't be the end of the world.
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Ken
Williams
*Sent:* Tuesday, October 02, 2007 10:48 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Zaptel slow dial out - TDM400P
Below is a copy of my log, zapata.conf extensions.conf that relate
to the ZAP lines. Basically when we dial out it takes on 10-12
seconds before the ZAP line actaully picks up. I'm hoping to find out
what the cause is for this as it's causing user grief with extremely
long connect times, and I believe it may be causing issues of cross
lines (an outgoing call gets mixed with an incoming call, both ending
up on the same line). Incoming calls are processed fairly quickly,
about 3 seconds which is perfectly acceptable.
[Oct 2 10:30:27] DEBUG[22199] chan_zap.c: Dialing 'xxx'
[Oct 2 10:30:27] DEBUG[22199] chan_zap.c: Deferring dialing...
[Oct 2 10:30:27] VERBOSE[22199] logger.c: -- Called 4/xxx
[Oct 2 10:30:35] DEBUG[22199] chan_zap.c: Engaged echo training on
channel 4
[Oct 2 10:30:38] DEBUG[22199] chan_zap.c: Echo cancellation already on
[Oct 2 10:30:38] VERBOSE[22199] logger.c: -- Zap/4-1 answered
SIP/717-08c387d0
ZAPATA.CONF
[channels]
language=en
echocancel=256
echocancelwhenbridged=256
echotraining=800
rxgain=6.0
txgain=0.0
faxdetect=no
signalling=fxs_ks
context=from-zaptel
group=0
channel = 2
signalling=fxs_ks
context=from-zaptel
group=0
channel = 3
---
EXTENSIONS.CONF
TRUNK_OPTIONS=rTt ;r here because of the 10-12 second delay
exten = _1NXXNXX,1,Dial(ZAP/2/${EXTEN},120,${TRUNK_OPTIONS})
exten = _1NXXNXX,n,Dial(ZAP/3/${EXTEN},120,${TRUNK_OPTIONS})
exten = _1NXXNXX,n,Dial(ZAP/4/${EXTEN},120,${TRUNK_OPTIONS})
exten = _1NXXNXX,n,Hangup()
---
Note that my extensions.conf used to have a single line exten =
_1NXXNXX,1,Dial(ZAP/g0/${EXTEN},120,${TRUNK_OPTIONS}) but I
changed it to see if this way sped things up at all, it doesn't.
Any suggestions would be greatly appreciated.
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Re: [asterisk-users] extensions.conf vs. AEL
Eric ManxPower Wieling wrote: Let us not forget that AEL cannot be stored in a database therefore rendering you unable to utilize realtime. AEL converted into standard extensions.conf syntax in the dialplan. Doesn't this render having used AEL pointless? -- Thank you and have a wonderful day, Anthony Francis ___ --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] extensions.conf vs. AEL
Its just a different way to express the same thing in a more fluid way. /b On Oct 3, 2007, at 10:33 AM, Anthony Francis wrote: Doesn't this render having used AEL pointless? ___ --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] extensions.conf vs. AEL
To each his own. I like the flat files personally, they are more fluid to me. Thanks, Steve Brian West wrote: Its just a different way to express the same thing in a more fluid way. /b On Oct 3, 2007, at 10:33 AM, Anthony Francis wrote: Doesn't this render having used AEL pointless? ___ --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] extensions.conf vs. AEL
I'm growing fond of XML. /b On Oct 3, 2007, at 10:39 AM, Steve Totaro wrote: To each his own. I like the flat files personally, they are more fluid to me. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best config for 12 FXO system?
I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot availability, are there any technical reasons to choose one of the above configurations over the others? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parking lot problems
Now on to another problem that we've had as far as I know since the
beginning of using Asterisk 9+ months ago. I've been trying very hard
to knock this problem out but regardless of what I do, it's still there.
So, the problem is, when a call is in the parking lot, it then times out
after whatever time frame and dials the extension that put it on hold.
After 60 seconds of ringing back, it's supposed to go to [park-dial] t
extension as far as I can tell, which it actually does seem to do.
However, before the t extension kicks in, the line is dropped with the
following error message on the CLI:
[Oct 3 08:45:31] WARNING[12621]: channel.c:2616 ast_indicate_data:
Unable to handle indication 3 for 'SIP/727-095c0348'
[Oct 3 08:46:31] WARNING[11487]: chan_sip.c:12037
handle_response_invite: Re-invite to non-existing call leg on other UA.
SIP dialog '[EMAIL PROTECTED] Giving up.
-- SIP/717-09570200 is circuit-busy
[Oct 3 08:46:31] NOTICE[12621]: cdr.c:434 ast_cdr_free: CDR on channel
'SIP/717-09570200' not posted
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/727-095c0348' status is 'CONGESTION'
[Oct 3 08:46:31] WARNING[12621]: channel.c:2616 ast_indicate_data:
Unable to handle indication 8 for 'SIP/727-095c0348'
[Oct 3 08:46:31] WARNING[11487]: chan_sip.c:12536 handle_response:
Remote host can't match request CANCEL to call
'[EMAIL PROTECTED]'. Giving up.
So the line hangs up, these errors are displayed, then I see the 't'
extension kick in. Notice this is all on the same network, SIP devices
only, no NAT or anything like that. I was initially testing on a
ZAP/SIP configuration, had the same type of errors and thought to reduce
complexity I'd keep it all SIP. I've tried canreinvite=yes and no on
the SIP devices, neither made a difference.
So, before I go the bug route I'd like someone to just verify my
configuration files make sure I'm not doing something stupid.
SIP.CONF:
[general]
callerid=Unknown Caller
disallow=all
allow=ulaw
allow=gsm
[717]
type=friend
dial=SIP/717
callerid=Ken Williams 717
[EMAIL PROTECTED]
allowsubscribe=yes
host=dynamic
context=from-internal
[727]
type=friend
secret=1234
dial=SIP/727
callerid=Conference Room 727
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
allowsubscribe=yes
host=dynamic
context=from-internal
EXTENSIONS.CONF:
[from-internal]
include = parkedcalls
exten = _20X,1,Goto(parkedcalls,${EXTEN},1)
[park-dial]
exten = t,1,Goto(from-internal,900,1)
FEATURES.CONF
[general]
parkext = 200
parkpos = 201-205
context = parkedcalls
parkingtime = 30
parkhints = yes
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Re: [asterisk-users] Configuration files inside SQLite3
GNUbie wrote: Hello all, Is it possible to store, read and write configuration files in an SQLite3 database instead of using the configuration files inside the /etc/asterisk/ directory? If it is then can you point me to the right documentation on how to do this or probably hints on how to do this? Thank you in advance. GNUbie It is possible to store configuration files in any relational database which has ODBC compatibility. Thus, sqlite qualifies. If you are using trunk, you won't even need to use ODBC, because Asterisk has native support for sqlite. If you're looking for an overview of the Asterisk Realtime Architecture (the means by which you can store configurations in a database) look in the doc directory of your asterisk source for realtime.txt and extconfig.txt, or search voip-info.org for asterisk realtime. If you're looking for more in-depth coverage of integrating Asterisk with a relational database, I suggest looking at the second edition of Asterisk: The Future of Telephony, available at book stores, or for download at http://openbooks.oreilly.com/ Specifically, check out chapter 12. It doesn't cover sqlite explicitly, but it's not much of a stretch to use it based on what's provided in the book. Mark Michelson ___ --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] Best config for 12 FXO system?
Tony Mountifield wrote: I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot availability, are there any technical reasons to choose one of the above configurations over the others? Cheers Tony If the Ericsson PBX has a T1 card already and you have one in your asterisk box, that would be the cleanest way of acheiving what you want. You could also use a channel bank with the analog and convert to T1 for the Asterisk connection. I try to use T1 (PRI if possible) whenever more than four analog trunks will be involved. I have had tons of bad experiences with analog cards, very few with ISDN. Thanks, Steve Totaro ___ --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] Best config for 12 FXO system?
Tony Mountifield wrote: I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot availability, are there any technical reasons to choose one of the above configurations over the others? No idea, but if you look further afield, if you buy a Sangoma A200 or an A400 you can have all 12 on one PCI (or PCI Express) slot (the former taking up 3 Spaces on your PCs backplane and the latter taking up only 1). If expandability is a concern, an A400 can support up to 48 FXO ports on one PCI (or PCI-Express) Slot (4 spaces) or an A200 can support up to 24 FXO ports. (6 spaces) I can't comment on how good they are, I've only got TDM400Ps myself. ___ --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] Best config for 12 FXO system?
Those are all analog though, aren't they? What about a channel bank into a digital card? Might that be cheaper than shelling out for 12 FXO ports and the cards to hold them? Just wanted to throw that out there before the discussion started :) Tony Mountifield wrote: I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot availability, are there any technical reasons to choose one of the above configurations over the others? Cheers Tony ___ --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] Best config for 12 FXO system?
None are great options. I'd use a T1 card and a channel bank. At minimum I'd do the single 2400P. IRQ problems are going to be a bear with multiple cards. -Darren -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Wednesday, October 03, 2007 12:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Best config for 12 FXO system? I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot availability, are there any technical reasons to choose one of the above configurations over the others? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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 This message was sent from D2 Technology, INC. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf vs. AEL
On Wed, 2007-10-03 at 09:33 -0600, Anthony Francis wrote: Eric ManxPower Wieling wrote: Let us not forget that AEL cannot be stored in a database therefore rendering you unable to utilize realtime. AEL converted into standard extensions.conf syntax in the dialplan. Doesn't this render having used AEL pointless? Absolutely not! Reasons to use AEL: 1. Several semantic checks are done on the AEL that are NOT done if you go straight to extensions.conf. We try to protect you... from yourself. 2. At least one security issue in USAGE is avoided by having AEL compile the corresponding code; as to how many more issues will automatically be handled via AEL in the future, is impossible to say. We'll see. If you keep coding via extensions.conf, be prepared to make corrections... if you do it in AEL, a restart of Asterisk will hopefully suffice, after AEL is updated. 3. Syntax errors are reported by AEL. It is pretty good at catching all omissions and commissions. Better than the extensions.conf parser is. For example, I don't know if we catch it now, but if you accidentally say extem = 3,... instead of exten = 3,... in extensions.conf, that line will silently be dropped. Sure, we could fix this, but to fix ALL possible problems will require an expensive rewrite of the config file parser, from the ground up. 4. You are insulated against any mods to extensions.conf; like the change to ',' instead of '|' in app arguments. No changes to AEL code are necessary. 5. In extensions.conf, you have to feed your dialplan to asterisk to find any problems. AEL provides the standalone parser, aelparse, so you can correct any problems BEFORE feeding it to a living asterisk. 6. AEL is easier to read, IF you take advantage of the ability to use tabs, etc. wisely. Especially for nested code. Staying away from goto as much as possible, and using the flow of control and looping statements will make your code easier to read, compose, and maintain in the future. It means fewer bugs in your code, and overall this all means lower cost. And higher profits. 7. Repetitious entry of extenname, priority, in your tabular extensions.conf can lead to subtle errors that could be hard to find, ESPECIALLY if you resort to using priority NUMBERS instead of n. And, if you ARE so foolish as to use just raw numbers, and you have to insert or delete a line or two, you have to renumber the remaining lines, and heaven help you if you make a simple error, and accidentally skip a number. 8. Work flow. Since aelparse allows you to dump the compiled dialplan in extensions.conf format, you can still use stuff like realtime. You can use this output against machines that don't even have pbx_ael loaded, then, and you should be able to use 1.4 compiled dialplans on 1.2 machines, as long as you are careful about what apps you call, and how you call them. 9. Easier to write code. Good Code. using Goto's in extensions.conf will allow you to do anything you need to do, but it also results in spaghetti style code. While the original author might be able to decrypt it, and maintain it, unless it's really well commented, the next guy to play with it, is going to have a hard time. Following the flow of control thru spaghetti can get your adrenalin flowing-- and side affects from strange cases and leakage in the spaghetti can make some devilishly hard to solve problems. Think of and treat extensions.conf like assembly code. Think of and treat AEL like a high(er) level language. For those who never did the computer science thing, I have just one piece of advise, and ignore this at your peril: your dialplan is a work of computer programming. It's software. If you don't treat it that way, and use good software methodologies, you'll pay your price. murf smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best config for 12 FXO system?
Thomas Kenyon wrote: Tony Mountifield wrote: I have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot availability, are there any technical reasons to choose one of the above configurations over the others? No idea, but if you look further afield, if you buy a Sangoma A200 or an A400 you can have all 12 on one PCI (or PCI Express) slot (the former taking up 3 Spaces on your PCs backplane and the latter taking up only 1). If expandability is a concern, an A400 can support up to 48 FXO ports on one PCI (or PCI-Express) Slot (4 spaces) or an A200 can support up to 24 FXO ports. (6 spaces) I can't comment on how good they are, I've only got TDM400Ps myself. Rhino also makes some very nice cards and have a good support staff. The Rhino cards are also made in the US. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best config for 12 FXO system?
Mojo with Horan Company, LLC wrote: Those are all analog though, aren't they? What about a channel bank into a digital card? Might that be cheaper than shelling out for 12 FXO ports and the cards to hold them? Just wanted to throw that out there before the discussion started :) It might well be, but try as I might, I can't find a channel bank that supports 12 FXO ports that would be less than an A400 with 6x2 FXO cards (£490 the first place I looked, with only software echo cancellation or £644 with HW-EC). Or as the above poster mentioned a Rhino card (would be much cheaper than sangoma if you required 24 ports and Hardware Echo cancellation). ___ --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] Configuration files inside SQLite3
Thank you very much, Mark. =) On 10/4/07, Mark Michelson [EMAIL PROTECTED] wrote: GNUbie wrote: Hello all, Is it possible to store, read and write configuration files in an SQLite3 database instead of using the configuration files inside the /etc/asterisk/ directory? If it is then can you point me to the right documentation on how to do this or probably hints on how to do this? Thank you in advance. GNUbie It is possible to store configuration files in any relational database which has ODBC compatibility. Thus, sqlite qualifies. If you are using trunk, you won't even need to use ODBC, because Asterisk has native support for sqlite. If you're looking for an overview of the Asterisk Realtime Architecture (the means by which you can store configurations in a database) look in the doc directory of your asterisk source for realtime.txt and extconfig.txt, or search voip-info.org for asterisk realtime. If you're looking for more in-depth coverage of integrating Asterisk with a relational database, I suggest looking at the second edition of Asterisk: The Future of Telephony, available at book stores, or for download at http://openbooks.oreilly.com/ Specifically, check out chapter 12. It doesn't cover sqlite explicitly, but it's not much of a stretch to use it based on what's provided in the book. Mark Michelson ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No audio on Zap (T1/PRI) channels
I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards. Each group of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback demo-congrats fine. The middle server (ts21) cannot -- just dead air. If I call via ZAP, dead air. If I call via IAX, I hear the file. I copied /etc/zaptel.conf, /etc/asterisk/*, /var/lib/asterisk/sounds/demo-congrats.gsm from ts20 to ts21 -- no joy. I have seen this in my system log file: Oct 2 18:41:49 WARNING[7477]: chan_zap.c:8087 zt_pri_error: [Span 0 D-Channel 0] PRI: !! Got reject for frame 95, but we have nothing -- resetting! I'm running asterisk-1.2.24, asterisk-addons-1.2.7, libpri-1.2.5, zaptel-1.2.20.1. show channel zap/?, zap show channel ? appear identical between working and non-working systems both on-hook and off-hook. Any clues or clues where to start looking? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parking lot problems
Ken Williams wrote:
Now on to another problem that we've had as far as I know since the
beginning of using Asterisk 9+ months ago. I've been trying very hard
to knock this problem out but regardless of what I do, it's still there.
[from-internal]
include = parkedcalls
I have this.
exten = _20X,1,Goto(parkedcalls,${EXTEN},1)
Is this really necessary? I only have the include. parked calls are
created on the fly. Our lot is from 90-99 and we don't need an entry in
the dial plan to handle it.
[park-dial]
exten = t,1,Goto(from-internal,900,1)
I have similar to this as well.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] No audio on Zap (T1/PRI) channels
On Wed, 3 Oct 2007, Steve Edwards wrote: I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards. Each group of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback demo-congrats fine. The middle server (ts21) cannot -- just dead air. If I call via ZAP, dead air. If I call via IAX, I hear the file. I copied /etc/zaptel.conf, /etc/asterisk/*, /var/lib/asterisk/sounds/demo-congrats.gsm from ts20 to ts21 -- no joy. I have seen this in my system log file: Oct 2 18:41:49 WARNING[7477]: chan_zap.c:8087 zt_pri_error: [Span 0 D-Channel 0] PRI: !! Got reject for frame 95, but we have nothing -- resetting! I'm running asterisk-1.2.24, asterisk-addons-1.2.7, libpri-1.2.5, zaptel-1.2.20.1. show channel zap/?, zap show channel ? appear identical between working and non-working systems both on-hook and off-hook. Any clues or clues where to start looking? Turning on pri debug span 1 yields: Oct 3 10:39:17 WARNING[20586]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0xb7c202f0', 9 retries! on the host that doesn't work for every call received. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get asterisk to take a dump?
I have an asterisk process that is consuming over 100mb (according to top). Show channels says 167 active channels and 53 active calls. It's an old install -- 1.2.7.1, but it has custom code that needs to be updated before moving to a more recent release. I'm assuming that 100mb is indicative of a memory leak (probably in my code). How can I get a dump (preferably without disrupting production) so I can poke around in it (using gdb) and what's a good strategy for finding memory leaks? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get asterisk to take a dump?
On Wednesday 03 October 2007 20:48:37 Steve Edwards wrote: install -- 1.2.7.1, but it has custom code that needs to be updated before moving to a more recent release. I'm assuming that 100mb is indicative of a memory leak (probably in my code). How can I get a dump (preferably without disrupting production) so I can poke around in it (using gdb) and what's a good strategy for finding memory leaks? Thanks in advance, I think, there's no way you can get a coredump without interrupting process. However you can do killall -5 asterisk. That would send a Trace/Breakpoint signal to asterisk and it would crash immediately to core - so you can play with it in gdb. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --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] ping
Steve Totaro wrote: must be blacklisted, i have posted like 4 messages and none are showing up. That's what I thought, too, but there's some weirdness going on with Digium's list server spam filtering. Anyway, you'll probably see this one :) -Stephen- ___ --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] No audio on Zap (T1/PRI) channels
Steve Edwards wrote: I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards. Each group of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback demo-congrats fine. The middle server (ts21) cannot -- just dead air. If I call via ZAP, dead air. If I call via IAX, I hear the file. I copied /etc/zaptel.conf, /etc/asterisk/*, /var/lib/asterisk/sounds/demo-congrats.gsm from ts20 to ts21 -- no joy. I have seen this in my system log file: Oct 2 18:41:49 WARNING[7477]: chan_zap.c:8087 zt_pri_error: [Span 0 D-Channel 0] PRI: !! Got reject for frame 95, but we have nothing -- resetting! I'm running asterisk-1.2.24, asterisk-addons-1.2.7, libpri-1.2.5, zaptel-1.2.20.1. show channel zap/?, zap show channel ? appear identical between working and non-working systems both on-hook and off-hook. Any clues or clues where to start looking? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Double check both zaptel.conf and zapata.conf and also call the telco to make sure they have they have the same NFAS scheme on all T1s setup correctly. Sometimes (let's face it, alot of times, the provider messes something up). Also check that all of your T1 cables are plugged into the correct T1 port. I have made that mistake myself when doing 28 T1s off a T3. I got dead air just as you described. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Executing commands even if user hangs up.
Greetings, I have a dialplan that calls the dictate application, but I want to do some post-processing on the RAW file created. The post processing is working fine as long as the dictation application exits gracefully, but fails when the user simply hangs up. How can I make sure the system() command is run regardless? Example: [test-dictation] exten = 123,1,Dictate(/tmp/dictate) exten = 123,2,System(post_processing_script.sh) exten = 123,3,Wait,1 exten = 123,4,Hangup Thanks -jc ___ --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] No audio on Zap (T1/PRI) channels
Steve Edwards wrote: On Wed, 3 Oct 2007, Steve Edwards wrote: I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards. Each group of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback demo-congrats fine. The middle server (ts21) cannot -- just dead air. If I call via ZAP, dead air. If I call via IAX, I hear the file. I copied /etc/zaptel.conf, /etc/asterisk/*, /var/lib/asterisk/sounds/demo-congrats.gsm from ts20 to ts21 -- no joy. I have seen this in my system log file: Oct 2 18:41:49 WARNING[7477]: chan_zap.c:8087 zt_pri_error: [Span 0 D-Channel 0] PRI: !! Got reject for frame 95, but we have nothing -- resetting! I'm running asterisk-1.2.24, asterisk-addons-1.2.7, libpri-1.2.5, zaptel-1.2.20.1. show channel zap/?, zap show channel ? appear identical between working and non-working systems both on-hook and off-hook. Any clues or clues where to start looking? Turning on pri debug span 1 yields: Oct 3 10:39:17 WARNING[20586]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0xb7c202f0', 9 retries! on the host that doesn't work for every call received. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 That does not look like PRI Debug output to me, just console verbose. From experience, I would say you crossed a couple of cables. Make sure you double check them and then LABEL them. If you find that is not the solution, call your provider. Thanks, Steve Totaro ___ --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] Executing commands even if user hangs up.
Jim Canfield wrote: Greetings, I have a dialplan that calls the dictate application, but I want to do some post-processing on the RAW file created. The post processing is working fine as long as the dictation application exits gracefully, but fails when the user simply hangs up. How can I make sure the system() command is run regardless? Example: [test-dictation] exten = 123,1,Dictate(/tmp/dictate) exten = 123,2,System(post_processing_script.sh) exten = 123,3,Wait,1 exten = 123,4,Hangup Thanks -jc If you're always running your post processing script after the call is over, I'd suggest moving the System command to the h extension. The h extension is called on hangup, so it should clear up your issue. Mark Michelson ___ --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] Executing commands even if user hangs up.
Have you tried adding an 'h' extension in addition? If the caller hangs up in the middle of priority 1 of extension 123, it should then jump to priority 1 of extension h and continue. ;Add to the test-dictation context: exten = h,1,System(post_processing_script.sh) OR ;Not tested, but maybe just the following single line instead? exten = h,1,Goto(123, 2) Jim Canfield wrote: Greetings, I have a dialplan that calls the dictate application, but I want to do some post-processing on the RAW file created. The post processing is working fine as long as the dictation application exits gracefully, but fails when the user simply hangs up. How can I make sure the system() command is run regardless? Example: [test-dictation] exten = 123,1,Dictate(/tmp/dictate) exten = 123,2,System(post_processing_script.sh) exten = 123,3,Wait,1 exten = 123,4,Hangup Thanks -jc ___ --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] Executing commands even if user hangs up.
Mojo with Horan Company, LLC wrote: Have you tried adding an 'h' extension in addition? If the caller hangs up in the middle of priority 1 of extension 123, it should then jump to priority 1 of extension h and continue. Thanks, That works perfectly. ___ --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] ping
Someone who's having trouble posting to the list should try placing [asterisk-users] or Re:in the subject line of a new email they send (near the END of the subject so it doesn't obscure the actual subject or have superfluous Re:'s near the beginning) to see if the spam filter is more likely to score the messages as ham example subject: SIP Re-registration does not occur after expiry, Asterisk 1.2 [asterisk-users] someone can smack me if this breaks some etiquette I'm not detecting, but it might be a back door :) Stephen Bosch wrote: Steve Totaro wrote: must be blacklisted, i have posted like 4 messages and none are showing up. That's what I thought, too, but there's some weirdness going on with Digium's list server spam filtering. Anyway, you'll probably see this one :) -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAXy and hook flash transfer
In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here Any help is appreciated. Here is features.conf: ; ; Sample Parking configuration ; [general] parkext = 700 ; What extension to dial to park parkpos = 701-720 ; What extensions to park calls on. These needs to be ; numeric, as Asterisk starts from the start position ; and increments with one for the next parked call. context = parkedcalls ; Which context parked calls are in ;parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds) ;transferdigittimeout = 3 ; Number of seconds to wait between digits when transferring a call ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep; to indicate an attended transfer is complete xferfailsound = beeperr ; to indicate a failed transfer ;adsipark = yes ; if you want ADSI parking announcements ;findslot = next ; Continue to the 'next' free parking space. ; Defaults to 'first' available ;pickupexten = *8 ; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = #1 ; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2; Attended transfer [applicationmap] ; Note that the DYNAMIC_FEATURES channel variable must be set to use the features ; defined here. The value of DYNAMIC_FEATURES should be the names of the features ; to allow the channel to use separated by '#'. For example: ;Set(DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3) ; ;testfeature = #9,callee,Playback,tt-monkeys ;Play tt-monkeys to ;callee if #9 was pressed Yours, Michael Munger, dCAP 404-438-2128 [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] Secondary Dialtone and selecting a specific line from Zap/g
It would be ugly, but you could prefix a zap channel or group number
before the phone number to dial. Using groups for an example:
exten = _*X*X.,1,Dial(ZAP/g${EXTEN:1:1}/${EXTEN:3})
exten = _*XX*X.,1,Dial(ZAP/g${EXTEN:1:2}/${EXTEN:4})
so dialing *4*18005551212 dials out over zap group 4...
bilal ghayyad wrote:
I need to select a line from the Zap group channel
using the SIP Phone (not FXO and not FXS ports).
ignorepat does not work?
Also, what is the method to let the second dial tone
has another tone frequency?
Regards
Bilal
No, ignorepat is for FXS ports (FXS ports use FXO
signaling). Also,
ignorepat does not apply to SIP phones, because SIP
phones provide
their
own dialtone, not a dialtone provided by Asterisk.
Al lists wrote:
Correction, on FXO port not FXS,
second, read his email first:
Also, how it will be possible to assign an
dedicated
line (connected to FXO) to an
button on the Polycom IP Phone or Broadtel IP Phone,
so if user select that button
then he will be sure that his outside call will be
via
that specific line.
Just assign a key on your phone to dial that
extension, and you will
have
dial tone on selected line,
then as a traditional PBX you can send any digits to
your provider.
On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:
ignorepat continues dialtone after a leading digit
has been dialed
on
FXS ports. How does ignorepat help this guy?
Al lists wrote:
ignorpat is your friend
On 9/30/07, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal
ghayyad wrote:
Dear List;
How can I place a call via Zap/g1 (group) but
need to
determine the line (FXO port)
that will go via it?
Simply don't use groups. Use channels directly.
To dial via the
specific
Zaptel channel NN, use Zap/NN
Am I missing anything?
Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for
today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow
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Re: [asterisk-users] IAXy and hook flash transfer
On Wednesday 03 October 2007 22:21:24 Michael Munger wrote: In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here Any help is appreciated. Do you have t and/or T flag set in Dial() options? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --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 and hook flash transfer
Michael Munger wrote: In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here…. Any help is appreciated. Those features are triggered via DTMF, not using a protocol-level transfer. The IAXy uses IAX2 to talk to Asterisk, so doing a flash-hook on the IAXY's FXS port will cause the IAXy to create a new IAX2 channel and handle the transfer itself. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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] No audio on Zap (T1/PRI) channels
On Wed, 3 Oct 2007, Steve Totaro wrote: Steve Edwards wrote: I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards. Each group of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback demo-congrats fine. The middle server (ts21) cannot -- just dead air. If I call via ZAP, dead air. If I call via IAX, I hear the file. I copied /etc/zaptel.conf, /etc/asterisk/*, /var/lib/asterisk/sounds/demo-congrats.gsm from ts20 to ts21 -- no joy. I have seen this in my system log file: Oct 2 18:41:49 WARNING[7477]: chan_zap.c:8087 zt_pri_error: [Span 0 D-Channel 0] PRI: !! Got reject for frame 95, but we have nothing -- resetting! I'm running asterisk-1.2.24, asterisk-addons-1.2.7, libpri-1.2.5, zaptel-1.2.20.1. show channel zap/?, zap show channel ? appear identical between working and non-working systems both on-hook and off-hook. Any clues or clues where to start looking? Double check both zaptel.conf and zapata.conf and also call the telco to make sure they have they have the same NFAS scheme on all T1s setup correctly. Sometimes (let's face it, alot of times, the provider messes something up). Also check that all of your T1 cables are plugged into the correct T1 port. I have made that mistake myself when doing 28 T1s off a T3. I got dead air just as you described. I think this is the problem, or at least a problem. When Qwest pointed a number to the group, I noticed that calls arrived on ascending channel numbers. I observed... Accepting call from '202239' to '866205' on channel 0/1, span 1 Executing Answer(Zap/1-1, ) in new stack Subsequent calls show the following: Channel SpanZap 0/1 1 1 . . . 0/231 23 1/1 1 73 . . . 1/231 95 2/1 1 25 . . . 2/241 48 3/1 1 49 . . . 3/241 72 So it looks to me like the T1 that is plugged into the second jack on the card really belongs in the 4th jack. Can I fix this by munging my configuration? (I'm in San Diego and the servers are in Phoenix.) Here's my zaptel.conf: # # span 1 span= 1,1,0,esf,b8zs bchan = 1-23 dchan = 24 # # span 2 span= 2,0,0,esf,b8zs bchan = 25-48 # # span 3 span= 3,0,0,esf,b8zs bchan = 49-72 # # span 4 span= 4,2,0,esf,b8zs bchan = 73-95 dchan = 96 # # (end of /etc/zaptel.conf) And my zapata.conf: [trunkgroups] trunkgroup = 1,24,96 spanmap = 1,1,0 spanmap = 2,1,2 spanmap = 3,1,3 spanmap = 4,1,1 [channels] context = block-ani echocancel = no echocancelwhenbridged = no echotraining= no group = 1 resetinterval = never signalling = pri_cpe switchtype = dms100 ; switchtype = 4ess ; span 1 (1-24) channel = 1-23 ; span 2 (25-48) channel = 25-48 ; span 3 (49-72) channel = 49-72 ; span 4 (73-96) channel = 73-95 ; ; (end of /etc/asterisk/zapata.conf) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXy and hook flash transfer
So what, then, is the procedure to transfer a call from a POTS phone on the FXS port of an IAXy? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, October 03, 2007 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAXy and hook flash transfer Michael Munger wrote: In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here Any help is appreciated. Those features are triggered via DTMF, not using a protocol-level transfer. The IAXy uses IAX2 to talk to Asterisk, so doing a flash-hook on the IAXY's FXS port will cause the IAXy to create a new IAX2 channel and handle the transfer itself. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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] No audio on Zap (T1/PRI) channels
Steve Edwards wrote: [trunkgroups] trunkgroup = 1,24,96 spanmap = 1,1,0 spanmap = 2,1,2 spanmap = 3,1,3 spanmap = 4,1,1 You caused the behavior you are seeing by configuring your spanmap this way; you've got physical span #4 configured as the second span in the trunkgroup, so Zaptel will treat physical channels 73-95 as logical channels 1/1 through 1/23. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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] Agent Callback Login in 1.4
Can you describe exactly what you lose by using the dynamic queue member alternative? We tried to ensure that no functionality was lost in this transition, so if there is something that was missed please let us know what it is and we'll try to take care of it. Now, i'm finally trying to migrate, and i see a problem here. When i was using Agent channels there was status Busy indicated in show queues, whenever agent was on call from queue. I'm trying to do all the stuff with RT queue members and Local channels, but i'm missing this. I have read about GROUP usage in Local channel - so that upon call arrival Local channel can indicate that it's busy, however this is not executed upon show queues - so no status changes occur. I believe this have some connection with ast_device_state_changed, but it's only available in chan_agent, that as i understand is deprecated. Is there any other way how i would get status indication in show queues? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --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 and hook flash transfer
When I was unable to figure out the IAXy's methods, I went with Asterisk's features.conf -- ## for blindxfer, and never looked back. That worked quite well. Michael Munger wrote: So what, then, is the procedure to transfer a call from a POTS phone on the FXS port of an IAXy? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, October 03, 2007 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAXy and hook flash transfer Michael Munger wrote: In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here Any help is appreciated. Those features are triggered via DTMF, not using a protocol-level transfer. The IAXy uses IAX2 to talk to Asterisk, so doing a flash-hook on the IAXY's FXS port will cause the IAXy to create a new IAX2 channel and handle the transfer itself. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension length
Hi list, Is there a limit on the length of an extension? I have an 18 byte long extension, when issuing goto, Asterisk comes back with invalid extension on the console. Anyone had this experience before? ___ --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] Best config for 12 FXO system?
If you want this to work nicely dont settle for anything else than a channel bank On 10/3/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Mojo with Horan Company, LLC wrote: Those are all analog though, aren't they? What about a channel bank into a digital card? Might that be cheaper than shelling out for 12 FXO ports and the cards to hold them? Just wanted to throw that out there before the discussion started :) It might well be, but try as I might, I can't find a channel bank that supports 12 FXO ports that would be less than an A400 with 6x2 FXO cards (£490 the first place I looked, with only software echo cancellation or £644 with HW-EC). Or as the above poster mentioned a Rhino card (would be much cheaper than sangoma if you required 24 ports and Hardware Echo cancellation). ___ --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] Extension length
I am assuming you mean 18 digits long. it shouldnt be a problem you mind posting your configs? On 10/3/07, Wai Wu [EMAIL PROTECTED] wrote: Hi list, Is there a limit on the length of an extension? I have an 18 byte long extension, when issuing goto, Asterisk comes back with invalid extension on the console. Anyone had this experience before? ___ --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] IAXy and hook flash transfer
It just dawned on me, that I can just press the hook button momentarily to open up a second IAX channel, dial the number, and hangup to complete the transfer. Thanks everyone! Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Wednesday, October 03, 2007 4:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAXy and hook flash transfer When I was unable to figure out the IAXy's methods, I went with Asterisk's features.conf -- ## for blindxfer, and never looked back. That worked quite well. Michael Munger wrote: So what, then, is the procedure to transfer a call from a POTS phone on the FXS port of an IAXy? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, October 03, 2007 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAXy and hook flash transfer Michael Munger wrote: In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here Any help is appreciated. Those features are triggered via DTMF, not using a protocol-level transfer. The IAXy uses IAX2 to talk to Asterisk, so doing a flash-hook on the IAXY's FXS port will cause the IAXy to create a new IAX2 channel and handle the transfer itself. ___ --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] No audio on Zap (T1/PRI) channels
Steve Edwards wrote: On Wed, 3 Oct 2007, Steve Totaro wrote: Steve Edwards wrote: I have 12 T1's going into 3 servers, 4 in each into Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02) cards. Each group of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback demo-congrats fine. The middle server (ts21) cannot -- just dead air. If I call via ZAP, dead air. If I call via IAX, I hear the file. I copied /etc/zaptel.conf, /etc/asterisk/*, /var/lib/asterisk/sounds/demo-congrats.gsm from ts20 to ts21 -- no joy. I have seen this in my system log file: Oct 2 18:41:49 WARNING[7477]: chan_zap.c:8087 zt_pri_error: [Span 0 D-Channel 0] PRI: !! Got reject for frame 95, but we have nothing -- resetting! I'm running asterisk-1.2.24, asterisk-addons-1.2.7, libpri-1.2.5, zaptel-1.2.20.1. show channel zap/?, zap show channel ? appear identical between working and non-working systems both on-hook and off-hook. Any clues or clues where to start looking? Double check both zaptel.conf and zapata.conf and also call the telco to make sure they have they have the same NFAS scheme on all T1s setup correctly. Sometimes (let's face it, alot of times, the provider messes something up). Also check that all of your T1 cables are plugged into the correct T1 port. I have made that mistake myself when doing 28 T1s off a T3. I got dead air just as you described. I think this is the problem, or at least a problem. When Qwest pointed a number to the group, I noticed that calls arrived on ascending channel numbers. I observed... Accepting call from '202239' to '866205' on channel 0/1, span 1 Executing Answer(Zap/1-1, ) in new stack So it looks to me like the T1 that is plugged into the second jack on the card really belongs in the 4th jack. Can I fix this by munging my configuration? (I'm in San Diego and the servers are in Phoenix.) I guess you could work around it by messing with your configuration and changing channel numbers and putting your backup D on the correct channels. I would highly advise against it though. You should call someone in Phoenix (just left there) and have them correct the cabling so it is done the proper way. Even if you do get it working with the channels all mangled out of order, it will bite you sometime down the road (or someone else) when some sort of problem arises and you are trying to troubleshoot it. Thanks, Steve Totaro ___ --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] No audio on Zap (T1/PRI) channels
Kevin P. Fleming wrote: Steve Edwards wrote: [trunkgroups] trunkgroup = 1,24,96 spanmap = 1,1,0 spanmap = 2,1,2 spanmap = 3,1,3 spanmap = 4,1,1 You caused the behavior you are seeing by configuring your spanmap this way; you've got physical span #4 configured as the second span in the trunkgroup, so Zaptel will treat physical channels 73-95 as logical channels 1/1 through 1/23. If it were configured as the second span, shouldn't is be channels 25-48 rather than 1-23? voip-info was very unclear about this when I looked at it over a year ago. I finally got it working by trying different combinations in spanmap. Digium should have it's own wiki that is maintained by Digium. Voip-info is ok but much of it is old and or incorrect at this point. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Where to download Junghanns ISDNguard software?
Hi list, I recently purchased an ISDNguard from Junghanns. It came with no software and there is no sign on their website or in any of their documentation where to download it. I have looked in http://www.junghanns.net/downloads/ and there is no sign of it there either. The only thing remotly close ther is isdnguard-asterisk-1.2.13.patch. Their documentation refers to /usr/sbin/ISDNguard. Where does one get this mysterious binary from? I have emailed their support a few times and get no response, needless to say I am NOT a happy customer. Can anyone help me with a download link? Thanks in advance.. - Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using PHP to reload extensions
I am trying to use PHP to reload the extensions in an Asterisk
installation. I keep getting this error:
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
when I run the script by visiting the URL; however, if I run the script
from the command line, it runs just fine (works perfect, actually).
I think it is permissions related. Does anyone have any ideas?
php
$output = shell_exec('asterisk -rxextensions reload');
echo $output;
?
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
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Re: [asterisk-users] Secondary Dialtone and selecting a specific line from Zap/g
Here is how i overcome this problem,
ignorpat = 9
exten = 9*,1,Dial(ZAP/1/w)
press 9* from your handset and after 1 second you have POTS line dial tone
on your phone,
On 10/3/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
It would be ugly, but you could prefix a zap channel or group number
before the phone number to dial. Using groups for an example:
exten = _*X*X.,1,Dial(ZAP/g${EXTEN:1:1}/${EXTEN:3})
exten = _*XX*X.,1,Dial(ZAP/g${EXTEN:1:2}/${EXTEN:4})
so dialing *4*18005551212 dials out over zap group 4...
bilal ghayyad wrote:
I need to select a line from the Zap group channel
using the SIP Phone (not FXO and not FXS ports).
ignorepat does not work?
Also, what is the method to let the second dial tone
has another tone frequency?
Regards
Bilal
No, ignorepat is for FXS ports (FXS ports use FXO
signaling). Also,
ignorepat does not apply to SIP phones, because SIP
phones provide
their
own dialtone, not a dialtone provided by Asterisk.
Al lists wrote:
Correction, on FXO port not FXS,
second, read his email first:
Also, how it will be possible to assign an
dedicated
line (connected to FXO) to an
button on the Polycom IP Phone or Broadtel IP Phone,
so if user select that button
then he will be sure that his outside call will be
via
that specific line.
Just assign a key on your phone to dial that
extension, and you will
have
dial tone on selected line,
then as a traditional PBX you can send any digits to
your provider.
On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:
ignorepat continues dialtone after a leading digit
has been dialed
on
FXS ports. How does ignorepat help this guy?
Al lists wrote:
ignorpat is your friend
On 9/30/07, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal
ghayyad wrote:
Dear List;
How can I place a call via Zap/g1 (group) but
need to
determine the line (FXO port)
that will go via it?
Simply don't use groups. Use channels directly.
To dial via the
specific
Zaptel channel NN, use Zap/NN
Am I missing anything?
Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's
updated for today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow
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Re: [asterisk-users] meetme conference using g729?
On Wed, 3 Oct 2007 08:35:06 -0500, Tilghman Lesher wrote: I invite you to try it. You could make a lot of really smart people look like fools if you're able to mix compressed audio together without decompressing, or you might make yourself look like a fool, because you get back garbage for attempting to mix compressed data. I wholly understand the problem here. You can't, at present, mix compressed audio stream, in compressed domain. You must decode them to baseband, do the manipulation, then re-encode. OK, we get that. That's today. Such things have parallels in my day job, which is television production transmission. At least in the US the signal that a TV station delivers to its DTV transmitter (ie the new digital one, not the old analog one that the feds will make us turn off in 2008) that is a compressed stream. Typically MPEG2 @ 19.2 MPBS. There was a time when that was a signal stream that could not be manipulated. It was just the transport mechanism from the last leg before the transmitter. Many companies wanted to be able to perform what seemed simple manipulations on the stream, for example to add a station logo, without taking the very significant quality hit of decompression and recompression. Such hardware systems have become available over time. Manipulation of the transmission streams in the compressed domain is possible, but its very compute intensive...and so expensive. It's done in massively parallel hardware architecture. There are a few vendors in the broadcast business who provide such systems. And that's for high bandwidth broadcast video. It would also be possible for voice streams, but the math is very complex. Hardware acceleration of encoding is already very common, witness Digium's own encode/decode board. Given the right motivation to spur the development this could be possible. In truth I suspect that there's little economic reason to do it. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using PHP to reload extensions
Michael Munger wrote:
I am trying to use PHP to reload the extensions in an Asterisk
installation. I keep getting this error:
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
when I run the script by visiting the URL; however, if I run the script
from the command line, it runs just fine (works perfect, actually).
I think it is permissions related. Does anyone have any ideas?
php
$output = shell_exec('asterisk -rxextensions reload');
echo $output;
?
I guess your web server does not run as root and thus is not
allowed to invoke asterisk. (Try
echo shell_exec('id');
or
echo get_current_user();
in PHP.)
A possible solution (although not nice): Add
www-data ALL=(ALL) NOPASSWD: ALL
to /etc/sudoers (depending on your distribution etc. the Apache user
might be www-data / apache / ...) and in the PHP script run
shell_exec('sudo asterisk -rx extensions reload');
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
My pick of the month: rfc 2822 3.6.5
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [asterisk-users] Using PHP to reload extensions
If you are running the script from a web server, the script gets
executed with the web server process permissions, hence, probably does
not have access to /var/run/asterisk.ctl.
You can give permissions to your web server, or better yet, dont
execute the command using shell_exec, better open a socket connection
to the Asterisk manager and execute Action: Command
Command: extensions reload
Regards
On 10/3/07, Michael Munger [EMAIL PROTECTED] wrote:
I am trying to use PHP to reload the extensions in an Asterisk installation.
I keep getting this error:
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
when I run the script by visiting the URL; however, if I run the script from
the command line, it runs just fine (works perfect, actually).
I think it is permissions related. Does anyone have any ideas?
php
$output = shell_exec('asterisk -rxextensions reload');
echo $output;
?
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
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Within C++, there is a much smaller and cleaner language struggling
to get out.
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