Re: [asterisk-users] dialed number notify at invalid dial situation
you could try to set a var to the exten maybe.. and then use that var ..
since when in exten = i , well i will be the exten..
On Tue, Apr 15, 2008 at 11:52 AM, Anonymous [EMAIL PROTECTED] wrote:
Originally posted by: mailto:
Hi all
Now I'm making IVR sequance that is customised [mainmanu].
I wish to notify invaid command like a following
exten = i,1,playback('your command is ...')
exten = i,2,playback(${EXTEN}) ; Say 'i' oops! ;-(
exten = i,3,playback(' is incorrect! please again ')
# This exten lines are figure for instruction.
# I know to use with gsm filename.
but ${EXTEN} meaning 'i' that isn't dialed number.
Does anyone have good idea?
please help
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Dairiten.com - an open source VoIP and Ubiquitus Portal site in Japan.
http://www.dairiten.com:81/modules/news/
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Re: [asterisk-users] My G729 problem re-visited
How do you get 11ms translation time on ulaw 729 ? we have 12ms and its dual xeons 2.6.. On 9/26/07, Scott Moseman [EMAIL PROTECTED] wrote: Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI core show version Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux *CLI show modules like 723 Module Description Use Count codec_g723.so G.723.1 Coder/Decoder 0 format_g723.so G.723.1 Simple Timestamp File Format 0 *CLI show modules like 729 Module Description Use Count codec_g729.so G.729 Coder/Decoder 0 format_g729.so Raw G729 data 0 *CLI show translation [truncated] g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 ulaw 5 2 - 1 2 2 1 3 7 - 11 2 - alaw 5 2 1 - 2 2 1 3 7 - 11 2 - g729 5 2 2 2 2 2 1 3 - - 11 2 - The configuration... [gateway] type=friend host=gateway context=default-inbound disallow=all allow=g729 [phone] type=friend context=sip host=dynamic username=phone secret=scott dtmfmode=RFC2833 disallow=all allow=g729 callerid=Scott qualify=yes canreinvite=no exten = 1266,1,Dial(SIP/[number],30,t) exten = 1266,2,Congestion exten = 1266,1,Dial(SIP/[number],30) exten = 1266,2,Congestion (The same results using both of the above dialplans...) The environment... PSTN - Gateway - Asterisk - Phone What I'm seeing works... With the gateway setup to send both G711 and G729, it sends an INVITE which includes both G711 and G729 codecs. Asterisk sends an INVITE to my phone with only G729. The call is made and there's a conversation in G711 with the gateway and G729 with the phone. I assume this means Asterisk is transcoding. What Im seeing fails... With the gateway setup to send only G729, it sends an INVITE to Asterisk which includes only G729. Asterisk send an INVITE to the phone using G729, too. The 200 OK from the phone to the Asterisk includes G729. The 200 OK going from Asterisk to the gateway doesn't include ANY codec. The call is dropped the moment I pickup the phone to answer the call. My question... Why does Asterisk not want to respond to my gateway in G729? Even if the gateway requests it, Asterisk seems to just ignore it. From the transcoding call, and phone to phone G729 calls, I have proof that Asterisk knows how to handle G729 calls. Where do I go from here??? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation Provided 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] les.net losing DID's
We can port most of these numbers, give us a call to see how fast we can switch this over, Meanwhile we know Les, so we can ask them to push temporariliy to our switches while it's being transfered. On 8/9/07, Jaswinder Singh [EMAIL PROTECTED] wrote: Please stop advertising your forums/services on every single chance u get on users list . On 08/08/07, Al Bochter [EMAIL PROTECTED] wrote: That is why you need to start posting info about the providers at http://www.bochterservices.com/phpbb/ so everyone knows This is a FREE SERVICE provided by Bochter Services and it is not going away any time soon. There will be more added by your request Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Join our forum. This is where you can talk about VOIP You can overview some providers others have used. http://bochterservices.com/phpbb/ --- Stephen Bosch wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -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 Inbound (clean). Database: 000764-2, 08/08/2007 - 8/8/2007 5:31:56 PM ___ --Bandwidth and Colocation Provided 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 -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation Provided 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] The quest for making hint more flexible continues - using Realtime now
is subscribe context an addiotional switch/field ?
or its the peer context ?
On 8/9/07, Mike [EMAIL PROTECTED] wrote:
I feared so, but I have already started working on this. Thanks for the
confirmation.
Too bad, the rest of my design was relatively elegant (IMO) and easily to
modify.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: Thursday, August 09, 2007 12:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] The quest for making hint more flexible
continues - using Realtime now
Mike wrote:
Ok, now that I've learned I cannot use any variables when using the
`hint` priority (for BLF), I figured I'd try to use the next best
thing: hardcoded values using realtime. This way I avoid variables
such as ${ACCOUNTCODE} but I can at least change the DB more easily
than text files. This is the appropriate line in the DB:
+--+--+---+--++-+
| id | context | exten | priority | app| appdata
|
+--+--+---+--++-+
| 2000 | hint-context | 705 | hint | SIP/test-1 |
|
+--+--+---+--++-+
This is what I put in mt hint-context in extensions.conf:
[hint-context]
switch = Realtime/[EMAIL PROTECTED]
mailto:Realtime/[EMAIL PROTECTED]
And this is what I get from the CLI:
Aug 9 11:34:14 NOTICE[19894]: chan_sip.c:11187
handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] from xx.xxx.xx.xx, but there is no hint for
that extension
Wellthere is! Is there any way I can do this?
Mike
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I personally opened a bug in the bugtracker about this and it was closed
as
wont fix. You simply cannot use the hint priority in realtime with out a
major change to the API. So until the code is changed, you are going to
have
to have a separate hint context with nothing but hint priority extensions
and set the subscribe context in sip.conf for all concerned devices to
that
context.
This is how I am running in production now.
Anthony
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Re: [asterisk-users] PRI Question
hmm from what i have seen this is not supposed to be.. the info is still
there but should not be used in case of privacy..
zap show channels always show last info till a span refresh.. but the
privacy should indeed replace those with Privacy.
Maybe it could be a bug ,
On 8/9/07, Jeremy Mann [EMAIL PROTECTED] wrote:
I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco,
Span 2 sends to my existing phone system(Nortel).
My Span1 gets sent to the context from-pri, detailed here:
[from-pri]
exten = _49XX,1,Set(CALLERID(all)=${CALLERID(all)})
exten = _49XX,2,Dial(Zap/g2/${EXTEN},,twk)
exten = _49XX,3,Congestion()
exten = _49XX,4,Set(CALLERID(all)=)
exten = _49XX,5,Hangup()
exten = _49XX,103,Congestion()
exten = _49XX,104,Set(CALLERID(all)=)
exten = _49XX,105,Hangup()
exten = h,1,Set(CALLERID(all)=)
exten = h,2,Hangup()
I'm receiving caller ID fine, and setting it on the outgoing channel the
same I received it, is my logic above wrong? Will Asterisk natively pass
through the caller ID, or is there a better way to set it?
The reason I ask, is that calls that are not coming in with CLID(blocked
or private) are showing up as the same number that was previously answered
on that channel.
Thanks.
Using Asterisk 1.4 FYI.
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Re: [asterisk-users] blades?
yes we have several of these... you need to use a PCI expensiont slot.. its an addon that piggy backs to a blade and takes 1 u ..so total blade will take 2 u's... but you can hook 2 PCIS on it.. sangoma or whatever... This way we can Redundantly failover 2 PRIS on each other with each blade have 2 cards A102D's that or cross linked to each seperate PRI Circuit.. So Blade A has 2 A102d's and so to B Each 102's has 2 ports.. A 1a BTN1 1b TF1 2a BTN2 2b TF2 B 1a BTN1 1b TF1 2a BTN2 2b TF2 --- ! A 1a ! ! 1b ! ! 2a ! ! 2b ! --- --- ! B 1a ! ! 1b ! ! 2a ! ! 2b ! then you put a T1 switch module.. then its all automatic .. PRI DROP X ( Locals) failvoer on carrier side to Y PRI DROP Y (toll frees) failover on carrier dide to X On 6/6/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Dean Collins [EMAIL PROTECTED]: http://www.theregister.co.uk/2007/06/06/sun_thinner_blades/ this article got me thinking - is anyone running asterisk on blade servers? We have a bunch of ibm blades, but the issue at least with the H series cabinets we have is that there is no where to put any pci cards of any sort so you would be limited to purely a voip setup. There is a T series cabinet that allows pci cards for just such purposes as asterisk (T is telephony), but the information out there about just what pieces you need is pretty vague. Anyone have a no bs description of how the bits actually work together in that setup ? Any lessons for us to learn? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outlook dialing
http://outcall.sourceforge.net/ we use outcall and modded the source directly for our apps.. 0$ fee.. 100% flexibility.. Works like a charm ! On 6/6/07, Martin Smith [EMAIL PROTECTED] wrote: We've been using SIPTAPI and love it for our call center. We originally used ASTTAPI, but liked the idea of not running AstManProxy. http://siptapi.sourceforge.net/ - website for SIPTAPI http://projects.bebr.ufl.edu/wiki/AsteriskTAPI - our external documentation, for the outside world in case it helps :) Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, June 05, 2007 11:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Outlook dialing The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone else have a favorite Outlook autodial application they use and love? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan inquiry using GotoIf()
yes on home pbx i love the s/CALLERID..
maybe you should
f($[${CALLERID(number)} = 15552221313]?15:5)
try to isolate string to strings.
this is not good i think
you need qhotes on the callerid part too if you evaluate to the 1555xxx
f($[${CALLERID(number)} = 15552221313]?15:5)
maybe im wrong need another cofee
On 6/6/07, Steve Murphy [EMAIL PROTECTED] wrote:
On Wed, 2007-05-30 at 20:05 -0400, Steve Finkelstein wrote:
Thanks for the help on this thread all.
It would make sense if I write an AGI and incorporate a DB backend to
check against numbers I want explicitly dropped. If anyone has such a
utility, I'd love to -not- reinvent the wheel. Otherwise, I'll go whip
it up and probably provide a web frontend for adding/removing numbers.
You can still use the dialplan with the DB func to check incoming
CID info. Also, the Dial() app has several options for call screening
and
privacy; these would be performed when dialing your extension.
You can have Dial keep a DB of callers, and remember whether to always
just patch them right thru, play them a polite go away and don't come
back,
or send them off to torture scripts, or just route them straight to VM.
And, Dial() will ask you what you want to do, on the first call. Read
thru the Dial doc you get with core show application dial. There's
an option to store an intro from each caller, where it records in a
sound file, who they say they are. I have several hundreds of these, and
play them as the
call comes in, so we know who's calling without having to run to a CID
display.
For those who have poor to no vision, this can be a cool feature.
murf
- sf
C F wrote:
It fails because the right function is ${CALLERID(num)}
On 5/30/07, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to
be
failing on me in my dial plan. All I basically wish to do is block a
particular caller. Sounds easy enough, but my ternary operator/plan
currently is not properly being implemented. Can anyone spot where
I'm
being a momo?
All extensions get forwarded to the following macro:
[macro-forward]
; arg1 = phone number
; arg2 = timeout
; arg3 = extension (voicemail)
; arg4 = mobile number
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,PrivacyManager
exten = s,3,Wait(1)
exten = s,4,GotoIf($[${CALLERID(number)} = 15552221313]?15:5)
exten = s,5,GotoIf($[${LEN(${CALLERID(number)})} = 4]?6:8)
exten = s,6,AGI(didextlookup.agi|${CALLERID(number)})
exten = s,7,Set(CALLERID(number)=${didlookup})
exten = s,8,GotoIf($[${LEN(${CALLERID(number)})} = 10]?9:10)
exten = s,9,Set(CALLERID(number)=1${CALLERID(number)})
exten = s,10,Dial(${ARG1},${ARG2})
exten = s,11,GotoIf($[${EXISTS(${ARG4})}]?11:12)
exten = s,12,Dial(${ARG4},${ARG2})
exten = s,13,Voicemail(u${ARG3})
exten = s,14,Playback(vm-goodbye)
exten = s,15,HangUp
exten = s,105,HangUp
As you can tell, exten = s,4,GotoIf($[${CALLERID(number)} =
15552221313]?15:5) is what I recently added.
Here's what I see in the CLI logs:
-- Executing [EMAIL PROTECTED]:1] Macro(IAX2/lime-3,
forward|SIP/5609|15|5609|IAX2/[EMAIL PROTECTED]/15164766201) in new
stack
-- Executing [EMAIL PROTECTED]:1] Zapateller(IAX2/lime-3,
answer|nocallerid) in new stack
-- Executing [EMAIL PROTECTED]:2] PrivacyManager(IAX2/lime-3,
)
in new stack
-- CallerID Present: Skipping
-- Executing [EMAIL PROTECTED]:3] Wait(IAX2/lime-3, 1) in new
stack
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/lime-3, 0?15:5)
in
new stack
-- Goto (macro-forward,s,5)
It evaluates to false, hence goes to s,5. I keep dialing from that
particular number (the one in the example is clearly masked as a
false
CID), and verified it's showing up as that number on callerID.
Also one last question. Say I need to add more numbers to block in
the
future, is there an easier way to do this than renumbering my entire
macro? Renumbering everything is just begging for a typo which can
effectively render my dial plan broken.
Thank you kindly, everyone!
- sf
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Re: [asterisk-users] Voip-info.org
wget -q -O - --connect-timeout=5 http://www.voip-info.org |grep '149461' gives me the string.. its up for now.. could of been just rebooted. On 6/6/07, Roger Schreiter [EMAIL PROTECTED] wrote: Ed Nuñez schrieb: Is anyone else having trouble going into voip-info.org today? Yes. Me. Roger. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk call quality detection
also note vnaks are iax i think On 6/6/07, Henry Cobb [EMAIL PROTECTED] wrote: On 6/6/07, Matt [EMAIL PROTECTED] wrote: I chart VNAKs per hour. Would you care to share how you accomplish this? What programs do you use? grep VNAK /var/log/asterisk/full | cut -d ' ' -f 4 | cut -d : -f 1 | uniq -c Needs a bit of an adjustment between the 1-9th and 10th-31st of the month so I'm looking for something to chomp this automatically. -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral TTS and app_swift
what versions of asterisk on both systems ? On 6/5/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: Have you tried something along the lines of: System(swift blah blah blah -o blah.wav) Playback(blah.wav) It does have an inherent delay for the generation step but maybe swift binary segfaults less? I've only used cepstral via swift binary, and it has never segfaulted for me. My swift and voice are version 4.2.0. I doubt different voices behave differently, but just in case, I use the $7 Damien voice. Moj Julian Lyndon-Smith wrote: We are having some major problems with app_swift since we went live. It is regularly segfaulting. I don't know if this is my fault or not, but here's the story: Installed the cepstral voices (at the time, 4.0) on our test system (2.6.9-42.0.10.ELsmp) and later added some extra voices (now 4.2). All worked fine - we stress tested (20+ simultaneous calls). Move to live ( 2.6.9-22.0.1.ELsmp) . Installed the cepstral voices (only 4.2). Started having problems with only 5 calls: swift by itself on the command line was fine (it worked) but app_swift complained that it couldn't find any voices. Looking into /opt/swift/lib, I saw that it was different to my test system. On live I had (snipped) -rwxrwxrwx 1 root root 139612 Jun 1 23:10 libceplang_en.so =rwxrwxrwx 1 root root 139612 Jun 1 23:11 libceplang_en.so.4 -rwxr-xr-x 1 root root 139612 Jun 1 07:09 libceplang_en.so.4.2 -rwxrwxrwx 1 root root 547624 Jun 1 23:11 libceplex_uk.so -rwxrwxrwx 1 root root 547624 Jun 1 23:11 libceplex_uk.so.4 -rwxr-xr-x 1 root root 547624 Jun 1 07:09 libceplex_uk.so.4.2 on test I had lrwxrwxrwx 1 999 20202 20 Apr 24 16:17 libceplang_en.so - libceplang_en.so.4.2 lrwxrwxrwx 1 999 20202 20 Apr 24 16:17 libceplang_en.so.4 - libceplang_en.so.4.2 -rwxrwxr-x 1 999 20202 315933 Aug 17 2006 libceplang_en.so.4.1 -rwxrwxr-x 1 999 20202 139612 Mar 15 18:21 libceplang_en.so.4.2 lrwxrwxrwx 1 999 20202 19 Apr 24 16:17 libceplex_uk.so - libceplex_uk.so.4.2 lrwxrwxrwx 1 999 20202 19 Apr 24 16:17 libceplex_uk.so.4 - libceplex_uk.so.4.2 -rwxrwxr-x 1 999 20202 591033 Aug 17 2006 libceplex_uk.so.4.1 -rwxrwxr-x 1 999 20202 547624 Mar 15 18:20 libceplex_uk.so.4.2 I then removed all the non 4.2 libs and created a symbolic link to the 4.2 libs to match test. fired it all up, and app_swift then worked. Or so I thought. segfault - but not on every call. what I would like to know is: A) has anybody got a later version of app_swift (0.9.1) B) does anyone else use cepstral, and how ? C) what is the story with the cepstral libraries ? many thanks Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] debug logs
means (I)f (I) (R)emember (C)orrectly On 6/4/07, ram [EMAIL PROTECTED] wrote: This notifies you that it has been used (IIRC). Hi what does that mean , it has been IIRC ? ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls being dropped
that becasue the reinvite is using a private ip probably.. sip debug pastebin the results.. look in the re-invite part.. On 6/4/07, Compnet Bobby [EMAIL PROTECTED] wrote: We have the latest version of asterisk, on a xeon dell server (2gb ram), with 6 snom320's(latest firmware) and 3 grandstream gxp 2000's (latest stable firmware) and are having a few problems. We have a basic menu that transfers calls to different extensions. The problems can be found on all extensions. We have 2 different incoming providers and the problem happens on both providers. I want your input on 2 problems, they are the following: 1. 60% of the time everything works fine and there are no problems, 40% of times when the calls are transferred to an extension, after a few seconds , the call drops. The log from the server is below(this is from pickup to hangup, the main area of concern is where it says warning). -- Executing [EMAIL PROTECTED]:1] Answer(SIP/9097406868-09e110f8, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/9097406868-09e110f8, menus/welcome-to-exec) in new stack -- SIP/9097406868-09e110f8 Playing 'menus/welcome-to-exec' (language 'en') == CDR updated on SIP/9097406868-09e110f8 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/9097406868-09e110f8, SIP/103|50|m) in new stack -- Called 103 -- Started music on hold, class 'default', on SIP/9097406868-09e110f8 -- SIP/103-09dedd68 is ringing [May 29 09:05:22] WARNING[2678]: chan_sip.c:1899 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 1 (Critical Response) [May 29 09:05:22] WARNING[2678]: chan_sip.c:1916 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. -- Stopped music on hold on SIP/9097406868-09e110f8 == Spawn extension (from-sip, 103, 1) exited non-zero on 'SIP/9097406868-09e110f8' 2. When a call comes in or is transferred(not on outgoing), there is a delay until the person on the incoming line can hear you. We can hear them, but they can't hear us. Sometimes there is no delay, sometimes for person calling in cant hear you for 6 seconds. Thanks for the help in advance!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zap fallback
i would force a timer on it.. dial (blah,30) maybe that would bypass , maybe not.. i actually think it wont.. another example of this problem is DNS echo '1.2.3.4your.favorite.itsp' /etc/hosts then Dial(SIP/[EMAIL PROTECTED]) DNS failing will BLOCK the call indefinitely... On 5/18/07, Steve Kennedy [EMAIL PROTECTED] wrote: On Fri, May 18, 2007 at 12:54:19PM -0500, Matthew Fredrickson wrote: On May 18, 2007, at 11:50 AM, Steve Kennedy wrote: I'm trying to get zap fallback to VoIP working. I dial the zap channel and if it fails I want to then try another route. If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if there's no dial-tone, it doesn't seem to detect this? Using a TDM400 with UK settings. Asterisk does not do dialtone detection before it starts using a zap channel. That's probably why you are seeing this behavior. Oh well, have to think of another way around this. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone that supports central provisioning?
may i add , eyebeams confnig file is xml and could be generated , BUT, the password is hashed in some way.. any idea on that ? its a pretty long hash On 4/25/07, Senad Jordanovic [EMAIL PROTECTED] wrote: Andrew Furey wrote: On 24/04/07, Senad Jordanovic [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: Dear Senad, The setup program for your soft phone can be downloaded from here: a href=http://malwareserver.com/malware.exe;http://LINK/a During the setup you will be asked for configuration file. Please use attached file. Tzafrir is referring to possible link that user can receive from someone... Since I was referring to SYSTEM email message generated from within PBXware, above is not possible without some serious hacking of the network, the box, the chroot etc... If one is at that level it then becomes a criminal issue. Not denying the criminal aspect, but who says the email has to really come from that box? If there's one thing SMTP is good at, it's allowing forged emails... it wouldn't take a decent phisher 10 minutes to craft an email that has all the same content including From addresses. Sure, the full headers would give up the game - but how many of your users would (a) check them, and (b) understand what they're seeing? I'd be surprised if it's more than 5% - and in many cases it only takes one person to fall for it... Andrew Hi Yeah, all valid points. Thanks for bringing this up. In order to eliminate above the setup program is actually in user self care on the local box. That is where the link refers to. The user self care is password protected. In addition, all of the above is on LAN. For someone to know there is installation going on at some LAN is very private matter so anyone wanting send these emails will have to be psychic. Regards, Senad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Purchasing a Sangoma A102 - should I get the hw echo cancellation or not?
echo cancel all the way, any company not including that in first place is just selling a car without the wheels.. i would see the a card without echo cancel as driving in winter with summer tires.. On 4/23/07, Erik Anderson [EMAIL PROTECTED] wrote: On 4/23/07, Tom [EMAIL PROTECTED] wrote: We have installed two of the Sangoma 2 port cards. Both had echo cancellation. The cost add on was about $450, not $800. I also had a single port T1 card without the echo cancellation. The extra money is worth it to me. Less CPU load and it just plain works. And if a customer is willing to spring for a PRI, they usually won't complain about the extra cost for better sound quality. Thanks for the info, Tom. I was thinking along those lines, but just wanted to get some verification. This is being installed in a fairly beefy server, and will only be handling ~20 calls at a time or so, so I don't envision CPU being a problem, but offloading the EC never hurt anyone. -Erik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SIP phones to buy?
From overall apprecation feedback : #1 Polycom (Any) #2 Aastra 480i #3 Cisco 7940+ #4 Linksys SPA-94x On 4/11/07, Stephen Bosch [EMAIL PROTECTED] wrote: I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge. Advice, anyone? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Blast over IP?
No entirely true.. but yeah realizing we are talking about asterisk .. you can't do reliable T38 faxing.. check out openpbx its embeded , all you need then is a ITSP to do TDM termination as T38 A bit more expensive then voip.. but you still get the bulk /automation part without the requirements and financials to get pri's Let me know what you decide to do. On 4/12/07, Wiley Siler [EMAIL PROTECTED] wrote: Thanks all... Looks like I will have to let them know that FOIP is a no go and that we can automate on Asterisk though... Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] www.education2020.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Thursday, April 12, 2007 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? Wiley Siler wrote: Basically, I want to send bulk faxes to a list of my clients. It is time consuming for a person to individually fax so a blast type solution seems best. Over IP is of course to save money... Thanks for the link, reading now... Any suggestions for the blast then? My suggestions are in the reading material. Basically it boils down to you not using VoIP for fax. Lee. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Thursday, April 12, 2007 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax Blast over IP? On Thu, 12 Apr 2007, Wiley Siler said something to this effect: Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? Asterisk can send faxes, if you make it interoperate with a few well-known open-source utilities and/or software packages, depending on what precisely you want to do: http://www.voip-info.org/wiki-Asterisk+fax -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sharing trunks between asterisk machines
No you are being misled.. SER can NOT DO IAX, SER = SIP only
but you would need SER to do that yes.
On 4/12/07, Alex Balashov [EMAIL PROTECTED] wrote:
Certainly. Any signaling / trunking protocol will do, in principle.
On Thu, 12 Apr 2007, Edgar Guadamuz said something to this effect:
Thank you Alex and It would be possible to do that using IAX too,
wouldn't it?
I mean something like
exten=_9NXX,1,Dial(Zap/g0/${EXTEN:1})
exten=_9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
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Re: [asterisk-users] IAX Trunk Failover
tried x+102 ?
On 4/5/07, Brent [EMAIL PROTECTED] wrote:
I'm trying to get an IAX trunk to failover to a local trunk it the trunk
is down.
This is what I've been working on:
[macro-forward1];
exten = s,1,Dial(IAX2/192.168.1.1/${ARG1},20)
exten = s,2,Goto(call-${DIALSTATUS},1)
exten = s-CONGESTION,1,Dial(LOCAL/${ARG2},20)
exten = s-CHANUNAVAIL,1,Dial(LOCAL/${ARG2},20
;end macro-forward1
exten = 6222626,1,Macro(forward1,6222626,6222627)
...in the debug, I never see dialstatus...the call just fails. Doesn't
ever try to dial the second extension.
Any ideas?
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Re: [asterisk-users] Off Topic: Open Source USB Softphone
sip would be the required one as iax..well.. also openwengo wont work.. to much overhead .. broswrer needed.. ie component + flash + css+js etc.. not viable.. so im also asking anyone have one ? since ihave a supply of around 2000 of the vonage usb stick OEM.. On 3/30/07, Michael Van Donselaar [EMAIL PROTECTED] wrote: Which USB Phone? I have written custom versions of iaxcomm for various people, and have a version that works with the Yealink phone. On Thu, 29 Mar 2007 11:33:07 -0300, Luis Claudio Santos [EMAIL PROTECTED] wrote: I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk
From what i was told via the GT guys at bell The customer should send I 1C Standard Facility Length = 21 9F Serv Discrim Networking Extensions PDU Component Begins (hex) 8B0100A10F020101... instead of: I 28 Display Length = 9 E Display .JIMBOB so we sending Ascii and they want HEXA.. can someone forward this to Matt Frederickson and the -dev list ? this is what fellow canadian telcos want to see.. PLEASE ! On 3/2/07, Webster, Andrew [EMAIL PROTECTED] wrote: On 2/28/07, Webster, Andrew [EMAIL PROTECTED] wrote: Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't support the name feature, so they've changed my link type). Version: Asterisk 1.2.14 svn rev 48468 Interesting, just finished a long day with a client that switched from 20+ POTS to a T1 using NI2 and CallerID Name (on inbound). So I guess NI2 does support CallerID Name. NI1/NI2 both support callerID name inbound. My issue is with CallerID outbound. There doesn't seem to be a consensus on the subject. Some people are saying outbound PRI name CallerID just doesn't exist on either NI1 or NI2, but the Telco thinks it can be done, but only on NI1 !?! I've seen dumps where the IE facility code shows the name (inbound), but would like to see an outbound one, if such a beast exists. -- Andrew Asterisk Log: Executing Set(SIP/304-091aafb8, CALLERID(all)=Andrewnn) in new stack (I've replaced the digits with n). PRI debug shows: Protocol Discriminator: Q.931 (8) len=42 Call Ref: len= 2 (reference 4/0x4) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 0c 21 80 nn nn nn nn nn nn nn nn nn nn] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) 'nn' ] From zapata.conf: callerid=asreceived ;Sangoma A101 port 1 [slot:12 bus:0 span: 1] switchtype=ni1 context=from-zaptel overlapdial=yes facilityenable=yes group=0 signalling=pri_cpe channel = 1-23 Thanks! -- Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: CallerID Name not being sent on outboundPRI trunk
hey sorry for caps my bad. and yes i will , sub to -dev however i tought that for that 1 post i shouldn't go trough the troubles of doing that. and that someone could already be on the -dev form this thread , i also assumed threads where like sub channels where people could isolate from the rest to talk on a specific topic and help each other out... On 3/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Mar 03, 2007 at 11:52:38AM -0500, Mike Lynchfield wrote: From what i was told via the GT guys at bell The customer should send I 1C Standard Facility Length = 21 9F Serv Discrim Networking Extensions PDU Component Begins (hex) 8B0100A10F020101... instead of: I 28 Display Length = 9 E Display .JIMBOB so we sending Ascii and they want HEXA.. can someone forward this to Matt Frederickson and the -dev list ? this is what fellow canadian telcos want to see.. Why won't you subscribe to the asterisk-dev list yourself and post it? Matt Frederickson does read it. PLEASE ! Screaming will not help you. For me you have already earned quite a few bad points with your previous post to this list. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] REMOTE CRASH FIX
Please note that we are available to fix the current REMOTE crash that affects Asterisk/openpbx/trixbox and crashes these systems via a malformed packet please contacts use if you need a hand to patch your systems. -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] REMOTE CRASH FIX
nope
http://svn.digium.com/view/asterisk/branches/1.2/channels/chan_sip.c?r1=56230r2=57475
is avail for free..
6574a6575,6579
if (uri == NULL) {
ast_log(LOG_WARNING, register_verify: URI is NULL!\n);
transmit_response_with_date(p, 503 Bad Request, req);
return -3;
}
is my patch
We just offering to support people that don't know how to mod stuff and
recompile..
On 3/2/07, BJ Weschke [EMAIL PROTECTED] wrote:
On 3/2/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
Please note that we are available to fix the current REMOTE crash that
affects Asterisk/openpbx/trixbox and crashes these systems via a
malformed
packet
please contacts use if you need a hand to patch your systems.
And you'll do it for free too? How gracious of you! If you were
charging money, I'd say you belong on the -biz list, but while you're
being so gracious, maybe your resources would like to volunteer for
some bug marshalling tasks too?
--
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http://www.btwtech.com/
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Re: [asterisk-users] Help: CallerID Name not being sent on outbound PRI trunk
nice one.. we have rogers and primus.. ni'2 and same.. let me know if this ni2 and ni1 thing is crap or not On 2/28/07, Webster, Andrew [EMAIL PROTECTED] wrote: Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't support the name feature, so they've changed my link type). Version: Asterisk 1.2.14 svn rev 48468 Asterisk Log: Executing Set(SIP/304-091aafb8, CALLERID(all)=Andrewnn) in new stack (I've replaced the digits with n). PRI debug shows: Protocol Discriminator: Q.931 (8) len=42 Call Ref: len= 2 (reference 4/0x4) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 0c 21 80 nn nn nn nn nn nn nn nn nn nn] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) 'nn' ] From zapata.conf: callerid=asreceived ;Sangoma A101 port 1 [slot:12 bus:0 span: 1] switchtype=ni1 context=from-zaptel overlapdial=yes facilityenable=yes group=0 signalling=pri_cpe channel = 1-23 Thanks! -- Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Paid support offered
We have decided to allow our tech's to do support for non-clients of voicemeup.com You can head to http://support.voicemeup.com/ and one will be in touch 8 to 6pm business hours. 3 levels of support are offered for Asterisk/compiling Trixbox , Ivr's etc. -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] read write or only read fields in cdr?
try not using dst.. maybe its a regex on te fieldname that matches for
reserved keywords..
try pre_dest instead
On 2/28/07, Bayrouni [EMAIL PROTECTED] wrote:
Hello,
I created a new field named pre_dst of type varchar(80) exactly like dst
field in cdr table.
In the dialplan I put:
exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})
and when I call, all goes fine except that pre_dst has always NULL value
in cdr.
Do you know why?
Is something wrong I did?
I know that original fields in cdr are only readable, but in this cas
pre_dst is one I created myself !!!
Thank you.
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Re: [asterisk-users] Newbie extensions.conf question
try putting near the exten = 1000,1,dial stuff
On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
I've installed Sven Slezak's Notify module. He gives the follow as an
example line to put into extensions.conf
exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/
sunnybook)
I understand what is going on with this line but I don't know where
in the extensions.conf file to put it?
Thanks,
Chris Griffin
[EMAIL PROTECTED]
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Re: [asterisk-users] Run-away Asterisk
You could try Fast agi.. then i think master agi deamon runs from services and replies to requests by including sub scripts. however i do see some connect failures sometimes... On 2/28/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Feb 28, 2007 at 10:56:14AM -0800, Yuan LIU wrote: After testing some AGI's, I noticed several extra Asterisk processes. An agi script is run by the same user running asterisk, but is not asterisk: it is a different program. What is the command name on those scripts? They are not zombies, but can't be killed by safe_asterisk. safe_asterisk attempts (poorly) to guard asterisk. Not really to guard all of its child processes. Nor will they die when CLI issues stop now. Then I read that each AGI spawns a separate Asterisk process. Huh? AGI? FastAGI? But all my AGI calls have apparently completed successfully. So there should be no reason for them to hang there. Several questions: 1) Under what conditions will an AGI hang a process? (My test scripts are pretty simple, almost directly derived from agi-test.agi.) An AGI may be an arbitrary subprocess. This subprocess can do basically everything. If it really wants to, (or if it misbehaves in the right way) it won't die. 2) How to detect run-away processes under 2.4 kernels? In this kernel, each thread clusters process space and it's very difficult to distinguish them without killing the main process. hmm, please attach the output of: ps auxww | grep asterisk 3) Any practical way to detect them from inside Asterisk - e.g., do some check after each AGI call? All my AGISTATUS reports success. I could use System() but isn't that cumbersome? Write/use better code, I guess. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple phones registered for the same user
you need also note that once you do that asterisk authorizes on first ip it sees in sip peers.. so client a and client b with same ip.. could cause problems unless you divide them . On 2/28/07, Bayrouni [EMAIL PROTECTED] wrote: Eric ManxPower Wieling a écrit : Yuan LIU wrote: From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] Date: Wed, 28 Feb 2007 10:57:43 -0600 Ricardo Carvalho wrote: Can't I register multiple phones with the same user/password? That's what I pretend to do, not ring groups... Ricardo, Any particular reason for not using ring groups? No, you cannot register multiple phones with the same user/password. Just curious: can I register multiple phones with one user name but different passwords? no. ___ Which is relevant for asterisk (like any other client/server based architecture), is the session. Your phone (hard||soft) is the client. Your PBX asterix is the server. Your session is defined by your agent confiuration (and configuration data is sent in SIP protocol over TCP/IP suite protocol) . But first there is a connection.(tcp/ip) And on the same IP/PORT there is only one connection. If you change username/password this is still one connection and the same connection. username password are mostly used to authenticate and not to connect. cheers Bayrouni ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] this i a test
or at http://asterisk.voicemeup.com for consolidated lists.. On 2/28/07, Rodrigo Gonzalez [EMAIL PROTECTED] wrote: Bayrouni wrote: Sorry for disturbing, but I sent some messages today and I am not seeing them on this list. Can sombody tell me, in case this message appear on the list. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yes, here Go to http://lists.digium.com/mailman/listinfo/asterisk-users Login and check that you have Receive your own posts to the list? in yes if you want to receive your own emails ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemeup @ 0.008 per minute USA /CAN
Just to let you all know we are offering 0.019 down to 0.008 automatic pricings on volume.. TDM termination/origination Unlimited SIP/IAX accounts g729/ulaw/alaw/gsm/etc 15 channels opened per account to start with Toll Free numbers / Local numbers Reseller rates Wholesale Rates Whitelabel -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk / ACT CRM Integration
Do You have a link to ACT CRM ? Thanks On 2/20/07, Cory Andrews [EMAIL PROTECTED] wrote: Has anyone ever been party to an integration of ACT CRM platform with Asterisk? Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Inbound Problem
Well, could be the fact provider not pushing as g729 or someting else. Can you set debug 999 and set verbose 999 then redump that ? you are missing the before the answer part also.. Also try G711 first then work your way to other codecs On 2/20/07, Rajeev Natarajan [EMAIL PROTECTED] wrote: Am working with Arun on this project - here's a longer description of the problem: We've been fighting with our service provider on this issue - we seem to be getting a BYE just after we receive an ACK. They claim that it is an asterisk issue! The service provider provides only IP based authentication for inbound. We have used username-password based authentication with the same setup with *no problems* whatsoever! If we configure an Audiocodes MEdia gateway to receive the calls, there is no issue - so there's something that asterisk is doing? or asterisk-Provider gateway combo? In our efforts to mask IP, I have used PROVIDER-IP for the IP of my service provider (host) and AsteriskIP to indicate my asterisk server sip.conf [PROVIDER] type=peer disallow=all allow=g729 context=default host= fromuser=y.y.y.y port=5060 insecure=very canreinvite=no nat=yes qualify=yes CLI output: -- Executing Answer(SIP/PROVIDER-IP-b7a076a8, ) in new stack We're at 124.7.195.102 port 47698 Adding codec 0x100 (g729) to SDP Reliably Transmitting (NAT) to PROVIDER-IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bK6bd3121243ee9f936c4aeb96d6785b7a;received=PROVIDER-IP From: sip:[EMAIL PROTECTED];tag=3380976385-794612 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:8009422419@'AsteriskIP' Content-Type: application/sdp Content-Length: 183 v=0 o=root 2172 2172 IN IP4 AsteriskIP s=session c=IN IP4 AsteriskIP t=0 0 m=audio 47698 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - --- -- Executing Playback(SIP/PROVIDER-IP-b7a076a8, park) in new stack -- Playing 'park' (language 'en') AstSQL*CLI -- SIP read from PROVIDER-IP:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 5 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 From: sip:[EMAIL PROTECTED];tag=3380976385-794612 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Via: SIP/2.0/UDP 221.135.102.100:5060;branch=z9hG4bK02505a1dcc5937d9a648eebc0052b422 Content-Length: 0 --- (9 headers 0 lines) --- AstSQL*CLI -- SIP read from PROVIDER-IP:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 5 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 From: sip:[EMAIL PROTECTED];tag=3380976385-794612 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE Via: SIP/2.0/UDP 221.135.102.100:5060 ;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f Content-Length: 0 --- (9 headers 0 lines) --- Sending to PROVIDER-IP : 5060 (NAT) Transmitting (NAT) to PROVIDER-IP:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f;received=PROVIDER-IP From: sip:[EMAIL PROTECTED];tag=3380976385-794612 To: sip:[EMAIL PROTECTED]:5060;tag=as52d36855 Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 The following is an ngrep of the traffic for an inbound call - 'U' marks the begin of the packet grabbed. U PROVIDER-IP:5060 - AsteriskIP:5060 INVITE sip:800942@AsteriskIP SIP/2.0..Max-Forwards: 5..Session-Expires: 3600;Refresher=uac..Suppor ted: timer..To: sip:[EMAIL PROTECTED]:5060..From: sip:PROVIDER-IP;tag=3380960452-790279..Co ntact: sip:PROVIDER-IP:5060..Remote-Party-Id: sip:PROVIDER-IP;party=calling;screen=no;privacy =off..Call-ID: [EMAIL PROTECTED]: 1 INVITE..Via: SIP/2.0/UDP 221. 135.102.100:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4..Allow-Events: telephone-event..Content-T ype: application/sdp..Content-Length: 206v=0..o=nextone-msw1 1774 4816 IN IP4 PROVIDER-IP..s=sip call..c=IN IP4 PROV-IP-2..t=0 0..m=audio 18932 RTP/AVP 18 19..a=ptime:20..a=rtpmap:19 CN/8000..a=fm tp:18 annexb=yes..a=rtpmap:18 G729/8000.. # U AsteriskIP:5060 - PROVIDER-IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4; received=PROVIDER-IP..From: sip:PROVIDER-IP;tag=3380960452-790279..To: sip:[EMAIL PROTECTED] 11.2:5060..Call-ID: [EMAIL PROTECTED]: 1 INVITE..User-Agent: Ast erisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:[EMAIL PROTECTED]..Content-Length: 0 # U AsteriskIP:5060 - PROVIDER-IP:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP PROVIDER-IP:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4
Re: [asterisk-users] Agents busy in queue
Yes first thing is not using 1.4 but as you probably won't budge , try hints. exten = 1001,hint,SIP/USER that will force it to poll status of that peer and reset the queue agent, of course replace values with actual ones On 2/20/07, Paul Hales [EMAIL PROTECTED] wrote: Are you using attended transfers? PaulH On Tue, 2007-02-20 at 15:37 +0500, Kashif Anwar wrote: I need some help with a problem which I'm facing with Asterisk 1.4 final release. I'm using static agents in a queue. Sometimes when an agent answers a call in queue and then releases it, the status for that agent in the queue remains busy where as there is not channel associated to that SIP client. For furthur calls in that queue that particular agent receives no more calls unless you unregister and then register that SIP client. This is occuring very regularly. Any one with a solution or idea?? Thanks, Kashif. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open CallerID Database?
With all other things said.. you might want a professional service for this like targusinfo.com Maintaining and running an operation like a cname web lookup thing is REALLY high overhead in terms of web traffic etc What happens when you get 30 ITSP/clients pulling 1000 calls each or 10 calls each per day.. that can easily go up to 1 mill requests per day , How will you pay for the bandwith/hardware/failover/load balance etc hardware for all this ? or if you are going to charge then why reinvent the wheel. targusinfo.com is what we would use.. Cname lookup is a really controversial matter , no one wants to absorb the costs , that is why some TELCOS charge 4.95 for callerid ( its basically the lookup service they are paying for) .. CNAME lookups is also not mandatory for TELCOS so some do it some don't , but FREE cname is just not going to happen untill some one has a Return on Investment strategy for this.. Take a look at Free 800 systems that went down , Any venture needs a capital source of income.. my 0.02 On 2/20/07, Robert Norton - SophMedia LLC [EMAIL PROTECTED] wrote: Hey Guys, I'm glad to see this ignited some discussion. I definitely understand there's some legal implications involved, both on a privacy level, and fraud prevention. Obviously an end-user (ie: the person controlling a listing) has to consent to some sort of release resolving the privacy concerns. I'm somewhat aware of the legal implications involved with storing such personally identifiable information (or whatever the legal term is) and have a concern in making sure such issues are resolved. In reality, how is it efficient for every provider to be running their own database? In my mind, this leaves the horribly evident inaccuracies, and even efficiency issues. Thank God these accuracies aren't integral to the operations of telephony systems. I do understand there is a price to pay for such infrastructure, and I believe that it's obvious the telephony world is riddled with racketeering, price gouging ventures, including companies that charge nearly a $0.01 for a lookup. I realize the following analogy is poor, but in mind this is as close as a internet search engine charging for a basic search query. Infact a basic internet query is much more complex, much more costly (ie: the infrastructure of said systems), and yet self-subsidizing. And to the poster who suggested that I was implying scrapping the results from 411.com, this is definitely not even a remote idea in my mind at all. The basis for my idea was a open, moderated, database that was user controlled and self-subsidized. I know this is way off topic, but I really feel that the telecom industry as a whole, and I'm sure I'm not the only one with this belief, is horribly bloated, running on business models that are clearly 30 years outdated. It is 2007, and with the help of the internet, the exchange of information, these telcos now have real, global competition, and real issues to deal with. Anyways guys, I'm curious to hear your thoughts. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Open CallerID Database?
Well caching is the way to go., bu then again most of the current solutions have this problem. John smit has a DID.. 514 555 1234 and closes account.. did sleeps for 3 months and new client Jane doe takes it.. Now how long should caching be ? this is a big problem ATM because some cache for 1 year others 1 day , they don't want to tell how long nor provider an API update method. On 20 Feb 2007 20:43:37 +0100, Benny Amorsen [EMAIL PROTECTED] wrote: RL == Richard Lyman [EMAIL PROTECTED] writes: RL TP'n to follow flow just like DNS, the 'root servers' would still RL see the high request hits, prior to passing off to local caching RL app. The DNS root servers are almost only loaded by irrelevant traffic. The root information is easily cacheable, so it is rare to have to actually ask the root servers. An ENUM-style solution would most likely not see much garbage traffic, and the relevant traffic is easily cacheable. I doubt that we will ever see such a solution though; there is too much invested in the old way of doing things. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Enterprise quality SIP provider
You can try us, http://www.voicemeup.com TDM in most areas , others offloaded white routes to L3 mainly. Cover most of usa , and canada. you can ping www.voicemeup.com to get an idea on location , we are directly on peer1,teleglobe,videotron with best quality bandwith only. Per minute pricing starts at 0.019 and goes down to 0.009 on volume, automatic and realtime adjustments starting at 2500 minutes. On 1/30/07, Martin Joseph [EMAIL PROTECTED] wrote: On 2007-01-28 08:37:43 -0800, Eric Germann [EMAIL PROTECTED] said: We LOVE Teliax. We're on a Time Warner business class fiber connection and avg 25ms latency from Ohio to Denver CO. With that connection I would love Teliax also. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Voipjet ...
also trixbox stop registering randomly on all versions.. confirmed with over 200 client accounts over here... all using trxibox.. asterisk vanilla is ok On 1/31/07, Alejandro Lengua [EMAIL PROTECTED] wrote: How many simultaneous calls per account are you sending ? On 1/31/07, Peter Halliday [EMAIL PROTECTED] wrote: That's interesting I use Voipjet cheap lines and I don't have a problem at all. Peter On 1/30/07, Alejandro Lengua [EMAIL PROTECTED] wrote: Hello, we have this problem with Trixbox 1.23 I have created an outgoing route where the 1st line has Voipjet and the 2nd an 3rd have voipcheap accounts. The problem is that at certain moments, when we call all the calls go through the voipcheap SIP accounts SIP, whose quality are not only not good enough but also consume a lot of bandwidth. The error message that returns Voipjet to Asterisk is that all circuits busy. What I asume from this? Thanks in advance ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digest of lists on forum asterisk.voicemeup.com
Just to let you know all we consolidated all posts on Asterisk/openpbx/freeswitch into 1 forum for ease of viewing.. threaded of course.. http://asterisk.voicemeup.com -- Mike Sales http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is 1.2.12.1 production ready
reboots are wiseOn 10/16/06, Tom Vile [EMAIL PROTECTED] wrote: fine for me here since it came out. We are running 15 extension all day long.On 10/16/06, shadowym [EMAIL PROTECTED] wrote:I am getting ready to image a production system.Right now I am planning on using Centos 4.4, Asterisk 1.2.12.1, Freepbx 2.1.3.I will be using aSangoma A200D card.I read of some people having problems with Asterisk 1.2.12.1 crashing.Isthis across the board or is there anyone out there with no problems.If youhave 24/7 uptime and no nightly reboot crons I would definitely appreciatehearingabout it.Cheers ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Managerhttp://www.theclubvoip.comMaking it happen 1.877.807.VOIP (8647) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some file aren't loaded its No file in that Directory.
sounds like itts missing the mysql -source libsOn 10/13/06, raviprakash sunkara [EMAIL PROTECTED] wrote:Hello Users,I Installed the Asterisk-1.2.11, For My Real time Use I'm Use MySql For Asterisk Database, By Using the Asterisk-addons - 1.2.4 in My Linux.For My Voice messages Storage , I want To Use the MySql. In Googled it shows me the ODBC integration..Is it need for that ODBC integration with MySql for my Voice Message storing in MySql.When I'm trying to integrate with ODBC + MySql. and Reinstall the Asterisk .. As per the Below Url..http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER When i followed the Step by Step. And While reinstall the Asterisk Server ...it Shows me errors...sql.h and sqltest.h is not found in /usr/src/asterisk-1.2.11/includes/asterisk/ Please Help me in this Issue or..Help in How to Store the Voice Messages without integrating the ODBCStorage. -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.877.807.VOIP (8647) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Asterisk : It calls me backup with a dial tone
Look into DeadAGI, should be easy enough that illl implement tomorow ;)On 10/13/06, Klaverstyn, David C [EMAIL PROTECTED] wrote: Can this be done? I call Asterisk using my mobile (cell), Asterisk then hangs up on me so I am not charged for the call. Asterisk then calls my mobile (cell) presenting me with a dial tone allowing me to make through the PBX. It does this based on caller ID and only allowing certain phone numbers the hang up and call back function. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.877.807.VOIP (8647) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: unix sysctl config for asterisk
Anyone ?On 10/12/06, Mike Lynchfield [EMAIL PROTECTED] wrote: Can anyone share sysctl tuning params for asterisk and unix ?trying to see if we have differences in them -- MikeSales Manager http://www.theclubvoip.com Making it happen1.877.807.VOIP (8647) -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.877.807.VOIP (8647) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How big is *your* ego?
Want to share these 13 packets ?On 10/13/06, Matt Loretitsch [EMAIL PROTECTED] wrote: http://www.elna-america.com/tech_al_reliability.php Capacitors are one of the components on that motherboard that have afinite life span.Other components are more or less tolerant of thesechanges over time.Eventually the caps WILL fail...this could be 5 years or 25, but it WILL happen with electrolytics.I have a wellmaintained, regulated (3 phase power distribution all ups'd generatoretc.) and vented data center (72F 40% relative humidity year round) and loose things once a week...typically hard disks, but power suppliesoften.I monitor and graph temperature PER SERVER.Cpu's also in fact also have a limited life span due toelectromigration.Keeping a processor cool certainly does slow this process, but does not eliminate it completely.This actually applies tomost IC's, but it is more significant in processors where the layeringprocess is extremely thin. http://en.wikipedia.org/wiki/ElectromigrationOften there is no symptom of these events before something criticalbreaks.I'm not siding with anybody here, but there is some glaringmis-information in this thread. -MattP.s. I do work with a lot of un-pro (claimed) dell equipment, but alsohp9000, sun enterprise 10k, old as/400 f40, dec alpha, and yes, pix andnetscreen.They all quit at some point! -Original Message-From: C F [mailto:[EMAIL PROTECTED]]Sent: Wednesday, October 11, 2006 11:18 PMTo: J. Oquendo; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How big is *your* ego?edited out for brevity and my point Motherboards in a well regulated maintained system that is ventilated good, don't just die. They don't? Funny, I've seen it happen from everything from AMD, Sun, HP, SGI, you name it.You are telling me that it was: A. Well regulated B. Well maintained C. good ventilation, and it died suddenly, without giving you any hintsbefore hand?I just don't believe you, I might have on one machine, but I'm not goingto believe you since you said you seen it on every machine. BTW, have you ever seen a machine that survived everything and was just taken tothe dump because it was outdated and wasn't needed anymore? Hard drives should be installed in an array (have you ever heard of RAID). CPUs when the heat is taken care of, don't just die. Oh really? Sounds like you live in hardware Nirvana. How long have you been in the computing environment?No they don't, they give some warnings like too hot. **The information contained in this E-mail message is intended for the personal andconfidential use of the designated recipient(s) named above. This message and all communication contained herein is privileged and confidential. If the reader ofthis message is not the intended recipient or an agent responsible for deliveringit to the intended recipient, you are hereby notified that you have received this E-mail message in error, and that any review, dissemination, distribution or copyingof this message is strictly prohibited. If you have received this message in error,please notify Bird Technologies immediately by calling (440) 248-1200. **___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.877.807.VOIP (8647) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] unix sysctl config for asterisk
Can anyone share sysctl tuning params for asterisk and unix ?trying to see if we have differences in them -- MikeSales Managerhttp://www.theclubvoip.com Making it happen1.877.807.VOIP (8647) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Test Call Script
yes.. actualy use 1 did for each proxy to check..then inbound for each use the method he described..On 10/12/06, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
on an analog Zap PSTN channel, you have no real way of determining ifthe remote side answered, because, as you discerned, it IS consideredanswered as soon as asterisk opens the channel.How about you contact another asterisk server through the PSTN, and dial
through to an extension on that remote asterisk server that, in turn,notifies the first asterisk server maybe via the internet that it wasreceived?for example, consider the following php script accessupdate.php
onprimary asterisk box:?phpif (!strcmp($_GET['update'], 'true')){touch(/etc/asterisk/secondary_server_last_access);}?then primary calls secondary box through PSTN, and through the magic of
DISA or CID or what-have-you, dials through to an extension that executesSystem(wget -q -O /dev/nullhttp://primary-server/access_update.php?update=true
)then hangs up.then primary server checks the last-access time of/etc/asterisk/secondary_server_last_access to make its decision, viacron script or bash script triggered through the dialplan subsequent to
the initial dial-out.This is of course a very rudimentary on-the-fly thing I came up with,but think outside the box and this may be the easiest way for you to dowhat you want.MojJohn Kane wrote:
I am trying to write a script to attempt to make a call on a Zap channel, and if it fails, send an alarm.I can generate the call, but because the Zap channel accepts the call, even though the other end
never answers, it sees it as a successful call, which it isn't. Anyone have any ideas on this?Thanks. !DSPAM:500,452d7fa8199221504517840!
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[EMAIL PROTECTED]Office Manager, Horan Company, LLC(907) 747- x112___--Bandwidth and Colocation provided by
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Re: [asterisk-users] unauthenticated calls
Actualy to have fun.. make it a playback(dialtone),300 ;)that will make your little hacker think they have a dialtone then Record the number dialed and put that in a db for further investigation..Actualy could also be a user that has no clue on how to configure the system. On 10/12/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Mark Quitoriano wrote: Hi list, i noticed from the cli my asterisk box is accepting unauthenticated calls how can i prevent this? CLI: -- Accepting UNAUTHENTICATED call from 192.168.0.2: requested format = gsm, requested prefs = (), actual format = ulaw, host prefs = (g729|ulaw|alaw), priority = mineIf a call is not authenticated then it SHOULD fall into theextensions.conf [whatever] context that is specified in sip.conf[general] context=whatever.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Managerhttp://www.theclubvoip.comMaking it happen 1.877.807.VOIP (8647) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LinkSys PAP2 ATA Siemens Cordless 3010
yeah...I Got a Siemens Phone and i can't hear the ringing. Try to change these settings in the pap2 device (Admin - Advanced Mode-Regional settings) Voltage = 90V frequency = 20 Hz impedance = 900 ohms waveform = trapezoidal not sure about the question but this is a must i think..On 9/5/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:Hi everyone, I'm having a problem using this cordless with this ATA.When I try to call that phone, the line is busy. When this phone tries to callsomeone, no line up.Ata is working with another phone that's not a cordless so it's configured correctly.Any clue about this problem?Best regardsOscar Bossi___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provider UNREACHABLE
hehe yeah.. still when you see that qualify breaks newer xlites' you would wonder why to use it anyhow ?On 7/11/06, Rick Smith [EMAIL PROTECTED] wrote:teliax had a 2.5 hour outage today. I wouldn't call that short. -Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of AndresPaglayanSent: Tuesday, July 11, 2006 1:58 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Provider UNREACHABLEthey had a short outage today, it was fixed already,dunno if related to your issue,___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Managerhttp://www.theclubvoip.comMaking it happen 1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server redundancy
#1.. most the failures and network bottle necks on asterisk in a 1k + user sip /iax are registrations polling'syou are right .. get SER ... dont be dumb.#2 the config file with asterisk hardcode ips is a simple matter of running a script that parses it and puts in whatever it needs #3 basic failover will actually steal the ip form other box.. wich in this case would steal ip of down box.#3b.. we need multiple listening addies.. since asterisk can only listen to one ip its sucks for now On 7/11/06, Alejandro Acosta [EMAIL PROTECTED] wrote: What about having the softphone/hardphones configured with a Main Asterisk server and an Alternative Asterisk Server?.I just did a couple of tests with a Cisco ATA 186 and worked quit well after changing the RegInterval and Alternative Proxy Timeout. thks,Alejandro,On Tuesday 11 July 2006 05:04 am, unplug wrote: I have asked about it here. As Douglas said, it doesn't support mult-asterisk in current version. However, I have questions about why multi-asterisk so difficult to implement. 1. As we can use ARA to store all information, sip user register info, dial plan ... to DB.All asterisks can use ARA to refer to the DB for necessary information even register information. 2. What is mean by multiple Asterisk systems can't reference the same MySQL database for SIP peers.?Does SIP peer information also store in DB? 3.Any difficulty to implement multiple asterisk? 4. If I want to implement multiple asterisk in some extent, how do I begin?Any reference? On 7/11/06, RR [EMAIL PROTECTED] wrote: Interesting points on both messages 1) as far as multiple asterisk servers talking to the same database is concerned, I will have to test this out. I know nothing about the database side of things, and a newbie on asterisk and linux so I have no idea what and where the development of either of these are. From your message it sounds like it's just how ARA is designed because I doubt it's to do with the ODBC driver itself. This will cause me a lot of grief if you're right about this for multiple * servers to not be able to access the same database for peer lookup. 2) Clustering of DB isn't an issue, not for me at least. Haven't tested this either but my DBs are clustered A/P providing a single entity to the internal systems. Might further look into a local DNS lookup to add to this. I believe it's possible to do this in the MySQL world with MySQL grid etc? 3) I don't believe frequent registration is that big of an issue for the network load it generates. Most providers out there set devices for a 30-40secs Reg. Refresh to support NAT'ed endpoints and the a reg refresh is hardly about 300-400Byte pkts (I think). The math doesn't add up for a major load esp. if you've got a load balancing mechanism in front of your * boxes. 4) I don't know enough about DUNDi to get into this discussion but DUNDi just lookup extensions? or it also have any part to play in registrations? If they just do extension lookup, then If DUNDi is implemented on an A/P pair of dedicated DUNDi lookup servers which access a clustered database, then barring #1 being true, each * server accesses the same database and pool of registrations. If registrations are refreshed frequently enough, the contact info in the database will always be current and one server dying won't affect anything. At the same time, they just consult the DUNDi lookup server for extension lookups instead of asking the database directly. 5) If you really want to improve on this, supplement your network with SER as proxies and have them deal with Registrations and load-balance feature requests to * servers etc. Once * has done whatever it needs to do (e.g. provide PBX features, voicemail, conference, IVR etc.) it passes the call back to the Proxy to deal with the endpoints. All depends on your scope and budget. If you want to have a SP grade service then you need to breakout your functions. I just hope #1 isn't true though. The only alternative then would be to have /etc/asterisk reside on an NFS share or a CFS for all servers to read massively huge conf files if you're catering for large number of endpoints. Dunno if it helps anyone or I'm just shooting sh*t ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it
Re: [asterisk-users] Rate or rank ITSP
something that you could drill into.. or even search.. hold on mate i got this..how about a master LCR system that would generate config for users in terms of filters..EX1: filter on qoswould return BEST QOS list of all terminations for providers
like..provider1=sip/[EMAIL PROTECTED]etc514XXX,1,dial(${provider1})etc..but in regards to qos..filter by rate you would get a conf file listed with all providers rates being lowest for each country
so basically you would get a config file dpeending on filters.. thing is you need accounts on all these providers.. so links to the signup on them ? that would be bad.. imagine keeping a 20$ balance on 100 providers .. ;)
the point is we can't uatomate this i think.. unless we keep this down to 5-6 providers..On 7/11/06, mike
[EMAIL PROTECTED] wrote:i'll be very interested in thatit would also be useful that every qos rate comply with some
deterministic criteriaalso, imho, keep in mind that a qos rating should be given on provider+destination countrybecause in my experience, qos varies very much depending on whichdestination country you are calling
that would add a lot of work to the list, which should be heavilycommunity drivenof course a 'resuming' score for each provider would be more readablesomeone have experience on determining an 'mean opinion score' value
with asterisk + some software solution ?i've been messing with app_milliwatt but my know-how is scarying empty.mikeOn Tue, 2006-07-11 at 09:40 -0400, Barry Fawthrop wrote: Hi There I know of wiki there is a list of VOIP providers, but is there a list or
can we create / suggest one that will list VoIP providers, their location and quality of service ? Too me this will be very valuable, plus looking at some of the requests of late I'm sure others would like that too?
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Re: [asterisk-users] So many configuration files!
larry each of these files do something for a specific needs.. hence the sip.conf is for sip related modules..iax etc etc..voicemail.conf if you need voicemailres_odbc etc for any database usage.. basically read the manual and look into each files to see what they do..asterisk will start and work without modding all of these but you could have a surprise if a demo user is by default in sip .conf and someone uses your system ;) On 7/11/06, Larry Alkoff [EMAIL PROTECTED] wrote: I'm working with Asterisk 1.2.5 to get a working system.There are 50 Asterisk configuration files in /etc/asterisk.Are they _all_ called by Asterisk or are some only used in a #include?Is there any way to get a list of which ones Asterisk uses by default? There is only a single #include file and it doesn't even exist.I have only messed with 4 files so far.Are there any more I should be editing?Which ones could be safely ignored?So far the system is just SIP with Zaptel to be added next. The 4 files I have changed are:sip.confextensions.confextensions_additional.confvoicemail.confMy list of files in /etc/asterisk - sorted most recent last: [EMAIL PROTECTED] asterisk # ls -1trzapata.confvpb.conftelcordia-1.adsiskinny.confsip_notify.confrtp.confrpt.confres_odbc.confqueues.confprivacy.confphone.confoss.confosp.conf musiconhold.confmodules.confmodem.confmisdn.confmgcp.confmeetme.confmanager.conflogger.confindications.confiaxprov.confiax.conffestival.conffeatures.confextensions.ael extconfig.confenum.confdundi.confdnsmgr.confcodecs.confcdr_tds.confcdr_pgsql.confcdr_odbc.confcdr_manager.confcdr_custom.confcdr.confasterisk.confasterisk.adsialsa.conf alarmreceiver.confagents.confadtranvofr.confadsi.confsip.confextensions.confextensions_additional.confvoicemail.conf--Larry Alkoff N2LA - Austin TXUsing Thunderbird on Slackware Linux ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Manager http://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provider UNREACHABLE
trxtel ping me.On 7/11/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Tue, 2006-07-11 at 14:10 -0400, Rick Smith wrote: teliax had a 2.5 hour outage today. I wouldn't call that short.its all relative, nufone had a 30 day outage :P--Trixter http://www.0xdecafbad.com Bret McDanelBelfast IE +44 28 9099 6461DE +49 801 777 555 3402Utrecht NL +31 306 553058US WA +1 360 207 0479US NY +1 516 687 5200FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you!-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.3 (GNU/Linux)iD8DBQBEs+72+1olxlzQw5cRApVnAKC4ob9F2SZDeU2DidVLwG7YK/xOlwCgrsOF BNqx9bUsHGBWeNCJUumQgdE==1VDH-END PGP SIGNATURE-___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server redundancy
that what i meantas in /16 etc..but That was case for asterisk 1.x is wrong too .. since 1.2.9.1is 1.x ;)my badOn 7/11/06, Douglas Garstang [EMAIL PROTECTED] wrote: I think it can listen either on a specific address, or on ALL addresses, not on a subset of available addresses. -Original Message- From: Aaron Daniel [mailto: [EMAIL PROTECTED]] Sent: Tuesday, July 11, 2006 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Server redundancy On Tue, 2006-07-11 at 14:34 -0400, Mike Lynchfield wrote: #3b.. we need multiple listening addies.. since asterisk can only listen to one ip its sucks for now Incorrect. Asterisk most definitely listens on multiple interfaces.We've got several asterisk boxes that are multi-homed... one public and one private interface, so that we can have external phones and internal phones.Works fine. I'm thinking this is a misconception.We even have heartbeat set up to switch ip's around.The server actually listens on the fly to the new ip address that comes up under it. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Manager http://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 legs and cdr's
if we use call out files in asterisk it only creates a cdr on the bottom leg or the callfile.ex: will have a cdr entry for the channel : but not the extension ;Anyidea how to fix that behaviour ? -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voting,suggestiuon,your input needed to all
ok, We are building the perfect voip company..we are trying...we need input on end-users:reply to my email with --ENDUSER in subject.with anything you would like to see your current voip provider offer online/offline ( don't say.. support, an answer on phone etc) be constructive.. reply to myt email with --WHOLESALE in subject with anything you would like to see for wholesale (resellling itsp like services, rebranding,whitelable, per client rates. etcreply to our email [EMAIL PROTECTED] only not to start an endless thread..we WILL make those features happen, we actually got 4 engineers that are doing only requests from clients.Please occupy them as we them anyhow. ;) push things you want to see. ( no non asterisk things) EG: T38 NOT GOOD. as if we do then wont be branched and all loose...only things you would want your provider/partner to have ..Only the things we always hear around here.. I WISH..Thanks and let's make it happen.PS BTW contact us for custom IVR/PBX/ANYTHING programming.-- MikeSales Manager http://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime SIP Registrations
can you elaborate on modify sip to update the status on the sip friends in realtimethanksOn 6/29/06, Doug G [EMAIL PROTECTED] wrote:What I did was modify sip to update the status on the sip friends in realtime. Then via FAGI dial them directly with the data found in real-time. (ie dial ( SIP/[EMAIL PROTECTED]:5060) Of course you need to check the status in realtime data before you dial.This allows MANY Asterisk servers to share the same SIP data.I then load balance with DNS SRV..Yes I have tested in failover it works. I too have been told that by many that this will not work.So I keep expecting to hit some problem with it, but to date I have not...Doug From: [EMAIL PROTECTED] on behalf of David ThomasSent: Thu 6/29/2006 1:05 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime SIP RegistrationsI think lots of us know about it... We're just not sure how to goabout fixing it. :-(I know it's been a thorn in my side since I started using Asterisk. I would suspect that many of those saying works for me have neveractually tested their system in failure scenarios, or they are workingin a controlled environment without NAT and such... regards,DavidOn 6/29/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Aaron Daniel [mailto: [EMAIL PROTECTED]] Sent: Thursday, June 29, 2006 9:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Realtime SIP Registrations On Thu, 2006-06-29 at 09:15 -0600, Douglas Garstang wrote: How about fixing realtime SIP so that multiple Asterisk boxes can reference the same database? Doug. That's kinda what I'm hoping to work towards :) I'm surprised you even knew about that. There seems to be a common misconception that this should work (caused by common sense maybe). Every time I bring it up, people go 'Of course it works!', or 'Works for me!' (still don't know why it works for some and not others.) Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000
HERE IS answer [EMAIL PROTECTED]had same problem..make the settings for 90 volt.. not 70 volt ringer.. make it trapezoidal not sinusoisalmake it 900 ohm not 600 impedence..that worked for pap2's seem siemens are made for europe style ring voltage not north american.On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hello, The main differences I can see: - in zaptel.conf you have span=1,0,0,ccs,hdb3, which means you ask Asterisk to serve as a timer for the PBX - on my setup the PBX is the master clock and Asterisk is the secondary one, so I have span=1,1,0,ccs,hdb3 (in fact, as I use CRC4 error correction, my setup is span=1,1,0,ccs,hdb3,crc4) - in zapata.conf I have switchtype=EuroISDN. Generally speaking, try using other switchtypes. Regards, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Josué ContiSent: 27 June 2006 14:41To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000 Silviu, thank's will be this attention. Below my configurations of zapata.conf and zaptel.conf #zapte.conf span=1,0,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31loadzone=usdefaultzone=us #zapata.conf [trunkgroups] [channels]language=pt_BRcontext=defaultswitchtype=qsigpridialplan=privateprilocaldialplan=privatefacilityenable = yessignalling=pri_cpe;rxwink=300usecallerid=yeshidecallerid=nocallwaiting=yesusecallingpres=yesrestrictcid=nocallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yes cancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yes rxgain=0.0txgain=0.0group=1callgroup=1immediate=nocallerid=asreceivedmusiconhold=defaultgroup=1channel=1-15channel=17-31 Best Regards Josué 2006/6/27, Herchi Silviu [EMAIL PROTECTED]: Hi, Could you post your /etc/zaptel.conf and zapata.conf? Also, is everything OK the other way round (i.e., from the SIP phones to the PBX)? Silviu Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9 , is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josué ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Managerhttp://www.theclubvoip.comMaking it happen 1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] best hardphone for Asterisk?
We use cisco 7960's but thats not cheap..BTW Doungyour signature :Those that sacrifice essential liberty to obtain a little temporary safetydeserve neither liberty nor safety. -- Ben Franklin (1759) is a good one.. tell that to your president..and the patriot act.s/patriot/cutallrights/PS Andrew.. penguins as in linux based ?or the phone just quacks all the time ?;) On 6/27/06, shadowym [EMAIL PROTECTED] wrote: Which public STUN servers are you using or did you setup your own? -Original Message- From: Cullin J. Wible [mailto:[EMAIL PROTECTED]] Sent: Monday, June 26, 2006 8:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; 'Iain Barker' Subject: RE: [Asterisk-Users] best hardphone for Asterisk? We've used a number of the polycom 301 and 501 phones in our office. We have also deployed a dozen of the Linksys SPA-1001 single-line FXS adapters using G726, SIP, NAT and STUN. They are extremely reliable and easy to deploy - $60-$70 US each. We tested a number of IAX hard phones and didn't find anything that was reliable and/or suitable for our corporate setting. We really wanted to run IAX for remote users, but eventually decided that SIP/STUN was easier to support. We also tested the IAXy device and found that it's inability to use DNS resolution, only be configured on Linux, and only run ulaw/alaw made and that it cost more then the SPA-1001, which can use DNS, G726/G729 and has web-based configuration for less money the more attractive option. We also tested the IAX hard phone made by AT-COM only to find that a number of features such as call transfer do not work. For home/remote users: setup STUN, and use a SPA-1001. For a corporate setting I highly recommend the Polycom phones. Cheers, Cullin J. Wible Co-Founder CTO Email Data Source, Inc. 212-514-8900 x1006 -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Doug Crompton Sent: Monday, June 26, 2006 11:49 PM To: Iain Barker Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] best hardphone for Asterisk? Iain,Thanks for the repsonse but you are kidding me right? From what I can see if I bought this phone and two remotes my outlay would be close to $800 US. This is NOT a home device unless you have nothing better to do with your money! You can buy a lot of single line wireless phones and FXS devices for that amount! Doug On Mon, 26 Jun 2006, Iain Barker wrote: Doug, What you are describing sounds like the Aastra 480-CT, a base Ethernet/SIP screenphone supporting multiple wireless handsets [but as this is a non-commercial list I won't go into more detail here, google for the above model number if you're interested in more info.] - Iain --- Message: 4 Date: Mon, 26 Jun 2006 00:08:48 -0400 (EDT) From: Doug Crompton [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] best hardphone for Asterisk? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII Still awfully pricey for home use and the styling is not there for a bedroom or many other areas of a modern home. What we need is a wireless sip phone modeled like the panasonic or uniden which allow multiple extension off of one base. The base would connect to the internet. The other problem is many of these phones require power, so even if you have backup for your central system the phone still needs to be on it. Power over ethernet would help. Doug Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety.-- Ben Franklin (1759) *Doug Crompton * *Richboro, PA 18954* *215-431-6307* ** * [EMAIL PROTECTED]* * http://www.crompton.com* ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Manager http://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Error in config sample for GoToIf?
BLAH=1BLAH=1On 6/27/06, Brian Capouch [EMAIL PROTECTED] wrote:
Jon Schøpzinsky wrote: Hello As far as ive understood, you can just write
Exten = s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and ${AVAILSTATUS} would return 1Through more testing, the double quotes I used seemed superfluous; if
you use them in both places, or in neither, it works the same.But your example above lacks the $ ahead of the left brace.It is*that* which I now believe is in error in the example.Plus there seems to be confusion, on the Wiki at least, as to what
values mean what for ${AVAILSTATUS}Thx.B.--This message has been scanned for viruses anddangerous content by MailScanner, and isbelieved to be clean.___
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Re: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000
not sure. however pap2 devices usually have htat in regional stttings.. for asterisk i don't know. im sales not tech ;)sorry.. but you got a heads upOn 6/27/06, Josué Conti [EMAIL PROTECTED] wrote: Hi Mike, all good? I thank its attention. Where I modify these parameters that you said? Best RegardsJosué 2006/6/27, Mike Lynchfield [EMAIL PROTECTED]: HERE IS answer [EMAIL PROTECTED]had same problem..make the settings for 90 volt.. not 70 volt ringer.. make it trapezoidal not sinusoisalmake it 900 ohm not 600 impedence..that worked for pap2's seem siemens are made for europe style ring voltage not north american. On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hello, The main differences I can see: - in zaptel.conf you have span=1,0,0,ccs,hdb3, which means you ask Asterisk to serve as a timer for the PBX - on my setup the PBX is the master clock and Asterisk is the secondary one, so I have span=1,1,0,ccs,hdb3 (in fact, as I use CRC4 error correction, my setup is span=1,1,0,ccs,hdb3,crc4) - in zapata.conf I have switchtype=EuroISDN. Generally speaking, try using other switchtypes. Regards, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Josué Conti Sent: 27 June 2006 14:41To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000 Silviu, thank's will be this attention. Below my configurations of zapata.conf and zaptel.conf #zapte.conf span=1,0,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31loadzone=usdefaultzone=us #zapata.conf [trunkgroups] [channels]language=pt_BRcontext=defaultswitchtype=qsigpridialplan=privateprilocaldialplan=privatefacilityenable = yessignalling=pri_cpe;rxwink=300usecallerid=yeshidecallerid=no callwaiting=yesusecallingpres=yesrestrictcid=nocallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yes rxgain=0.0txgain=0.0group=1callgroup=1immediate=nocallerid=asreceivedmusiconhold=defaultgroup=1channel=1-15channel=17-31 Best Regards Josué 2006/6/27, Herchi Silviu [EMAIL PROTECTED]: Hi, Could you post your /etc/zaptel.conf and zapata.conf? Also, is everything OK the other way round (i.e., from the SIP phones to the PBX)? Silviu Hello all. I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade in asterisk-1.0.9 , is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is happening he is that the calls originated for PABX Siemens and destined to SIP phones asterisk are being without audio, nor Ring, is dumb. They could help in this case me? Best Regards Josué ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen 1.888.470.7253 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Managerhttp://www.theclubvoip.comMaking it happen 1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Vs SIP cpu load
dual support yes.. however i read a few articles on the fuct that single with double the ram is better..something about the bus or sshare between both processors i think.i would go AMD opteron, but that me. or sunOn 6/13/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: HelloIs it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads?Also, does Asterisk support and use multiprocessor architectures, such as Xeon? RegardsJon--No virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Managerhttp://www.theclubvoip.comMaking it happen 1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Vs SIP cpu load
taskset does not seem to exist on redhad 9 nor freebsd..;)On 6/13/06, Zoa [EMAIL PROTECTED] wrote:
When i did this test ages ago, i found out that iax was worse than sip,but sip was worse than trunked iax.
Joachimolin Anderson wrote: I use IAX2 quite a bit and I haven't really noticed any difference between IAX2 and SIP. CPU usage in Asterisk is aggravated by transcoding, changing one audio format to another, and SIP or IAX2 is simply the protocol used to
carry the audio. Any function of Asterisk will be affected by high system load; if you have a loadaverage of 3, for example, your box is in trouble regardless of the protocol used. Although this may have changed in the newer
1.2.X series of Asterisk, I believe that Asterisk does not support SMP from the perspective of dispatching *internal* processes to different CPU's, instead, *external* processes such as AGI's are balanced out and dispatched automatically to
different CPU's - but this is a kernel thing. It's generally well-known that a fake SMP machine such as a HyperThreading CPU affects Asterisk negatively, and best practice is to disable
HyperThreading. However, real SMP machines have no trouble (I use a 4 way Xeon). It's possible to pin a process to a specific CPU, and in fact, I do this to force Asterisk to it's own CPU, and pin all other processes to a
specific CPU that Asterisk does *not* use: setasteriskaffinity.sh: #!/bin/bash ASTERISKPID=`ps -A | grep -a -A0 asterisk` taskset 0x0003 -p${ASTERISKPID:0:5}
This pins Asterisk to CPU # 4 on a 4 way system. Repeat for all other processes, to different CPU's with the affinity mask: 0x = CPU 1 0x0001 = CPU 2
0x0002 = CPU 3 0x0003 = CPU 4 -Original Message- From: Jon Schøpzinsky [mailto:[EMAIL PROTECTED]] Sent: Tuesday, June 13, 2006 8:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAX2 Vs SIP cpu load Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that
IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? Regards Jon
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Re: [Asterisk-Users] IAX2 Vs SIP cpu load
No way.. im keeping it.. be uptime since.. let me see..uptime13:12:56 up 587 days, 13 users, load average: 0.11, 0.02, 0.01i think i got an old sun laying around in the noc that been just compiling since 1973..
lol nah but i remember the old days when a password leak on our hosting clients would generate loads of 400-500 and would take around 10 minutes to type stop httpd10 chars..but then again if i cold shut .. would take 21 minutes to reboot..
the darn thing weighted around 50 pounds and was the best thing iv seen as far as load ocud handle..that 500 load was around 25000 simult users grabbing web pages..when we replaced with pentium crap it could handle a load of 10-20 not more.
last uptime on it was 800+ days lol.. soon its 3rd birtday.. never changed a thing on it.. actually i rebooted it then for moving it.On 6/13/06,
Colin Anderson [EMAIL PROTECTED] wrote:
2002
called. They want their operating system back. :- )
-Original Message-From: Mike Lynchfield
[mailto:[EMAIL PROTECTED]]Sent: Tuesday, June 13, 2006 9:42
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] IAX2 Vs SIP cpu
loadtaskset does not seem to exist on redhad 9 nor
freebsd..;)
On 6/13/06, Zoa
[EMAIL PROTECTED] wrote:
When
i did this test ages ago, i found out that iax was worse than sip,but
sip was worse than trunked iax. Joachimolin Anderson
wrote: I use IAX2 quite a bit and I haven't really noticed any
difference between IAX2 and SIP. CPU usage in Asterisk is aggravated
by transcoding, changing one audio format to another, and SIP or
IAX2 is simply the protocol used to carry the audio. Any function
of Asterisk will be affected by high system load; if you have a
loadaverage of 3, for example, your box is in trouble regardless of
the protocol used. Although this may have changed in the
newer 1.2.X series of Asterisk, I believe that Asterisk does not
support SMP from the perspective of dispatching *internal* processes
to different CPU's, instead, *external* processes such as AGI's are
balanced out and dispatched automatically to different CPU's - but
this is a kernel thing. It's generally well-known that a
fake SMP machine such as a HyperThreading CPU affects Asterisk
negatively, and best practice is to disable HyperThreading.
However, real SMP machines have no trouble (I use a 4 way Xeon).
It's possible to pin a process to a specific CPU, and in fact, I
do this to force Asterisk to it's own CPU, and pin all other
processes to a specific CPU that Asterisk does *not*
use: setasteriskaffinity.sh:
#!/bin/bash ASTERISKPID=`ps -A | grep -a -A0 asterisk`
taskset 0x0003 -p${ASTERISKPID:0:5} This
pins Asterisk to CPU # 4 on a 4 way system. Repeat for all other
processes, to different CPU's with the affinity mask:
0x = CPU 1 0x0001 = CPU 2 0x0002 = CPU
3 0x0003 = CPU 4 -Original
Message- From: Jon Schøpzinsky [mailto:[EMAIL PROTECTED]] Sent: Tuesday, June
13, 2006 8:14 AM To: Asterisk Users Mailing List - Non-Commercial
Discussion Subject: [Asterisk-Users] IAX2 Vs SIP cpu
load Hello Is it correct that IAX2
uses more CPU, than SIP? Also, can it be true that IAX2 is much
more sensitive against high CPU loads? Also, does Asterisk support
and use multiprocessor architectures, such as
Xeon? Regards Jon
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happen1.888.470.7253
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Re: [Asterisk-Users] SIP 486 Busy Here
how baout codecs ?try enabling all for testing ..then limit..On 6/9/06, Jason Lixfeld [EMAIL PROTECTED] wrote:Kinda confused by this...I have a Cisco 7960 configured with a couple SIP extensions configured on the phone.Just trying to dialone extension from the other on the same phone, but when I do, I get: -- Remote UNIX connection -- Executing Dial(SIP/2001-ffd4, SIP/2002) in new stack -- Called 2002 -- Got SIP response 486 Busy here back from xxx.xx.xx.xxx -- SIP/2002-f29b is busy == Everyone is busy/congested at this time (1:1/0/0) -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from xxx.xx.xx.xxxAny ideas?# sip.conf[2001]type=friendusername=2001secret=hjksdfg23ASDFcontext=ario-extensionshost=dynamicnat=yesregister=yesqualify=yesdisallow=all allow=ulaw[EMAIL PROTECTED][2002]type=friendusername=2002secret=hjksdfg23ASDFcontext=ario-extensionshost=dynamicnat=yesregister=yesqualify=yesdisallow=allallow=ulaw [EMAIL PROTECTED]# extensions.conf[ario-extensions]exten = 2000,1,GoTo(2001,1)exten = 2001,1,Dial(SIP/2001)exten = 2002,1,Dial(SIP/2002)# asterisk -rx sip show peers Name/usernameHostDyn Nat ACL Port Status2002/2002xxx.xx.xx.xxxD N5060 OK(198 ms)2001/2001xxx.xx.xx.xxxD N5060 OK (167 ms)2 sip peers [2 online , 0 offline] -- Remote UNIX connection___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MikeSales Managerhttp://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Plainvoip problem.
could it be IPP VS digium implementation ?On 6/8/06, William Piper [EMAIL PROTECTED] wrote: Send an email to support@plainvoip.com. They are normally quite helpful. bp On 6/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote: Do you have the g729 codec? On 6/8/06, Henry J. Cobb [EMAIL PROTECTED] wrote: Jun8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729... Jun8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to 256 Yes, and that works fine when talking with the phone itself, as you seethe connection to the phone is g729.Then it changes from g729 to g729?--Henry J. Cobb http://www.io.com/~hcobb/ ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Managerhttp://www.theclubvoip.comMaking it happen 1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No CID on ZAP
yes you need to waiti assume you are using latest ?WAIT(5) should work..also i guess you need an answerOn 6/9/06, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: Mine has usecallerid=yes and caller id works. Not sure if that's the problem or not. On 6/9/06, Curt Shaffer [EMAIL PROTECTED] wrote: [channels]language=en#include zapata_additional.confcontext=from-zaptelsignalling=fxs_ks faxdetect=incomingusecallerid=asreceivedechocancel=yescallprogress=nobusydetect=noechocancelwhenbridged=noechotraining=800group=0channel=1-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of Tom VileSent: Friday, June 09, 2006 11:53 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] No CID on ZAPThanks for sharing that info.How about sharing your zapata.conf configuration so that someone can look at it and maybe see if there is a problem.I'm guessing you want help with this.On 6/9/06, Curt Shaffer [EMAIL PROTECTED] wrote: I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP routes inbound are providing CID with no issue. Here is the output from the log when a call is coming in: -- Starting simple switch on 'Zap/1-1' -- Executing NoOp(Zap/1-1, Entering from-zaptel with DID == ) in new stack -- Executing Set(Zap/1-1, DID=s) in new stack -- Executing NoOp(Zap/1-1, DID is now s) in new stack -- Executing GotoIf(Zap/1-1, 1?zapok:notzap) in new stack -- Goto (from-zaptel,s,7) -- Executing NoOp(Zap/1-1, Is a Zaptel Channel) in new stack -- Executing Set(Zap/1-1, CHAN=1-1) in new stack -- Executing Set(Zap/1-1, CHAN=1) in new stack -- Executing Macro(Zap/1-1, from-zaptel-1|s|1) in new stack -- Executing NoOp(Zap/1-1, Returned from Macro from-zaptel-1) innew stack -- Executing Goto(Zap/1-1, ext-did|s|1) in new stack -- Goto (ext-did,s,1) -- Executing Set(Zap/1-1, FROM_DID=s) in new stack -- Executing Goto(Zap/1-1, ext-local|200|1) in new stack -- Goto (ext-local,200,1) -- Executing Macro(Zap/1-1, exten-vm|200|200) in new stack -- Executing Macro(Zap/1-1, user-callerid) in new stack -- Executing GotoIf(Zap/1-1, 0?report) in new stack -- Executing GotoIf(Zap/1-1, 0?start) in new stack -- Executing Set(Zap/1-1, REALCALLERIDNUM=) in new stack -- Executing NoOp(Zap/1-1, REALCALLERIDNUM is ) in new stack -- Executing Set(Zap/1-1, AMPUSER=) in new stack -- Executing Set(Zap/1-1, AMPUSERCIDNAME=) in new stack -- Executing GotoIf(Zap/1-1, 1?report) in new stack -- Goto (macro-user-callerid,s,9) -- Executing NoOp(Zap/1-1, Using CallerID ) in new stack -- Executing Set(Zap/1-1, FROMCONTEXT=exten-vm) in new stack -- Executing Set(Zap/1-1, VMBOX=200) in new stack -- Executing Set(Zap/1-1, EXTTOCALL=200) in new stack -- Executing Set(Zap/1-1, CFUEXT=) in new stack -- Executing Set(Zap/1-1, RT=25) in new stack -- Executing Macro(Zap/1-1, record-enable|200|IN) in new stack -- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Zap/1-1, recordingcheck|20060609-095557|1149864957.408) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060609-095557|1149864957.408: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(Zap/1-1, No recording needed) in new stack -- Executing GotoIf(Zap/1-1, 0?dolocaldial|1) in new stack -- Executing Macro(Zap/1-1, dial|25|tr|200) in new stack -- Executing AGI(Zap/1-1, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi --dialparties.agi: priority is 1dialparties.agi: Caller ID name is 'unknown' number is 'unknown' dialparties.agi: Methodology of ring is'none' --dialparties.agi: Added extension 200 to extension map --dialparties.agi: Extension 200 cf is disabled --dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 --dialparties.agi: Checking CW and CFB status for extension 200 --dialparties.agi : DbSet CALLTRACE/200 to unknown -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(Zap/1-1, SIP/200|25|tr) in new stack Any help would be appreciated. Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone:
Re: [Asterisk-Users] shutting down a mysql server renders cdr_mysql dead and asterisk nolonger makes or receives calls
well.. whats the question...if you shut down an essential servic eof course you will have this..its like saying i removed the drive and all crashed.On 6/9/06, Andrew Kirch [EMAIL PROTECTED] wrote: At approximately 3:15pm I shut down the office MySQL server to changeout some hardware.Shortly after I received a call from one of twocustomers whose asterisk servers output CDR data to that server.Theycould not place or receive calls.Shortly after that I received a call from the other customer.I'm below providing output from the messagelog (At debug level).I don't see much of use and would greatlyappreciate any help that could be given.AndrewJun9 15:16:05 ERROR[29101] cdr_addon_mysql.c: cdr_mysql: Unknown connection error: (2013) Lost connection to MySQL server during queryJun9 15:29:49 ERROR[29382] cdr_addon_mysql.c: cdr_mysql: Unknownconnection error: (2013) Lost connection to MySQL server during queryJun9 15:31:51 WARNING[4918] channel.c: Avoided initial deadlock for'0x2aaab35009c0', 10 retries!Jun9 15:31:51 WARNING[4918] channel.c: Avoided initial deadlock for'0x2aaab35009c0', 10 retries!Jun9 15:32:58 ERROR[29382] cdr_addon_mysql.c: cdr_mysql: cannot connect to database server 208.64.32.55.Jun9 15:36:08 ERROR[29647] cdr_addon_mysql.c: cdr_mysql: cannotconnect to database server 208.64.32.55 .Jun9 15:36:08 NOTICE[4928] pbx.c: Cannot find extension context'did-incomig'Jun9 15:39:41 ERROR[29796] cdr_addon_mysql.c: cdr_mysql: cannotconnect to database server 208.64.32.55 .Jun9 16:19:21 NOTICE[22239] cdr.c: CDR simple logging enabled.Jun9 16:19:22 WARNING[22239] cdr_addon_mysql.c: MySQL database sockfile not specified.Using defaultJun9 16:19:23 WARNING[22239] pbx_config.c: No closing parenthesis found? 'MeetMe(50667|Msipr}'Jun9 16:19:23 WARNING[22239] pbx_config.c: No closing parenthesisfound? 'MeetMe(31391|Msipr}'Jun9 16:19:23 WARNING[22239] pbx.c: Unable to register extension'50667', priority 1 in 'did-incoming', already in use Jun9 16:19:23 WARNING[22239] pbx_config.c: Unable to registerextension at line 103Jun9 16:19:23 WARNING[22239] pbx.c: Unable to register extension'82612', priority 1 in 'did-incoming', already in use Jun9 16:19:23 WARNING[22239] pbx_config.c: Unable to registerextension at line 104Jun9 16:19:23 WARNING[22239] pbx.c: Unable to register extension'61908', priority 1 in 'did-incoming', already in useJun9 16:19:23 WARNING[22239] pbx_config.c: Unable to register extension at line 105Jun9 16:19:23 WARNING[22239] pbx.c: Unable to register extension'24104', priority 1 in 'did-incoming', already in useJun9 16:19:23 WARNING[22239] pbx_config.c: Unable to register extension at line 106Jun9 16:19:23 WARNING[22239] pbx.c: Unable to register extension'83416', priority 1 in 'did-incoming', already in useJun9 16:19:23 WARNING[22239] pbx_config.c: Unable to registerextension at line 107 Jun9 16:19:23 WARNING[22239] pbx.c: Unable to register extension'90780', priority 1 in 'did-incoming', already in useJun9 16:19:23 WARNING[22239] pbx_config.c: Unable to registerextension at line 108 Jun9 16:19:23 WARNING[22239] pbx.c: Unable to register extension'77252', priority 1 in 'did-incoming', already in useJun9 16:19:23 WARNING[22239] pbx_config.c: Unable to registerextension at line 110Jun9 16:19:23 WARNING[22239] pbx.c: Unable to register extension'77604', priority 1 in 'did-incoming', already in useJun9 16:19:23 WARNING[22239] pbx_config.c: Unable to registerextension at line 111Jun9 16:19:23 WARNING[22239] pbx.c : Unable to register extension'25647', priority 1 in 'did-incoming', already in useJun9 16:19:23 WARNING[22239] pbx_config.c: Unable to registerextension at line 112___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Managerhttp://www.theclubvoip.comMaking it happen 1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disabling debug output
but what if the ast_log function is conditioned ast_log(LOG_WARNING, Autodestruct on call '%s' with owner in place\n, p-callid);whats in there ?On 6/9/06,
BJ Weschke [EMAIL PROTECTED] wrote:
On 6/9/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jun 08, 2006 at 10:44:39PM -0400, BJ Weschke wrote: On 6/8/06, Nick Hoffman
[EMAIL PROTECTED] wrote: Hi guys. I'm trying to disable all debug output, but am not having any success:[EMAIL PROTECTED]:~ sudo asterisk -r
Asterisk 1.2.8, Copyright (C) 1999 - 2006 Digium, Inc. and others. ..snip... certain conditions. Type 'show license' for details. =
Connected to Asterisk 1.2.8 currently running on asterisk-dev1 (pid = 7216) Verbosity is at least 19 asterisk-dev1*CLI sip no debug SIP Debugging Disabled
asterisk-dev1*CLI debug level 0 Debugging level set to 0, file 'any' asterisk-dev1*CLI asterisk-dev1*CLI Jun9 12:34:57 DEBUG[7225]: chan_sip.c:1323 __sip_autodestruct: Auto
destroying call '[EMAIL PROTECTED]'What am I doing wrong? I noticed that Asterisk said that the verbosity
level is = 19, but the debug message that appeared should've been suppressed by ``sip no debug'', no? In theory, yes, In practice:
p-autokillid = -1; ast_log(LOG_DEBUG, Auto destroying call '%s'\n, p-callid); append_history(p, AutoDestroy, );
if (p-owner) { ast_log(LOG_WARNING, Autodestruct on call '%s' with owner in place\n, p-callid); ast_queue_hangup(p-owner); } else {
Should that ast_log be conditioned? Yes. Absolutely.--Bird's The Word Technologies, Inc.http://www.btwtech.com/___
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1.888.470.7253
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Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install
if on freebsd..stop asteriskkillall mpg123cd /usr/ports/audio/madplay/make make installedit musiconhold.conf[default]mode=customdirectory=/usr/local/share/asterisk/mohmp3 application=/usr/local/bin/madplay -Q -o raw:- --mono -R 8000 -a -12then restart asterisk.. Mikehttp://www.theclubvoip.com On 6/7/06, turby [EMAIL PROTECTED] wrote: convert the moh sounfile to pcm or sln save the file to /var/lib/asterisk/moh/default set the musiconhold.conf [default]mode=filesdirectory=/var/lib/asterisk/moh/default turby@ www.canistec.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Richard ReinaSent: Wednesday, June 07, 2006 5:30 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install Thank you very much for your relply. No I did not install mpg123 as the instructions at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conffor version 1.2 say the mpg123 is no longer needed.| Rurouni Alucard | [EMAIL PROTECTED] wrote: Did you check your mpg123 version ?, asterisk needs a specific version in order to work... - Original Message - From: Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02 AM Subject: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install I have followed the instructions provided at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.confincluding installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now.However, no sound is heard and I get this message from the CLI when accessing MOH:-- Started music on hold, class 'default', on channel 'Zap/19-1'-- Stoped music on hold on Zap/19-1This happens whether it's a parked call or whether I access MOH directly via:exten = 800,1,Answerexten = 800,2,MusicOnHold()Any help would be greatly appreciated.Thank you very much.Richard __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unlock / install of Cisco 7940 IP Phone ?
cisco topic .. is there a sip image for 7980's yet ?On 6/7/06, Aaron Daniel [EMAIL PROTECTED] wrote: You have to press settings, then **#, and wait a moment to make sure itunlocks.Then you can configure a tftp server to use. The alternative is to configure your dhcp server with a tftp server.Onlinux, that would be next-server ip/host in the subnet section.TFTPserver on windows.On Wed, 7 Jun 2006, Mateo Meier wrote: Hello Guys, I just got my new Cisco 7940* IP Phone Unfortunately I can't find out how to setup this phone. Am I really that stupid ;-) ? I've read the Cisco Manuel, but everything I tried did not help anything. 1.When I turn on the phone it will display Configuring VLANConfiguring IP.. This message will not disappear. 2.I can see that the phone has a local IP. I can also access the IP over my LAN with http (only http, telnet does not work) 3 My Main menu will this show Configuring VLANConfiguring IP.. But if I click on settings, network settings it will show me the local IP of the phone Now, my question, what do I do wrong ? how can I get that phone installed with a sip image ? I tried to unlock the phone with **# but that does not do anything. Also, there is no unlock function in the phone menu (phone settings) This is a new Cisco phone, no sip image on it. Thank you for the help Matt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--Aaron DanielComputer Systems TechnicianSam Houston State University[EMAIL PROTECTED](936) 294-4198 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- MikeSales Manager http://www.theclubvoip.comMaking it happen1.888.470.7253 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF feedthru again...
in Fact we saw similar problems with all sipura products. We think its a default value thats not quite right for the north american market, these units are built and tested in asia mostly.one simple test to check it out is call this number
www.nextwavetitaniumplus.com Toll-Free Account Information Line: 888-252-9535it just seemd that even the cisco is not passing the dtmf ..Can anyone confirm ?
On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:
Also to expand on this... when listening to opposing phone in a connectedcall over PSTN you hear a click followed by a very short burst of DTMFaudible energy. Same in both directions.I can't be the only one having this problem!
DougOn Tue, 6 Jun 2006, Tom Vile wrote: try setting dtmf playback length to .5 in the admin section of the Sipura and try again. On 6/6/06, Doug Crompton
[EMAIL PROTECTED] wrote: Ok trying this again... is there anyone using the SPA-3000 with * I am not sure if this is a specific problem to it or not. This is something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine.
I checked by calling myself. Listening to either end on a completed call, and pressing a DTMF button on the opposing phone results in an audible click and very little if any audible DTMF energy being heard.
What is muting the DTMF??? Does * have anything to do with this? I am not using any dial flags. I tried 'inband' in all places with no difference. At one point this
seemed like a * feature problem and I thought removing dial flags fixed it but that does not now seem to be the case. How can I fix this??? Doug
*Doug Crompton * *Richboro, PA 18954* *215-431-6307* **
* [EMAIL PROTECTED]* * http://www.crompton.com*
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-- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205
Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersThose that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety.-- Ben Franklin (1759)*Doug Crompton **Richboro, PA 18954**215-431-6307***
* [EMAIL PROTECTED]** http://www.crompton.com*___
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Re: [Asterisk-Users] Zork and Asterisk
i got cepstral loaded and cmu recognition server waiting for comands..i instaled basic rezrov and got my main screenhow do we enable the speech onto it ?On 6/6/06, John Todd [EMAIL PROTECTED] wrote: http://www.boingboing.net/2006/06/05/play_zork_by_phone.htmlLet me preface this idea with one comment: I don't have the time todo this - I don't even have time to eat these days.But someone out there has the cycles to do this... and it would be very cool.OK, so now Zork is attached to Asterisk, but using theless-than-clear Festival engine.There are beta tests of theLumenVox speech recognition engine out there which tie directly into Asterisk.Allison Smith (the voice of Asterisk) would almostCERTAINLY do a great dramatic reading of all of the text blockswithin Zork.I see an excellent opportunity for a demo server onsome CLEC who would love to get some $ by opening up a few DIDs to a huge recip comp traffic load.Even if it's just available via IAX2or SIP, this would be one of those legends of the Net in the nextfew years...JT___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
