Re: [asterisk-users] GotoIf not working with ${EXTEN} for me in 1.4.8
On 8/19/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Sat, 18 Aug 2007, voiplist wrote:
I am using GotoIf all over the place in 1.4.8 but for some reason, the
following in my dial plan:
#
exten = _1NXXNXX,1,GotoIf([${EXTEN} = 15554441212]?100)
Missing $ before the [
Gordon
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Doh! Thanks, I guess I missed it when comparing my working examples to
this non working one.
Thanks all, I will give this a shot.
Regards,
Todd R.
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[asterisk-users] Blacklisting Toll-Free etc.
I have always been able to block toll-free numbers by catching them with a line similar to this for each DID I have on my system: exten = 5554441212/_888NXX,n,Playback(GoAway) Where 15554441212 is one of the DIDs that rings into our Asterisk box. The problem with this approach that I have to create a line like this for every pattern I want to block multiplied by every DID on my system, this gets old. So for example, if I have 20 DIDs that ring into my box, and I want to block 10 caller-id's or caller-id patterns I need like 200 lines in my extensions.conf. What I want to do is something like this: exten = _NXXNXX/_888NXX,n,Playback(GoAway) This never seems to work, why? Is there another way to block all calls from a particular caller-id or caller-id pattern without specifying the exact DID/Extension? Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best way to detect unknown and/or private incoming caller-id?
I am aware of how to match a particular caller-id or a caller-id pattern and do something with the call like this: exten = 15554441212/_888NXX,n,Playback(GoAway) What I am curious about, is the best way to block unknown, private and 000-000- calls. I know I can do this for 000-000- calls: exten = 15554441212/00,n,Playback(GoAway) Is there a better way to catch calls which are purposely blocked by the calling party? Sometimes they come through as 000-000- and as I recall sometimes just blank or unknown. Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blacklisting Toll-Free etc.
On 8/18/07, Arnaud Ligot [EMAIL PROTECTED] wrote: Hello, You can generate a file using bash for example which will be included inside extension.conf. for example, in bash: for i1 in `seq -f %02g 0 99` do for i2 in `seq -g %06g 0 99` echo 'exten = _N$i1N'$i2'/_888NXX,n,Playback(GoAway)' done; done extension-include.conf ... you will probably want to reduce the range of number if you don't want a really really really long file... an other way would be to execute a macro/agi for each call which will make your checks. best regars, Arnaud. On Sat, 2007-08-18 at 14:06 -0500, voiplist wrote: I have always been able to block toll-free numbers by catching them with a line similar to this for each DID I have on my system: exten = 5554441212/_888NXX,n,Playback(GoAway) Where 15554441212 is one of the DIDs that rings into our Asterisk box. The problem with this approach that I have to create a line like this for every pattern I want to block multiplied by every DID on my system, this gets old. So for example, if I have 20 DIDs that ring into my box, and I want to block 10 caller-id's or caller-id patterns I need like 200 lines in my extensions.conf. What I want to do is something like this: exten = _NXXNXX/_888NXX,n,Playback(GoAway) This never seems to work, why? Is there another way to block all calls from a particular caller-id or caller-id pattern without specifying the exact DID/Extension? Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Arnaud, Thanks but I already knew how to do it with an AGI and I know how to create the text file with some script. I was really just looking for a better way to do it within Asterisk logic itself, mostly because I am curious. Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blacklisting Toll-Free etc.
On 8/18/07, Trevor Peirce [EMAIL PROTECTED] wrote:
voiplist wrote:
I have always been able to block toll-free numbers by catching them
with a line similar to this for each DID I have on my system:
exten = 5554441212/_888NXX,n,Playback(GoAway)
Where 15554441212 is one of the DIDs that rings into our Asterisk box.
The problem with this approach that I have to create a line like this
for every pattern I want to block multiplied by every DID on my
system, this gets old.
Yup, try something like this -
[macro-blocktollfree]
exten = s,1,GotoIf($[${MACRO_EXTEN:3} = 800]?goaway)
exten = s,n,GotoIf($[${MACRO_EXTEN:3} = 888]?goaway)
(etc...)
exten = s,n,MacroExit
exten = s,n(goaway),Playback(goaway)
Then, if this will always be your first priority, you can include it
globally with something like
exten = _NXXNXX,1,Macro(blocktollfree)
Just make sure all other exten = lines start at priority 2 so you don't
have a conflict.
HTH,
Trevor
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Why I didn't think of this, I don't know :)
I will not do it quite like you suggested but you gave me an idea
which will work.
Gotta test a little now.. Thanks again.
Regards,
Todd R.
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Re: [asterisk-users] Best way to detect unknown and/or private incoming caller-id?
On 8/18/07, Andres Jimenez [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 2007/8/18, voiplist : Is there a better way to catch calls which are purposely blocked by the calling party? Sometimes they come through as 000-000- and as I recall sometimes just blank or unknown. The problem here is How can you be sure the calling PERSON is purposely blocking its own CALLERID? I don think you shouldn't be punishing, for example, SkypeOut users or people using dodgy carriers. If your Playback says you don't accept any anonymous call at least they would be able to change it. - -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: http://firegpg.tuxfamily.org iD8DBQFGx0me8SZxpGYWwpYRAoR7AKCdZGX8//GfdPCZovRuQN87hQh90QCdHgBl 5tTHq8WRiTjum3GIEwgkeAs= =tFmP -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I hadn't really decided what to do with those callers but I usually send them to voice mail rather than hanging up on them. This is really just for hobby use, I am just messing around with a personal number at this point, not my business numbers. Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GotoIf not working with ${EXTEN} for me in 1.4.8
I am using GotoIf all over the place in 1.4.8 but for some reason, the
following in my dial plan:
#
exten = _1NXXNXX,1,GotoIf([${EXTEN} = 15554441212]?100)
exten = _1NXXNXX,n,Dial(SIP/provider1/${EXTEN},60)
exten = _1NXXNXX,n,Dial(SIP/provider2/${EXTEN},60)
exten = _1NXXNXX,n,Hangup
exten = _1NXXNXX,100,NoOp(Calling my cell w/special CID)
exten = _1NXXNXX,n,Set(CALLERID(all)=Dude 5551112233)
exten = _1NXXNXX,n,Dial(SIP/provider1/${EXTEN},60)
exten = _1NXXNXX,n,Dial(SIP/provider2/${EXTEN},60)
exten = _1NXXNXX,n,Hangup
#
Results in this in my CLI:
##
dalint1*CLI
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing [EMAIL PROTECTED]:1]
GotoIf(SIP/5556685598-b7dc7f90, [1800555 = 15554441212]?100)
in new stack
-- Goto (out-personal,1800555,100)
-- Executing [EMAIL PROTECTED]:100]
NoOp(SIP/5556685598-b7dc7f90, Calling my cell w/special CID) in
new stack
-- Executing [EMAIL PROTECTED]:101]
Set(SIP/5556685598-b7dc7f90, CALLERID(all)=Dude 5551112233) in
new stack
-- Executing [EMAIL PROTECTED]:102]
Dial(SIP/5556685598-b7dc7f90, SIP/provider1/1800555|60) in new
stack
-- Called provider1/1800555
Clearly 1800555 does NOT equal 15554441212 so why is it jumping
to priority 100?
I always hate to scream BUG but I can't see my mistake here to save my life..
Anyone? Maybe I am just tired..
Of course I have changed the numbers involved to protect the innocent :-)
Regards,
Todd R.
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Re: [asterisk-users] GotoIf not working with ${EXTEN} for me in 1.4.8
On 8/18/07, C F [EMAIL PROTECTED] wrote:
You are missing a dollar sign $
On 8/18/07, voiplist [EMAIL PROTECTED] wrote:
I am using GotoIf all over the place in 1.4.8 but for some reason, the
following in my dial plan:
#
exten = _1NXXNXX,1,GotoIf([${EXTEN} = 15554441212]?100)
exten = _1NXXNXX,n,Dial(SIP/provider1/${EXTEN},60)
exten = _1NXXNXX,n,Dial(SIP/provider2/${EXTEN},60)
exten = _1NXXNXX,n,Hangup
exten = _1NXXNXX,100,NoOp(Calling my cell w/special CID)
exten = _1NXXNXX,n,Set(CALLERID(all)=Dude 5551112233)
exten = _1NXXNXX,n,Dial(SIP/provider1/${EXTEN},60)
exten = _1NXXNXX,n,Dial(SIP/provider2/${EXTEN},60)
exten = _1NXXNXX,n,Hangup
#
Results in this in my CLI:
##
dalint1*CLI
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing [EMAIL PROTECTED]:1]
GotoIf(SIP/5556685598-b7dc7f90, [1800555 = 15554441212]?100)
in new stack
-- Goto (out-personal,1800555,100)
-- Executing [EMAIL PROTECTED]:100]
NoOp(SIP/5556685598-b7dc7f90, Calling my cell w/special CID) in
new stack
-- Executing [EMAIL PROTECTED]:101]
Set(SIP/5556685598-b7dc7f90, CALLERID(all)=Dude 5551112233) in
new stack
-- Executing [EMAIL PROTECTED]:102]
Dial(SIP/5556685598-b7dc7f90, SIP/provider1/1800555|60) in new
stack
-- Called provider1/1800555
Clearly 1800555 does NOT equal 15554441212 so why is it jumping
to priority 100?
I always hate to scream BUG but I can't see my mistake here to save my
life..
Anyone? Maybe I am just tired..
Of course I have changed the numbers involved to protect the innocent :-)
Regards,
Todd R.
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Where?
I the only variable I am using is ${EXTEN} and as far as I can see I
have a dollar sign on each ${EXTEN}.
Regards,
Todd R.
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Re: [asterisk-users] How strip +1 from caller id on inbound call
On 8/14/07, James Collier [EMAIL PROTECTED] wrote:
What if it is an international call? Then your callerID won't work.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de C F
Enviado el: lunes, 13 de agosto de 2007 3:21
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] How strip +1 from caller id on inbound call
After rereading this post, I belive that this could also be
acomplished doing this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than 10 digits grab the last 10 digits of the CIDNUM
exten = _X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):-10})
;this grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
On 8/12/07, C F [EMAIL PROTECTED] wrote:
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-
10]});this
grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
Hope this helps.
On 8/12/07, voiplist [EMAIL PROTECTED] wrote:
From some of our telecom providers we get the caller-id as:
NXXNXX
From others we get:
+1NXXNXX
We are trying to standardize the way our caller-id comes in so we
would like to strip off the +1 from the inbound caller id.
Can anyone offer any suggestions?
I have tried:
;exten = +18664918575,1,Set(CALLERID(all)=${CALLERIDNAME}
${CALLERIDNUM:2})
but it just yacks..
Thanks in advance for any help.
Regards,
Todd R.
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Yeah, that's a problem.. I guess I may be stuck with the +1 because it
more right than wrong. Maybe I can just add a +1 to the others which
will be much easier and make it all standard..
Thanks for that, I am sure I would have run into it eventually but
it's always nice to not just run into things :-)
Regards,
Todd R.
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Re: [asterisk-users] How strip +1 from caller id on inbound call
On 8/12/07, C F [EMAIL PROTECTED] wrote:
After rereading this post, I belive that this could also be
acomplished doing this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than 10 digits grab the last 10 digits of the CIDNUM
exten = _X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):-10})
;this grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
On 8/12/07, C F [EMAIL PROTECTED] wrote:
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-10]});this
grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
Hope this helps.
On 8/12/07, voiplist [EMAIL PROTECTED] wrote:
From some of our telecom providers we get the caller-id as:
NXXNXX
From others we get:
+1NXXNXX
We are trying to standardize the way our caller-id comes in so we
would like to strip off the +1 from the inbound caller id.
Can anyone offer any suggestions?
I have tried:
;exten = +18664918575,1,Set(CALLERID(all)=${CALLERIDNAME}
${CALLERIDNUM:2})
but it just yacks..
Thanks in advance for any help.
Regards,
Todd R.
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Thanks for all the feedback fellas, I will give all of these a try and
see what happens. I will report back my results.
Regards,
Todd R.
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[asterisk-users] How strip +1 from caller id on inbound call
From some of our telecom providers we get the caller-id as:
NXXNXX
From others we get:
+1NXXNXX
We are trying to standardize the way our caller-id comes in so we
would like to strip off the +1 from the inbound caller id.
Can anyone offer any suggestions?
I have tried:
;exten = +18664918575,1,Set(CALLERID(all)=${CALLERIDNAME} ${CALLERIDNUM:2})
but it just yacks..
Thanks in advance for any help.
Regards,
Todd R.
--
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Re: [asterisk-users] Query
You can set the caller-id in many different ways but the easiest in by setting it in the sip.conf profile for the extension. So you can just add a line like this to your sip.conf under the extension: callerid=Your Name 5554441212 Hope this helps.. Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com On 8/8/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I am running asterisk PBX ( digium TE120P card configured) on one system. It is connected to E1 card running application on the other system. After establishing sync between two card, I am able to place call from sip phone to E1 card running application. I want to pass the callerid, when calling from sip phone to E1 card running application. Which all configuration files is to be changed in the asterisk. I am doing the following changes in extensions.conf exten=115,1,Dial(ZAP/g1/115,20) So, extension 115 is received at other end as callerid. Is it correct. Can any body help in how to configure for callerid with digium card. thanks and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] turn off music on hold for a single sip user
You could set a variable in the users sip.conf details like: setvar=PlayMOH=NO or setvar=PlayMOH=NO Then in your extensions.conf setup a GoToIf() which reads the variable PlayMOH and either sets the m or the r in the dial command.. This should work fine and I know it will work in 1.4.x but not sure about earlier versions only because I am not sure how far back version wise you can set a variable in the sip.conf. Maybe it's always been possible, not sure. Hope this helps. Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com On 8/7/07, Damon Estep [EMAIL PROTECTED] wrote: Is there a clean way to disable music on hold for a specific user sip user? I have seen one example that creates a class called [none] that points to an empty directory, which creates log errors that are annoying (but not harmful?) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A simple IVR extension problem
Might want to start by proving out your DTMF by just sending the calls to something like VoiceMailMain(). When going into the voicemail system, see if you can reliably get DTMF to work while entering mailbox numbers and password and moving around the VM system.. At first glance it sure sounds to me like a DTMF issue of some sort. Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com On 8/2/07, Vincent Li [EMAIL PROTECTED] wrote: Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf context=incoming signalling=fxs_ks channel = 4 context=internal signalling=fxo_ks channel = 1 - extensions.conf: [office] exten = s,1,Dial(Zap/1,30) [home] exten = s,1,Macro(stdexten,106,SIP/ht286,t) [incoming] ; incoming calls from the FXO port are directed to this context from zapata.conf exten = s,1,Answer exten = s,1,Background(enter-ext-of-person) exten = s,n,WaitExten(20) exten = 100,1,Dial(Zap/1,30) exten = 106,1,Macro(stdexten,106,SIP/ht286) exten = 101,1,Macro(stdexten,101,SIP/vli) exten = 107,1,AGI(math.agi) exten = 108,1,Playback(12) ;exten = s,1,GotoIfTime(9:00-16:30|mon-fri|*|*?office,s,1) ;exten = s,n,GotoIfTime(17:00-9:00|*|*|*?home,s,1) When I call my PSTN number, I can hear the enter-ext-of-person message, but once I press any one of the extension number, Asterisk sometime execute the relevant extension application, sometime not at all. If I comment the IVR extensions config and simply use : exten = s,1,GotoIfTime(9:00-16:30|mon-fri|*|*?office,s,1) exten = s,n,GotoIfTime(17:00-9:00|*|*|*?home,s,1) I can always get call My console message: ( Asterisk did not execute relevant extension in the last two call after I entered the extension digit) -- Starting simple switch on 'Zap/4-1' [Aug 2 13:46:38] NOTICE[4429]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:46:40] NOTICE[4429]: chan_zap.c:6373 ss_thread: Got event 2 (Ring/Answered)... -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [ [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] Macro(Zap/4-1, stdexten|101|SIP/vli|t) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, SIP/vli|20) in new stack -- Called vli -- SIP/vli-08353298 is ringing -- SIP/vli-08353298 answered Zap/4-1 == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/4-1' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' [Aug 2 13:47:32] NOTICE[4437]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:47:33] ERROR[4437]: callerid.c:564 callerid_feed: fsk_serie made mylen 0 (-168) [Aug 2 13:47:33] WARNING[4437]: chan_zap.c:6405 ss_thread: CallerID feed failed: Success [Aug 2 13:47:33] WARNING[4437]: chan_zap.c:6505 ss_thread: CallerID returned with error on channel 'Zap/4-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] Dial(Zap/4-1, Zap/1|30) in new stack -- Called 1 -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 answered Zap/4-1 -- Native bridging Zap/4-1 and Zap/1-1 -- Hungup 'Zap/1-1' == Spawn extension (incoming, 100, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' [Aug 2 13:48:22] NOTICE[]: chan_zap.c:6373 ss_thread: Got event 18 (Ring Begin)... [Aug 2 13:48:23] ERROR[]: callerid.c :564 callerid_feed: fsk_serie made mylen 0 (-9) [Aug 2 13:48:23] WARNING[]: chan_zap.c:6405 ss_thread: CallerID feed failed: Success [Aug 2 13:48:23] WARNING[]: chan_zap.c:6505 ss_thread: CallerID returned with error on channel 'Zap/4-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(Zap/4-1, enter-ext-of-person) in new stack -- Zap/4-1 Playing 'enter-ext-of-person' (language 'en') -- Executing [EMAIL PROTECTED]:3] WaitExten(Zap/4-1, 20) in new stack == CDR updated on Zap/4-1 -- Executing [EMAIL PROTECTED]:1] AGI(Zap/4-1, math.agi) in new stack --
Re: [asterisk-users] Royalty for On Hold Music ?
So is there a simple way to license decent, up to date music? Can I just go to a website, click a buy button, pay my money and download the song? It seems idiotic that you need 15 lawyers and a million bucks use decent on hold music. Maybe I just don't know the procedure. I am all for paying the license fees and doing it right but they sure don't make it easy to give them money. Any help would be appreciated. On 7/31/07, Deepak Naidu [EMAIL PROTECTED] wrote: Thanks Jared, Yes I am using with Asterisk only. So I am using the inbuilt music from Asterisk for onhold. -- Deepak Jared Smith [EMAIL PROTECTED] wrote: On Tue, 2007-07-31 at 06:36 +0100, Deepak Naidu wrote: I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. I'm no lawyer, but here's what I understand. (Please consult with an attorney in your area, and don't consider this legal advice.) The hold music that comes with Asterisk is provided by Digium under license from Freeplay Music Corporation for use in conjunction with the Asterisk software only. It's my understanding that you don't have to pay any kind of royalties to use it, as long as you're using it with Asterisk. You *do* have to pay royalties on music (or MP3 files) by commercial artists. These royalties vary by country. Using commercial music as hold music is considered broadcasting the music, which requires different licensing arrangements with the copyright holder. In the United States, you can buy a license from ASCAP (the American Society of Composers, Authors, and Publishers) to be able to broadcast music from the major record labels. There are also several other places you can get royalty-free music for hold music. I've had good luck looking online, especially at sites like MagnaTune. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Royalty for On Hold Music ?
I have done this in the past and I don't recall ever finding any popular music by popular artist. For example, if I wanted to play oh I don't know an original song performed by the original artist such as Nora Jones or The Beatles will I find this sort of thing at a Royalty Free Site? On 7/31/07, john beaman [EMAIL PROTECTED] wrote: Just Google for: royalty free music, and will find plenty of sites that will serve your needs. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 7/31/2007 12:49:45 PM So is there a simple way to license decent, up to date music? Can I just go to a website, click a buy button, pay my money and download the song? It seems idiotic that you need 15 lawyers and a million bucks use decent on hold music. Maybe I just don't know the procedure. I am all for paying the license fees and doing it right but they sure don't make it easy to give them money. Any help would be appreciated. On 7/31/07, Deepak Naidu [EMAIL PROTECTED] wrote: Thanks Jared, Yes I am using with Asterisk only. So I am using the inbuilt music from Asterisk for onhold. -- Deepak Jared Smith [EMAIL PROTECTED] wrote: On Tue, 2007-07-31 at 06:36 +0100, Deepak Naidu wrote: I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. I'm no lawyer, but here's what I understand. (Please consult with an attorney in your area, and don't consider this legal advice.) The hold music that comes with Asterisk is provided by Digium under license from Freeplay Music Corporation for use in conjunction with the Asterisk software only. It's my understanding that you don't have to pay any kind of royalties to use it, as long as you're using it with Asterisk. You *do* have to pay royalties on music (or MP3 files) by commercial artists. These royalties vary by country. Using commercial music as hold music is considered broadcasting the music, which requires different licensing arrangements with the copyright holder. In the United States, you can buy a license from ASCAP (the American Society of Composers, Authors, and Publishers) to be able to broadcast music from the major record labels. There are also several other places you can get royalty-free music for hold music. I've had good luck looking online, especially at sites like MagnaTune. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today. ___ --Bandwidth and Colocation Provided 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 - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Live Answering Service with Direct SIP Connections and Light Accounts starting at $14.95/mo
A while back I spent days searching for an answering service that would allow us to forward our calls via SIP or IAX but to my surprise found nothing. I have been a member of the Asterisk community for a few years and had Asterisk users in mind as we created our offering. I am now proud to announce the launch of Prestige Messaging: http://www.PrestigeMessaging.com We are a US based answering service with 100% of our operators located within the US. In the near future there will be many features that make us different from the traditional answering service, for now the two most exciting are outlined below. -Forward your calls to us using SIP or a toll-free number -Light use plans for the small business or consultant that start at just $14.95 Please have a look and if you have any questions please submit the contact form on our website or call us at (888) 229-1185. We do not monitor this list for questions regarding our service and want to be sure to respond in a timely manner. This is basically my personal list email address I use and I do not check it regularly. Thanks all and hope that we can be of service to the Asterisk community and take a few more calls off the PSTN :-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Live Answering Service with Direct SIP Connections and Light Accounts starting at $14.95/mo
A thousand apologies, I use gmail and when I type Asterisk both the users and biz list drop down. Somehow I picked the wrong one and I am terribly sorry about that. Please accept my apologies for this error, believe it or not it was an honest mistake. I have been on the list long enough to know that I will get flamed anyhow but all I can do is try to apologize. -- Forwarded message -- From: voiplist [EMAIL PROTECTED] Date: Jul 31, 2007 10:50 PM Subject: Live Answering Service with Direct SIP Connections and Light Accounts starting at $14.95/mo To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com A while back I spent days searching for an answering service that would allow us to forward our calls via SIP or IAX but to my surprise found nothing. I have been a member of the Asterisk community for a few years and had Asterisk users in mind as we created our offering. I am now proud to announce the launch of Prestige Messaging: http://www.PrestigeMessaging.com We are a US based answering service with 100% of our operators located within the US. In the near future there will be many features that make us different from the traditional answering service, for now the two most exciting are outlined below. -Forward your calls to us using SIP or a toll-free number -Light use plans for the small business or consultant that start at just $14.95 Please have a look and if you have any questions please submit the contact form on our website or call us at (888) 229-1185. We do not monitor this list for questions regarding our service and want to be sure to respond in a timely manner. This is basically my personal list email address I use and I do not check it regularly. Thanks all and hope that we can be of service to the Asterisk community and take a few more calls off the PSTN :-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues with logged in agents that are not reachable
Hello, I am using 1.4.8 and have a question about Queues. I noticed that if I have an agent logged in using AgentCallBackLogin and that agent is unreachable for some reason (SIP phone unplugged) calls to him/her will completely yack. For example: 1-Agent 500 is the only one logged into queue number 1. 2-A call comes into queue number 1 3-The call is pushed to agent 500 at extension 21 which is unreachable because the ethernet cable is unplugged to extension 21's handset. 4-The caller gets hungup on entirely instead of the call going to another agent or leaving the caller in the queue I don't expect this to happen but I want to be sure all bases are covered on light days during shift changes etc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.8 jabber integration
I have searched the Wiki, this list and Google for instructions on setting up Jabber with 1.4. All I see if a jabber.conf which I have setup but beyond that I am stuck. I see references to commands like jabber show connected and such but when I try them I just get No such application. It seems there is a module somewhere and maybe it needs to be loaded? Specifically I am trying to connect to Google Talk using Jabber. I am using 1.4.8 and I only want text messages not voice. Some sort of example as to how to send messages would be great too. For example, send a message with the caller-id info to a Google Talk user. Any help would be appreciated. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Received mini frame before first full voice frame
Can someone give me a little detail as to what this error message means and why it may be occuring? I keep seeing tons of these roll by on the CLI on one of our systems. Thanks! Apr 9 11:05:40 WARNING[19263]: chan_iax2.c:7538 socket_read: Received mini fra me before first full voice frame ___ --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 on mini-itx
How many simultaneous calls will this device support and with which codecs/transcoding? Do you sell the hardware stand-alone without your software so we can load our own version of Asterisk/Gui? On 3/12/07, Ioan Biris [EMAIL PROTECTED] wrote: Hi , We have done exactly that … fan less , VIA processor , flash card , firewall. http://www.allo.com/products/micropbx.php We sell wholesale. Ioan at allo.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mail Lists Sent: Saturday, March 10, 2007 11:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk on mini-itx Hello, I'm trying to put together a low cost - low powers PBX appliance for several customers. I have purchased a couple of the soekris net4801 boards and have asterisk up and running on them fine but they just don't quite cut it in the processing power department. I've been able to get about 10 simultaneous SIP calls with simple ulaw (no encoding decoding). While this might be OK for a very small business or home I just don't think it leaves a lot of overhead to do anything else. I've had a look around and I think I have settled on one of the VIA EPIA fanless boards. Does anyone have any experience with these running asterisk as far as performance and reliability is concerned? Has anyone run asterisk with any compressed codecs on this setup? I am going to TRY to run the system from flash memory one way or another - I realize the hoops I might have to jump through to prevent a large number of read/write cycles but I'd really like to have the whole thing solid state... Maybe someone has a better idea regarding program storage? Also, I would really like to run this as a router/firewall appliance as well so that that the box can sit on a public IP if the client only has one. For this reason I kind of have my heart set on openbsd. The routing and firewall utilities on openbsd are very simple to configure and easy to use. Does anyone know what limitations asterisk might have on openbsd (besides lack of zaptel.. ) ? I have run asterisk 1.2.? on openbsd before and found it worked pretty well. Failing that I suppose I would settle for running the routing/firewalling on linux. I've just found the linux networking tools very awkward up until now - perhaps someone know of a linux distribution - or tool - that makes routing/firewall/NAT as painless as on openbsd? Maybe I just need to sit down for a day and learn the tool properly ;) Anyways, I know there are a lot of questions in here but perhaps someone has done one or all of these things? Thanks for any advice or warnings! Steve Glaus ___ --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
Re: [asterisk-users] Asterisk 1.4.1 Released
Um, is it possible to patch 1.2.4? We have some pretty busy production systems and are not exactly excited about having to upgrade from this version. Is there no other way to protect our systems from this hole? On 3/3/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Michelle Dupuis wrote: Can one do an in-place update to 1.4.1 from 1.4.0 ? (Just compile and install) ./configure make update worked here. ___ --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
[asterisk-users] SIP remote crash bug
We run 1.2.4, do we have to upgrade? Is there a patch for this version? At this point we REALLY don't want to upgrade and potentially introduce a bunch of new issues and problems including our AGI's all breaking due to SET and SETVAR etc.. Thanks in advance for any insight on this issue. ___ --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] Timing, use analog card, ZT Dummy etc.
Hello, we are setting up another system that will run either 1.2.4, the latest version of 1.2 or 1.4. We have not yet decided on the version. Anyhow, this is a higher volume system (dual processor) which will handle 30-50 simultaneous calls with 60 to 100 simultaneous channels lit up. Most calls are g711 with very little g729 and a little gsm mixed in. We have a similar system doing exactly this, quite well. With our existing system we have a single span Digium T1 card installed, which we never ended up using. Nice it is in there though because Asterisk uses it for timing. The new system will be pure IP with no need for Analog or T1 circuits. Questions are: 1- Can I really get away with using ZT Dummy on a high volume system like this and put no card in? 2- If I can, should I even risk it or just put a card in? 3- I obviously don't want to put a $500 T1 card in but I do have a Digium Analog card with 2 FXO modules. I also have some clone cards. The question is, should I use the clone cards and will they work reliably just for timing. OR should I use the Digium card? 4- If I use the Digium card in, do I need to also waste a module or can I just put the bare card in with no modules since it's just for timing? Thanks for any help, I will be moving forward today more than likely and thought I would get a little advice from the list first. ___ --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] Registrations, how many is too many?
Anyone have any idea if there is some sort of limitation to the number of SIP or IAX end points which can register to an Asterisk system (2.8Ghz dual processor, 2GB ram) while also handling 30-50 simultaneous calls without getting into trouble? Of course the 30-50 simultaneous calls end up being 60-100 channels of mostly G711 VoIP. We have seen issues where our Asterisk just gets all crazy and SIP quits working all together, to the point sometimes where we can't even fix it with a restart. At one point we were using all realtime for IAX and SIP clients, then we went to text files (more or less), still we are seeing this issue. When this happens we can't even do simple things with SIP like sip show peers etc. because Asterisk just says that the application doesn't exist. This has been a battle for a few months and we can't put our finger on it. Can't seem to figure out when it's going to happen either which is VERY tough on the nerves to say the least. This happens during peak times but also in the middle of the night when call volume is slow to non existent. The only thing that's constant during both peak and non peak times is the amount of registrations the system deals with. We have approx 1500-1800 end points registering to this particular system at any one time. This is a split between IAX and SIP not sure what the percentage of each is at the moment. It's been a long time since a problem has beat me/us and this one has won so far. Any help in getting my sanity back would be REALLY appreciated. ___ --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] Registrations, how many is too many?
That only helps on the SIP side.. :( Although it would help some.. Before we go making changes, we are really just trying to determine what the cause is. On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote: voiplist wrote: Anyone have any idea if there is some sort of limitation to the number of SIP or IAX end points which can register to an Asterisk system (2.8Ghz dual processor, 2GB ram) while also handling 30-50 simultaneous calls without getting into trouble? Of course the 30-50 simultaneous calls end up being 60-100 channels of mostly G711 VoIP. We have seen issues where our Asterisk just gets all crazy and SIP quits working all together, to the point sometimes where we can't even fix it with a restart. At one point we were using all realtime for IAX and SIP clients, then we went to text files (more or less), still we are seeing this issue. When this happens we can't even do simple things with SIP like sip show peers etc. because Asterisk just says that the application doesn't exist. This has been a battle for a few months and we can't put our finger on it. Can't seem to figure out when it's going to happen either which is VERY tough on the nerves to say the least. This happens during peak times but also in the middle of the night when call volume is slow to non existent. The only thing that's constant during both peak and non peak times is the amount of registrations the system deals with. We have approx 1500-1800 end points registering to this particular system at any one time. This is a split between IAX and SIP not sure what the percentage of each is at the moment. It's been a long time since a problem has beat me/us and this one has won so far. Any help in getting my sanity back would be REALLY appreciated. Time for SER? Thanks, Steve ___ --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
Re: [asterisk-users] Registrations, how many is too many?
We do not use dyndns for anything, not sure what we would even use it for. We do have lots of hostnames to different systems in our sip.conf, I have changed them all to IP to see if this helps. So, you think that maybe when DNS gets hosed up that it could cause SIP to just tank on a high volume system? On 2/28/07, C F [EMAIL PROTECTED] wrote: any dns in the sip channel could do this not only dynamic On 2/28/07, voiplist [EMAIL PROTECTED] wrote: That only helps on the SIP side.. :( Although it would help some.. Before we go making changes, we are really just trying to determine what the cause is. On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote: voiplist wrote: Anyone have any idea if there is some sort of limitation to the number of SIP or IAX end points which can register to an Asterisk system (2.8Ghz dual processor, 2GB ram) while also handling 30-50 simultaneous calls without getting into trouble? Of course the 30-50 simultaneous calls end up being 60-100 channels of mostly G711 VoIP. We have seen issues where our Asterisk just gets all crazy and SIP quits working all together, to the point sometimes where we can't even fix it with a restart. At one point we were using all realtime for IAX and SIP clients, then we went to text files (more or less), still we are seeing this issue. When this happens we can't even do simple things with SIP like sip show peers etc. because Asterisk just says that the application doesn't exist. This has been a battle for a few months and we can't put our finger on it. Can't seem to figure out when it's going to happen either which is VERY tough on the nerves to say the least. This happens during peak times but also in the middle of the night when call volume is slow to non existent. The only thing that's constant during both peak and non peak times is the amount of registrations the system deals with. We have approx 1500-1800 end points registering to this particular system at any one time. This is a split between IAX and SIP not sure what the percentage of each is at the moment. It's been a long time since a problem has beat me/us and this one has won so far. Any help in getting my sanity back would be REALLY appreciated. Time for SER? Thanks, Steve ___ --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
Re: [asterisk-users] Registrations, how many is too many?
That is interesting.. Not sure though how getting rid of IAX could have fixed your SIP issues, seems odd. We can't really get rid of IAX, our customers would flip their lids. The big difference we have is that this has happened on more than one occasion when there was little to no call volume. On 2/28/07, Alejandro Kauffmann [EMAIL PROTECTED] wrote: Anyone have any idea if there is some sort of limitation to the number of SIP or IAX end points which can register to an Asterisk system (2.8Ghz dual processor, 2GB ram) while also handling 30-50 simultaneous calls without getting into trouble? Of course the 30-50 simultaneous calls end up being 60-100 channels of mostly G711 VoIP. We have seen issues where our Asterisk just gets all crazy and SIP quits working all together, to the point sometimes where we can't even fix it with a restart. At one point we were using all realtime for IAX and SIP clients, then we went to text files (more or less), still we are seeing this issue. When this happens we can't even do simple things with SIP like sip show peers etc. because Asterisk just says that the application doesn't exist. This has been a battle for a few months and we can't put our finger on it. Can't seem to figure out when it's going to happen either which is VERY tough on the nerves to say the least. This happens during peak times but also in the middle of the night when call volume is slow to non existent. The only thing that's constant during both peak and non peak times is the amount of registrations the system deals with. We have approx 1500-1800 end points registering to this particular system at any one time. This is a split between IAX and SIP not sure what the percentage of each is at the moment. It's been a long time since a problem has beat me/us and this one has won so far. Any help in getting my sanity back would be REALLY appreciated. ___ We've seen the same behavior since the 1.0.x version and have been unable to track it down. What I can tell you is that we used to peak around 40 calls (SIP/IAX to zap over E1 PRI all using alaw) and SIP would crash. I saw a peak of 97 calls today (216 channels with chanspy accounting for most of the difference) and this is on a single core 3.2Ghz with 1Gb ram. What changed? We got rid of IAX. I can't tell you if this will work for you or not, but my nerves are doing better now that I don't get calls at random hours of the day telling me that we have no phones. As a side note, inbound ZAP(PRI) would still work, but only to IAX endpoints. ___ --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
Re: [asterisk-users] Newbie extensions.conf question
What does this module do?
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] Newbie extensions.conf question
Thanks for the link.. As for Google, I know how to use it. I searched for Sven Slezak's Notify and other variations and got Squat.. On 2/28/07, Patrick [EMAIL PROTECTED] wrote: On Wed, 2007-02-28 at 22:04 -0600, voiplist wrote: What does this module do? On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote: I've installed Sven Slezak's Notify module. http://mezzo.net/asterisk/app_notify.html Google is your friend. Regards, Patrick ___ --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
[asterisk-users] strip + sign from incoming ${EXTEN} var?
Is it possible to strip the plus sign from the ${EXTEN} var on an incoming call?
We have our system setup to deal with incoming calls to numbers
without a plus sign, lots of AGIs and databases we don't want to have
to change.
We have seen things like this ${EXTEN:1} which you can use in the dial
command but we want to basically change the ${EXTEN} var right off
when it comes into extensions.conf before we do anything else.
I have read that since this is a built in Asterisk variable and it can
only be read, not written to.
We know there are other ways to handle this but we were just hoping
for a simple solution like resetting the variable.
Any help would be appreciated.
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Re: [asterisk-users] Freepbx changes dont reflect in asterisk
We ran into a Beta version of FreePBX a few weeks ago that was doing this.. So, if you are running a beta version, upgrade or downgrade and see if that does the trick. On 11/17/06, Alex Robar [EMAIL PROTECTED] wrote: I think you guys are all misunderstanding the problem here. Unless I'm misunderstanding, Pedro's issue is that when he makes changes in FreePBX, those changes are not written out to the config files. Alex On 11/17/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: You can't do any modifications in extensions_additional.conf and sip_additional.conf. Same is true for extensions.conf and sip.conf, and other original trixbox files. As soon as you press the red bar, they are returned to their original state. For modifications, you create your own files or use sip_customs.conf and extensions_custom.conf. Please don't mix trixbox with asterisk just because its based on asterisk. Its a completely customized solution of various software packages configured to make it work according to its own requirements. For help, post on trixbox.org forums. ___ --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 -- Alex Robar [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
[asterisk-users] CDR shows NO ANSWER when call is really ANSWERED
Tonight I made 3 calls, all which were answered at the remote end. All three calls showed up in the CDR but only one showed a disposition of ANSWERED the other 2 had a disposition of NO ANSWER. Few other things to note, on the calls with no answer the bill seconds is of course 0. After further testing we noticed that only calls to this one particular number over 16 seconds will show up as ANSWERED calls under 16 seconds show as NO ANSWER. On the calls which show ANSWERED, the bill seconds is always 16 seconds less than the duration. There seems to be some bug that was submitted about this but I don't see any further info. Not sure if this is really a bug and if it was resolved, which version, was there a patch etc. The bug is here: http://bugs.digium.com/view.php?id=8221 These calls are not getting billed and we are paying our carrier for them so this is a problem to say the least. Thanks for any help you (the list) can provide. ___ --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] Load balance Asterisk servers?
We are looking to be able to put a device in front of an array of Asterisk systems which would do the job of load balancing them. We would store all the particulars on one or more MySQL servers. What want to accomplish is to have all calls sent to/from a single IP, then push the calls off to another Asterisk server in the array. If one server goes out, we are hoping there will be no effect other than we have reduced capacity until it's fixed. If possible we would like to do this with either a low cost device or an open source solution which can run on a Linux box. Can anyone suggest something that would be reliable in a production environment? We would like to make this solution scale to at least a few hundred simultaneous calls. We have looked at some ready made devices but many of them only support SIP, we need a solution that will support both IAX and SIP. Any advice would be most appreciated. ___ --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] Getting Music On Hold working in * 1.2.12.1 with Fedora?
We are aware of the MPG123 tweaks that were always needed with Fedora in the past. We have MOH working on all other systems. We just installed a new system with a clean install of 1.2.12.1. It seems that there is info on the Wiki which states that there is a new way to do MOH using some internal Asterisk method. Says we have to install the add-ons package which we have done. I see no other hints or instructions on making MOH work with this version of Asterisk and Fedora 4. We only get silence where the MOH should be. Have I missed something? ___ --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] Linksys SPA-941 Message Waiting Indicator
Greetings.. I have a few Linksys SPA-941 IP phones running the latest firmware 4.1.12(a). I tried turning on the Message Waiting indicator but it doesn't seem to work correctly for me. This phone is connecting to Asterisk 1.24 running Realtime. Not sure if it matters but rtcachefriends=yes is set. Basically, as soon as I turn the item labeled Message Waiting to yes the red light turns on on my phone, I get stutter tone and little envelope icons show up on my phone. Doesn't matter if I have voicemail or not. I have tried filling in Mailbox ID: and VoiceMailServer: with various things but nothing seems to help. Any ideas? Would be nice to use this feature as we have with other Sipura products in the past. ___ --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] Check call duration on active call in CLI?
Is there a command to check the call duration of an active call in the CLI? ___ --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] Rate engine AGI?
Is there an AGI out there which we can call from extensions.conf which will lookup a rate in a MySQL db based on the number the callerer dialed? We don't want anything with tons of features as we are doing all our coding, we just want something that will give us the rate and maybe permission to call or not call that country. We don't want it to bill for us or anything else because we have all that worked out, just need to get the rate. Any help would be greatly appreciated. ___ --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] Rate engine AGI?
Is there an AGI out there which we can call from extensions.conf which will lookup a rate in a MySQL db based on the number the callerer dialed? We don't want anything with tons of features as we are doing all our coding, we just want something that will give us the rate and maybe permission to call or not call that country. We don't want it to bill for us or anything else because we have all that worked out, just need to get the rate. Any help would be greatly appreciated. ___ --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] Two phone numbers, one SIP provider
On 7/20/06, Mat Stace [EMAIL PROTECTED] wrote: I'm not exactly sure on the /how/ * mathes items from the sip.conf (I suspect it goes to the latter for whichever provider), but the way configured my extenions.conf to handle multiple incoming accounts from sipgate is like this (obviously much simplified for ease of explanation): [incoming_sipgate] exten = ,1,Answer exten = ,2,Dial(SIP/ciscophone,12) exten = ,1,Answer exten = ,2,Dial(SIP/pcsoftphone,12) Also, in the sip.conf, each peer has context=incoming_sipgate in it. HTH, Mat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Stocker Sent: 20 July 2006 16:05 To: Asterisk Users Mailing List Subject: [asterisk-users] Two phone numbers, one SIP provider Hi I have two phone numbers from my SIP provider sippro.com, say and . I use two sip.conf entries to register this phone numbers: register = :[EMAIL PROTECTED]/ register = :[EMAIL PROTECTED]/ [] type=friend username= secret=pass insecure=very host= sip.sippro.com context=incoming- [] type=friend username= secret=pass insecure=very host=sip.sippro.com context=incoming- Now, from my dialplan I can use them to do outgoing calls, like Dial(SIP/[EMAIL PROTECTED]). That works pretty fine. The problem are incoming calls. According to [1] asterisk should lookup a match in sip.conf when somebody (outside sippro.com) calls or . For example, a call to should look for a extension in context 'incoming-'. A call for should go to context incoming-. But in the above scenario, asterisk always gets a match on ''. As a result, context 'incoming-' is always used. How does asterisk search for a match in sip,conf for incoming calls and how can I get it to use the context specified in the account settings? 1. http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf I think it might be finding a match on the host= field. I could be totally wrong here but this might be worth a try. If it IS the host which is matching, you might try splitting up the incoming and outgoing context. This way, you can remove the host entry from the incoming context completely. So something like this: [] type=friend username= secret=pass insecure=very context=incoming- [] type=friend username= secret=pass insecure=very context=incoming- [sippro_out] type=peer host=sip.sippro.com username= ;OR secret=pass Then in your dial string use: Dial(SIP/[EMAIL PROTECTED]) Let us know how it works out.. ___ --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] Keep Zap Channel from answering
Anyone know how to keep an Analog Zap channel from answering? I know I can answer it and send it to voicemail or do any number of other things with it once it's answered. I want to keep Asterisk from answering it, completely ignoring it while still having the line connected for outgoing purposes. Reason is, I have Vonage line I am going to be porting and for now it works horribly for inbound calls hooked up from the Cisco 186 - Wildcard. What I have done is setup an instant forward with Vonage to another number with another provider. Problem is, Vonage still rings the ATA once causing the call to be picked up by Asterisk instead of being forwarded as intended. I know, I could just unplug the ATA but it's bugging me and I would like to use it for outbound until I port the number and close the account. Looked around quite a bit but I can't find much on this topic. ___ --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] SPA-2000, Asterisk 1.2.4 Incoming call success? Anyone?
We still have a miserable problem trying to figure out why ALL of our SPA-2000 ata's which work fine on other versions of Asterisk do not allow incoming calls from Asterisk 1.2.4 I have gone into specifics of this problem in other post so I won't do that here. The purpose of this post is to identify how many, if any are actually using this combination with success. Thanks! ___ --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] Testing 911?
It seems that 911 is important enough that when setting up an Asterisk box, it should be tested. How do you go about testing 911 dialing without getting fined for calling for a non-emergency reason? Is there some circumstances where you can ask permission from the city ahead of time? I realize this may be a real stupid question but I have not seen this discussed and I am curious. ___ --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 SPA 2002 ATA hardphone UNREACHABLE...!!!
On 7/17/06, | Rurouni Alucard | [EMAIL PROTECTED] wrote: When using Grandstreamg Handytone ATA everything works fine incoming/ougoing but when using Linksys SPA 2002 ATA 'sip show peers' marks those extensions as UNREACHABLE and can't receive calls, but they can call out. Any Idea about possible reasons ?... Rurouni, ___ --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 We have 6 or 7 SPA-2000's which all work with other installs of Asterisk but can't get a single one to receive calls using Asterisk 1.2.4. So far we have not been able to solve this problem and calling Sipura support was just more work than I could deal with. We are still looking for a solution to this problem also. ___ --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 SPA 2002 ATA hardphone UNREACHABLE...!!!
On 7/17/06, Luki [EMAIL PROTECTED] wrote: We have 6 or 7 SPA-2000's which all work with other installs of Asterisk but can't get a single one to receive calls using Asterisk 1.2.4. Ha! You're right. I just got some too and didn't even think of testing the ringer. Outgoing calls work fine, but incoming calls say Call 1 State: Ringing on the web interface and the call details are displayed but the phone does not ring. It obviously gets the SIP message that it should ring but it does not. Asterisk CLI also confirms that device is ringing. Increasing the ring voltage did not help either. Needless to say the same phone works fine with SPA 1000, 1001 and Grandstream. Interesting... any ideas what the heck is up with that? This is software version 3.1.9(LSa). I can't upgrade the software because the unit thinks it's not idle and hence does not start the upgrade process. Kind of disappointing. --Luki ___ --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 Not sure this is exactly the problem we have, our call gets rejected by the device for some odd reason. Not 100% sure at this point because it's been a while, I am going to do more testing in a few minutes I think. Can those having trouble confirm their Asterisk version? The version I am having issues with is 1.2.4, I have a slightly older version of Asterisk which rings these ATAs just fine.. ___ --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: Wrong account code from iax_buddies
Trixbox is only used as the client to simulate what we already saw happening with a customer. I don't think the fact that we used a Trixbox on the client side has anything to do with the problem on the server side which is not using Trixbox. The on the server side Asterisk only sees the Trixbox as a client coming in under an iax_buddies record which has an accountcode assigned to it. The server side is taking the call, then inserting the CDR with a correct account code, dst, src and so on but the channel is wrong. On 7/15/06, Tim Panton [EMAIL PROTECTED] wrote: On 15 Jul 2006, at 19:26, voiplist wrote: On 7/15/06, Tim Panton [EMAIL PROTECTED] wrote: On 15 Jul 2006, at 17:24, voiplist wrote: After further testing, here is what we found.. The account code was actually right after all, what made us think it was incorrect was the fact that * was reporting the wrong channel for the call. Test 2: We turned off the softphones mentioned above. We then setup a test Asterisk box with the same IAX accounts mentioned above. Each account registered to our server remotely just as the softphones did. We then made 10 phone calls with each account dialing totally different phone numbers so we could identify the records from each IAX account. Results (Test 2): We found that the majority of calls now had wrong/mismatched channels in the CDR. Basically we would show a channel like IAX2/user2 for a call that we are certain came from the IAX user user1. And... That's my story and I'm stickin' to it :) Hmm, I'd like to know more about this, as it is likely to bite me too :-( Can you describe the setup of the 'originating' asterisk in your second test? If possible please share the relevant bits of extensions.conf and iax.conf. Thanks. Tim. Tim Panton [EMAIL PROTECTED] Sure it was TrixBox 1.1 running on a VmWare virtual machine (great for testing). We basically setup two Trunks as they are known is TrixBox ([EMAIL PROTECTED]), one with each of the IAX accounts we setup on the server end. Just to keep the confusion to a minimum, the TrixBox was the client which was sending the calls to the Asterisk Server which was terminating the calls to the PSTN. If you are familiar with TrixBox Trunk Configs I can send you the basic details of the settings in the GUI Trunk screen. Ah, no, I am using the base asterisk facilities, plus a bit of AGI magic. You would need to strip the test case down to just asterisk before you could be sure where the problem is. Tim Panton www.mexuar.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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Wrong account code from iax_buddies
Actually, at this point this info was more for the community as a whole. We don't need to fix this now because the account code is right and that's what is important. On 7/16/06, Tim Panton [EMAIL PROTECTED] wrote: On 16 Jul 2006, at 19:00, voiplist wrote: Trixbox is only used as the client to simulate what we already saw happening with a customer. I don't think the fact that we used a Trixbox on the client side has anything to do with the problem on the server side which is not using Trixbox. The on the server side Asterisk only sees the Trixbox as a client coming in under an iax_buddies record which has an accountcode assigned to it. The server side is taking the call, then inserting the CDR with a correct account code, dst, src and so on but the channel is wrong. I still think that if you want your problem fixed, you'll need to reduce it to a test case that an asterisk maintainer can reproduce, without using trixbox. (or fix it yourself) The other alternative would be to capture some IAX traces either with ethereal or asterisk's iax2 debug of a call where this goes wrong. This might give clues as to how your problem happens. If you do produce IAX traces I'd be happy to look them over as I'm going to likely run into this problem myself in a couple of weeks. Tim Panton www.mexuar.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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Wrong account code from iax_buddies
After further testing, here is what we found.. The account code was actually right after all, what made us think it was incorrect was the fact that * was reporting the wrong channel for the call. For example we may have a call from iaxuser1 which comes in with the correct account code but it seems that the channel that shows for the call is something like IAX2/user2. In our testing we made 10 calls each to two separate IAX accounts to two different phone numbers. We did this so we could identify the records in the CDR with certainty. What we found is that we would have records with the correct to and from numbers, correct account code and so on but the channel column was many times flat out wrong and didn't match the iax account which to call came in on. We set two test and I will explain both below. We have only seen this bahavior when the IAX accounts in question are coming from the same IP address. So for example a NAT setup with either multiple IAX softphones, multiple Asterisk servers or one Asterisk server with multiple IAX accounts to our server. Test 1: We setup two IAX softphones (Diax) behind a NAT. Each softphone had a totally different account code in their configuration (on the server side in iax.conf). Each phone registered using their respective usernames/secrets.. Results (Test 1): When calls were made each call was correctly inserted into the CDR with the correct channel account code etc. Test 2: We turned off the softphones mentioned above. We then setup a test Asterisk box with the same IAX accounts mentioned above. Each account registered to our server remotely just as the softphones did. We then made 10 phone calls with each account dialing totally different phone numbers so we could identify the records from each IAX account. Results (Test 2): We found that the majority of calls now had wrong/mismatched channels in the CDR. Basically we would show a channel like IAX2/user2 for a call that we are certain came from the IAX user user1. And... That's my story and I'm stickin' to it :) On 7/14/06, Matt Riddell (NZ) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 voiplist wrote: I wish it were that simple.. We see the username coming in, it's in the channel etc.. We see the call come into one account and we see * set an account code for another account.. Really.. It seems that it has something to do with the fact that accounts registering from the same IP get mixed up. Anyone else experience this? We are using 1.2.4 on this particular box. Are you sure you don't have an account with no password or something? - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuG+NS6d5vy0jeVcRAh/vAJ0cArD+Zs2fmKYmZZf+VumBVh0CUwCfRYa4 rywZhhYMlLFNWRCMc/4nrkw= =9gIG -END PGP SIGNATURE- ___ --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
Re: [asterisk-users] Re: Wrong account code from iax_buddies
On 7/15/06, Tim Panton [EMAIL PROTECTED] wrote: On 15 Jul 2006, at 17:24, voiplist wrote: After further testing, here is what we found.. The account code was actually right after all, what made us think it was incorrect was the fact that * was reporting the wrong channel for the call. Test 2: We turned off the softphones mentioned above. We then setup a test Asterisk box with the same IAX accounts mentioned above. Each account registered to our server remotely just as the softphones did. We then made 10 phone calls with each account dialing totally different phone numbers so we could identify the records from each IAX account. Results (Test 2): We found that the majority of calls now had wrong/mismatched channels in the CDR. Basically we would show a channel like IAX2/user2 for a call that we are certain came from the IAX user user1. And... That's my story and I'm stickin' to it :) Hmm, I'd like to know more about this, as it is likely to bite me too :-( Can you describe the setup of the 'originating' asterisk in your second test? If possible please share the relevant bits of extensions.conf and iax.conf. Thanks. Tim. Tim Panton [EMAIL PROTECTED] Sure it was TrixBox 1.1 running on a VmWare virtual machine (great for testing). We basically setup two Trunks as they are known is TrixBox ([EMAIL PROTECTED]), one with each of the IAX accounts we setup on the server end. Just to keep the confusion to a minimum, the TrixBox was the client which was sending the calls to the Asterisk Server which was terminating the calls to the PSTN. If you are familiar with TrixBox Trunk Configs I can send you the basic details of the settings in the GUI Trunk screen. ___ --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: Wrong account code from iax_buddies
Anyone have any thoughts on this? On 7/13/06, voiplist [EMAIL PROTECTED] wrote: We have a situation where the wrong account code is being passed from Asterisk to our AGI and then on into the accountcode field in the CDR. Here is the situation, best I can explain it.. We have 3 user records in the iax_buddies table which all come from the same IP address and possibly the same Asterisk server (client side). The accountcode field in the iax_buddies records look like this: name accountcode ipaddr user1 155112.223.225.114 user2 156112.223.225.114 user3 157112.223.225.114 When user1, user2 or user3 terminates a call through the * box the account code doesn't match the accountcode assigned to that user in iax_buddies most of the time. As far as we can tell it only gets mixed up with iax users coming from the same IP. We have lots of other records which show the correct account code on every call. We have searched around and tried for hours to understand how this is possible. All we can come up with is that Asterisk is somehow associating the IP with any of the users names sort of willy nilly regardless of the IAX user the call comes in as. Any help would be appreciated. ___ --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: Wrong account code from iax_buddies
I wish it were that simple.. We see the username coming in, it's in the channel etc.. We see the call come into one account and we see * set an account code for another account.. Really.. It seems that it has something to do with the fact that accounts registering from the same IP get mixed up. Anyone else experience this? We are using 1.2.4 on this particular box. On 7/14/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Sounds to me that the incoming call is providing the wrong userid/password. voiplist wrote: Anyone have any thoughts on this? On 7/13/06, voiplist [EMAIL PROTECTED] wrote: We have a situation where the wrong account code is being passed from Asterisk to our AGI and then on into the accountcode field in the CDR. Here is the situation, best I can explain it.. We have 3 user records in the iax_buddies table which all come from the same IP address and possibly the same Asterisk server (client side). The accountcode field in the iax_buddies records look like this: name accountcode ipaddr user1 155112.223.225.114 user2 156112.223.225.114 user3 157112.223.225.114 When user1, user2 or user3 terminates a call through the * box the account code doesn't match the accountcode assigned to that user in iax_buddies most of the time. As far as we can tell it only gets mixed up with iax users coming from the same IP. We have lots of other records which show the correct account code on every call. We have searched around and tried for hours to understand how this is possible. All we can come up with is that Asterisk is somehow associating the IP with any of the users names sort of willy nilly regardless of the IAX user the call comes in as. Any help would be appreciated. ___ --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 -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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
[asterisk-users] Wrong account code from iax_buddies
We have a situation where the wrong account code is being passed from Asterisk to our AGI and then on into the accountcode field in the CDR. Here is the situation, best I can explain it.. We have 3 user records in the iax_buddies table which all come from the same IP address and possibly the same Asterisk server (client side). The accountcode field in the iax_buddies records look like this: name accountcode ipaddr user1 155112.223.225.114 user2 156112.223.225.114 user3 157112.223.225.114 When user1, user2 or user3 terminates a call through the * box the account code doesn't match the accountcode assigned to that user in iax_buddies most of the time. As far as we can tell it only gets mixed up with iax users coming from the same IP. We have lots of other records which show the correct account code on every call. We have searched around and tried for hours to understand how this is possible. All we can come up with is that Asterisk is somehow associating the IP with any of the users names sort of willy nilly regardless of the IAX user the call comes in as. Any help would be appreciated. ___ --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] Mutiple Homes one asterisk box
Yes, you want to use different context for each house.
In your sip.conf:
[house1]
username=house1
secret=house1pass
context=house1
---Other sip options here---
[house2]
username=house2
secret=house2pass
context=house2
---Other sip options here---
In your extensions.conf:
[house1]
;House1 Local Calls out through pots line 1, replace 555 with your area code
exten = _1555NXX,1,Dial(Zap/1/${EXTEN},60,r)
;House1 Long distance calls out through a VoIP provider
exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
[house2]
;House2 Local Calls out through pots line 2, replace 555 with your area code
exten = _1555NXX,1,Dial(Zap/2/${EXTEN},60,r)
;House2 Long distance calls out through a VoIP provider
exten = _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
If you need the VoIP piece with reliable support http://www.VoIPstreet.com
Hope this helps.
On 7/10/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Jul 10, 2006, at 10:48 AM, Andrew Niemantsverdriet wrote:
Is that the standard way of doing things? I found a bunch of asterisk
hosting providers in my search on the best way to do this. Is this
what they are doing?
Yes,l I think that's what contexts are for... I am also relatively new
at this, and experimenting using the contexts for separate locations
and separate users. This works although it takes a moment to understand
it.
You can also use separate prepaid accounts for the VOIP long distance
calls...
I don't really see that separate trunking is needed in you case,
although I admit I a not clear on what he means by this... Since you
have such a small amount of traffic I don't see it as a big deal...
Marty
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[Asterisk-Users] More g729 calls than licenses?
What happens when/if your Asterisk server is asked to handled more g729 calls than it has licenses? Does it fall back to an alternate codec or does the call get rejected? ___ --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] More g729 calls than licenses?
Any way to monitor this? Send an email to admins? Something? On 7/4/06, Thomas Kenyon [EMAIL PROTECTED] wrote: voiplist wrote: What happens when/if your Asterisk server is asked to handled more g729 calls than it has licenses? Does it fall back to an alternate codec or does the call get rejected? Well IME you get around 15 notices a second in the console stating that you have run out of licenses and there is no sound in either direction for the caller/callee. ___ --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
[Asterisk-Users] Sipura SPA-2000 Asterisk 1.24 w/incoming calls
We have issues with all of the SPA-2000 ATAs we have where incoming calls from only one of our Asterisk servers do not complete. Details: 1- On the CLI we see that when the call is pushed to the ATA it shows Busy/Congested 2- We can make calls to this same server just fine 3- We can receive calls from other Asterisk servers running older CVS versions of Asterisk with the same exact ATA configuration on the same exact network, behind the same exact NAT routers. 4- We have tried putting the ATAs in front of the NAT routers using the DMZ setting, no help. 5-The ATAs do register just fine 6- This happens on multiple networks behind multiple different NAT routers 7- We have tried turning off the firewall on the server side temporarily 8-We have tried a Grandstream 486 at one of these locations and all is well with receiving incoming from this server Remember, these ATAs work fine currently at their current locations on their current networks, behind their current NAT routers when communicating with a different Asterisk server. Other ATAs in these same locations have no trouble receiving calls from the newer server. Any ideas? ___ --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] Sipura SPA-2000 Asterisk 1.24 w/incoming calls
Yes, I have limited access to one SPA-2000 at the moment. Anyone else seeing this? When you say open a bug on this do you mean with Asterisk or Sipura? I guess that's part of the problem, not sure if we should be troubleshooting on Asterisk or the Sipura device.. On 6/17/06, Rich Adamson [EMAIL PROTECTED] wrote: voiplist wrote: We have issues with all of the SPA-2000 ATAs we have where incoming calls from only one of our Asterisk servers do not complete. Details: 1- On the CLI we see that when the call is pushed to the ATA it shows Busy/Congested 2- We can make calls to this same server just fine 3- We can receive calls from other Asterisk servers running older CVS versions of Asterisk with the same exact ATA configuration on the same exact network, behind the same exact NAT routers. 4- We have tried putting the ATAs in front of the NAT routers using the DMZ setting, no help. 5-The ATAs do register just fine 6- This happens on multiple networks behind multiple different NAT routers 7- We have tried turning off the firewall on the server side temporarily 8-We have tried a Grandstream 486 at one of these locations and all is well with receiving incoming from this server Remember, these ATAs work fine currently at their current locations on their current networks, behind their current NAT routers when communicating with a different Asterisk server. Other ATAs in these same locations have no trouble receiving calls from the newer server. Interesting... I just ran into the same problem with no firewalls or nating involved. This one is an spa3k but with exactly the same issue. Had been working just fine, but now fails with SVN-branch-1.2-r34400. All other sip phones are functioning fine. An ethereal trace only indicates the spa3k is returning busy here. I don't see anything wrong with the initial sip packets, but didn't have the time to dig to deeply into it either. I also made sure DND, Call Forwarding, etc, was not the issue. Looks like we need to open a bug on this. I'll be out of town for the next week and won't have any access to the boxes for testing. R. ___ --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
[Asterisk-Users] Size limitations of extensions.conf
Can someone tell me the size (or any other) limitations for the extensions.conf? We have managed to keep our file pretty small thanks to AGI but we are about to setup a bunch of call restrictions based on area and country code. One line per area code in the US alone adds a LOT of text to this file. Is it a bad thing to have 5 or 6000 lines of text in your extensions.conf on a production system? Will it affect the performance? ___ --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] Size limitations of extensions.conf
So what are the smart folks doing when it comes to retricting/allowing which area/country codes can and can't be called? AGI? We can go AGI but we are trying to avoid yet more calls to AGI apps for obvious reasons. So, is it smarter to use AGI or have it in the text file? Thanks.. On 6/3/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Sat, 2006-06-03 at 16:03 -0500, voiplist wrote: Can someone tell me the size (or any other) limitations for the extensions.conf? We have managed to keep our file pretty small thanks to AGI but we are about to setup a bunch of call restrictions based on area and country code. One line per area code in the US alone adds a LOT of text to this file. Is it a bad thing to have 5 or 6000 lines of text in your extensions.conf on a production system? Will it affect the performance? it adds memory and increases load time, it also causes asterisk to walk a longer tree each time it has to do something in that context at least rather than not ... -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQBEggCT+1olxlzQw5cRAqmvAJwOCa9atTESuky3rxvE9H9+gexqXwCfWdf8 UoVBXbLKIPOL1TbXuFCvlo0= =/Ar4 -END PGP SIGNATURE- ___ --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
[Asterisk-Users] All non US 48 area codes?
Is there a list somewhere or a way to find the following: 1- All non US 48 area codes which can be dialed as 1+10 2- All strange area codes which are used for premium services such as 900-XXX- 3- Anything else that should be restricted if one was to restrict all calls to US 48 only I have found many list but it's tough looking at the entire list of area codes and pulling out each of them one at a time. Thanks! ___ --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] All non US 48 area codes?
Thanks for the reply, I was going in the direction you mentioned of denying all and allowing specific area codes etc.. I was really trying to cut down on the amount of data I had to mess with and store. I just thought I would hit the list with this since it seems many would have come across the same problem already. Thanks again! On 6/2/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Fri, 2006-06-02 at 18:37 -0500, voiplist wrote: Is there a list somewhere or a way to find the following: 1- All non US 48 area codes which can be dialed as 1+10 2- All strange area codes which are used for premium services such as 900-XXX- 3- Anything else that should be restricted if one was to restrict all calls to US 48 only The best way is to restric everything then allow what you know to be good. If you try to take the approach of allowing everything except what is denied then you are in effect saying you know everything bad and can accurately restrict it. This often proves to be a bad decision. There is a listing at astbill.com that I started and presumably they have maintained since I basically abandoned it. It has US as well as other countries in there. I know for a fact that its not complete on the global numbers, but believe it to be reasonably complete for NANPA numbers (although some wireless providers are listed as geographic, but that shouldnt affect this application). NANPA assigns the numbers, you can goto them for the information (they have lists of all assigned codes, which carrier they are assigned to and what geographic region they are from). http://www.nanpa.com/reports/reports_cocodes_assign.html That list is all the assignments from NANPA. It would be your best bet since they are much more authoritative on this than my list. I would be wary of any codes that have X for their rate center, as those may be premium since they arent assigned to a specific rate center. They may be something else as well, however once a NPA-NXX is assigned to a rate center it cannot change rate centers, so if it isnt listed odds are its some special premium service number. As for premium numbers, yes 900 is a NPA used for premium but there are local ones, while typically 976- that isnt always the case and some numbers, are assigned to lcoal carriers and have no rate center listed and can be a pay per call service (such as some in NJ that now verizon owns). If your numbers are in LIDB that should prevent most if not all billing from occuring on those numbers, and the call to those numbers will be rejected. Your carrier should be able to insert your numbers into LIDB for you. The carrier may not know LIDB by name (or at least the sales drone you talk to) but they should be familiar with call blocking for pay per call services. hope this answers your question :) -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQBEgNIE+1olxlzQw5cRAmgTAJ41QcgrpFGp4KQU2YngdauwgJMHagCfbqn5 pyPKDXzqjwHvED0bVV3nM7w= =pNzJ -END PGP SIGNATURE- ___ --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
