Re: [asterisk-users] Paging to external speaker like in airports etc...
Thats AMAZING! This google you have shown me is truly a modern marvel
of the interwebs.
You know what would be EVEN BETTER though?
If idiots (such as you and I) would find something better to do with our
time than mock others on mailing lists in a pitiful attempt to appear
more knowledgeable/cool/hip/popular/what have you.
Nobody likes you or thinks you're pretty or will ask you to prom.
Welcome to the asshole club, we get badges... and black eyes.
Lacy Moore - Aspendora wrote:
On 9/13/07, *Deepak Naidu* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi, I have a production asterisk-1.2.8 system with FreePBX PRI
Digium card.
I am looking for a paging system to an external speaker. I can
page to internal Polycom 501 VoIP.
But, what hardware or system do I need to integrate with the
asterisk to have this acheived.
You know what would be even better? If we had a search engine that
you could type something into and it would produce a list of pages
related to this.
Oh wait, maybe that's what this does:
http://www.google.com/search?hl=enq=Asterisk+paging
http://www.google.com/search?hl=enq=Asterisk+paging
Google is a wonderful tool, learn to use it...
--
Deepak
*Linux your Life,** Don't Window it [[]] *
*{ All for the best }*
Yahoo! Answers - Get better answers from someone who knows. Try it
now
http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc2VjA21haWwEc2xrA3RhZ2xpbmU.
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Re: [asterisk-users] Feedback on Linksys SPA-921 and GrandStream GXP-2000
On the GXP-2000 press the Mute/DEL button while the phone is ringing, and it will return 486 (Busy). This works to bounce new incoming calls while already in a call as well (call waiting). - Anthony Kepler Andrew Joakimsen wrote: My main complaint about both phones is there is no way to reject a call once the phone starts to ring. ___ --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 1.4.0-beta3 spandsp rxfax woes (or me being hard of thinking)
Awww... This is when I feel stupid, and for the sake of others... I will expose my shame: Be sure you run `autoconf` after applying the patch (and making the required changes to configure.ac) Since it's altering configure.ac afterall, and not configure; then of course run configure and etc. I did so and now it works... except I now have the disappointment of realizing that ulaw over SIP isn't really all that well suited to fax. (hey... 1/3 of a page is better than nothing... r-right?) Anthony Kepler wrote: Did you ever find a solution for this? I'm in the same boat with 1.4.0-beta3 and SpanDSP - Anthony Kepler Matt Gibson wrote: Okay, So, More updates after testing some more 1. with the free line commented out of app_rxfax.c, and recompiled, asterisk seems to work on non-fax incoming calls to my fax extension. Doesn't send a file obviously, but does seem to actually reach the right place and do what it's supposed to do. ___ --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 1.4.0-beta3 spandsp rxfax woes (or me being hard of thinking)
Did you ever find a solution for this? I'm in the same boat with 1.4.0-beta3 and SpanDSP - Anthony Kepler Matt Gibson wrote: Okay, So, More updates after testing some more 1. with the free line commented out of app_rxfax.c, and recompiled, asterisk seems to work on non-fax incoming calls to my fax extension. Doesn't send a file obviously, but does seem to actually reach the right place and do what it's supposed to do. 2. Here is all the debug output I have while trying to get this to receive the faxes: Error #1 : -- *** glibc detected *** asterisk: double free or corruption (out): 0x0820ae98 *** === Backtrace: = /lib/libc.so.6[0xb72e8d20] /lib/libc.so.6(__libc_free+0x84)[0xb72ea364] /lib/libc.so.6(closedir+0x28)[0xb730bf48] /usr/lib/asterisk/modules/app_voicemail.so[0xb66d1ef5] /usr/lib/asterisk/modules/app_voicemail.so[0xb66d26a4] /usr/lib/asterisk/modules/chan_zap.so[0xb697304e] asterisk[0x80ecb99] /lib/libpthread.so.0[0xb7e6b294] /lib/libc.so.6(__clone+0x5e)[0xb7341c9e] === Memory map: 08048000-0813b000 r-xp 03:03 2474075/usr/sbin/asterisk 0813b000-08148000 rw-p 000f3000 03:03 2474075/usr/sbin/asterisk 08148000-0822b000 rw-p 08148000 00:00 0 [heap] b5e0-b5e21000 rw-p b5e0 00:00 0 b5e21000-b5f0 ---p b5e21000 00:00 0 b5fc1000-b5fc2000 ---p b5fc1000 00:00 0 b5fc2000-b5ffd000 rwxp b5fc2000 00:00 0 b602e000-b602f000 ---p b602e000 00:00 0 b602f000-b606a000 rwxp b602f000 00:00 0 b606a000-b609b000 r-xp 03:03 889758 /usr/lib/libcurl.so.3.0.0 b609b000-b609c000 rw-p 00031000 03:03 889758 /usr/lib/libcurl.so.3.0.0 b60a4000-b60a8000 r-xp 03:03 1066578/usr/lib/libogg.so.0.5.2 b60a8000-b60a9000 rw-p 3000 03:03 1066578/usr/lib/libogg.so.0.5.2 b60a9000-b60b4000 r-xp 03:03 1130913 /usr/lib/libvorbisenc.so.2.0.2 b60b4000-b61a3000 rw-p a000 03:03 1130913 /usr/lib/libvorbisenc.so.2.0.2 b61a3000-b61a5000 rw-p b61a3000 00:00 0 b61a5000-b61bf000 r-xp 03:03 1132171 /usr/lib/libvorbis.so.0.3.1 b61bf000-b61cd000 rw-p 0001a000 03:03 1132171 /usr/lib/libvorbis.so.0.3.1 b61d2000-b61d4000 r-xp 03:03 2376646 /usr/lib/asterisk/modules/func_curl.so b61d4000-b61d5000 rw-p 1000 03:03 2376646 /usr/lib/asterisk/modules/func_curl.so b61d5000-b61d6000 ---p b61d5000 00:00 0 b61d6000-b6211000 rwxp b61d6000 00:00 0 b6211000-b6212000 ---p b6211000 00:00 0 b6212000-b6252000 rwxp b6212000 00:00 0 b6252000-b6253000 ---p b6252000 00:00 0 b6253000-b6293000 rwxp b6253000 00:00 0 b6293000-b629b000 r-xp 03:03 1002454/lib/libnss_nis-2.4.so b629b000-b629d000 rw-p 7000 03:03 1002454/lib/libnss_nis-2.4.so b629d000-b62a3000 r-xp 03:03 1002470 /lib/libnss_compat-2.4.so b62a3000-b62a5000 rw-p 5000 03:03 1002470 /lib/libnss_compat-2.4.so b62a9000-b62ac000 r-xp 03:03 2376631 /usr/lib/asterisk/modules/format_ogg_vorbis.so b62ac000-b62ad000 rw-p 2000 03:03 2376631 /usr/lib/asterisk/modules/format_ogg_vorbis.so b62ad000-b62b r-xp 03:03 1343898 /usr/lib/pwlib/codecs/audio/g726_audio_pwplugin.so b62b-b62b1000 rw-p 3000 03:03 1343898 /usr/lib/pwlib/codecs/audio/g726_audio_pwplugin.so b62b1000-b62b9000 r-xp 03:03 1343897 /usr/lib/pwlib/codecs/audio/lpc10_audio_pwplugin.so b62b9000-b62ba000 rw-p 8000 03:03 1343897 /usr/lib/pwlib/codecs/audio/lpc10_audio_pwplugin.so b62ba000-b62c7000 r-xp 03:03 1343896 /usr/lib/pwlib/codecs/audio/ilbc_audio_pwplugin.so b62c7000-b62ca000 rw-p c000 03:03 1343896 /usr/lib/pwlib/codecs/audio/ilbc_audio_pwplugin.so b62ca000-b62e4000 r-xp 03:03 1343895 /usr/lib/pwlib/codecs/audio/speex_audio_pwplugin.so b62e4000-b62e9000 rw-p 0001a000 03:03 1343895 /usr/lib/pwlib/codecs/audio/speex_audio_pwplugin.so b62e9000-b62f2000 r-xp 03:03 1343907 /usr/lib/pwlib/codecs/audio/gsm0610_audio_pwplugin.so b62f2000-b62f3000 rw-p 8000 03:03 1343907 /usr/lib/pwlib/codecs/audio/gsm0610_audio_pwplugin.so b62f3000-b633d000 r-xp 03:03 2376522 /usr/lib/asterisk/modules/chan_h323.so b633d000-b6341000 rw-p 00049000 03:03 2376522 /usr/lib/asterisk/modules/chan_h323.so b6341000-b639b000 r-xp 03:03 3244836 /opt/swift/lib/libceplex_us.so.4.1 b639b000-b639e000 rw-p 00059000 03:03 3244836 /opt/swift/lib/libceplex_us.so.4.1 b639e000-b63ba000 r-xp 03:03 3244833 /opt/swift/lib/libceplang_en.so.4.1 b63ba000-b63c rw-p 0001b000 03:03 3244833 /opt/swift/lib/libceplang_en.so.4.1 b63c-b643c000 r-xp 03:03 3244839 /opt/swift/lib/libswift.so.4.1 b643c000-b6443000 rw-p 0007b000 03:03 3244839 /opt/swift/lib/libswift.so.4.1 b6443000-b6446000 r-xp 03:03 2376595 /usr/lib/asterisk/modules/app_txfax.so b6446000-b6447000 rw-p 2000 03:03 2376595 /usr/lib/asterisk/modules/app_txfax.so b6447000-b644a000 r-xp 03:03 2376580 /usr/lib/asterisk/modules/app_rxfax.so b644a000-b644b000 rw-p 2000 03:03 2376580 /usr/lib
Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
I have been using an approach such as this but am looking for something
else because of some limitations it has. The phone thinks it dialed,
and was connected to 011 (which it was)
As such, that will be stored in the phones dial history (redial if
nothing else).
I'm not even certain what I want is possible, which is why I'm asking
the list.
Thank you for your help once again though.
- Anthony Kepler
[EMAIL PROTECTED] | SIP/EMail
Doug Crompton wrote:
Well that is certainly an option but not all phones would have a send key
especially if you are using analog phones. I guess the # keys
functions in
that way on many of those.
I still like my wired phones to work like they use to. You dial a
number
and it executes the call immediately.
Ok I came up with one that I think would work, maybe needs some
refinement
[out-international]
exten = _011,1,goto(process-international,s,1)
[process-international]
exten = s,1,read(number)
exten = s,2,Dial(SIP/[EMAIL PROTECTED],120,T)
exten = s,3,Macro(failann,${DIALSTATUS})
This accepts the 011 prefix and then any number of following digits.
Terminator is timeout period OR # key to send. Change obviously for your
provider.
The read command has many options including saying a file. You could for
instance hear Country Code after dialing 011. This would clue you into
the fact that you were dialing and international call. There are also
digit limits and timeouts that can be set.
So if you use early dial this would be the only rule that would require a
wait or # key to send. I could certainly live with that.
Can anyone supply some international test numbers??? Say in the UK or
Germany or wherever outside the US.
Doug
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
Do you, Gordon or Doug, happen to place international calls with
early-dial enabled? What kind of extensions.conf magic do you work to
allow this?
I have been trying for some time to get this to work. (My message from
2006.11.03 regarding this is quoted just below)
On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:I am trying to allow users to
place outgoing international calls from a
GXP-2000 with early dial enabled, connected to Asterisk 1.2.12.1
http://1.2.12.1
I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
When I attempt to place a call to a number in, for instance, Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new stack
-- Called g1/0112
It is attempting to dial out as soon as it receives a single digit to
represent the .
What I need is for it to wait a reasonable amount of time for additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following in the
asterisk console:
-- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5) in new stack
-- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.
I assume there must be something relatively obvious I'm missing here...
if anyone can shed some light on this, it would be greatly appreciated.
Thank you,
- Anthony Kepler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] | SIP/Email
Gordon Henderson wrote:
On Sun, 5 Nov 2006, Doug Crompton wrote:
On the Budgetone 200 it is in the account tab settings of the web setup
and it does work here with asterisk and my dialplans..
On the GPX2000's it's via the web interface under each of the 4 Line
configuration tabs. (so you'd have to set it on each account you were
using on the phone)
Gordon
Doug
On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:
Hi,
Where can I find that option?
Thanks
Jesus
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Gordon
Henderson
Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
On Wed, 1 Nov 2006, Henry.L.Coleman wrote:
I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that when
dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the pattern is
match ed so it saves a second or two. Maybe they will fix this?
Set the Early Dial option - it's on a per-line basis, then as soon
as Asterisk gets a number it can dial, it will. No need to wait the 4
seconds or press the send button...
Gordon
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
I understand how early dial works (484 response and all that jazz), I
also understand the NANP and how to keep my extensions from
overlapping... but thank you for the tips.
My question was: Do you place international calls from phones with
early-dial enabled?
If so, might you be willing to share the relevant portions of your dial
plan that are concerned with placing said international calls?
Thanks again,
- Anthony Kepler
[EMAIL PROTECTED] | SIP/Email
Doug Crompton wrote:
Early dial is a real nice feature BUT it requires that you carefully plan
and design your extensions. Each digit is accepeted by Asterisk and if a
match exists up to that point it will be accepted and dialed.
As an example, my internal extensions are 4xx and my internal special
extensions are 5xx. I chose those because they do not conflict with local
area codes or other first 3 digit sequences.
However if a call come in from, say, area code 512 (without the 1
prepended), and I have a local 512 extension, I would not be able to dial
that person back. It would instead go to the local 512, as this is
satisfied first.
Often callerID does not come in with the 1 before the area code. This is
what prompted me to put code in to append a 1 if none existed on the
incoming callerID. With the 1 appended there is no problem as 151 does not
match any local extension and I can use redial without problems.
Using 4 digit extensions would mostly eliminate this problem although you
still could not use 1xxx extensions.
Wildcard extension matches like X. or using the '.' anywhere in the
matches would not work.
You just have to use it and fix things as they come up. I think I have
most all cases trapped now!
Doug
On Tue, 19 Dec 2006, Anthony Kepler wrote:
Do you, Gordon or Doug, happen to place international calls with
early-dial enabled? What kind of extensions.conf magic do you work to
allow this?
I have been trying for some time to get this to work. (My message from
2006.11.03 regarding this is quoted just below)
On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:I am trying to allow users to
place outgoing international calls from a
GXP-2000 with early dial enabled, connected to Asterisk 1.2.12.1
http://1.2.12.1
I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
When I attempt to place a call to a number in, for instance, Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new stack
-- Called g1/0112
It is attempting to dial out as soon as it receives a single digit to
represent the .
What I need is for it to wait a reasonable amount of time for additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following in the
asterisk console:
-- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5) in new stack
-- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.
I assume there must be something relatively obvious I'm missing here...
if anyone can shed some light on this, it would be greatly appreciated.
Thank you,
- Anthony Kepler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] | SIP/Email
Gordon Henderson wrote:
On Sun, 5 Nov 2006, Doug Crompton wrote:
On the Budgetone 200 it is in the account tab settings of the web setup
and it does work here with asterisk and my dialplans..
On the GPX2000's it's via the web interface under each of the 4 Line
configuration tabs. (so you'd have to set it on each account you were
using on the phone)
Gordon
Doug
On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:
Hi,
Where can I find that option?
Thanks
Jesus
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Gordon
Henderson
Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
On Wed, 1 Nov 2006, Henry.L.Coleman wrote:
I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that when
dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the pattern is
match ed so it saves a second or two. Maybe they will fix this?
Set the Early Dial option - it's on a per-line basis, then as soon
as Asterisk gets a number it can dial, it will. No need to wait the 4
seconds or press the send button...
Gordon
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
I am located on the west coast of the united states. In order to dial an international number from within the US, we must first dial the special international access code that tells the PSTN the following call is an international one - in the US that is 011, followed by the country code, and then the actual number for our destination within that country. (which would include whatever their concept of area code, prefix, and destination number are - which varies widely from country to country) If you're generally interested in this, then you might find the following reading interesting as well: http://en.wikipedia.org/wiki/North_American_Numbering_Plan and http://en.wikipedia.org/wiki/Area_code - Anthony Kepler [EMAIL PROTECTED] | SIP/Email Doug Crompton wrote: Sorry, I did not read the original message completely. The answer is no I do not make international calls. I do not know anyone in any other country to call! I do not have a rule for that but it should be easy to implement as 01x would not match anything I currently have for early dial. Would you always dial a 0 first for all international mumbers? Give me an example? Are you outside the US? If so give me your number and I will try it! Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International dialing with GPX-2000 and early dial
Early dial is a feature on the phone that makes use of the 484 (Address
Incomplete) response.
This is desired for in-office, local (PSTN), and long distance dialing.
I'm really hoping to find a best-of-both-worlds solution to this.
Andrew Joakimsen wrote:
Does the GXP-2000 not have its own dialplan? Use that and disable
early dial
On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I am trying to allow users to place outgoing international calls
from a
GPX-2000 with early dial enabled, connected to Asterisk 1.2.12.1
http://1.2.12.1
I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
When I attempt to place a call to a number in, for instance, Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new stack
-- Called g1/0112
It is attempting to dial out as soon as it receives a single digit to
represent the .
What I need is for it to wait a reasonable amount of time for
additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following
in the
asterisk console:
-- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5) in
new stack
-- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.
I assume there must be something relatively obvious I'm missing
here...
if anyone can shed some light on this, it would be greatly
appreciated.
Thank you,
- Anthony Kepler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] |
SIP/Email
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[asterisk-users] International dialing with GPX-2000 and early dial
I am trying to allow users to place outgoing international calls from a
GPX-2000 with early dial enabled, connected to Asterisk 1.2.12.1
I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
When I attempt to place a call to a number in, for instance, Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new stack
-- Called g1/0112
It is attempting to dial out as soon as it receives a single digit to
represent the .
What I need is for it to wait a reasonable amount of time for additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following in the
asterisk console:
-- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5) in new stack
-- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.
I assume there must be something relatively obvious I'm missing here...
if anyone can shed some light on this, it would be greatly appreciated.
Thank you,
- Anthony Kepler
[EMAIL PROTECTED] | SIP/Email
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