Re: [asterisk-users] Soundfiles adding during phone calls
Ronald Wiplinger wrote:
Ronald Wiplinger wrote:
Tom Lynn wrote:
Ron,
The guy is trying to help you. Go to the link and read it. There is
a feature that you can use to play a recording into the voice
channel. Mine is set so when you press #9, the caller hears the
lots of monkeys recording.
The best part of it is that you can hang up and the recording will
continue to play to the caller. When it expires, so does the call
I tried this:
features.conf
[featuremap]
blindxfer = ##; Blind transfer was #1 - now press # twice
disconnect = *0; Disconnect
automon = *1; One Touch Record
atxfer = *2; Attended transfer
[applicationmap]
tortore= *9,callee,Playback,tt-monkeys
extensions.conf
exten = 601,1,Set(DYNAMIC_FEATURES=hangup#play#tortore#automon) ;
enable One-touch
exten = 601,2,Dial(${PHONE_601},30,tTwWr)
I make a call from 615 to 601
601 hits *9 but nothing happens!
when 601 hits *1 it records the conversion.
vpbx*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8Blind Transfer
# ##Attended Transfer *2One Touch
Monitor *1Disconnect Call * *0
Dynamic Feature Default Current
--- --- ---
(none)
Call parking
Parking extension : 750
Parking context : parkedcalls
Parked call extensions: 751-770
I added already in extenions.conf:
include = featuremap
bye
Ronald Wiplinger
What do I miss?
bye
Ronald Wiplinger
On 11/11/06, * Ronald Wiplinger* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Andrew Joakimsen wrote:
http://www.voip-info.org/wiki-Asterisk+config+features.conf
... and where exactly did you see this feature
bye
Ronald Wiplinger
On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
I want to add some sound filed on demand during a phone call
only
possible on some extension numbers.
I get many phone calls from local companies, but don't
understand
Chinese! I would like to record the call, but also ask the
caller some
questions, which should be added into the call with some
keys on the
phone, ... e.g. *66554 should add into the call: How are
you? or What
is your phone number?
But I do have another application for that too.
I get many fake phone calls, where Chinese people tell you
that your
phone bill is not paid, your court fee is not paid, and
ask the
caller to go to the ATM machine and key in a series of key
strokes,
most likely it will clear out your account.
For such fake callers I would like to add a terrible noise
to the
call
and make scare them as much as possible.
Such fake calls I get now for each of my phone lines at
least 10
each!!!
Either the caller-id is not set, is 0 or is a tollfree number.
bye
Ronald Wiplinger
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Bails
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[asterisk-users] Re: unable to get channel lock BAD BAD BAD
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Deadlocks are not a config or Trixbox issue. I'm confirming this one! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] queue management
Don't forget to analyze the QUEUESTATUS variable. :) -- i've just set joinempty=no and then in extension.conf : exten = 2701,1,Queue(2701|t|||10) exten = 2701,2,Background(orario_2701) Thanks to all ___ --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] sip forward behind a nat
On 11/14/06, Vicky [EMAIL PROTECTED] wrote: One more thing i would like to point out is that softphones like sjphone use some freeware stun server to detect nat on network (as a client ) . Asterisk ( as client ) cannot use external stun server to detect nat type automatically so i think thats why it isnt able to make calls while softphone works . Port forwarding should help here .Also edit sip_nat.conf after port forwarding but it will be a burden to setup if asterisk is on dynamic ip . Ok, i've resolved editing sip.conf and adding: externip= my_public_ip localnet=192.168.100.0/255.255.255.0 and the opening the port 5060 on the firewall. Many thanks to all ___ --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] Siemens Gigaset SL75
Hi, Has anyone tested Siemens Gigaset SL75 with Asterisk ? How would you rate its performances ? Cheers ___ --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] Nokia E70
Michiel van Baak a écrit : Hi, Hello Anyone here has any experience with the Nokia E70 and asterisk ? I read on the nokia website this phone is capable of talking SIP and do Presence based on SIP/SIMPLE. Please share your experience, I'm thinking of getting one but want to be sure I can use it with * before I do. For me E70 is at this moment the most viable solution. But it depend on firmware version you have -here in Europe-: if it's a 05/2006 one, registration and others SIP stuff, upgrade to the last one which is dated from 08/2006. This one seems much more stable. If you can see the revision by pressing *## -- Daniel ___ --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] splitting a PRI using TDMoE
Hi, I have an application where we need to route several channels of a PRI going into the Digium TE110P card on one Asterisk server to another Asterisk server. I am thinking that this might be possible by using TDMoE and somehow bridge the corresponding channels from the Zap channels of the TE110P to the Zap channels of the TDMoE. One requirement is that the secondary server should be able to receive all the PRI signals about the channels that is being routed over like busy, connect, etc. Is this possible? I cannot find any documentation or mentioning about this from the config files of the zaptel drivers. Thanks. King ___ --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: T.38 - make conclusion
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I want to guess that it's your SIP provider. Faxing via VoIP (SIP) is not reliable unless you are using T38, my guess is your sip provider is providing this feature. Hi Doug! Have I understand it right. You are saying that my provider, when it detects FAX stream, he is trying to use T.38. And since my Asterisk doesn't support it I get the error message? So, it does nothing to do with FAX machine that is (over ATA) plugged to my Asterisk, or with FAX that is on the other side? Please confirm or deny this. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] dialplan * and 0 key detection, not working
Eric ManxPower Wieling[EMAIL PROTECTED] Wrote on: 11/16/2006 7:36 PM: Special extensions like a, o, i, etc do not seem to be read from include = 'ed contexts. Is this a bug or as designed? In this case, it is not an include. The i seems to work fine. From an off list discussion, it appears that a and o only work within the Voicemail function. ?? Since the testing I am doing is outside that function, am I forced to re-configure my dialplan? joe a. ___ --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] Problem with Asterisk 1.4.0-beta3 and Digium TE405P
Hope this is the right place to report/ask for help...
Have have a 1.2.7.1 installation running reasonably happily for a while.
Thought we might give 1.4.0b3 a go. Ran it on a local test machine (that
has the single port card) and all was well. However, when I run it on a
machine with the 4-port card, it has trouble with Zaptel. I attach some
diagnostics and welcome any suggestions (including requests for more
information). We're running this on a Del SC430 with Ubuntu 5.10.
Thanks!
Alex
-
[EMAIL PROTECTED]:/etc/modprobe.d# /etc/init.d/asterisk start
Starting Asterisk PBX: Notice: Configuration file is /etc/zaptel.conf
line 22: Unable to read Zaptel version information.
Zaptel Version: ôoô·´vô·ÿwHVä¿Èó·Vä¿`vô·
Echo Canceller:
Configuration
==
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
124 channels configured.
ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)
[EMAIL PROTECTED]:/etc/modprobe.d# strace ztcfg
execve(/sbin/ztcfg, [ztcfg], [/* 23 vars */]) = 0
uname({sys=Linux, node=m900a, ...}) = 0
brk(0) = 0x80a2000
access(/etc/ld.so.nohwcap, F_OK) = -1 ENOENT (No such file or
directory)
old_mmap(NULL, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0xb7fa6000
access(/etc/ld.so.preload, R_OK) = -1 ENOENT (No such file or
directory)
open(/etc/ld.so.cache, O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=40467, ...}) = 0
old_mmap(NULL, 40467, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f9c000
close(3)= 0
access(/etc/ld.so.nohwcap, F_OK) = -1 ENOENT (No such file or
directory)
open(/lib/tls/i686/cmov/libm.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0`3\0\000...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0644, st_size=136976, ...}) = 0
old_mmap(NULL, 139424, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE,
3, 0) = 0xb7f79000
old_mmap(0xb7f9a000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0xb7f9a000
close(3)= 0
access(/etc/ld.so.nohwcap, F_OK) = -1 ENOENT (No such file or
directory)
open(/lib/tls/i686/cmov/libc.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\260O\1...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0644, st_size=1229936, ...}) = 0
old_mmap(NULL, 1236124, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE,
3, 0) = 0xb7e4b000
old_mmap(0xb7f73000, 16384, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x128000) = 0xb7f73000
old_mmap(0xb7f77000, 7324, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0xb7f77000
close(3)= 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0xb7e4a000
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0xb7e49000
mprotect(0xb7f73000, 4096, PROT_READ) = 0
set_thread_area({entry_number:-1 - 6, base_addr:0xb7e496c0,
limit:1048575, seg_32bit:1, contents:0, read_exec_only:0,
limit_in_pages:1, seg_not_present:0, useable
:1}) = 0
munmap(0xb7f9c000, 40467) = 0
open(/dev/zap/ctl, O_RDWR)= 3
brk(0) = 0x80a2000
brk(0x80c3000) = 0x80c3000
open(/etc/zaptel.conf, O_RDONLY) = 4
fstat64(4, {st_mode=S_IFREG|0744, st_size=256, ...}) = 0
mmap2(NULL, 131072, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1,
0) = 0xb7e29000
read(4, span=1,1,0,ccs,hdb3\nbchan=1-15\nd..., 131072) = 256
read(4, , 131072) = 0
close(4)= 0
munmap(0xb7e29000, 131072) = 0
ioctl(3, 0x40244a12, 0x80a0a60) = 0
ioctl(3, 0x40244a12, 0x80a0a84) = 0
ioctl(3, 0x40244a12, 0x80a0aa8) = 0
ioctl(3, 0x40244a12, 0x80a0acc) = 0
ioctl(3, 0x80844a05, 0xbfcbaaec)= 0
ioctl(3, 0x404c4a13, 0x808daac) = -1 ENOTTY (Inappropriate ioctl
for device)
write(2, ZT_CHANCONFIG failed on channel ..., 71ZT_CHANCONFIG failed
on channel 1: Inappropriate ioctl for device (25)
) = 71
close(3)= 0
exit_group(1) = ?
---
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
span=2,2,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
span=3,0,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,0,0,ccs,hdb3
bchan=94-108
dchan=109
bchan=110-124
loadzone=uk
defaultzone=uk
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Re: [asterisk-users] spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
A little progress on this problem
Examining the logs i found a weird looking 'soft hangup' which
reminded me on an earlier issue we had. (and the reason why we were
still on the 'i' release of bristuff). It looked as if the channel
hung up just before rxfax actually could begin to work.
Normally we would let the faxdetection be automatic and let ast jump
to the 'fax' extension, which in turn calls the faxreceive macro as
described in my original post.
Bypassing all that and directly tie-ing an extension to the
faxreceive macro (not even answering the channel), gives a successful
fax reception. RxFax has a check whether the channel is answered, and
it will answer it, if not already.
I think this was a lucky shot in the dark though. The problem seems
to be a mismatch between the state of the channel (answered) and what
it actually is. Getting data from the channel wont work then i guess.
This is a suitable workaround for our little setup for now. The only
thing we miss at this point is that we wont be able to receive faxes
at every extension anymore, just the one.
I do not have enough knowledge of the sources to suggest a fix for
this. It looks like either the specific stuff for our card (quadbri)
or asterisk itself would be the area to look into, but again, i am
not (yet) capable of doing so myself.
Hope this helps anyone fixing the real problem.
marcel
On 16 nov 2006, at 11:27, Marcel van der Boom wrote:
Hi,
I'm using spandsp-0.0.3
[http://www.soft-switch.org/downloads/snapshots/spandsp/
spandsp-20061116.tar.gz]
on a bristuffed asterisk (1.2.13)
[http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0-
PRE-1v.tar.gz]
libtiff is at version 3.6.0
Running on: Linux router2 2.6.17-2-686 #1 SMP Wed Sep 13 16:34:10
UTC 2006 i686 GNU/Linux
Debian testing distro.
I've tried many combinations of bristuffed ast and spandsp
versions, but all fail at the same point. The last combination i
got to work was bristuffed 0.3.0-PRE-1i with spandsp-0.0.2-pre25
(on an earlier kernel)
The app_rxfax.c in use is from:
[http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-
asterisk-1.2/app_rxfax.c]
On reception of a fax through RxFax, i get the exception. The
relevant part of the dialplan is
[macro-faxreceive]
; Receive a fax
exten = s,1,Set(FAXFILE=${FAXSPOOL}/${UNIQUEID}.tif) ; Save the
fax in a tif file
exten = s,2,RxFAX(${FAXFILE}) ; Receive it
exten = s,3,NoOp(Fax reception complete) ;
exten = s,4,Hangup
Running asterisk (with the above versions) through gdb and doing a
backtrace gives me:
#0 0xa7d45947 in raise () from /lib/tls/libc.so.6
#1 0xa7d470c9 in abort () from /lib/tls/libc.so.6
#2 0xa7d7afda in __fsetlocking () from /lib/tls/libc.so.6
#3 0xa7d8289f in mallopt () from /lib/tls/libc.so .6
#4 0xa7d82942 in free () from /lib/tls/libc.so.6
#5 0xa75efd68 in rxfax_exec (chan=0x818c5f8, data=0xa74a4798) at
app_rxfax.c:327
#6 0x08090088 in pbx_extension_helper (c=0x818c5f8, con=value
optimized out, context=value optimized out, exten=0x818c83c s,
priority=2,
label=0x0, callerid=0x0, action=1) at pbx.c:554
#7 0xa762cb05 in macro_exec (chan=0x818c5f8, data=0xa74aafe8) at
app_macro.c:221
#8 0x08090088 in pbx_extension_helper (c=0x818c5f8, con=value
optimized out, context=value optimized out, exten=0x818c83c s,
priority=1,
label=0x0, callerid=0x0, action=1) at pbx.c:554
#9 0x08091dee in __ast_pbx_run (c=0x818c5f8) at pbx.c:2231
#10 0x08092a1c in pbx_thread (data=0x818c5f8) at pbx.c:2518
#11 0xa7f0d0bd in start_thread () from /lib/tls/libpthread.so.0
#12 0xa7de892e in clone () from /lib/tls/libc.so.6
This seems to indicate that the ast_frfree(inf); at line 327 of
app_rxfax.c causes the problem chain?
I'm a bit lost on how to debug this further. Is this actually a
spandsp problem or is another package the cause?
Any tips?
marcel
--
Marcel van der Boom
HS-Development BV -- http://www.hsdev.com
So! webapplicatie framework -- http://make-it-so.info
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Marcel van der Boom
HS-Development BV -- http://www.hsdev.com
So! webapplicatie framework -- http://make-it-so.info
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[asterisk-users] Need help on Music on Hold
I am testing asterisk (version 1.2.12.1) on a Dell 1950 server and have this strange problem on music on hold. When I called into a queue using SIP from PSTN line which goes through our cisco gateway (cisco 5300), asterisk will start play music on hold. But this MOH seems at voice activation mode. That is only when I make noice on my end then I can hear music otherwise I will hear silence. I have another asterisk (version 1.2.9.1) running on an older Dell server and MOH works fine for call from PSTN. So my guess is that maybe there is some settings in asterisk cause this problem. Any suggestion about this problem? GG ___ --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] Need help on Music on Hold
Hi, Do you have vad disabled in your dial-peer voice XX voip dial-peer? What kind of MOH are you using; asterisk native or an external player like mpg123? --basv On Fri, Nov 17, 2006 at 08:41:49AM -0500, gc wrote: I am testing asterisk (version 1.2.12.1) on a Dell 1950 server and have this strange problem on music on hold. When I called into a queue using SIP from PSTN line which goes through our cisco gateway (cisco 5300), asterisk will start play music on hold. But this MOH seems at voice activation mode. That is only when I make noice on my end then I can hear music otherwise I will hear silence. I have another asterisk (version 1.2.9.1) running on an older Dell server and MOH works fine for call from PSTN. So my guess is that maybe there is some settings in asterisk cause this problem. Any suggestion about this problem? GG ___ --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 pgpOMULkKCmYV.pgp Description: 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
Re: [asterisk-users] Asterisk on Solars?
Andrew Joakimsen wrote: Has anyone gotten Asterisk to compile on Solaris 10? I have tried both 1.2 and 1.4 and I get errors about editline. Actually it seems that 1.4 goes through more of the process, but thats not good enough Last login: Wed Oct 25 09:18:02 2006 from 208.47.125.33 Sun Microsystems Inc. SunOS 5.10 Generic January 2005 $ su Password: # asterisk # asterisk -r Asterisk 1.2.7.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Connected to Asterisk 1.2.7.1 currently running on unknown (pid = 667) unknown*CLI show version Asterisk 1.2.7.1 built by root @ unknown on a sun4u running SunOS on 2006-07-02 05:05:54 UTC unknown*CLI exit # uname -a SunOS unknown 5.10 Generic_118822-25 sun4u sparc SUNW,Sun-Fire-280R -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic 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
[asterisk-users] redhat enterprise 3
Hi i am planning to install Asterisk with a WildCard TDM400P on a redhat enterprise 3. I will use the last stable source, 1.2.13 Asterisk and 1.2.11 Zaptel. Do you know if there are some issue with this version? Can i compile Asterisk with the classic make make install make sample? Do you suggest to use pre build rpm? ___ --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 Solars?
Would you mind explaining how you got it to compile? Regards, Andrew On 11/17/06, J. Oquendo [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: Has anyone gotten Asterisk to compile on Solaris 10? I have tried both 1.2 and 1.4 and I get errors about editline. Actually it seems that 1.4 goes through more of the process, but thats not good enough Last login: Wed Oct 25 09:18:02 2006 from 208.47.125.33 Sun Microsystems Inc. SunOS 5.10 Generic January 2005 $ su Password: # asterisk # asterisk -r Asterisk 1.2.7.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Connected to Asterisk 1.2.7.1 currently running on unknown (pid = 667) unknown*CLI show version Asterisk 1.2.7.1 built by root @ unknown on a sun4u running SunOS on 2006-07-02 05:05:54 UTC unknown*CLI exit # uname -a SunOS unknown 5.10 Generic_118822-25 sun4u sparc SUNW,Sun-Fire-280R -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --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 on Solars?
Andrew Joakimsen wrote: Would you mind explaining how you got it to compile? Regards, Andrew I didn't do anything out of the ordinary. I followed https://svn.sunlabs.com/svn/solaris-asterisk/README to the letter... PATH=/usr/sbin:/usr/bin:/usr/ccs/bin:/usr/sfw/bin svn co https://svn.sunlabs.com/svn/solaris-asterisk/zaptel-solaris/trunk zaptel-solaris svn co https://svn.sunlabs.com/svn/solaris-asterisk/asterisk/trunk asterisk cd zaptel-solaris gmake cd .. cd asterisk gmake gmake pkg cd .. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic 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
Re: [asterisk-users] Asterisk on Solars?
Andrew Joakimsen wrote: Would you mind explaining how you got it to compile? Regards, Andrew http://www.infiltrated.net/asteriskSol.tar.gz if it makes it easier for you -bash2-2.05b$ ls -ltha asteriskSol.tar.gz |md5 38e6eb68e2ef29968a61b16fca2e3c78 -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic 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
Re: [asterisk-users] Asterisk on Solaris (last message)
Andrew Joakimsen wrote: Would you mind explaining how you got it to compile? Regards, Andrew See what happens when you're overdosing on coffee.. Anyhow: SunOS *sun4u sparc *SUNW,Sun-Fire-280R Take note for others downloading... It's not x86 Solaris before someone shoots off It won't pkgadd for me -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic 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
Re: [asterisk-users] Attempting native bridge of
El vie, nov 17 de 2006 a las 04:34 +0530, Vicky comentaba: Thats really strange .. if you have made canreinvite=no then it should not even attampt native bridging and should transcode codecs ..something's fishy here .. Also try to put canreinvite=no in testulaw exntension too . So why do I have audio in both ways using 2 IP Phones with 2 different codecs and getting 'Attempting native bridging' at the same time? I've always had canreinvite=no in all my extensions. This is my sip.conf: [testgsm] type=friend host=dynamic username=testgsm context=astertest canreinvite=no disallow=all allow=gsm [testulaw] type=friend host=dynamic username=testulaw context=astertest canreinvite=no disallow=all allow=ulaw [testalaw] type=friend host=dynamic username=testalaw context=astertest canreinvite=no disallow=all allow=alaw [testg723] type=friend host=dynamic username=testg723 context=astertest canreinvite=no disallow=all allow=g723 [testg729] type=friend host=dynamic username=testg729 context=astertest canreinvite=no disallow=all allow=g729 [1001] type=friend host=dynamic username=1001 context=astertest canreinvite=no disallow=all allow=g729 [1010] type=friend host=dynamic username=1010 context=astertest canreinvite=no disallow=all allow=ulaw Thanks again! ___ --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] 1 FXO termination device
Hi List, I am looking for a 1 FXO analog termination device, other than the obvious PC + FXO card, and which can achieve decent call quality. The SPA-3000 seems an option... have you got any other ideas? Cheers, Jean-Michel. ___ --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] Understanding the CDR with forwards...
Hello, I have a few users using Polycom 501's and some are using the forward function on the phone. When a call comes in the system, he/she gets the standard welcome to abc inc., bla bla bla message. When the dial an extension, they get forward to the phone, which forwards them to a cell phone. That much is clear. My CDR entries look like this: (columns are calldate, accountcode, src, dst, billsec and lastapp) 2006-11-17 11:37:56 | 514555 | Unknown | 702 | 28 | Dial | | 2006-11-17 11:38:27 | 51455 | Unknown | 416123 | 89 | Dial | My question is, if the caller spends 28 seconds listening to options before dialing an extension, and the call last 89 seconds...Should the first leg have a billsec of 89+28=117sec and the second 89 seconds? Mike ___ --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] specify codec by domain?
Hi, My pstn provider is currently set up so that when asterisk sends an outbound SIP call to them, if sip.conf says: [general] allow=g729,ulaw then it always picks ulaw, even though g729 is listed first. However, if sip.conf says: [general] allow=g729 then g729 is chosen. Dial([EMAIL PROTECTED] How can I force calls to this sip domain to use g729? Mark ___ --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 as an analog call recording solution (was: Recording outbound analog calls with X100P)
John Novack and Time Bandit, Thank you for your excellent advice and for correcting me on the 12V power connector issue. I feel confident to move forward on this project now. Thanks, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] 1 FXO termination device
Jean-Michel Hiver a écrit : Hi List, I am looking for a 1 FXO analog termination device, other than the obvious PC + FXO card, and which can achieve decent call quality. The SPA-3000 seems an option... have you got any other ideas? Tiger G104 has PSTN to VoIP and vice versa. Didn't had time to test it. -- Daniel ___ --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] 1 FXO termination device
*I suggest the digitnetworks fxo oem it has a good cuality of sound and a good delay* 2006/11/17, Jean-Michel Hiver [EMAIL PROTECTED]: Hi List, I am looking for a 1 FXO analog termination device, other than the obvious PC + FXO card, and which can achieve decent call quality. The SPA-3000 seems an option... have you got any other ideas? Cheers, Jean-Michel. ___ --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] some questions about atxfer usage
Hi I just press * to retrieve the caller again - Have you tried that? No, I haven't. Thanks, it's perfect for me. Conrad ___ --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] some questions about atxfer usage
OK
Thank you very much.
On 11/16/06, Alberto Pastore [EMAIL PROTECTED] wrote:
Antonio Almodóvar ha scritto:
Hi all.
I have enabled the attended transfer feature in features.conf. I'm
using it and I want to resolve some questions, I hope someone can help
me :)
When I transfer a call to an extension:
- The extension rings during 15 seconds and the call returns to the
transferer. Is there any possibility to recover the call before the
timeout of 15 seconds expires?
I mean, I would like to personalize the way of making transfers using
the feature of atxfer. How can I do that?
Thanks in advance.
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Hi Antonio.
Taking a look at the following code line from res_features.c:
newchan = ast_feature_request_and_dial(transferer,
Local,
ast_best_codec(transferer-nativeformats),
dialstr,
15000, // ---
outstate,
cid_num,
cid_name);
I assume that 15000 msecs is a hardcoded value...
You might want to replace it with some variable taken from
pbx_builtin_getvar_helper() results
but it involves recompiling at least the res_features.c module;
something more or less
like this (I haven't tested it!!!):
//these two lines go at the beginning of the if {} block
char *transfer_timeout_str;
int transfer_timeout = 15; //default value
//these lines replace the newchan = ast_feature_request_and_dial(...) one
//read the value (if any) from TRANSFER_TIMEOUT
//can be set in extensions.conf's [globals] (TRANSFER_TIMEOUT = 30)
transfer_timeout_str = pbx_builtin_getvar_helper(transferer,
TRANSFER_TIMEOUT);
if (transfer_timeout_str) {
transfer_timeout = atoi(transfer_timeout_str);
//sanity check
if (transfer_timeout = 0) transfer_timeout = 15;
}
newchan = ast_feature_request_and_dial(transferer,
Local,
ast_best_codec(transferer-nativeformats),
dialstr,
transfer_timeout * 1000, // ---
outstate,
cid_num,
cid_name);
Bye,
Alberto.
--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
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[asterisk-users] voice quality of Aastra 480i CT and cordless
Hi Folks, Looking for feedback on the cordless phones with the Aastra 480i CT handset. Is voice quality comparable to standard consumer residential 2.4GHz cordless phones in the US$30 - $50 price range, or better/worse? How about handset and speakerphone quality for the main phone? Seems like there have been various big issues with firmware in past, but is it pretty stable now? Thanks, Scott ___ --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] Freepbx changes dont reflect in asterisk
Hello, From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS. ___ --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] voice quality of Aastra 480i CT and cordless
Scott - I find the phone to be a great product, as it stands today with current firmware. I use it extensively for customer deployments, and everyone seems really pleased with the features and performance. We have hundreds of these in the field. I much prefer DECT to WIFI for client sites where a few users need basic in-office mobility. Cory Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Keagy Sent: Friday, November 17, 2006 2:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voice quality of Aastra 480i CT and cordless Hi Folks, Looking for feedback on the cordless phones with the Aastra 480i CT handset. Is voice quality comparable to standard consumer residential 2.4GHz cordless phones in the US$30 - $50 price range, or better/worse? How about handset and speakerphone quality for the main phone? Seems like there have been various big issues with firmware in past, but is it pretty stable now? Thanks, Scott ___ --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] voice quality of Aastra 480i CT and cordless
We have had good results mostly from this unit except for one issue that is currently being looked into by Aastra. The issue is if a second call comes in and the cordless answers then puts the call on hold audio drops one way on the handset. Aastra was able to reproduce this and is working on it. From time to time we get reports of bad feedback on the cordless unit but most of the time it is fine. Also the buttons on the cordless are easily mashed with a chubby face :-) Overall we are very pleased with the unit and the ability to have the cordless off of the handset is a great thing. I have not been able to find another unit that has this same feature. Curt _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Keagy Sent: Friday, November 17, 2006 1:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voice quality of Aastra 480i CT and cordless Hi Folks, Looking for feedback on the cordless phones with the Aastra 480i CT handset. Is voice quality comparable to standard consumer residential 2.4GHz cordless phones in the US$30 - $50 price range, or better/worse? How about handset and speakerphone quality for the main phone? Seems like there have been various big issues with firmware in past, but is it pretty stable now? Thanks, Scott ___ --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] metermaid and 1.2.13?
It is unclear to me if the metermaid patch should be in 1.2.13 or not. Please advise. Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org ___ --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] voice quality of Aastra 480i CT and cordless
Scott, I've used the phone for 9 months. It's a truly outstanding phone. The cordless handset sounds great. It is limited to two ongoing calls at one time, but that has not been an issue for me. The range on the cordless is comparable to the Panasonic KX-TG4000 KSU that I used to use and a Panasonic 5.8GHz corldess that I use via an ATA. The ability to sync the directory of the base and the handset makes the 480i CT much more convenient than the ATA/Cordless combination. Michael --Original Message Text--- From: Scott Keagy Date: Fri, 17 Nov 2006 14:09:02 -0500 Hi Folks, Looking for feedback on the cordless phones with the Aastra 480i CT handset. Is voice quality comparable to standard consumer residential 2.4GHz cordless phones in the US$30 - $50 price range, or better/worse? How about handset and speakerphone quality for the main phone? Seems like there have been various big issues with firmware in past, but is it pretty stable now? Thanks, Scott ___ --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] Freepbx changes dont reflect in asterisk
With FreePBX you can not modify certain conf files - many are overwritten at reload Bart Pedro Silva wrote: Hello, From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS. ___ --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] Freepbx changes dont reflect in asterisk
Hi Pedro, Did you press the red bar at the top of the page? Until you do this, the config files are not written out. Alex On 11/17/06, Pedro Silva [EMAIL PROTECTED] wrote: Hello, From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS. ___ --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
[asterisk-users] wget from within asterisk?
What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? ___ --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] wget from within asterisk?
-Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] wget from within asterisk?
hi, On Fri, 2006-11-17 at 13:32 -0700, Damon Estep wrote: Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. agi is your friend. Matteo ___ --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] wget from within asterisk?
I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] wget from within asterisk?
Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] wget from within asterisk?
Thx! I saw a note about Curl vs. CURL, is there a difference? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, November 17, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] wget from within asterisk?
They both seem to work, but the Curl spits out warnings about being deprecated. Ours are all configured using CURL. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:52:35 -0700 Subject: RE: [asterisk-users] wget from within asterisk? Thx! I saw a note about Curl vs. CURL, is there a difference? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] TDM2400p and HW echo canceller
Do the zaptel drivers need to be told NOT to use the software echo canceller when using a TDM2400p card with hardware echo canceller, or does the driver figure this out by itself? Thanks, Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] automated response
I will be out of the office from November 17, returning November 27. Mark Stayt Director of IT Ocean Optics, Inc +1-727-733-2447 (Phone) +1-727-733-3962 (Fax) ___ --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] wget from within asterisk?
Thanks a bunch, this seems to be a simple solution, I just did not have CURL installed before I built asterisk. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, November 17, 2006 2:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? They both seem to work, but the Curl spits out warnings about being deprecated. Ours are all configured using CURL. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:52:35 -0700 Subject: RE: [asterisk-users] wget from within asterisk? Thx! I saw a note about Curl vs. CURL, is there a difference? From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, November 17, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] wget from within asterisk?
Options I am aware of for installing curl are yum install in FC4 or download from curl.haxx.se, neither option distinguishes between curl and CURL, can someone offer me the slap in the head I need? Damon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Friday, November 17, 2006 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Thanks a bunch, this seems to be a simple solution, I just did not have CURL installed before I built asterisk. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, November 17, 2006 2:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? They both seem to work, but the Curl spits out warnings about being deprecated. Ours are all configured using CURL. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:52:35 -0700 Subject: RE: [asterisk-users] wget from within asterisk? Thx! I saw a note about Curl vs. CURL, is there a difference? From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, November 17, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] Extension Response Slow
Here is my Extensions.conf file (Default Context). When an
individual calling in dials the extension, the response time seems
very slow. It doesn't immediately go to the next step, but hangs out
for a few seconds (silence)... Suggestions?
Thanks in advance... /pj
[default]
exten = _XX.,1,Wait,2 ; Wait a second, just for fun
exten = _XX.,n,Answer ; Answer the line
exten = _XX.,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to
10 seconds
exten = _XX.,n(restart),BackGround(securerad-welcome2); Play
some instructions
exten = _XX.,n,WaitExten ; Wait for an extension to be dialed.
;Directory
exten = 9,1,Directory(default)
;Sales
exten = 1,1,Dial(SIP/linksys, 15)
exten = 1,2,Voicemail([EMAIL PROTECTED])
exten = 1,3,PlayBack(vm-goodbye)
exten = 1,4,HangUp()
;Customer Service
exten = 2,1,Dial(SIP/linksys, 15,m)
exten = 2,2,Voicemail([EMAIL PROTECTED])
exten = 2,3,PlayBack(vm-goodbye)
exten = 2,4,HangUp()
;Operator
exten = 0,1,Dial(SIP/linksys, 15,m)
exten = 0,2,Voicemail([EMAIL PROTECTED])
exten = 0,3,PlayBack(vm-goodbye)
exten = 0,4,HangUp()
;Jackson
exten = 1000,1,Dial(SIP/linksys, 15,m)
exten = 1000,2,Voicemail([EMAIL PROTECTED])
exten = 1000,3,PlayBack(vm-goodbye)
exten = 1000,4,HangUp()
;Voicemail extension
exten = 2000,1,VoiceMailMain
;Record greetings extension
exten = 1005,1,Answer
exten = 1005,2,Wait(2)
exten = 1005,3,Record(asterisk-recording%d:gsm)
exten = 1005,4,Wait(2)
exten = 1005,5,Playback(${RECORDED_FILE})
exten = 1005,6,Wait(2)
exten = 1005,7,Hangup
___
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RE: [asterisk-users] wget from within asterisk?
The Curl/CURL is an asterisk dialplan distinction. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 15:06:49 -0700 Subject: RE: [asterisk-users] wget from within asterisk? Options I am aware of for installing curl are yum install in FC4 or download fromcurl.haxx.se, neither option distinguishes between curl and CURL, can someone offer me the slap in the head I need? Damon From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfDamon Estep Sent: Friday, November 17, 2006 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Thanks a bunch, this seems to be a simple solution, I just did not have CURL installed before I built asterisk. From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 2:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? They both seem to work, but the Curl spits out warnings about being deprecated. Ours are all configured using CURL. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:52:35 -0700 Subject: RE: [asterisk-users] wget from within asterisk? Thx! I saw a note about Curl vs. CURL, is there a difference? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf OfRuss Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] Do Not Call List
A quick google search says there isn't anything written yet. But looking at the database itself, it seems pretty easy to import data into a sql table or do xml pulls from them directly.. https://www.donotcall.gov/FAQ/FAQBusiness.aspx#download https://www.donotcall.gov/FAQ/FAQBusiness.aspx#download It wouldn't be hard to code up at all actually... a little perl magic and voila. ;) Who needs a weekend project? -Original Message- From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Wednesday, November 15, 2006 9:18 PM To: Asterisk-Users Subject: [asterisk-users] Do Not Call List The US has a Do Not Call list to which people can subscribe to prevent being called by advertisers. Federal laws (strengthened by some state and more local laws) assign penalties for calling people/phones on the DNCL. Is there a query gateway that Asterisk (or an app using Asterisk) can filter through to ensure a number is OK to call (not on the list) before calling it? -- (C) Matthew Rubenstein ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users _ ella for Spam Control has removed 3785 Spam messages and set aside 117 Newsletters for me You can use it too - and it's FREE! www.ellaforspam.com http://www.ellaforspam.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
Re: [asterisk-users] TDM2400p and HW echo canceller
Webster, Andrew wrote: Do the zaptel drivers need to be told NOT to use the software echo canceller when using a TDM2400p card with hardware echo canceller, or does the driver figure this out by itself? If I'm recalling correctly, if the drivers see the hardware E.C. it'll use it instead. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] wget from within asterisk?
On version 1.2.12.1 running on FC4 with curl.i386 installed the asterisk CURL function is not registered, perhaps in need something else (curl-devel.i386 ?) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, November 17, 2006 1:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] wget from within asterisk? Make sure the curl library/package is installed, then re-compile asterisk. We're using it on 1.2. -Original Message- From: Damon Estep [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:40:40 -0700 Subject: RE: [asterisk-users] wget from within asterisk? I saw CURL, but it does not register appear in show functions or show applications, deprecated or add-on? From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Russ Beaupre Sent: Friday, November 17, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] wget from within asterisk? -Original Message- From: Damon Estep [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 17 Nov 2006 13:32:53 -0700 Subject: [asterisk-users] wget from within asterisk? What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions? Look at the CURL function. ___ --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] Extension Response Slow
Phil Jackson wrote: Here is my Extensions.conf file (Default Context). When an individual calling in dials the extension, the response time seems very slow. It doesn't immediately go to the next step, but hangs out for a few seconds (silence)... Suggestions? [default] exten = _XX.,n(restart),BackGround(securerad-welcome2); Play some instructions That doesn't look correct. What is it supposed to do? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Do Not Call List
It wouldn't be hard to code up at all actually... a little perl magic and voila. ;) Who needs a weekend project? The Perl magic would be easy. Writing the check to pay for all of that data is what is so hard... -MC ___ --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] Destar release!
Grettings! The 0.2.1 version of Destar has been released. Destar is a simple web-based interface to manage Asterisk. It supports different types of trunks and phones, many asterisk applications, and Vitual/hosted PBXs. It can be downloaded from: http://destar.berlios.de/ Or directly from: http://prdownload.berlios.de/destar/destar-0.2.1.tar.gz This new version contains many bug fixes present in the 0.2.0 version. There is a first attempt to manage the button style of the operator panel and now is possible to jump to a Virtual PBX from the incoming context of a trunk. You can write us to [EMAIL PROTECTED] or chat in #destar at irc.freenode.net. Give it a try! Bye bye. ___ --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] Do Not Call List
I'm surprised someone doesn't come up with a consortium for all the asterisk users to poll a central location or does the data come with restrictions about sharing the data? Duane from e164.org says he's already built the application you are looking for to deal with Australian databases if that helps. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Friday, 17 November 2006 6:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Do Not Call List It wouldn't be hard to code up at all actually... a little perl magic and voila. ;) Who needs a weekend project? The Perl magic would be easy. Writing the check to pay for all of that data is what is so hard... -MC ___ --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] Do Not Call List
Depending on your organization, you're allowed up to 5 area codes for free. -Original Message- From: Michael Collins [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Friday, November 17, 2006 3:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Do Not Call List It wouldn't be hard to code up at all actually... a little perl magic and voila. ;) Who needs a weekend project? The Perl magic would be easy. Writing the check to pay for all of that data is what is so hard... -MC ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users _ ella for Spam Control has removed 3789 Spam messages and set aside 117 Newsletters for me You can use it too - and it's FREE! www.ellaforspam.com http://www.ellaforspam.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
[asterisk-users] Ringing a group of phones but not if they are busy
I need to ring a group of 8 phones, but not if they are already on another call. How can I determine which of those 8 phones are busy so I only ring the others? -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chàvez Prats Director de Tecnologìa +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Ringing a group of phones but not if they are busy
Carlos Chavez wrote: I need to ring a group of 8 phones, but not if they are already on another call. How can I determine which of those 8 phones are busy so I only ring the others? chanIsAvail 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
Re: [asterisk-users] outgoing works, incoming fails on asterisk passthrough to inter-tel
Alright, I've figured out that by adding a wait to the dial I can get it
to connect to the inter-tel pbx. I still can't get it to either a) pass
the caller id, or b) talk to the correct extension. The inter-tel box
always redirects the call to the operator.
When the call comes in on the T1 line it looks like this:
*801555*154 where 801-555- is the incoming caller id and 154 is
the extension I'm dialling (the first 4 digits get cut off before I ever
receive a connection). When I duplicate this using
Dial(Zap/g3/w*${CALLERID}*${EXTEN},15,or) it won't connect (Exiting with
DIALSTATUS=NOANSWER) at all. If I leave it with just
Dial(Zap/g3/w${EXTEN},15,or) it will connect to the operator with no
caller id being set.
Any help would be greatly appreciated. Thanks
Nathan Bell
Nathan Bell wrote:
Hi everybody,
Well, I've finally got asterisk to to talk nicely with my Intertel
pbx. Currently there is a outside T1 line (em wink start, esf, b8zs)
connected to asterisk, and then asterisk connected similarly to my
Intertel pbx. For right now all asterisk is doing is passing calls
between the two.
When I call out from the pbx, I can connect perfectly to the outside
world. When I call from outside, I can talk to the asterisk box, but
asterisk fails to pass the call to the pbx. The following is the log
of the connection (numbers scrambled to protect the innocent). At the
end I've included my extensions.conf file. The incoming phone number
is 801-555-, and I'm calling 555-5154
I've tried changing the exten =
_X.,2,Dial(Zap/g3/${EXTEN},15,r) line that transfers the call to
the pbx to exlude the , add in the caller id, and various other
things, but the results are always identical. If anyone has any
experience with talking to inter-tel pbx's, please let me know what
trick is necessary.
Thanks a million.
call log follows:
*** call log cut to save space, see original message for log ***
extensions.conf follows:
; --- First all the incoming routes ---
; from outside T1
[from-ptsn]
exten = s,1,Answer()
include = intertel-ext
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup()
; from intertel-axxess box
[from-intertel]
include = internal
; generic interal route
[internal]
exten = s,1,Answer()
include = intertel-ext
include = to-ptsn
; --- next all the outgoing routes ---
; send call to outside world
[to-ptsn]
; Trunk group '4' is the outgoing T1
exten = _X.,1,SetTransferCapability(SPEECH)
exten = _X.,2,Dial(Zap/g4/${EXTEN},15,r)
exten = _X.,3,Playback(vm-nobodyavail)
exten = _X.,4,Hangup()
exten = _X.,103,Playback(vm-nobodyavail)
exten = _X.,104,Hangup()
; check if extension is to intertel
[intertel-ext]
; I think this is all of our DID numbers...
; internal extensions
exten = _1XX,1,Goto(to-intertel,${EXTEN},1)
; main number
exten = 033,1,Goto(to-intertel,${EXTEN},1)
; customer service number
exten = 418,1,Goto(to-intertel,${EXTEN},1)
; fax number
exten = 096,1,Goto(to-intertel,${EXTEN},1)
; other did numbers no one seems to know anything about
exten = _2[2-3]X,1,Goto(to-intertel,${EXTEN},1)
; send call to intertel
[to-intertel]
; Trunk '3' is the intertel box
exten = _X.,1,SetTransferCapability(SPEECH)
exten = _X.,2,Dial(Zap/g3/${EXTEN},15,r)
exten = _X.,3,Playback(vm-nobodyavail)
exten = _X.,4,Hangup()
exten = _X.,103,Playback(vm-goodbye)
exten = _X.,104,Hangup()
; --- lastly all of the macros we'll be using ---
; none as of now
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Re: [asterisk-users] Freepbx changes dont reflect in asterisk
Hi, 2006/11/17, Alex Robar [EMAIL PROTECTED]: Hi Pedro, Did you press the red bar at the top of the page? Until you do this, the config files are not written out. Yes, i press the red bar and freepbx dont return any error. For example, If i add a new extension, the files extensions_addicional.conf and sip_addicional.con are supposed to be updated and are not. Best regards, PS. Alex On 11/17/06, Pedro Silva [EMAIL PROTECTED] wrote: Hello, From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS. ___ --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
RE: [asterisk-users] Do Not Call List
Oddly enough, there's really nothing stopping one from doing so in the material I just scan through at: http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm http://www.ftc.gov/bcp/rulemaking/tsr/tsrrulemaking/index.htm In regards to the fee, here is the latest: The amended rule increases the annual fee for access to the Registry for each area code of data to $62 per area code, or $31 per area code of data during the second six months of an entity's annual subscription period. The maximum amount that would be charged to any single entity for accessing 280 area codes of data or more is increased to $17,050. In addition, the amended rule retains the provisions regarding free access by exempt organizations, as well as free access to the first five area codes of data by all entities. In particular, here is the part on the usage... If a central database (external from the FTC) does start up, they'll have to register who uses the database. --- § 310.9 Fee for access to do-not-call registry. (c) Access to the do-not-call registry is limited to telemarketers working on their own behalf or working on behalf of other sellers or telemarketers. Prior to accessing the do-not-call registry, a telemarketer must provide the identifying information required by the operator of the registry to collect the user fee, and must certify, under penalty of law, that the telemarketer is accessing the registry solely to comply with the provisions of this rule. If the telemarketer is accessing the registry on behalf of other sellers or telemarketers, that telemarketer also must identify each of the other sellers or telemarketers on whose behalf it is accessing the registry, and it must certify, under penalty of law, that the other sellers or telemarketers will be using the information gathered from the registry solely to comply with the provisions of this rule. -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Friday, November 17, 2006 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Do Not Call List I'm surprised someone doesn't come up with a consortium for all the asterisk users to poll a central location or does the data come with restrictions about sharing the data? Duane from e164.org says he's already built the application you are looking for to deal with Australian databases if that helps. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael Collins Sent: Friday, 17 November 2006 6:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Do Not Call List It wouldn't be hard to code up at all actually... a little perl magic and voila. ;) Who needs a weekend project? The Perl magic would be easy. Writing the check to pay for all of that data is what is so hard... -MC ___ --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 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 http://lists.digium.com/mailman/listinfo/asterisk-users _ ella for Spam Control has removed 3790 Spam messages and set aside 117 Newsletters for me You can use it too - and it's FREE! www.ellaforspam.com http://www.ellaforspam.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
Re: [asterisk-users] Ringing a group of phones but not if they are busy
On Fri, 2006-11-17 at 16:03 -0800, Steven Ringwald wrote: Carlos Chavez wrote: I need to ring a group of 8 phones, but not if they are already on another call. How can I determine which of those 8 phones are busy so I only ring the others? chanIsAvail The problem with ChanIsAvail is that if ig give it a line like this: s,1,ChanIsAvail(SIP/100SIP/101SIP/102SIP/103SIP/104SIP/105SIP106) the resulting variable only lists the first available channel and not all the available channels so I cannot ring all the available channels. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chàvez Prats Director de Tecnologìa +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Jitter Buffers in Zapata
Hi, Does anybody know how exactly the jitter buffer in the zap channels work? Is it adaptive or fixed? ; Configure jitter buffers in zapata (each one is 20ms, default is 4) ; jitterbuffers=30 This setting puts 600ms of jitter buffer but the call does not sound as if it had a .6 second delay which leads me to believe the buffer is adaptive. If so, can I hardcode it to be fixed? Thanks, -- Andres Technical Support http://www.telesip.net ___ --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] Freepbx changes dont reflect in asterisk
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
Re: [asterisk-users] voice quality of Aastra 480i CT and cordless
Aastra is a great phone for sound quality and other features. I didn't have any problems with it and didn't go back to Grandstream once installed Aastra. My only concern was some problem with its web UI bugs, but that will be eventually fixed. ___ --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] Need help on Music on Hold
Its Cisco. Please disable VAD and voice compression in your Cisco equipment. I had exactly the same problem which haunted me for more than a year and I tried everything, asked everyone, and no one could solve the problem, until the service provider told me they had some voice compression feature enabled on their Cisco equipment for bandwidth saving. Once they turned it off and let G711 pass through as is, MoH started to work perfectly 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] specify codec by domain?
You can use SIP_CODEC variable. Read README.variables file. On 11/18/06, Mark Price [EMAIL PROTECTED] wrote: Hi, My pstn provider is currently set up so that when asterisk sends an outbound SIP call to them, if sip.conf says: [general] allow=g729,ulaw then it always picks ulaw, even though g729 is listed first. However, if sip.conf says: [general] allow=g729 then g729 is chosen. Dial([EMAIL PROTECTED] How can I force calls to this sip domain to use g729? Mark ___ --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] Freepbx changes dont reflect in asterisk
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
Re: [asterisk-users] Soundfiles adding during phone calls
bails wrote: Ronald Wiplinger wrote: Ronald Wiplinger wrote: Tom Lynn wrote: Ron, The guy is trying to help you. Go to the link and read it. There is a feature that you can use to play a recording into the voice channel. Mine is set so when you press #9, the caller hears the lots of monkeys recording. The best part of it is that you can hang up and the recording will continue to play to the caller. When it expires, so does the call I tried this: features.conf [featuremap] blindxfer = ##; Blind transfer was #1 - now press # twice disconnect = *0; Disconnect automon = *1; One Touch Record atxfer = *2; Attended transfer [applicationmap] tortore= *9,callee,Playback,tt-monkeys Yap, that magic word helped! I got still some problems with it. I understand that I do not hear the sound, but wonder if I should get the call back after the playback or not anymore. In my experience the caller hang up and my phone remains on the status connected I have only the choice to power cycle the phone. Anything I can do ? bye Ronald Wiplinger ___ --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
Re: [asterisk-users] strip + sign from incoming ${EXTEN} var?
voiplist wrote:
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 ${EXTEN} var right off
when it comes into extensions.conf before we do anything else.
I have read that since this is a built in Asterisk variable and it can
only be read, not written to.
We know there are other ways to handle this but we were just hoping
for a simple solution like resetting the variable.
Any help would be appreciated.
exten = _+NXXNXX,1,Goto(${EXTEN:1},1)
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[asterisk-users] spc.exe
Does anyone have a copy of spc.exe they could send me? ___ --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
