Re: [Asterisk-Users] logging performance, important impact?
Moises Silva ha scritto: How important is the impact i could have if I have a single entry log file in /etc/asterisk/logger.conf wich loggs everything, even debug level. This is sometimes important to us because it helps us to make a track of the issues some times we have with the system. I just want to know if there is a considerable impact in performance because of the writing of the logs. I haven't made benchmarks, but speaking out of my experience and knowing that asterisk debug level is very verbose I think it will have a sensible impact. I can remember a very slow samba installation due to the sysadmin forgetting to turn off the debug level of logging, it made the difference between "we can use it" and "we switch back to windows", and I'm talking about a dozen of users, not big numbers. Are you sure debug level will help you tracking the issues ? Usually debug level info is for debug like "what is the bottleneck ?", "why my prepaid agi isn't doing the update on hangup ?", nothing you need to keep tracking once you are in production. Is better to log as few expected stuff as possible and as much unexpected stuff as possible. Anyway autoanswering your question is pretty simple, put an agi which timestamps the first line of each extension and one for the last one, send a lot of calls in the system with and without debugging and look at the results. ___ --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] Error when compiling asterisk
Jourdan, What Distro are you using? do you have gcc installed?On 12/6/05, jourdan lemieux <[EMAIL PROTECTED]> wrote: Any help on this pleaseHi, I am getting this error when compiling asterisk `ls *.c`: unrecognized option h -DBUSYDETECT_MARTIN `ls *.c`Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ... GNU long options: --debug --dump-po-strings --dump-strings --help --login --noediting --noprofile --norc --posix --rcfile --rpm-requires --restricted --verbose --version --wordexp Shell options: -irsD or -c command (invocation only) -abefhkmnptuvxBCHP or -o optionmake: *** [.depend] Error 2 Any ideas of what the problem might be. Thank you -- Regards,Mark Quitoriano, CCNAFan the flame...http://www.spreadfirefox.com/?q=user/register&r=19441 ___ --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] OH323 user configuration
Hi all, I installed OH323 successfully, But i am in confiused where i should create the H.323 IP Phones User Configuration, like sip.conf, and what will be the H.323 user configuration format. I already checked oh323.conf, but i did not find any H.323 user configuration example. Please advise me how i can register my H.323 IP Phone? == Parsing '/etc/asterisk/oh323.conf': Found == Registered channel type 'OH323' (InAccess Networks OpenH323 Channel Driver) == OpenH323 Channel Ready (v0.6.7) Thank You -- Thank You, Code Lover ___ --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] Grandstream NTP
Rod Bacon wrote: It now appears to be server specific. The shipped default, time.nist.gov, appears to work OK. Does anyone know of anything specific required by these grandstream phones as far as NTP server support goes? I also have GXP2000 pones and use a 'standard' ntp.conf, nothing fancy at all: driftfile /etc/ntp/drift server pool.ntp.org server 127.127.1.1 fudge 127.127.1.1 stratum 10 restrict 10.10.0.0 mask 255.255.255.0 nomodify nopeer notrap ; allow local lan to use the ntp server restrict 127.0.0.1 ___ --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] How to restric user to call only specified country
Hi i have local extensions and i have connected sip provider account to call out side but i have account can call any part of the world how to restrict some of users should call only USA or any Other or restrict to call USA ram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo cancellation over satellite link
In software asterisk can support more than that, no? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Tuesday, 6 December 2005 17:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Echo cancellation over satellite link On 12/6/05, funny guy <[EMAIL PROTECTED]> wrote: > Hi, > > Just wondering, is the echo canceller in the TE411P capable of cancelling > the echo caused by the delay over satellite link (i.e. approx 400 ms delay)? > > Does anyone have any success story to share? > > I'm kinda stuck with a really2 annoying echo... adjusting the gain didn't > help... and what should my zapata.conf look like for effective echo > cancellation? > > Thanks in advance ^_^ > No. Neither Digium nor Sangoma I believe are putting in hardware cans that would support a 400ms+ tail. I think the most you're going to get is 128ms. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation over satellite link
Good luck. I am running dedicated echo cans and they only go to 192ms. I do not think 400ms would be regarded as toll quality which is what most links strive for. Byt the time the echo cans buffer enough so they can cancel 400 you would have some extreme latency. My .02 anyway. On Dec 6, 2005, at 12:03 AM, BJ Weschke wrote: On 12/6/05, funny guy <[EMAIL PROTECTED]> wrote: Hi, Just wondering, is the echo canceller in the TE411P capable of cancelling the echo caused by the delay over satellite link (i.e. approx 400 ms delay)? Does anyone have any success story to share? I'm kinda stuck with a really2 annoying echo... adjusting the gain didn't help... and what should my zapata.conf look like for effective echo cancellation? Thanks in advance ^_^ No. Neither Digium nor Sangoma I believe are putting in hardware cans that would support a 400ms+ tail. I think the most you're going to get is 128ms. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation over satellite link
On 12/6/05, funny guy <[EMAIL PROTECTED]> wrote: > Hi, > > Just wondering, is the echo canceller in the TE411P capable of cancelling > the echo caused by the delay over satellite link (i.e. approx 400 ms delay)? > > Does anyone have any success story to share? > > I'm kinda stuck with a really2 annoying echo... adjusting the gain didn't > help... and what should my zapata.conf look like for effective echo > cancellation? > > Thanks in advance ^_^ > No. Neither Digium nor Sangoma I believe are putting in hardware cans that would support a 400ms+ tail. I think the most you're going to get is 128ms. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Echo cancellation over satellite link
Hi, Just wondering, is the echo canceller in the TE411P capable of cancelling the echo caused by the delay over satellite link (i.e. approx 400 ms delay)? Does anyone have any success story to share? I'm kinda stuck with a really2 annoying echo... adjusting the gain didn't help... and what should my zapata.conf look like for effective echo cancellation? Thanks in advance ^_^ Yahoo! Personals Let fate take it's course directly to your email. See who's waiting for you Yahoo! Personals___ --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] EAGI Audio Capture
Hello Everyone,
Why EAGI is made so complex? The audio captured with
the EAGI-perl script on voip-info.org is almost not
useful. The clarity of the audio is pathetic. Am i
missing something??? I have Digium TDM 12B. I can get
calls to my VoIP phone ok thru TDM and asterisk. But
when i use EAGI-perl script, neither GSM nor RAW audio
file created after capture sounds clear. Lot of noise
and voice can not even be heard.
Please Help!!!
Thanks
Frank
EAGI-perl script:
===
#!/usr/bin/perl
#
# Note that this example doesn't check the results of
AGI calls, and doesn't use
# Asterisk::AGI in an attempt to keep it simple and
dependency free.
#
# This program is free software; you can redistribute
it and/or modify
# it under the same terms as Perl itself.
#
# Author: Simon P. Ditner / http://uc.org/simon
#
# Usage:
# - Create an AGI in /var/lib/asterisk/agi-bin, i.e.:
perl.eagi
# - Call using EAGI from your dialplan: exten =>
100,1,EAGI(perl.eagi)
#
use warnings;
use strict;
use IO::Handle;
$| = 1; # Turn of I/O Buffering
my $buffer = undef;
my $result = undef;
my $AUDIO_FD = 3; # Audio is delivered on file
descriptor 3
my $audio_fh = new IO::Handle;
$audio_fh->fdopen( $AUDIO_FD, "r" ); # Open the audio
file descriptor for reading
# Skip over the preamble that Asterisk sends this AGI
while( ) {
chomp($_);
last if length($_) == 0;
}
# Playback beep
print "STREAM FILE beep \"#\"\n"; $result = ;
# Record 5 seconds of audio at 8,000 samples/second
(uses 16 bit integers)
# 5 seconds x 8000 samples/second x ( 16 bits /
8bits/byte ) = 8 bytes
my $bytes_read = $audio_fh->read( $buffer, 8 );
$audio_fh->close();
# Playback beep
print "STREAM FILE beep \"#\"\n"; $result = ;
# Write the raw audio to a file for later analysis
my $fh;
open( $fh, ">/tmp/recording.raw" );
print $fh $buffer;
close( $fh );
# Also convert the raw audio on-the-fly to the GSM
format using 'sox', so that
# we can play it back to the user right now.
open( $fh, "|/usr/bin/sox -t raw -r 8000 -s -w -c 1 -
/tmp/recording.gsm" );
# | | | | | | |
# | | | | | | '-- Write to this file
# | | | | | '-- Read from STDIN
# | | | | '-- Mono Audio
# | | | '-- Samples are words (a word is 2 bytes = 16
bit audio)
# | | '-- The audio is signed (32766..-32766)
# | '-- The sample rate is 8,000 samples/second
# '-- The input format is SLIN, which is 'raw' audio
print $fh $buffer;
close( $fh );
# Playback /tmp/recording.gsm
print "STREAM FILE /tmp/recording \"#\"\n"; $result =
;
exit;
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Re: [Asterisk-Users] Messages button on a Polycom 501
Yes, but I'll bet you have it set the the telephones extension which is why you receive the vm greeting. You need to configure a different extnsion setup to retrieve messages. I do not use @home so not sure how it is setup On Dec 5, 2005, at 6:29 PM, Brent Bloodworth wrote: Actually I think that is how it is setup now. I configured the phone through the web interface. Callback mode is set to "contact" and the Callback contact is set to the extension. On 12/5/05, Jerry Jones <[EMAIL PROTECTED]> wrote: I assume you wish to have the button retrieve your vm - if so then time to edit your config file, or use web interface. msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi. 1.callBack="xxx" xxx=extension to dial to retrieve vm On Dec 5, 2005, at 5:38 PM, Brent Bloodworth wrote: > Need a little help. Just set up an [EMAIL PROTECTED] box with 5 Polycom > 501 phones. Everything works great except the messages button which > when pressed results in asterisk responding "Person at extension > 102 is on the phone. Please leave a message after the tone. I have > searched the web and several of the the asterisk mailing list > archive pages - but I haven't had any luck. Anyone have a suggestion? > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
Ok. I will give one more shot on that. Last time I had one-way-audio issue with that. Thanks. Isamar On Tue, 6 Dec 2005, Boris Bakchiev wrote: No, max we used is 30 channels. But according to voip-info its faster protocol because it offloads media processing to asterisk (which is a better choice I think) and only looks after H323 call setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 11:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] h323 vs oh323 Ok.. how many channels are you using? More than 100? Maybe it can be good only for 10 or 20 simultaneous connections... Isamar On Tue, 6 Dec 2005, Boris Bakchiev wrote: I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Realtime SIP Lookups
All, I have a feeling this question has already been addressed, but the alternative to me asking is searching through six years of list archive messages... month-by-month. We have two asterisk boxes that are using realtime for sip.conf, both static and ... I guess 'realtime' realtime. A polycom phone registers with the first Asterisk box (well actually it registers with OpenSER who forwards the registration to the first Asterisk box... but that shouldn't be too relevant). I check the sip_buddies table in mysql and I can see that the contact info has been updated for this phone. First question, if I do not set rtcachefriends, then the fullcontact field is not updated. Why? There's also another polycom phone that came in through the other Asterisk box. It also updated that phone's contact info in sip_buddies. When I make a call, while running ngrep on the mysql server, I can see the asterisk box doing a 'select * from sip_buddies where username=' and 'select * from sip_buddies where username='. Under certain circumstances that I don't yet understand, it fails periodically it can't find the contact details for the callee. Why? Also, if I DO set rtcachefriends to yes, then it updates /var/lib/asterisk/astdb, and the calls FAILS EVERY time eventhough it is still doing the same two select queries above. Why? Why doesn't it use the values it just pulled out of the database? The info is there! This is pretty basic stuff. Why doesn't it work? Trying to implement a HA solution with Asterisk. CrAzY me thought that realtime might be the solution. If I can't get this to work (and it would have been cool if it did), then it's back to registering with SER, and having SER forward the registration to _all_ the Asterisk boxes, such that every Asterisk box knows the contact details for every user agent. This works well in small test situations, but I don't know how well it will scale. How fast would retrievals be on a fully loaded (ie 120 calls) Asterisk box trying to pull a key/value pair out of 16,000 records in astdb? Help appreciated. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 leg bridged call not hanging up until both legs hangup
Hello everyone,
I am somewhat new to asterisk but am hoping someone can help me. I have
an application that sends the following commands to asterisk's telnet
port:
Action: Originate
Channel: Zap/g1/5551239876
Timeout: 3
Context: from-agent
Exten: 5559871234
Priority: 1
Variable: call_id=1234
Variable: origination=5551239876
This causes the 5551239876 number to be called and when the agent
answers, the extension number is called from the dial plan and the two
calls are bridged. The dial plan to do this looks like this:
[from-agent]
exten => _X.,1,Answer()
exten => _X.,n,AGI(route_call.php)
exten => _X.,n,SetCIDNum(${GATEWAY_NUMBER})
exten => _X.,n,Monitor(wav,${CALL_ID},b)
exten => _X.,n,Dial(Zap/g1/${EXTEN},,HM(setchannel^${CALL_ID}))
exten => _X.,n,Hangup()
exten => t,1,Hangup()
exten => i,1,Hangup()
exten => h,1,StopMonitor()
exten => h,n,SoftHangup(${CHANNEL})
exten => h,n,SoftHangup(${BRIDGEPEER})
exten => h,n,System(/usr/bin/soxmix
/var/spool/asterisk/monitor/${CALL_ID}-in.wav
/var/spool/asterisk/monitor/${CALL_ID}-out.wav
/var/spool/asterisk/monitor/${CALL_ID}.wav)
exten => h,n,System(/usr/bin/lame -b 16
/var/spool/asterisk/monitor/${CALL_ID}.wav
/var/recordings/${CALL_ID}.mp3)
exten => h,n,System(/bin/rm -rf
/var/spool/asterisk/monitor/${CALL_ID}*.wav)
exten => h,n,DeadAGI(cleanup_call.php)
The call is monitored once the bridge starts and after hangup converted
to an mp3 and that all works great.
The problem I am having is the call and recording continues after the
agent hangs up their phone until the extension number hangs up their
phone. This is especially evident when an answering machine/voicemail
is called and the call recording can last from 10 to 60 seconds beyond
the real person hanging up depending on how long the answering machine
stays on the line.
All of the hangup commands in the dial plan are futile attempts to
shorten this time. What I am really looking for is for the 2nd leg of
the call to be forcibly hung up whenever the first leg of the call is
detected as hung up so my dial plan execution can continue.
I would imagine this would have to be something in the "bridging" code
but there doesn't seem to be a bridge command, only the Dial command.
Any thoughts?! Thanks so much.
Aaron Bostick
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[Asterisk-Users] Please help in writing AGI script
Hi,
I'm new in writing AGI script and actually newbie in Asterisk. I'm
writing a small script that will read the number inputed by the
caller of the extension 123. First he will dial number 123 then a
voice prompt will be played (welcome) then he should press number on
the softphone and the script will echo the number to the caller. Here
is my script:
#!/usr/bin/perl
use Asterisk::AGI;
$|=1;
$AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
$AGI->stream_file('welcome');
while(length($num) != 3) {
$num = $AGI->get_data("sayme", "1", "3");
$saythis = $num;
}
$AGI->say_number($saythis);
please correct my script if there is something wrong. but i think there is.
thanks,
ryan
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Re: [Asterisk-Users] diax not working properly
Thank you gentlemen for your prompt replies and help i really appreciate that. Actually i have only used asterisk-1.0.3 :) I will definitely visit these webpages and see if they help. I really like DIAX and i was to stick to it so if you can help solve my problem with diax??? Thanx Amna On 12/5/05, Time Bandit <[EMAIL PROTECTED]> wrote: > Hi!> I have been using Asterisk-1.0.3 for quite some time now.My main aim> nowadays is to make iax-iax calls for which i am usin DIAX soft phone.The> problem is that sometimes the phone doesn`t register and at others it gets> out of the registration(after being registere for some time).Can anyone tell> me what can be the problem ,what other iax phones are available ? I don't think your problem is DIAX, Dan is making a great phone and hetest it carefully. But anyway, since you asked, here is a short list :- MediaX (my own) : http://www.marccharbonneau.com/asterisk/mediaxphone.php- Idefix : http://www.asteriskguru.com/tools/idefisk_beta.php- IAX phone : used to be at this address : http://www.sokol-associates.com/IaxPhone.htm but the site changed andI lost track of it- MozIAX : plugin for Firefox/Mozilla : http://moziax.mozdev.org/- iaxComm : http://iaxclient.sourceforge.net/iaxcomm/hth>> Thanx and Regards,> Amna> ___ > --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>> http://lists.digium.com/mailman/listinfo/asterisk-users>>>___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
No, max we used is 30 channels. But according to voip-info its faster protocol because it offloads media processing to asterisk (which is a better choice I think) and only looks after H323 call setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 11:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] h323 vs oh323 Ok.. how many channels are you using? More than 100? Maybe it can be good only for 10 or 20 simultaneous connections... Isamar On Tue, 6 Dec 2005, Boris Bakchiev wrote: > I like the chan_ooh323. > I like the idea of selfcontained H323 channel that doesn't rely external > libraries, often with specific versions that conflict with something ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Preventing incoming calls from ringing SIP lines
Paul Redstone wrote: Hi We're using three line SIP phones (X-lite), very nice, with Asterisk 1.2 But we want to prevent either direct incoming calls or calls from other extensions from ringing if the user is in another incoming call (i.e incoming into the extension), making an outgoing call or even checking their voicemail. Just a newbie response, but what about the incominglimit= option in your /etc/asterisk/sip.conf? JES 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: Re: [Asterisk-Users] Best Switch for VOIP Applications
We installed a HP 2626 PWR for a customer about 18 months ago, and it seemed OK. There are probably much more options now though PaulH > Michael Welter <[EMAIL PROTECTED]> wrote: > > Ok, what's the best VoIP switch with PoE? Does anyone have experience > with the D-Link DES-1526? > > Wiley Siler wrote: > > What is your port density requirement? > > > > For 24 ports the LinkSys SRW2024 is awesome. > > They street for less than $500 and have good QoS. > > For a smaller switch, they make a 12 port variant. > > > > Wiley > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of calvis > > Sent: Monday, December 05, 2005 3:12 PM > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: [Asterisk-Users] Best Switch for VOIP Applications > > > > > > I need to replace my switch. Does anyone have any recommendations for > a > > switch that is VoIP friendly? I want it to be a managed gigabyte > > switch. > > There are lots of brands out there, but would prefer some > > recommendations from the list. > > > > > > -Charles > > > > ___ > > --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 > > > > > > > -- > Michael Welter > Telecom Matters Corp. > Denver, Colorado US > +1.303.414.4980 > [EMAIL PROTECTED] > www.TelecomMatters.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 ___ --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] Asterisk on PPC & chan_capi issue
Hi all,
I have a PPC box (IBM RS6000 43P-150, bigendian afaik) which runs Fedora
Core 5 Test1 and zaptel, libpri and asterisk 1.2.0. I also installed
chan_capi (0.6.1) so I can use my Eicon Diva Server BRI card. Asterisk
was compiled with DEBUG=-g and DEBUG_THREADS = -DDUMP_SCHEDULER
-DDEBUG_SCHEDULER-DDEBUG_THREADS -DDO_CRASH -DDETECT_DEADLOCKS. Next I
did make clean, make valgrind, make install. Asterisk runs as user/group
asterisk/asterisk.
SIP <--> SIP calls are fine, Calls from SIP out to the PSTN via
CAPI/ISDN are fine too. ISDN/CAPI --> SIP calls don't work. Example
output of the issue is below. Anyone have an idea how I fix this?
Thanks and regards,
Patrick
chan_capi registers fine:
**
[chan_capi.so] => (Common ISDN API for Asterisk)
== This box has 1 capi controller(s).
== Reading config for BRI1
-- ast_capi_pvt BRI1-pseudo-D (,,capi-in,0,2) (1,4,128)
-- ast_capi_pvt BRI1 (,,capi-in,0,2) (1,4,128)
-- ast_capi_pvt BRI1 (,,capi-in,0,2) (1,4,128)
-- listening on contr1 CIPmask = 0x1fff03ff
== Registered channel type 'CAPI' (Common ISDN API Driver ($Revision:
1.115 $) )
== Registered application 'capiCommand'
== Registered custom function VANITYNUMBER
Call from my GSM to a SIP phone (exten 1003) via ISDN/CAPI (MSN2):
**
== BRI1: Incoming call '' -> ''
-- Executing Macro("CAPI/BRI1/-0", "stdexten|1003|SIP/1003")
in new stack
-- Executing Dial("CAPI/BRI1/-0", "SIP/1003|10|TtwW") in new
stack
Dec 6 02:30:47 WARNING[28889]: channel.c:2494 ast_request: No
translator path exists for channel type SIP (native 65535) to 0
Dec 6 02:30:47 NOTICE[28889]: app_dial.c:1011 dial_exec_full: Unable to
create channel of type 'SIP' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("CAPI/BRI1/-0", "s-CHANUNAVAIL|1") in new
stack
-- Goto (macro-stdexten,s-CHANUNAVAIL,1)
-- Executing Goto("CAPI/BRI1/-0", "s-NOANSWER|1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing Answer("CAPI/BRI1/-0", "") in new stack
== BRI1: Answering for 703241494
-- Executing Wait("CAPI/BRI1/-0", "1") in new stack
Dec 6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping
incompatible voice frame on CAPI/BRI1/-0 of format alaw since our
native format has changed to unknown
Dec 6 02:30:47 NOTICE[28889]: channel.c:1893 ast_read: Dropping
incompatible voice frame on CAPI/BRI1/-0 of format alaw since our
native format has changed to unknown
[snipped tons more of these]
Dec 6 02:30:48 NOTICE[28889]: channel.c:1893 ast_read: Dropping
incompatible voice frame on CAPI/BRI1/-0 of format alaw since our
native format has changed to unknown
-- Executing VoiceMail("CAPI/BRI1/", "u1003") in new stack
Dec 6 02:30:48 WARNING[28889]: channel.c:2313 set_format: Unable to
find a codec translation path from unknown to gsm
Dec 6 02:30:48 WARNING[28889]: file.c:820 ast_streamfile: Unable to
open vm-theperson (format unknown): No such file or directory
== BRI1: CAPI Hangingup
> CAPI INFO 0x3490: Normal call clearing
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[Asterisk-Users] sipura "Vertical Service Activation Codes"
Do the Sipura "Vertical Service Activation Codes" have any meaning to the phone itself? It doesn't seem like they do anything, but that leaves me with the question why are they listed at all? I'm trying to reconcile asterisk's idea of some features like group pickup being on "*8" with the sipura's desire to have it on "*37". Which one is more common these days? Can I just make an extension and assign the pickup code to it? exten => *37,1,pickup(SOMETHING_TBD); -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --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] uip200 phone not work with 1.2
I have a handful of phones that work with 1.0.9. I was trying to upgrade to 1.2 and the UIP200 phones dont ring. below is my config for 1 phone. I tried it with and without the qualify=yes or qualify=no and did not seem to make a difference. still no ring. Any ideas on what might be the issue? THanks, Jerry ; Jerry Phone [528] type=friend dtmfmode=rfc2833; Choices are inband, rfc2833, or info username=something secret=something disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid="Jerry" <528> ___ --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] logging performance, important impact?
How important is the impact i could have if I have a single entry log file in /etc/asterisk/logger.conf wich loggs everything, even debug level. This is sometimes important to us because it helps us to make a track of the issues some times we have with the system. I just want to know if there is a considerable impact in performance because of the writing of the logs. Best Regards-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.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] Best Switch for VOIP Applications
Ok, what's the best VoIP switch with PoE? Does anyone have experience with the D-Link DES-1526? Wiley Siler wrote: What is your port density requirement? For 24 ports the LinkSys SRW2024 is awesome. They street for less than $500 and have good QoS. For a smaller switch, they make a 12 port variant. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of calvis Sent: Monday, December 05, 2005 3:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Best Switch for VOIP Applications I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --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 -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.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] Is a BUG ? Hints and incominglimit
which version of Asterisk do you have ?, Becouse when i change the function to your code, every time that one phone with call-limit the Asterisk crash.
I have 1.2.0
On 12/3/05, Paradise Dove <[EMAIL PROTECTED]> wrote:
hi,This is the new update_call_counter() which works fine for me:/*! \brief update_call_counter: Handle call_limit for SIP users
* Note: This is going to be replaced by app_groupcount* Thought: For realtime, we should propably update storage with inusecounter... */static int update_call_counter(struct sip_pvt *fup, int event){
char name[256]; int *inuse, *call_limit; int outgoing = ast_test_flag(fup, SIP_OUTGOING); struct sip_user *u = NULL; struct sip_peer *p = NULL; if (option_debug > 2) ast_log(LOG_DEBUG, "Updating call counter for %s call\n",
outgoing ? "outgoing" : "incoming"); /* Test if we need to check call limits, in order to avoid realtime lookups if we do not need it */ if (!ast_test_flag(fup, SIP_CALL_LIMIT))
return 0; ast_copy_string(name, fup->username, sizeof(name)); /* Check the list of users */ // paradise dove p = find_peer(name, NULL, 1); if (p) { inuse = &p->inUse;
call_limit = &p->call_limit; } else if (!u) { /* Try to find user */ u = find_user(name, 1); if (u) { inuse = &u->inUse; call_limit = &u->call_limit;
} else { if (option_debug > 1) ast_log(LOG_DEBUG, "%s is not a local user, no calllimit\n", name); return 0; } } switch(event) {
/* incoming and outgoing affects the inUse counter */ case DEC_CALL_LIMIT: if ( *inuse > 0 ) { (*inuse)--; } else { *inuse = 0; }
if (option_debug > 1 || sipdebug) { ast_log(LOG_DEBUG, "Call %s %s '%s' removed from calllimit %d\n", outgoing ? "to" : "from", u ? "user":"peer"
} break; case INC_CALL_LIMIT: if (*call_limit > 0 ) { if (*inuse >= *call_limit) { ast_log(LOG_ERROR, "Call %s %s '%s' rejected due
to usage limit of %d\n", outgoing ? "to" : "from", u ? "u // paradise dove if (p) ASTOBJ_UNREF(p,sip_destroy_peer);
else if (u) ASTOBJ_UNREF(u,sip_destroy_user); return -1; } } (*inuse)++; if (option_debug > 1 || sipdebug) {
ast_log(LOG_DEBUG, "Call %s %s '%s' is %d out of%d\n", outgoing ? "to" : "from", u ? "user":"peer", name, *in } break;
default: ast_log(LOG_ERROR, "update_call_counter(%s, %d) calledwith no event!\n", name, event); } // paradise dove if (p) ASTOBJ_UNREF(p,sip_destroy_peer);
else if (u) ASTOBJ_UNREF(u,sip_destroy_user); return 0;}Paradise DoveOn 12/2/05, Alvaro Parres <[EMAIL PROTECTED]> wrote:> Could you send it patch please.
> On 11/30/05, Paradise Dove <[EMAIL PROTECTED]> wrote:> >> > btw, i've patched this part of code and now its working fine for me.
> > i'm going to upload it.> >> > Paradise Dove> >> > On 11/30/05, Kevin Hanson <[EMAIL PROTECTED]> wrote:> > > Paradise Dove wrote:
> > >> > > >>Yes with version 1.2. I have tried already with call-limit and the> same.> > > >>> > > >>> > > >i agree with you, it seems to be a bug which i've submited before (bug
> > > >#5281) but it's now closed by bug marshals!> > > >> > > >> > > >> > > It's not closed. It's suspended waiting input from you:> > >
> > > "Closing until the appropriate debug/trace output can be provided."> > >> > > On 10/30 you said you were still trying to get the debug output.> > >> > > Cheers,
> > > Kevin> > > ___> > > --Bandwidth and Colocation provided by Easynews.com --> > >> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:> > >> http://lists.digium.com/mailman/listinfo/asterisk-users> > >
> > ___> > --Bandwidth and Colocation provided by Easynews.com --> >> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:> >> http://lists.digium.com/mailman/listinfo/asterisk-users> >>
>> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users>>>___
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Re: [Asterisk-Users] Messages button on a Polycom 501
Actually I think that is how it is setup now. I configured the phone through the web interface. Callback mode is set to "contact" and the Callback contact is set to the extension. On 12/5/05, Jerry Jones <[EMAIL PROTECTED]> wrote: I assume you wish to have the button retrieve your vm - if so thentime to edit your config file, or use web interface. msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi. 1.callBack="xxx"xxx=extension to dial to retrieve vmOn Dec 5, 2005, at 5:38 PM, Brent Bloodworth wrote:> Need a little help. Just set up an [EMAIL PROTECTED] box with 5 Polycom> 501 phones. Everything works great except the messages button which > when pressed results in asterisk responding "Person at extension> 102 is on the phone. Please leave a message after the tone. I have> searched the web and several of the the asterisk mailing list > archive pages - but I haven't had any luck. Anyone have a suggestion?> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
make http://www.voip-info.org your friend.. http://www.voip-info.org/wiki-Asterisk+H323+channels Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: So, we have h323, oh323 and ooh323 I knew about h323 and oh323 but didn't know about ooh323. What is URL of ooh323, I want to know more about them. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 6 Dec 2005 09:16:05 +1100 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] h323 vs oh323 I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something else. OOH323 works "right out of box" and since we started using it to interconnect Asterisk to Samsung OfficeServ 500 we had no problems whatsoever. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 08:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
Ok.. how many channels are you using? More than 100? Maybe it can be good only for 10 or 20 simultaneous connections... Isamar On Tue, 6 Dec 2005, Boris Bakchiev wrote: I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something else. OOH323 works "right out of box" and since we started using it to interconnect Asterisk to Samsung OfficeServ 500 we had no problems whatsoever. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 08:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Grandstream NTP
It now appears to be server specific. The shipped default, time.nist.gov, appears to work OK. Does anyone know of anything specific required by these grandstream phones as far as NTP server support goes? On Tue, 6 Dec 2005 10:34 am, Rod Bacon wrote: > All my BT101's and GXP2000's are failing NTP update. My NTP server is on my > local LAN (and I've tried external ones), DNS is OK (and I've used IP > address instead of DNS name). > > tcpdump on NTP server shows valid request, AND a valid response, yet the > phones still display 02-01-1900. > > I have tried latest (and BETA firmware). > > Does anyone have any ideas? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --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] Messages button on a Polycom 501
I assume you wish to have the button retrieve your vm - if so then time to edit your config file, or use web interface. msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi. 1.callBack="xxx" xxx=extension to dial to retrieve vm On Dec 5, 2005, at 5:38 PM, Brent Bloodworth wrote: Need a little help. Just set up an [EMAIL PROTECTED] box with 5 Polycom 501 phones. Everything works great except the messages button which when pressed results in asterisk responding "Person at extension 102 is on the phone. Please leave a message after the tone. I have searched the web and several of the the asterisk mailing list archive pages - but I haven't had any luck. Anyone have a suggestion? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADIT 600 T1 with DNIS digits problem
Well we just finished turning up the first additional
T1 and now I'm seeing problems with DNIS digits.
We have a T1 split into 3 trunk groups, the second
two trunk groups are to be connected to other
equipment. The first trunk group (8 channels) is
for inbound traffic to Asterisk.
telco <-> ADIT 600 <-> T4XXP(*)
The trunk group is setup with 4 DNIS digits to
be passed to the PBX.
zaptel.conf:
span=1,0,0,esf,b8zs,yellow
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs
span=4,0,0,esf,b8zs
# Span 1 - Not working on card
unused=1-24
# Span2 - ADIT 600 - Modular unit, first card is FXO
fxsks=25-32# FXO Card
fxogs=33-40# FXS Card
# not connected
unused=41-48
# Span 3
e&m=49-56 # inbound trunk 1 - Office voice
unused=57-64 # inbound trunk 2 - Transaction Modems
unused=65-72 # inbound trunk 3 - FTP partial T1
# Span 4
unused = 73-96
loadzone = us
defaultzone=us
zapata.conf:
group = 3
signalling=em_w ; featd
usecallerid=no
context = inbound
channel => 49-56
If in exentions.conf I have:
[inbound]
exten => _5948,1,Goto(incoming,s,2)
I get an invalid extension response, using this:
[inbound]
exten => _594,1,Goto(incoming,s,2)
It gets to the incoming context then complains about
"8" being an invalid extension.
Using this:
[inbound]
exten => _X.,1,Wait(1)
exten => _X.,2,NoOp(${DNIS})
exten => _X.,3,NoOp(${EXTEN})
exten => _X.,4,SetVar(EXTEN="s")
exten => _X.,5,Goto(incoming,s,2)
I still get the invalid extension from the incoming
context.
Trying to use the featd signaling gives:
Dec 5 16:58:43 WARNING[9713]: chan_zap.c:4786 ss_thread: Got a
non-Feature Group D input on channel 56. Assuming E&M Wink instead
Using Asterisk 1.0.9
Any suggestions?
Thanks,
William.
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Re: [Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message
Andrew Kohlsmith wrote: On Monday 05 December 2005 13:39, Colin Anderson wrote: That appears to work *perfectly* but I don't get it. With the 'r' option on, how can Asterisk determine that the user has answered the phone as opposed to the carrier? Is it a signal that the carrier is sending? Anyway, thanks. Works like a hot damn. With the carrier voicemail turned off (not subscribed to) the carrier does not answer the line to say "this person is out of the service area or has their phone off" -- it's the same trick (early audio) used with digital lines to inform the caller of a problem without charging them for the privilege. This is the ONLY use for the "r" option of Dial that I have found. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Switch for VOIP Applications
Michiel van Baak wrote: On 14:42, Mon 05 Dec 05, snacktime wrote: On 12/5/05, calvis <[EMAIL PROTECTED]> wrote: I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. We use the Fastiron workgroup swiches and really like them. Very solid but a tad expensive. little expensive but also good are the cisco's. They play very nice with the 79XX series. Add PoE to that and you can really see why I like setups like that. I have no experience with the gbit line of cisco's though. We don't need GigE. We use Cat 5505 and 5509 switches. Dirt cheap from eBay. ___ --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] hierarchical VoIP system
And about the protocol used to create this hierarchical network? Should I use SIP (routing between SERs) or should I use IAX (routing between Asterisks)? About ENUM, Isnt the managing of the ENUM tree going to be very complicated and heavy when we reach the millions of users? Joao Jan Saell wrote: Hi there! We have kind of the same setup but are using a few number of SER boxes for the on net calls - using enum for the lookup would be a great idea so that you can get the numbers to do sip calls and move over slowly. And for the central routing voip server make the routing use SIP redirects as the central server then can handle a lot of calls as its only doing the routing decisions. Best regards jan --On Wednesday, November 30, 2005 05:45:21 PM + Joao Pereira <[EMAIL PROTECTED]> wrote: Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao Pereira www.fccn.pt ___ --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 ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ --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] Messages button on a Polycom 501
Need a little help. Just set up an [EMAIL PROTECTED] box with 5 Polycom 501 phones. Everything works great except the messages button which when pressed results in asterisk responding "Person at extension 102 is on the phone. Please leave a message after the tone. I have searched the web and several of the the asterisk mailing list archive pages - but I haven't had any luck. Anyone have a suggestion? ___ --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] Grandstream NTP
All my BT101's and GXP2000's are failing NTP update. My NTP server is on my local LAN (and I've tried external ones), DNS is OK (and I've used IP address instead of DNS name). tcpdump on NTP server shows valid request, AND a valid response, yet the phones still display 02-01-1900. I have tried latest (and BETA firmware). Does anyone have any ideas? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Switch for VOIP Applications
Cisco owns Linksys so they have some good features now. 64 VLANs, 8 port trunking groups, console port, 802.1p CoS support Four Quality of Service egress queues per port let you prioritize traffic via 802.1p. http://www1.linksys.com/products/product.asp?grid=35&scid=40&prid=673 This can be found for close to $400. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Monday, December 05, 2005 3:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best Switch for VOIP Applications Wiley Siler wrote: >What is your port density requirement? > >For 24 ports the LinkSys SRW2024 is awesome. >They street for less than $500 and have good QoS. >For a smaller switch, they make a 12 port variant. > > Does the SRW2024 support port mirroring? I was shopping around, but couldn't find any Linksys switch that support port mirroring. I ended with the DLINK DES-1226G which retails for a lot less than the SRW2024 (over here we can get it for Wiley > > >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of calvis >Sent: Monday, December 05, 2005 3:12 PM >To: 'Asterisk Users Mailing List - Non-Commercial Discussion' >Subject: [Asterisk-Users] Best Switch for VOIP Applications > > >I need to replace my switch. Does anyone have any recommendations for >a switch that is VoIP friendly? I want it to be a managed gigabyte >switch. >There are lots of brands out there, but would prefer some >recommendations from the list. > > >-Charles > >___ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >___ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Switch for VOIP Applications
On 14:42, Mon 05 Dec 05, snacktime wrote: > On 12/5/05, calvis <[EMAIL PROTECTED]> wrote: > > > > I need to replace my switch. Does anyone have any recommendations for a > > switch that is VoIP friendly? I want it to be a managed gigabyte switch. > > There are lots of brands out there, but would prefer some recommendations > > from the list. > > We use the Fastiron workgroup swiches and really like them. Very > solid but a tad expensive. little expensive but also good are the cisco's. They play very nice with the 79XX series. Add PoE to that and you can really see why I like setups like that. I have no experience with the gbit line of cisco's though. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called 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] Include a variable from another file in configfiles
JP Carballo wrote: amaury BOSSE wrote: Thanks for your answer but I don't want to include a file, I only want to include a variable. Is it possible to execute linux commands like grep or top in a .conf file in order to parse a file and get a variable? Look into the System() command: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+System Oops, I missed the "get a variable" part. Your best bet is to use AGI. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Switch for VOIP Applications
On 12/5/05, calvis <[EMAIL PROTECTED]> wrote: > > I need to replace my switch. Does anyone have any recommendations for a > switch that is VoIP friendly? I want it to be a managed gigabyte switch. > There are lots of brands out there, but would prefer some recommendations > from the list. We use the Fastiron workgroup swiches and really like them. Very solid but a tad expensive. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Switch for VOIP Applications
Wiley Siler wrote: What is your port density requirement? For 24 ports the LinkSys SRW2024 is awesome. They street for less than $500 and have good QoS. For a smaller switch, they make a 12 port variant. Does the SRW2024 support port mirroring? I was shopping around, but couldn't find any Linksys switch that support port mirroring. I ended with the DLINK DES-1226G which retails for a lot less than the SRW2024 (over here we can get it for 802.1q) and port mirroring. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of calvis Sent: Monday, December 05, 2005 3:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Best Switch for VOIP Applications I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
So, we have h323, oh323 and ooh323 I knew about h323 and oh323 but didn't know about ooh323. What is URL of ooh323, I want to know more about them. Thanks, -- You don't have any choice, you already made it before you came here. > -Original Message- > From: [EMAIL PROTECTED] > Sent: Tue, 6 Dec 2005 09:16:05 +1100 > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] h323 vs oh323 > > I like the chan_ooh323. > I like the idea of selfcontained H323 channel that doesn't rely external > libraries, often with specific versions that conflict with something > else. > > OOH323 works "right out of box" and since we started using it to > interconnect Asterisk to Samsung OfficeServ 500 we had no problems > whatsoever. > > regards > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: Tuesday, 6 December 2005 08:11 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] h323 vs oh323 > > > Try chan_oh323 and if it is not ok, try chan_h323 > Both work well in different situations/equipments. > > > Isamar > > On Mon, 5 Dec 2005, Innocent Evil wrote: > >> Hello, >> >> Would you please share your experience regarding h323 and oh323 in > asterisk. >> I am confused to choose one. >> >> Thanks, >> >> >> -- >> You don't have any choice, you already made it before you came > here.___ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --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] Best Switch for VOIP Applications
I have a 24 port that is doing well for us. I will check out the LinkSys. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Monday, December 05, 2005 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Best Switch for VOIP Applications What is your port density requirement? For 24 ports the LinkSys SRW2024 is awesome. They street for less than $500 and have good QoS. For a smaller switch, they make a 12 port variant. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of calvis Sent: Monday, December 05, 2005 3:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Best Switch for VOIP Applications I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Switch for VOIP Applications
What is your port density requirement? For 24 ports the LinkSys SRW2024 is awesome. They street for less than $500 and have good QoS. For a smaller switch, they make a 12 port variant. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of calvis Sent: Monday, December 05, 2005 3:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Best Switch for VOIP Applications I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --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] h323 vs oh323
I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something else. OOH323 works "right out of box" and since we started using it to interconnect Asterisk to Samsung OfficeServ 500 we had no problems whatsoever. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 08:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: > Hello, > > Would you please share your experience regarding h323 and oh323 in asterisk. > I am confused to choose one. > > Thanks, > > > -- > You don't have any choice, you already made it before you came here.___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Best Switch for VOIP Applications
I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. -Charles ___ --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] Include a variable from another file in configfiles
amaury BOSSE wrote: Thanks for your answer but I don't want to include a file, I only want to include a variable. Is it possible to execute linux commands like grep or top in a .conf file in order to parse a file and get a variable? Look into the System() command: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+System -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] PRI indications.
Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk "unallocated number" but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16 bchan=1-10 alaw=1-10 loadzone=pl defaultzone=pl My /etc/asterisk/zapata.conf: [channels] language=en context=from-pstn switchtype=euroisdn signalling=pri_cpe pridialplan=local usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=no cancallforward=no callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no priindication=outofband group = 1 channel => 1-10 Regards, Adam Rybak ___ --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 + WiFi Phones
I'm curious if anything new has been determined on this phone? Is it SIP compatible with Asterisk and, say, Broadvoice? I'm a little wary that this may be vaporware. The phone doesn't seem to be listed by the FCC. But, I would preorder one if it's Asterisk and Broadvoice compatibile. Phil PS- Contact us form on the viopsupply site seems to be broken? Just spins for me. Cory Andrews wrote: The F3000 is also a clamshell, "flip" type phone. I should be receiving an eval unit shortly and will post my findings after we work it over in the lab. Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Luki wrote: UTStarCom has the F3000 coming in December, which will have according to their spec * WEP (64 and 128 bit )/WPA/MD5 Auth * Handover/Roaming between different AP and SSID So what else is different compared to the F1000? The 1000 also does WEP 64/128 and WPA with the newest firmware. Not sure about MD5 auth, but SIP nonce/MD5 response certainly is implemented. Roaming kind of works, but could be improved. In one place I made it from 4th floor -> elevator -> lobby while on the phone and without any noticeable dropouts (ulaw codec). But the building was covered with access points, on average NetStumbler saw 6 at the same time. So it works, but not always. Don't get me wrong, the phone does have issues and in my opinion is not production quality, meaning it will freak out unexpectedly and only a reboot helps, which hardly ever happens to any Sipura adapters or phones. Hopefully the new 3.6 firmware performs better. Luki ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Linksys SPA-941 DTMF failure with asterisk v.1.2
One other piece of information that I just stumbled on while doing a packet capture which may explain the whole thing:
The Cisco packet shows the RTP event as this:
RFC 2833 RTP Event
Event ID: DTMF Pound # (11)
End of Event: True
Reserved: False
Volume: 10
Event Duration: 1600
The Linksys packet shows the following:
RFC 2833 RTP Event
Event ID: DTMF Pound # (11)
End of Event: True
Reserved: False
Volume: 0
Event Duration: 1760
Notice the volume setting in the Linksys packet. Could this be
the issue? I have changed every DTMF-related setting in the
Linksys that I can think of with no change in behavior.
What still doesn't make sense to me is that why would this not work
with asterisk 1.2 yet still work when used with asterisk 1.0.x?
On 12/5/05, tracinet <[EMAIL PROTECTED]> wrote:
Been working on testing asterisk 1.2 before upgrading our production
systems from 1.0.x and have found a few issues. The one I am
working on
now involves DTMF failure with the following setup:
*Linksys SPA-941* ---SIP---> *asteriskA(v.1.2)* IAX2> *asteriskB(v.1.2)* SIP-> *Global Crossing* (PSTN)
g711 with RFC 2833 out of band DTMF is used throughout the entire setup
from the Linksys to Global Crossing. Asterisk servers are using
asterisk SVN 1.2 from Friday. asteriskA is used as a SIP
registrar server for SIP devices to connect and asteriskB is used as a
gateway to our SIP provider.
In order to test DTMF at each stage, I set up the following so asterisk could playback which digits I entered:
; Test DTMF
exten => 123,1,Read(NUMBER)
exten => 123,2,SayDigits(${NUMBER})
exten => 123,3,Goto(1)
Here are the tests I ran and the results
*Linksys SPA-941* ---SIP---> *asteriskA(v.1.2)*
Test Passed - DTMF detected with no problem
*Linksys SPA-941* ---SIP---> *asteriskA(v.1.2)* IAX2> *asteriskB(v.1.2)*
Test Passed - DTMF detected with no problem
*Linksys SPA-941* ---SIP---> *asteriskA(v.1.2)* IAX2> *asteriskB(v.1.2)* SIP-> *Global Crossing* (PSTN)
Test Failed - poor DTMF accuracy
I then trying reverting asteriskB to version 1.0.x of asterisk and surprisingly, DTMF worked fine:
*Linksys SPA-941* ---SIP---> *asteriskA(v.1.2)* IAX2> *asteriskB(v.1.0)* SIP-> *Global Crossing* (PSTN)
Test Passed - DTMF detected with no problem
I then tried using a Cisco 7960 in place of the Linksys SPA-941 and all worked fine there as well:
*Cisco 7960* ---SIP---> *asteriskA(v.1.2)* IAX2> *asteriskB(v.1.2)* SIP-> *Global Crossing* (PSTN)
Test Passed - DTMF detected with no problem
One would think the issue is with the SIP provider (Global Crossing)
but what makes it odd is that DTMF fails only when using the
Linksys and only when using version 1.2 of asterisk. So for now I am ruling out Global Crossing.
Any thoughts?
PS: Bug 5780 states that it is related to g729, not g711 which is in use here.
___
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RE: [Asterisk-Users] h323 vs oh323
I am still having a non-solved problem with Oh323/h323 and checking Digium homepage after a long time, it looks like they need some dimes now to support me in this case. I have 46(2 T1) PSTN channels receiving calls through H323 protocol. With oh323, after 40 channels in use, It crashes due to some bug related to the limit of file handles. Even playing with some high values in /proc/sys/fs/file-max, didn't solve. With chan_h323, I don't have this problem but, I have this one: localhost*CLI> show channels Channel Location State Application(Data) Zap/20-1 [EMAIL PROTECTED]:1 Up Bridged Call(H323/ip$a.b.c.d) 1 active channel 5 active calls I have only one active channel but 5 active calls?! Asterisk version 1.2.0 with H323 and the same pwlib/H323 libs recommended by the README. Checking the logs, I have tons of these errors: Dec 6 00:36:17 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:18 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:19 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:20 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:21 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:22 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! And this one too: Dec 6 00:36:18 WARNING[31530] channel.c: Prodding channel 'H323/ip$202.83.196.25:32791/31907' failed How to solve this problem? Isamar On Mon, 5 Dec 2005, David Waugh wrote: Prely subjective, but I first installed h323 and it worked. Somewhere along the line something happened and it no longer worked. Recompiling it etc seemed to have no effect. I then tried oh323 and it worked first time and has stayed working. I probably did soemthing wrong, but oh323 seems to work for me. Thanks David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Innocent Evil Sent: 05 December 2005 14:36 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] h323 vs oh323 Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys SPA-841 Missing Calls
> Subject: RE: [Asterisk-Users] Linksys SPA-841 Missing Calls "Dave Morrow" <[EMAIL PROTECTED]> wrote: > I've narrowed it down to the phones dislike for my older 3COM switch. > I noticed on the weekend that when these missed calls occur, if I ping > the phone, the first few packets are dropped..almost like it's > gone to sleep.. We have had some network issues with our SPA-841's as well. We ended up having to take the phone off our standard network. Even though it was a completely switched network, we believe sufficient ARP broadcasts packets were being sent to the phones to slow them down. Our symptom was choppy or "robotic" sound similar to what you'd expect with high packet loss, accompanied by extremely high "decode latency" numbers on the System page. Even that wasn't enough: we needed higher quality switches than the cheap ones we expected to be able to use, to avoid other sound quality issues which continued to crop up. This is good evidence as to why they didn't put 2 ethernet ports on the phone: it would only make things worse if you shared the port with a PC or workstation, I'd expect. Alan ___ --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] h323 vs oh323
Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys SPA-941 DTMF failure with asterisk v.1.2
Been working on testing asterisk 1.2 before upgrading our production
systems from 1.0.x and have found a few issues. The one I am
working on
now involves DTMF failure with the following setup:
*Linksys SPA-941* ---SIP---> *asteriskA(v.1.2)* IAX2> *asteriskB(v.1.2)* SIP-> *Global Crossing* (PSTN)
g711 with RFC 2833 out of band DTMF is used throughout the entire setup
from the Linksys to Global Crossing. Asterisk servers are using
asterisk SVN 1.2 from Friday. asteriskA is used as a SIP
registrar server for SIP devices to connect and asteriskB is used as a
gateway to our SIP provider.
In order to test DTMF at each stage, I set up the following so asterisk could playback which digits I entered:
; Test DTMF
exten => 123,1,Read(NUMBER)
exten => 123,2,SayDigits(${NUMBER})
exten => 123,3,Goto(1)
Here are the tests I ran and the results
*Linksys SPA-941* ---SIP---> *asteriskA(v.1.2)*
Test Passed - DTMF detected with no problem
*Linksys SPA-941* ---SIP---> *asteriskA(v.1.2)* IAX2> *asteriskB(v.1.2)*
Test Passed - DTMF detected with no problem
*Linksys SPA-941* ---SIP---> *asteriskA(v.1.2)* IAX2> *asteriskB(v.1.2)* SIP-> *Global Crossing* (PSTN)
Test Failed - poor DTMF accuracy
I then trying reverting asteriskB to version 1.0.x of asterisk and surprisingly, DTMF worked fine:
*Linksys SPA-941* ---SIP---> *asteriskA(v.1.2)* IAX2> *asteriskB(v.1.0)* SIP-> *Global Crossing* (PSTN)
Test Passed - DTMF detected with no problem
I then tried using a Cisco 7960 in place of the Linksys SPA-941 and all worked fine there as well:
*Cisco 7960* ---SIP---> *asteriskA(v.1.2)* IAX2> *asteriskB(v.1.2)* SIP-> *Global Crossing* (PSTN)
Test Passed - DTMF detected with no problem
One would think the issue is with the SIP provider (Global Crossing)
but what makes it odd is that DTMF fails only when using the
Linksys and only when using version 1.2 of asterisk. So for now I am ruling out Global Crossing.
Any thoughts?
PS: Bug 5780 states that it is related to g729, not g711 which is in use here.
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[Asterisk-Users] Preventing incoming calls from ringing SIP lines
Hi We're using three line SIP phones (X-lite), very nice, with Asterisk 1.2 But we want to prevent either direct incoming calls or calls from other extensions from ringing if the user is in another incoming call (i.e incoming into the extension), making an outgoing call or even checking their voicemail. In 1.0 the SetGroup and CheckGroup commands could do this but you have to build it into all parts of the dial plan. In 1.2 these do not exist and the Set(Group type commands with GotoIf are supposed to be used. But I still have not seen anywhere a full example of this. There is the call-limit setting in SIP - beautiful, works at the SIP level so easier than the dial plan. BUT with this you cannot do attended or blind transfers - not sensible. This must be a very common requirement, certainly is judging from the posts but in hours of searching I have not see the sort of complete solution which looks feasible. Thanks and sorry if I've missed it. Alternatively I'd be happy to use single line SIP softphones but cannot find one which feels good. TIA Paul R ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfers from Polycom 501 involving Sipura 300 and asterisk 1.2
When transferring a call that came in on the Sipura and picked up by a Polycom 501 (sip 1.52), then transferred to another polycom using the transfer button on the polycom (havn't tried with the blind transfer from the polycom phone), then as soon as the transfer is complete (after pressing transfer again on the polycom) then the caller on the Sipura side can hear the new polycom caller, but the polycom cannot hear the sipura caller. This is all on a flat network, no nat, no gateways, between any of the points. If I change canreinvite=no for the sipura then everyting works fine. I'm assuming this is a bug in 1.2, but before I jump to conclusions I would like to know if anyone else has seen this? I did not yet have a chance to capture the output, but will do so if needed. Thank You ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error when compiling asterisk
Any help on this pleaseHi, I am getting this error when compiling asterisk `ls *.c`: unrecognized option h -DBUSYDETECT_MARTIN `ls *.c`Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ...GNU long options: --debug --dump-po-strings --dump-strings --help --login --noediting --noprofile --norc --posix --rcfile --rpm-requires --restricted --verbose --version --wordexpShell options: -irsD or -c command (invocation only) -abefhkmnptuvxBCHP or -o optionmake: *** [.depend] Error 2 Any ideas of what the problem might be. Thank you Appe l audio GRATUIT partout dans le monde avec le nouveau Yahoo! MessengerTéléchargez le ici ! ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Biz mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-biz Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! MessengerTéléchargez le ici ! ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez le ici ! ___ --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] Anyone know anything about the new Linksys One product - does it use Asterisk?
You can find more information at http://www.linksysone.com/ Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Chuck Bunn wrote: Hi, Does anyone have any details about the Linksys one product that was just announced? Does it use Asterisk? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Anyone know anything about the new Linksys One product - does it use Asterisk?
No it does not user Asterisk. It is a proprietary system based around the Call Manager products. Linksys sells the system to a service provider who then offers the service to end users. Basically, LinksysOne is a means by which service providers can offer a hosted PBX solution. Kerry Garrison Publisher - GeekGazette.com - VOIPSpek.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, December 05, 2005 11:28 AM To: Asterisk - Users Subject: [Asterisk-Users] Anyone know anything about the new Linksys One product - does it use Asterisk? Hi, Does anyone have any details about the Linksys one product that was just announced? Does it use Asterisk? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] DISA function
I had a problem with DTMF with DISA.. I am using a Sipura SPA 3000 for the line. I set the FXO port impedance (on the PSTN line tab) to 900 as advised by others and it worked. Having said that, I'm sure you will be using some other FXO adapter.. Just thought I'd tell. - Original Message - From: Richard Smith To: asterisk-users@lists.digium.com Sent: Monday, December 05, 2005 01:44 Subject: [Asterisk-Users] DISA function Hi all, I was wondering whether the DISA function on the latest asterisk 1.2 stable release actually works better than the other prior releases. Basically the [EMAIL PROTECTED] version 2.0 BETA 4 I'm using does not recognise the DTMF tones all the time and sometime when it does, it disconnects. Cheers, Richard. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone know anything about the new Linksys One product - does it use Asterisk?
Hi, Does anyone have any details about the Linksys one product that was just announced? Does it use Asterisk? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] video phones
Anyone using any H.263+ video phones and want to relay their experiences? -Jonathan ___ --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] Looking for advice on cell carrier's default "Un avaliable" message
On Monday 05 December 2005 13:39, Colin Anderson wrote: > That appears to work *perfectly* but I don't get it. With the 'r' option > on, how can Asterisk determine that the user has answered the phone as > opposed to the carrier? Is it a signal that the carrier is sending? > Anyway, thanks. Works like a hot damn. With the carrier voicemail turned off (not subscribed to) the carrier does not answer the line to say "this person is out of the service area or has their phone off" -- it's the same trick (early audio) used with digital lines to inform the caller of a problem without charging them for the privilege. -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] Panasonic DBS DISA
Hopefully, someone here has dealt with a Panasonic DBS in this way. I have put an Asterisk server in front of our Panasonic DBS phone system. The goal is to phase out our DBS, but during the transition, I still need to have asterisk extensions access some features of our Panasonic. The two features in question are paging though the Panasonic DBS and pickup of parked calls. The T1 card in my Panasonic sees Asterisk as a CO, but is also configured to send 56XX and 57XX directly out the T1, so I can call from system to system transparently. Also, (I have not decided yet) I may keep the Panasonic indefinitely just for paging and for the analog extensions for fax, etc. I assume that I have two options: 1. Use DISA in the Panasonic DBS and have an *9001 (Panasonic code for pickup park pos. 1) extension in Asterisk to dial into the Panasonic, log into DISA and dial *9001 in the Panasonic. Then do similar for other park positions and paging. I am having trouble figuring out DISA in the Panasonic. 2. Configure an analog station port on asterisk and connect it directly to an analog extension on the Panasonic to send these Panasonic codes. The catch here is that I only have so many analog extensions on the Panasonic and may not have one available. Also, I have no more slots in my Asterisk to put in an analog card to do this with. Also, I think that the iaxy, etc. can only be used as analog CO ports. Factoring the issues with above, the DISA over T1 would seem the best if I could get it to work. Has anyone here dealt with DISA on a Panasonic DBS? -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message
>Turn off voicemail on his cell phone, give out his DID instead of his cell #. >Send an SMS to his cellphone when new voicemail is left. That's what we do now. Works fine. >As far as Dial()ing his cell goes, use 'r' (this is exactly what it's designed >for) so that when the carrier is saying "The person you're calling is out of >the calling area or has his phone off" all the caller hears is ringing. That appears to work *perfectly* but I don't get it. With the 'r' option on, how can Asterisk determine that the user has answered the phone as opposed to the carrier? Is it a signal that the carrier is sending? Anyway, thanks. Works like a hot damn. ___ --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 1.2 problems
Thanks! It looks like you were right. We placed the phones and PBX on a minimal, physically separate network and have had no problems. We were using a 3com unmanaged switch but have ordered an HP managed switch with VLANs and VoIP QoS capabilities. We couldnt find anything about Shadow ping, is this an app? Is it useful? Also, this issue sounds like a good argument against the use of soft phones since you would be unable to segregate voice and data, right? Thanks, Tim > On Fri, 2005-12-02 at 14:22, [EMAIL PROTECTED] wrote: >> Help! I've encountered some problems with Asterisk that IÂm unable to >> solve. We have been running Asterisk version 1.0.9 for many months >> using a few local network connected Cisco 7960 phones as SIP clients. >> All our phones are currently internal so there is no NAT involved. We >> were not having any problems until last week when some strange issues >> started to crop up. I started experiencing calls that I initially >> believed were being dropped, but discovered that only one side of the >> conversation had dropped. The other party could hear me but I couldn't >> hear them. This seems to happen more often on longer calls but is not >> consistent. I am also seeing issues where incoming or local extension >> calls that are hung up by the originator before being answered will >> continue to ring the SIP phone. At the time the errors occur, the >> Asterisk console displays a variety of "...retrans_pkt: Maximum retries >> exceeded on call.." messages. I scoured the forums for an answer, found >> many refere > nce >> s to these errors, tried every suggested fix that I could find, but >> none have resolved these problems. After working on the problem for >> several days, I finally built a new box and installed Asterisk 1.2 on >> it. Using this new 1.2 box I no longer see the "Maximum retries >> exceeded on call" warnings on the console but still experience the >> strange behavior. Unfortunately, the errors occur randomly so I am >> unable to reproduce the error on demand. I turned on SIP debugging and >> set console logging to debug and captured an instance of the problem >> with the hang up not being recognized. The details are below: >> >> I dial in from my cell phone. My Cisco phone begins to ring. I then >> hang up my cell phone. Asterisk acknowledges the hang up, but the Cisco >> phone continues to ring. After a minute or so, or if I pickup the >> phone, Asterisk display the following message "That's odd... Got a >> response on a call we donÂt know about. Cseq 102 Cmd SIP/2.0" I've >> included a copy of the console output when this occurs that shows both >> the SIP message and the Asterisk debug output. > > Odds are you have local network congestion -- Dropped packets or delayed > packets. Try moving your phone and asterisk server to an isolated network > switch - no other traffic (certainly no computers) - then test. > > If the problems go away, then update your virus scanners and check your > computers. > > Good Luck > > Jon Carnes > > ___ --Bandwidth and Colocation > provided by Easynews.com -- > > Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk 1.2 problems ([EMAIL PROTECTED])
We are using firmware version 6.3. Dont we need a service agreement to get the latest drivers? We let ours expire since we werent having any problems. Isnt it also true that once you upgrade the firmware there is no way to revert to an earlier version? This is worrisome because we have heard of "bad versions" and do not want to upgrade without having a back out plan. Thanks, Tim > What version firmware are you running on your Cisco Phones? We are > running Asterisk 1.2 with the 7.4 firmware. The latest is 7.5 but there > are some strange things that happen with this firmware. If I were you I > would try a different firmware on the phones. Hope this helps. Jeremiah > > > >> Help! I've encountered some problems with Asterisk that Iâm unable to >> solve. We have been running Asterisk version 1.0.9 for many months >> using a few local network connected Cisco 7960 phones as SIP clients. >> All our phones are currently internal so there is no NAT involved. We >> were not having any problems until last week when some strange issues >> started to crop up. I started experiencing calls that I initially >> believed were being dropped, but discovered that only one side of the >> conversation had dropped. The other party could hear me but I couldn't >> hear them. This seems to happen more often on longer calls but is not >> consistent. I am also seeing issues where incoming or local extension >> calls that are hung up by the originator before being answered will >> continue to ring the SIP phone. At the time the errors occur, the >> Asterisk console displays a variety of "...retrans_pkt: Maximum retries >> exceeded on call.." messages. I scoured the forums for an answer, found >> many reference s to these errors, tried every suggested fix that I could >> find, but none have resolved these problems. After working on the >> problem for several days, I finally built a new box and installed >> Asterisk 1.2 on it. Using this new 1.2 box I no longer see the "Maximum >> retries exceeded on call" warnings on the console but still experience >> the strange behavior. Unfortunately, the errors occur randomly so I am >> unable to reproduce the error on demand. I turned on SIP debugging and >> set console logging to debug and captured an instance of the problem >> with the hang up not being recognized. The details are below: >> >> I dial in from my cell phone. My Cisco phone begins to ring. I then >> hang up my cell phone. Asterisk acknowledges the hang up, but the Cisco >> phone continues to ring. After a minute or so, or if I pickup the >> phone, Asterisk display the following message "That's odd... Got a >> response on a call we donât know about. Cseq 102 Cmd SIP/2.0" I've >> included a copy of the console output when this occurs that shows both >> the SIP message and the Asterisk debug output. >> >> Let me know if any more information would be of use and thanks in >> advance! >> >> The Cisco phone is on IP 192.168.2.203 The Asterisk switch is on IP >> 192.168.2.30 >> >> >> <-- SIP read from 192.168.2.203:50237: SIP/2.0 408 Request Timeout Via: >> SIP/2.0/UDP 192.168.2.30:5060;branch=z9hG4bK3dd277f1;rport From: "JOHN >> DOE " ;tag=as78389007 To: >> ;tag=001380df7eee002b0c2db83c-5ecedbb5 >> Call-ID: [EMAIL PROTECTED] Date: Fri, 02 Dec >> 2005 17:04:49 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: >> Content-Length: 0 >> >> >> Dec 2 09:04:37 VERBOSE[3842] logger.c: --- (10 headers 0 lines)Dec 2 >> 09:04:37 VERBOSE[3842] logger.c: --- (10 headers 0 lines)--- Dec 2 >> 09:04:37 DEBUG[3842] chan_sip.c: That's odd... Got a response on a >> call we dont know about. Cseq 102 Cmd SIP/2.0 > > > ___ --Bandwidth and Colocation > provided by Easynews.com -- > > Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Queues Tutorial updated...
Hello, Just a note to say the Asterisk Queues Tutorial at http://www.orderlyq.com/asteriskqueues.html has been updated to take account of changes in the 1.2.0 release. Anybody who has used our tutorial to create their queues, or uses queues and is thinking of upgrading, will probably find this new version useful. Comments & feedback welcome - though message me privately please to avoid bugging the list Many thanks, Matt King Managing Director, Orderly Software Ltd. http://www.orderlyq.com - the world's most advanced queue system. P.S. You can also check out our new statistics package, OrderlyStats, at http://www.orderlyq.com/statistics.html ___ --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] Restore logging functionality...
Hi, I deleted the files and ran 'logger restart' - no dice, 'logger rotate' - no dice, 'reload' - no dice, 'restart gracefully' - no dice. Logs are not recreated??? Any other ideas Thanks Marco Supino wrote: The user running asterisk doesnt have permission to write on the files, delete them , and asterisk will recreate them as user asterisk, or chown them, or change them to 777 best of all, delete them! Marco. Chuck Bunn wrote: Hi, A while back I made the stupid mistake of deleting my log files 'full' and 'messages' for asterisk. I recreated the files by 'touch' filename and I have gone into the Asterisk CLI and tried both 'logger restart' and 'logger rotate' but the logs still show nothing. I run 'logger show channels' and the output below shows up. I have recompiled Asterisk 1.2 and still the logs do not show up. I am getting data into the 'queue_log' and the 'events' logs however so I know logger is running. Any suggestions to fix this??? CLI output tomato*CLI> logger show channels Channel Type StatusConfiguration --- --- tomato*CLI> tomato*CLI> Output from /var/log/asterisk directory [EMAIL PROTECTED] asterisk]# ls -la total 140 drwxr-xr-x 4 asterisk asterisk 4096 Dec 5 08:22 . drwxr-xr-x 11 root root 4096 Dec 4 04:03 .. drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-csv drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-custom -rw-r--r-- 1 root root 0 Dec 5 08:22 event_log -rw-r--r-- 1 asterisk asterisk 1186 Nov 12 07:43 event_log.0 -rw-r--r-- 1 root root 0 Nov 17 06:41 event_log.1 -rw-r--r-- 1 root root 0 Nov 17 06:45 event_log.2 -rw-r--r-- 1 root root 0 Nov 18 06:38 event_log.3 -rw-r--r-- 1 root root 0 Nov 18 06:37 full -rw-r--r-- 1 root root 0 Nov 18 06:37 messages -rw-r--r-- 1 asterisk asterisk 111711 Dec 5 08:23 queue_log *** Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
solved (Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i)
I solved it by registering the phone in the sip.conf. ___ --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] Transfer/take call to/from other phone
Hi, To pick up another persons phone that is ringing dial '*8' followed by their extension. To do an attended transfer dial '*2' followed by the extension... Hope that helps Denny Schierz wrote: hi, Quoting Chuck Bunn <[EMAIL PROTECTED]>: Push the '#' key followed by the extension for a blind transfer. absolut perfect, thanks :-) .Is there also a shortcut, to take a phone call from other phones to me? cu denny This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Looking for advice on cell carrier's default "Un avaliable" message
On Monday 05 December 2005 12:09, Colin Anderson wrote: > Everything works 100%, except when the user shuts his cell phone off. When > that happens, and he doesn't pick up his SIP/IAX extension, it hits his > cell phone, and the cell carrier's default Unavailable message is played. > Asterisk detects this as the call being "answered" and completes the call. Turn off voicemail on his cell phone, give out his DID instead of his cell #. Send an SMS to his cellphone when new voicemail is left. As far as Dial()ing his cell goes, use 'r' (this is exactly what it's designed for) so that when the carrier is saying "The person you're calling is out of the calling area or has his phone off" all the caller hears is ringing. I just described how I have my own system working and it seems to work just fine. :-) -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
RE: [Asterisk-Users] Include a variable from another file in configfiles
Thanks for your answer but I don't want to include a file, I only want to include a variable. Is it possible to execute linux commands like grep or top in a .conf file in order to parse a file and get a variable? -Message d'origine- De : Administrator TOOTAI [mailto:[EMAIL PROTECTED] Envoyé : lundi 5 décembre 2005 12:43 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Include a variable from another file in configfiles Amaury BOSSE a écrit : > I would like to know if it is possible to include a variable in > sip_nat.conf. > > I have a file with my network configuration and I want to parse it and > to use LAN IP in sip_nat.conf. > > Is there a way to parse a file and include variables in a .conf file. > > > > Amaury > In your sip.conf #include /path/to/the/file/you/want/to/include In this file Asterisk will find the command, eg localnet= -- 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] Transfer/take call to/from other phone
This is what I use. You pre-pend a '4' to the extension number (I used
that because that is how our old pbx worked). There is a number you can
use that will pickup any ringing extension but I forgot what that is. It
should be listed on the asterisk wiki for Pickup.
exten => _4XXX,1,Pickup(${EXTEN:1})
exten => _4XXX,1,Hangup
- James
Denny Schierz wrote:
hi,
Quoting Chuck Bunn <[EMAIL PROTECTED]>:
Push the '#' key followed by the extension for a blind transfer.
absolut perfect, thanks :-) .Is there also a shortcut, to take a phone
call from
other phones to me?
cu denny
This message was sent using IMP, the Internet Messaging Program.
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RE: [Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message
Neat macro but not quite what I’m looking for if I force call recipients to press 1 to accept a call they will scream bloody murder. Good idea though. -Original Message- From: Joe Pukepail [mailto:[EMAIL PROTECTED] Sent: Monday, December 05, 2005 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message Look into the findme feature, this will require the person receiving the call to push a button "hit 1 to accept this call" before a call gets transfered to a cell phone (or home phone for that matter), if nobody hits "1" it continues in the dialplan, this will prevent calls from being transfered to cell phone voicemail or the caller getting the unavailable message from the cell phone carrier. On 12/5/05, Colin Anderson <[EMAIL PROTECTED]> wrote: In our dialplan, we use centralized voicemail for SIP, IAX and cell phones. This means, if a caller calls a user's DID, it tries his SIP/IAX extension, then if he doesn't answer there, it tries his cell, then it goes to Comedian Mail. Everything works 100%, except when the user shuts his cell phone off. When that happens, and he doesn't pick up his SIP/IAX extension, it hits his cell phone, and the cell carrier's default Unavailable message is played. Asterisk detects this as the call being "answered" and completes the call. However, this is undesirable behavior. We want it to go to Comedian mail instead. Note that this is contrary to what the carrier said would happen. The carrier indicated to us that it would just ring and ring and ring forever, which is what we want. Now they are saying: "too bad, this is the way it works, deal with it" In order to have the desired behavior, there are three options: 1. Carrier makes it ring forever (not gonna happen) 2. I set the call forward/Unavailable on the cell to a DID that points to Comedian Mail and do some Caller ID stuff to make it go to the right mailbox. This isn't practical from a management standpoint, it would be troublesome and error prone to maintain 3. When the cell is off, the carrier's Unavailable message plays right away, within 2 seconds of the call being dialed. So, somehow magically modify the dialplan so that if a cell is answered within 2 seconds, go to Comedian Mail. Of these options, 3) would provide the optimum workaround, but I don't think it's possible to express this in an Asterisk dialplan. Anyone have any advice or dialplan magic on how to do 3) ? ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk won't answer malformed caller id
Hello,
Hopefully someone can advise me on the last problem I have in my config.
Among my trunks I have an spa-3000 with the pstn connected to an
ata-186 that I am trying to bring into asterisk. All works perfectly
except apparently when I receive a malformed caller id from this
outside service like below. There is no closing quote on this caller
id and that's apparently the way it's passed in from the ata-186 to
the spa-3000.
Asterisk will just not answer this call apparently. Is there any
mechanism for asterisk to deal with this?
Dec 5 11:14:41 WARNING[8118] chan_sip.c: No closing quote found in
'"WIRELESS CALLE ;tag=3957b3bfa5fe1a2o1'
Dec 5 11:14:41 WARNING[8118] chan_sip.c: Huh? Not a SIP header
("WIRELESS CALLE
;tag=3957b3bfa5fe1a2o1)?
thanks for any help.
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Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i
Dave Cotton wrote: One thing is to do a factory reset to reinit everything, I did that with my 9112i after upgrading the firmware. I just did that. Now Asterisk is giving me the follow error: (0.99 is my Asterisk server and 0.111 is the phone) Dec 5 12:04:10 NOTICE[14222]: chan_sip.c:10817 handle_request_register: Registration from 'No User ' failed for '192.168.0.111' - Username/auth name mismatch -- Registered SIP '3006' at 192.168.0.111 port 5060 expires 300 ___ --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] Looking for advice on cell carrier's default "Un avaliable" message
Look into the findme feature, this will require the person receiving the call to push a button "hit 1 to accept this call" before a call gets transfered to a cell phone (or home phone for that matter), if nobody hits "1" it continues in the dialplan, this will prevent calls from being transfered to cell phone voicemail or the caller getting the unavailable message from the cell phone carrier. On 12/5/05, Colin Anderson <[EMAIL PROTECTED]> wrote: In our dialplan, we use centralized voicemail for SIP, IAX and cell phones.This means, if a caller calls a user's DID, it tries his SIP/IAX extension, then if he doesn't answer there, it tries his cell, then it goes to ComedianMail.Everything works 100%, except when the user shuts his cell phone off. Whenthat happens, and he doesn't pick up his SIP/IAX extension, it hits his cell phone, and the cell carrier's default Unavailable message is played.Asterisk detects this as the call being "answered" and completes the call.However, this is undesirable behavior. We want it to go to Comedian mail instead. Note that this is contrary to what the carrier said would happen.The carrier indicated to us that it would just ring and ring and ringforever, which is what we want. Now they are saying: "too bad, this is the way it works, deal with it"In order to have the desired behavior, there are three options:1. Carrier makes it ring forever (not gonna happen)2. I set the call forward/Unavailable on the cell to a DID that points to Comedian Mail and do some Caller ID stuff to make it go to the rightmailbox. This isn't practical from a management standpoint, it would betroublesome and error prone to maintain3. When the cell is off, the carrier's Unavailable message plays right away, within 2 seconds of the call being dialed. So, somehow magically modify thedialplan so that if a cell is answered within 2 seconds, go to ComedianMail.Of these options, 3) would provide the optimum workaround, but I don't think it's possible to express this in an Asterisk dialplan.Anyone have any advice or dialplan magic on how to do 3) ? ?___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message
In our dialplan, we use centralized voicemail for SIP, IAX and cell phones. This means, if a caller calls a user's DID, it tries his SIP/IAX extension, then if he doesn't answer there, it tries his cell, then it goes to Comedian Mail. Everything works 100%, except when the user shuts his cell phone off. When that happens, and he doesn't pick up his SIP/IAX extension, it hits his cell phone, and the cell carrier's default Unavailable message is played. Asterisk detects this as the call being "answered" and completes the call. However, this is undesirable behavior. We want it to go to Comedian mail instead. Note that this is contrary to what the carrier said would happen. The carrier indicated to us that it would just ring and ring and ring forever, which is what we want. Now they are saying: "too bad, this is the way it works, deal with it" In order to have the desired behavior, there are three options: 1. Carrier makes it ring forever (not gonna happen) 2. I set the call forward/Unavailable on the cell to a DID that points to Comedian Mail and do some Caller ID stuff to make it go to the right mailbox. This isn't practical from a management standpoint, it would be troublesome and error prone to maintain 3. When the cell is off, the carrier's Unavailable message plays right away, within 2 seconds of the call being dialed. So, somehow magically modify the dialplan so that if a cell is answered within 2 seconds, go to Comedian Mail. Of these options, 3) would provide the optimum workaround, but I don't think it's possible to express this in an Asterisk dialplan. Anyone have any advice or dialplan magic on how to do 3) ? ? ___ --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] Transfer/take call to/from other phone
hi, Quoting Chuck Bunn <[EMAIL PROTECTED]>: Push the '#' key followed by the extension for a blind transfer. absolut perfect, thanks :-) .Is there also a shortcut, to take a phone call from other phones to me? cu denny This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i
On Mon, 2005-12-05 at 11:27 -0500, Robert La Ferla wrote: > Pete Barnwell wrote: > > I wasted a lot of time getting 9112is to work with almost identical > > setup. The problem I eventually found was that the 9112is look for the > > config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas > > the documentation says they look for lower case, so they were ignoring > > my tftp settings. The 9133i may well be the same. > > > > The other thing I had to do was to provide the line > > > > next-server ; > > > > in dhcpd.conf to get them to pick everything up. (IIRC that last bit was > > only to do with time&date format though). > > > > I read about the mac address case sensitivity so I used an all uppercase > filename which works fine. The downloading of the firmware works fine > too. I also have the ntp time/date working. I just can't get Asterisk > to respond to the phone! Help! One thing is to do a factory reset to reinit everything, I did that with my 9112i after upgrading the firmware. -- Dave Cotton <[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] Getting started with Asterisk and Aastra 9133i
Pete Barnwell wrote: I wasted a lot of time getting 9112is to work with almost identical setup. The problem I eventually found was that the 9112is look for the config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas the documentation says they look for lower case, so they were ignoring my tftp settings. The 9133i may well be the same. The other thing I had to do was to provide the line next-server ; in dhcpd.conf to get them to pick everything up. (IIRC that last bit was only to do with time&date format though). I read about the mac address case sensitivity so I used an all uppercase filename which works fine. The downloading of the firmware works fine too. I also have the ntp time/date working. I just can't get Asterisk to respond to the phone! Help! ___ --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] DISA function
I tried to use DISA 1.2 with regular asterisk (not [EMAIL PROTECTED]), and had problems with it (losing the last digit or occasionally other digits), YMMV. On 12/4/05, Richard Smith <[EMAIL PROTECTED]> wrote: Hi all, I was wondering whether the DISA function on the latest asterisk 1.2 stable release actually works better than the other prior releases. Basically the [EMAIL PROTECTED] version 2.0 BETA 4 I'm using does not recognise the DTMF tones all the time and sometime when it does, it disconnects. Cheers, Richard.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i
On Mon, 2005-12-05 at 11:15 -0500, Robert La Ferla wrote: > Let me simplify my problem. I have a single Aastra 9133i SIP phone and > latest Asterisk from SVN source running on Fedora Core 4. The phone > currently says "No Service" I would like to be able to dial "1234" from > the phone and get Asterisk to play back an audio message or say some > digits. I can't get this to work with either SayDigits or Playback. > Please help. > > == > sip.conf > == > > [general] > port = 5060 > bindaddr = 0.0.0.0 > context=tutorial > > [3006] > type=friend > username=3006 > secret=mypassword > host=dynamic > canreinvite=no > permit=192.168.0.0/24 > allow=all > mailbox=3006 > > === > extensions.conf > === > > [tutorial] > exten => 1234,1,Answer > exten => 1234,2,SayDigits(123456789) > > > > ** TFTP directory ** > > = > mymacaddress.cfg > = > > sip line1 auth name: 3006 > sip line1 password: mypassword > sip line1 user name: 3006 > sip line1 display name: "myname" > sip line1 screen name: "myname" > > === > aastra.cfg > === > > dhcp: 1# DHCP enabled. > sip silence suppression: 2 # "0" = off, "1" = on, "2" = default > sip proxy port: 5060 # 5060 is set by default. > sip registrar ip: 192.168.0.99# IP of registrar. --- > THIS IS THE IP of my Asterisk and tftp server > sip registrar port: 5060 # 5060 is set by default. > sip digit time out: 6 > time server disabled: 0 # Time server disabled. > time server1: 192.168.0.99# Enable time server and enter at I wasted a lot of time getting 9112is to work with almost identical setup. The problem I eventually found was that the 9112is look for the config file mymacaddress.cfg in upper case (eg 00085D035BC1.cfg) whereas the documentation says they look for lower case, so they were ignoring my tftp settings. The 9133i may well be the same. The other thing I had to do was to provide the line next-server ; in dhcpd.conf to get them to pick everything up. (IIRC that last bit was only to do with time&date format though). Cheers Pete ___ --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] Getting started with Asterisk and Aastra 9133i
One more thing. I upgraded the firmware of the 9133i to 1.3. ___ --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] Getting started with Asterisk and Aastra 9133i
Let me simplify my problem. I have a single Aastra 9133i SIP phone and latest Asterisk from SVN source running on Fedora Core 4. The phone currently says "No Service" I would like to be able to dial "1234" from the phone and get Asterisk to play back an audio message or say some digits. I can't get this to work with either SayDigits or Playback. Please help. == sip.conf == [general] port = 5060 bindaddr = 0.0.0.0 context=tutorial [3006] type=friend username=3006 secret=mypassword host=dynamic canreinvite=no permit=192.168.0.0/24 allow=all mailbox=3006 === extensions.conf === [tutorial] exten => 1234,1,Answer exten => 1234,2,SayDigits(123456789) ** TFTP directory ** = mymacaddress.cfg = sip line1 auth name: 3006 sip line1 password: mypassword sip line1 user name: 3006 sip line1 display name: "myname" sip line1 screen name: "myname" === aastra.cfg === dhcp: 1# DHCP enabled. sip silence suppression: 2 # "0" = off, "1" = on, "2" = default sip proxy port: 5060 # 5060 is set by default. sip registrar ip: 192.168.0.99# IP of registrar. --- THIS IS THE IP of my Asterisk and tftp server sip registrar port: 5060 # 5060 is set by default. sip digit time out: 6 time server disabled: 0 # Time server disabled. time server1: 192.168.0.99# Enable time server and enter at ___ --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] Restore logging functionality...
Chuck Bunn wrote: drwxr-xr-x 4 asterisk asterisk 4096 Dec 5 08:22 . -rw-r--r-- 1 root root 0 Dec 5 08:22 event_log -rw-r--r-- 1 asterisk asterisk 1186 Nov 12 07:43 event_log.0 -rw-r--r-- 1 root root 0 Nov 18 06:37 full -rw-r--r-- 1 root root 0 Nov 18 06:37 messages -rw-r--r-- 1 asterisk asterisk 111711 Dec 5 08:23 queue_log you can: delete your logfiles, * will re-create them I think or: change the owner to asterisk. (chown asterisk.asterisk /var/log/asterisk/ -R) 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
[Asterisk-Users] kernel lockup with Fedora Core 4.0 2.6.14-1.1637
I have an Asterisk system with Fedora Core 4.0, kernel 2.6.14-1.1637. It sometimes locks up with heavy load (e.g., lots of HDLC messages). This requires a hard reboot. I saw some other reports of hard lockups under load. I have disabled as much as possible in the BIOS and as much as possible in the modules (e.g., removing USB, turning off lots of not-needed services, etc.) Could this be a Fedora problem, zaptel problem, or other? This is reproducible on several systems. I am using ZAPTEL 1.0.9.2. My next test is to try the 1644 kernel update. ___ --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] Restore logging functionality...
Hi, A while back I made the stupid mistake of deleting my log files 'full' and 'messages' for asterisk. I recreated the files by 'touch' filename and I have gone into the Asterisk CLI and tried both 'logger restart' and 'logger rotate' but the logs still show nothing. I run 'logger show channels' and the output below shows up. I have recompiled Asterisk 1.2 and still the logs do not show up. I am getting data into the 'queue_log' and the 'events' logs however so I know logger is running. Any suggestions to fix this??? CLI output tomato*CLI> logger show channels Channel Type StatusConfiguration --- --- tomato*CLI> tomato*CLI> Output from /var/log/asterisk directory [EMAIL PROTECTED] asterisk]# ls -la total 140 drwxr-xr-x 4 asterisk asterisk 4096 Dec 5 08:22 . drwxr-xr-x 11 root root 4096 Dec 4 04:03 .. drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-csv drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-custom -rw-r--r-- 1 root root 0 Dec 5 08:22 event_log -rw-r--r-- 1 asterisk asterisk 1186 Nov 12 07:43 event_log.0 -rw-r--r-- 1 root root 0 Nov 17 06:41 event_log.1 -rw-r--r-- 1 root root 0 Nov 17 06:45 event_log.2 -rw-r--r-- 1 root root 0 Nov 18 06:38 event_log.3 -rw-r--r-- 1 root root 0 Nov 18 06:37 full -rw-r--r-- 1 root root 0 Nov 18 06:37 messages -rw-r--r-- 1 asterisk asterisk 111711 Dec 5 08:23 queue_log *** Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting 2 Asterisk using SIP
username= did it. Thanks, Waldo On Dec 5, 2005, at 2:14 AM, Luki wrote: Any ideas on how to correctly set this up? Try adding authuser= and/or username= to the configuration. Do a SIP DEBUG and see what peer asterisk looks for when trying to authenticate the INVITE. It probably can't find the right peer; authuser on the initiating end should help in this case. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Connecting 2 Asterisk using SIP
This worked perfectly.
Thanks,
Waldo
On Dec 5, 2005, at 4:32 AM, xcel wrote:
Try this
___
1st Machine sip.conf
[box2]
username=box1
type=friend
host=10.0.0.2
secret=*
in extensions.conf
exten => _XX,1,Dial(SIP/box2/${EXTEN})
__
2nd Machine sip.conf
[box1]
username=box2
type=friend
host=10.0.0.1
secret=*
in extensions.conf
exten => _X,1,Dial(SIP/box1/${EXTEN})
--xce
*** REPLY SEPARATOR ***
On 12/5/2005 at 12:11 AM Waldo Rubinstein wrote:
I have 2 Asterisk servers running 1.2.0. One of them is a PSTN
gateway. Currently they are connected using IAX2. I wanted to play
with SIP.
I setup a sip entry (type=friend) in the PSTN gateway box and a sip
entry (type=user) in the second box in order to send calls using SIP
to the second box. This works fine. However, when I setup the second
box as type=friend in order for it to be able to send calls back to
the gateway box, then calls no longer work from gateway box to the
second box. The reported error is:
Dec 5 00:07:14 NOTICE[203]: chan_sip.c:9514 handle_response_invite:
Failed to authenticate on INVITE to '"2125551212" ;tag=as0698b1b9'
In the gateway box, my sip.conf looks like this:
[general]
allowguest=yes
autocreatepeer=no
[secondbox]
type=friend
host=10.0.0.2
secret=mysecret
In the second box, my sip.conf looks like this:
[general]
allowguest=yes
autocreatepeer=no
[secondbox]
type=user
host=10.0.0.1
secret=mysecret
Any ideas on how to correctly set this up?
Thanks,
Waldo
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Re: [Asterisk-Users] Transfer/take call to/from other phone
Hi, Push the '#' key followed by the extension for a blind transfer. Thanks Denny Schierz wrote: hi, my asterisk 1.2 works very well with ISDN and SIP, but how can i transfer calls from my phone (for example ISDN, MSN 400) to the another phone (ISDN, MSN 401)? Or when the phone 401 rings, but my boss is not there, how can i take the phonecall from 401 to 400? Do i need special options in my extensions.conf or is that feature from the isdn phone? cu denny This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Ambient Modem
Hi to all i'm finding the procedures for install the ambient md 3200 chipset modem to make tests, anybody have a link or the procedure to do that?? thanks to all Vladimir __ Visita http://www.tutopia.com y comienza a navegar m�s r�pido en Internet. Tutopia es Internet para todos. ___ --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] h323 vs oh323
Prely subjective, but I first installed h323 and it worked. Somewhere along the line something happened and it no longer worked. Recompiling it etc seemed to have no effect. I then tried oh323 and it worked first time and has stayed working. I probably did soemthing wrong, but oh323 seems to work for me. Thanks David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Innocent Evil Sent: 05 December 2005 14:36 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] h323 vs oh323 Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 vs oh323
Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer/take call to/from other phone
hi, my asterisk 1.2 works very well with ISDN and SIP, but how can i transfer calls from my phone (for example ISDN, MSN 400) to the another phone (ISDN, MSN 401)? Or when the phone 401 rings, but my boss is not there, how can i take the phonecall from 401 to 400? Do i need special options in my extensions.conf or is that feature from the isdn phone? cu denny This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] warning message
Hi I got this warning message repeating itself in the log this morning Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:52 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position Dec 5 08:52:53 WARNING[25686]: format_wav.c:247 update_header: Unable to find our position I had to disable logging to be able to use the console Anybody seen this one ? Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP INVITE with no 'Contact' field and RealTime support.
lokotes wrote: When sip device sends to Asterisk INVITE with no 'Contact' field, the server should respond with all information it holds about client. When reading database fields, 'fullcontact' is empty. So, whole procedure ends with 'chan_sip.c:6393 register_verify: Failed to parse contact info'. Interesting thing, internal database (CLI> databse show SIP/Registry x) holds all valid information about this client, so why it's not used? This is completely wrong; if the SIP peer sends an INVITE with no Contact information, the request is invalid. Are you talking about REGISTER? If so, that's a known problem, that Asterisk does not currently support registration queries. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys SPA-841 Missing Calls
Dave Morrow wrote: Thanks all for the replies. I've narrowed it down to the phones dislike for my older 3COM switch. I noticed on the weekend that when these missed calls occur, if I ping the phone, the first few packets are dropped..almost like it's gone to sleep.. Not likely to be the switch if everything continues to function through that switch. It is entirely possible for the ping function to miss one or two attempts while your system conducts the normal arp discovery process; that's fairly normal, particularly for older equipment. ___ --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] VegaStream 400
Hi All Apologise if this has been previously asked but I am fairly new to the list. I have a VegaStream 400 and have succesfully connected the asterisk to the box to make outgoing calls with no problems. I cannot for the life of me work out how to recieve incoming calls. I have looked around and cannot find any information regarding this, can someone help? Thanks Scott Pinhorne ___ --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
