I have several sip stations that on a that are on a nat'd network behind a
nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc.
However, I can't get any of my phones to Transfer or Blind Transfer..
I search and search, and well, just about gone nuts on this one.
Here is
: Thursday, January 07, 2010 11:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sip REFER failes w/603 Decline (Policy),
Polycom Phone
7 jan 2010 kl. 17.15 skrev William Stillwell (Lists):
I have several sip stations that on a that are on a nat'd
Has there been any improvement with app_fax ?
I stopped using it as I had a high failure rate with inbound faxes (10%+)
1000 faxes a week ,with over a 100 failures can get quite annoying from
people complaining.. I could get it to fail everytime I tried sending a
solid black fax page.
(ie, take
?
On 01/08/2010 06:05 AM, William Stillwell (Lists) wrote:
Has there been any improvement with app_fax ?
I stopped using it as I had a high failure rate with inbound faxes (10%+)
1000 faxes a week ,with over a 100 failures can get quite annoying from
people complaining.. I could get it to fail
In version prior to 1.6, timing is very critical for faxing, and the use of
a timing source improves fax sending/receiving., and if no timing source was
used, then you would use zt_dummy, but I am not sure how reliable that is or
was..
And from what I am reading, v1.6 is far better with faxing,
internal_timing = yes in asterisk.conf
yes or no for faxing? Or is this option irrelevant for app_fax/spandsp ?
2010/1/8 William Stillwell (Lists) william.stillwell-li...@ablebody.net:
In version prior to 1.6, timing is very critical for faxing, and the use
of
a timing source improves fax sending
I have the issue where they hit me, get no where, and then my box tells them
invalid context, and it timeouts connecting back to them..
And I get these :(
[Jan 10 19:49:06] WARNING[4103] chan_sip.c: Maximum retries exceeded on
transmission 209673377-00012714169-309054...@117.34.72.42 for seqno
Here is an exert of my speed dial system that pulls a phone number from a
database, and then connects the caller.
$AGI-verbose(Record found in database.,3);
$AGI-exec('Playback','/var/lib/asterisk/agi-bin/speeddial/trsf-call');
my $stmnt = $db-prepare(select phone from phonebook where
;);
I have several extensions in the Central Timezone, the Server is in the Eastern
Timezone. all the voicemail files have a datetimestamp of EST not of the tz=
option under the usermail ...
voicemail.conf
under [general]
tz=EST
under [default]
mailbox_a,password,,,tz=CST6CDT
ok, I figured it out..
tz=zonename from zonemessages
all fixed.
- Original Message -
From: William Stillwell ( Lists )
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, January 12, 2010 9:59 PM
Subject: [asterisk-users] Odd Voicemail Issue
I
I have a dialplan entry that takes a did, and sends it to a group of
stations Dial(Sip/ExtSip/ExtSip/Ext) etc.
However, cdr only shows dst = 5000 (given) and lastdata shows the dial
context, however I see no cdr entry for who actually answered the phone. , I
can see dstchannel as
Most important thing is to PLAN your solution out.. flowcharts,
understanding where calls go, etc.
Project planning, and good ideas on how the calls should be handled, and
coming up with testing scenarios, to make sure everything flows correctly.
From:
Here is the 1.4.x version on centos 5 walk through.
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Nik
Sent: Friday, January 15, 2010 3:15 PM
I know in v1.6 its part of logger.c but I noticed this:
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=11625
However, it doesn't seem to ever been applied to any version of 1.4.x
branch..
Nor can I figure out what it was applied to?
This is over 3 years old, you
logger.conf
[general]
queue_log = yes
queue_log_name = queue_log
Thanks,
Best regards!!
Cristian Arguello.
- Original Message -
From: William Stillwell (Lists)
mailto:william.stillwell-li...@ablebody.net
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell (Lists)
Sent: Tuesday, January 19, 2010 1:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] ast_queue_log to mysql asterisk 1.4 ?
Yeah, I know all that.. I am just saying
http://lists.digium.com/pipermail/asterisk-users/2005-July/110220.html
phone is using old authentication challenge, you may have restarted
asterisk, or did a sip reload, if the message is driving you batty, reboot
the phone.
-Original Message-
From:
I use the 331, and only have 1 line assigned, and each phone has a call
limit of 10, if another call comes in, they can answer it, and it would put
the other caller on hold, you can then switch between callers by using the
up/down keys.
-Original Message-
From:
What is the configuration of the TDM400?
Sangoma makes a nice card as well., I think the A200 is available in PCIe
and supports from 2-4 and I think the A400 does 2-24
If you just answer 4 lines.. you could always just use a SIP Gateway, and
not use any PCIe card.
If you have a pbx, maybe a
Let me know if you figure it out, I am interested in this as well.
Right now I have a cron job that executes this every 5 minutes..
UPDATE cdr SET userfield = MID( dstchannel, 1 , LOCATE( '-', dstchannel )-1)
WHERE disposition = 'ANSWERED' AND LOCATE( '-', dstchannel ) 0 and lastapp
= 'Queue'
setinterfacevar=yes
Needs to be under each queue
What does your dialplan end up looking like?
I would like to add to mine, and stop running a cron job..
exten = 5000,1,Answer
exten = 5000,n,Queue(5000|rn)
exten = 5000,n,VoiceMail(5000,u)
exten = 5000,n,Hangup
-Original Message-
From:
23, 2010 at 12:14 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
setinterfacevar=yes
Needs to be under each queue
What does your dialplan end up looking like?
I would like to add to mine, and stop running a cron job..
exten = 5000,1,Answer
exten = 5000,n,Queue
This is how I did it..
I have to Servers, SRV1 and SRV2
In SRV1 iax.conf
[SRV1-SRV2]
type=peer
username=SRV1-SRV2
secret=Password1
host=IP OF SRV2
qualify=yes
[SRV2-SRV1]
type=user
username=SRV2-SRV1
secret=Password2
context=from-iax
host=IP OF SRV2
quailfy=yes
If
Your inbound context needs to have access to your outbound context.
[iax-inbound]
Include = outbound-conext
[outbound-context]
Exten = _1NXXNXX,1,Dial(DAHDI\g1\${EXTEN})
Something like that.
From: asterisk-users-boun...@lists.digium.com
) PBX (pbx2)?
3- i am using two identical dialplan's is this gonna confuse the
communication process (contextes's name are duplicated over the two servers)
thank you very much for making it clear for me!
2010/1/28 William Stillwell (Lists) william.stillwell-li...@ablebody.net
Your inbound
track your call over your asterisk system.
I wrote about this in old post and submit an complete solution.
Regards,
On Sun, Jan 24, 2010 at 1:14 PM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
Yeah, after hours of trying Friday, I got working by a macro.. I didn't
like
Box #1
faxserver*CLI core show version
Asterisk 1.4.21.2 built by root @ faxserver.localhost on a i686 running Linux
on 2008-08-07 20:30:54 UTC
faxserver*CLI core show uptime
System uptime: 21 weeks, 2 days, 22 hours, 43 minutes, 42 seconds
faxserver*CLI
this box gets about 200 faxes a day,
servers, the crashing
went away.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell
(Lists)
Sent: Monday, February 08, 2010 8:43 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
Polycom 331's are also in the same price range, and offer good features as
well.
All my polycoms are provisions with option 66 on dhcp, and an ftp site with
cfg files that are build from a mysql database from sip users table.
From: asterisk-users-boun...@lists.digium.com
Do a qeuee, add each as a station in the quee..
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Anness
Sent: Tuesday, February 16, 2010 10:04 AM
To: asterisk-users@lists.digium.com
Subject:
What is the easiest method or can someone point me in the direction I need
to look to do remote agent login..
Ie, Caller calls in with a cell or home phone, authenticates himself, select
a queue to be added too, hangs up, and then any calls coming into said queue
would ring their home or cell
Anybody work out how to fix this?
Asterisk 1.4.26.3
Sip Trunk inbound - to Queuee - Outbound to two sip stations, and one sip
trunk.
sip trunk caller answers, queue shows ring+inuse , core show channels
shows inbound/outbound
after caller hanges up, no channels in use, queue still
http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
Sent: Wednesday, April 14, 2010 10:52 AM
To: asterisk-users@lists.digium.com
Subject:
What ports to you have available on the ESI ?
Analog Trunk Lines?
Analog Station Lines?
PRI?
You could bridge with maybe a small 4 or 8 port FXO/FXS device depending on
what you have available in on your ESI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Anybody know a reliable fax solution for 1.4.30 branch?
I am using PikaFax on another server and works very well (about 3000 faxes
a week), but it appears they no longer offer their product to open source
asterisk, only for there WARP appliance.
NOT really looking to migrate from 1.4.x
Subject: Re: [asterisk-users] Need fax solution for 1.4.xx
William Stillwell (Lists) wrote:
Anybody know a reliable fax solution for 1.4.30 branch?
That would be HylaFAX+ along with iaxmodem
http://hylafax.sourceforge.net
http://iaxmodem.sourceforge.net
Doug
--
Ben Franklin quote:
Those
something like the
MyFax service.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell (Lists)
Sent: Tuesday, May 11, 2010 2:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users
==
On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists)
william.stillwell-li...@ wrote:
Anybody know a reliable fax solution for 1.4.30 branch?
I am using PikaFax on another server and works very well (about 3000
faxes
a week), but it appears they no longer
] On Behalf Of William
Stillwell (Lists)
Sent: Wednesday, May 12, 2010 11:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Need fax solution for 1.4.xx
But that can't handle the call volume, and doesn't support (2) 23B+D now
does it?
-Original
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Need fax solution for 1.4.xx (Resolution)
On 05/13/2010 10:48 PM, William Stillwell (Lists) wrote:
Ok, I ended up upgrading 2 of my 5 boxes to 1.6.2.7 , and using spandsp
0.0.6pre17, dahdi-linux-complete
Don't some thin clients run on WindowsCE or Linux/rdesktop?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Thursday, May 20, 2010 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial
You would need to see if there is a hook flash hold.
Try playing with a hook/flash ( ie do a flash, wait, then hangup phone, it
may send the onhold message ) it may also ring back.
Or you will have to park the call
Hook flash , Dial 700 (if that's your park extension), hangup, then
I would think AGI would be better. ?
I don't think system() returns anything, except maybe a success/fail ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Monday, June 14, 2010 12:00 PM
To: Asterisk
I know on my polycom phones, I just press the conf button, dial, and then
hit join, and all done, no special programming required on dialplan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
I have several remote phones that experience a slight call delay when
answering phones, ie, they will answer, speak a few words, and then the
remote caller will hear them, and the first half is cutoff?
Any idea what could be causing this?
Thanks,
Bill.
--
I use SecureCRT+FX , and use ansi graphics.
Putty is nice w/WinSCP as well.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Tuesday, June 29, 2010 10:17 AM
To: Asterisk Users Mailing List
Also, technically your 101This is a salt is stronger than your SHA1 Hash.
Let's say you stick with the 17 character password
You are using 0-9, a-z, A-Z, and space.
0-9 = 10
a-z = 26
A-Z = 26
Space = 1
Total Possible Values = 63
17^63 = 3.2982384238829760312713680399948e+77
Your sha1 is
http://www.coffer.com/mac_find/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Church
Sent: Monday, July 12, 2010 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Another tool, to search by company.
http://standards.ieee.org/regauth/oui/index.shtml
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C.
Bailey
Sent: Monday, July 12, 2010 11:53 AM
To: Asterisk Users
Zoiper seems to have a software update every other week, and annoys you to
death on updates, and sometime the update breaks it.
I am looking myself for a good windows softphone, Zoiper is nice, never
tried the pay for version.
-Original Message-
From:
I am trying to configure a channel bank with 24 ports of FXS., but appear to
be hitting a roadblock? This worked on v1.4.xx but now just get
SimpleSwitch and immediate=no/yes don't seem to make a difference?, no
matter if under top section, under channel, etc.
Chan_dahdi.conf:
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell (Lists)
Sent: Tuesday, October 26, 2010 8:58 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Channel Bank ? Simple Switch Hangup?
I am trying
Use Network LCR and PRI Card...
I interfaced two Samsung iDCS 500 r2 into an asterisk box this way. I don't
think Samsung has gotten SIP 100% open yet.
I have since scrapped all the Samsung gear, and have a big pile of parts :D
If you're talking 10 or 20 users on the 100, just scrap it, go
How many lines are we talking here?
Get a two port T1/PRI Card, use a channel bank, and get your lines from your
provider on a PRI. (this way you can start off with 10 numbers, and add up
to 300+ and never have to add any extra lines at a per line price.
If you looking to save money with SIP
For those who don't know, (as I just figured out by reading the sourcecode),
that all settings for a particular channels must be placed before the
channel = entry.
Ie,
Immediate=no
Channel=1-24
Immediate=yes
Channel=25-48
Immediate=no
Channel=49-72
1-24 will have
On Wed, Nov 03, 2010 at 06:30:09PM -0400, William Stillwell (Lists)
wrote:
For those who don't know, (as I just figured out by reading the
sourcecode),
that all settings for a particular channels must be placed before
the
channel = entry.
Immediate=no
Channel=1-24
Immediate=yes
Im using
asterisk-1.6.2.13
asterisk-addons-1.6.2.2
dahdi-linux-complete-2.4.0+2.4.0
libpri-1.4.11.4
spandsp-0.0.6
Sangoma Hardware, using wanpipe-3.5.17
Extensions.conf:
[fax-in]
exten = s,1,Answer()
exten = s,n,Wait(1)
exten =
Why not just use tiff2pdf ?
tiff2pdf input.tif -o output.pdf
William Stillwell
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Michael
Sent: Friday, November 19, 2010 9:43 AM
To:
I know there was a patch for dahdi to fix server lockups on time shift. (not
sure what version, but if you changed the time, the server would just go
crash.)
Do you have the latest version ?
Check your ntpd settings to make sure your time isn't bouncing all over the
place.
59 matches
Mail list logo
