I've never used an 850, but I had similar problem on the 750 when I had the channel
configured wrong in the 750 console. Have you tried reseting the config and making
sure everything is FXS Loopstart.
Also, have you tried another AMP-50 cable with your bank. I had a bad cable that was
First , you need to see what your insurance policy covered. If it covered
replacement, then the easist thing for you to do is make the claim and replace your
old pbx through a local service provider(asterisk or not).
Second if you know next to nothing about pbx's and phone, then the time it
If there is already an existing phone system in place, you could easily migrate to an
asterisk based solution if your internal phones are analog. The big question for you
is not number of phone lines, but peak utilization. Here's what I have.
141 Analog Phone Lines
15 SIP IP Phones (Mix Cisco
I've got Asterisk STABLE-CVS-4/19/04 with 12 Cisco 7960 phones 6.0 Firmware using
ulaw, 6 Polycom IP500 ulaw phones, and 192 Zap channels. I have Gig-E Copper to my
server and 100Mbit-Full to all my phones. I haven't had any choppy audio at all. My
switch is a Cisco 4500.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam
Goryachev
Sent: Wednesday, April 21, 2004 2:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Repeated Notice: (UN/REACHABLE)
Should this actually attempt more than a single ping before claiming
In laymans terms.
To use your telco's T-1 as the timing source
span=1,1,0,esf,b8zs,yelllow
To use the internal clock of the card you would use (I'm pretty sure that this would
only be used for channel banks, or connections to other PBX hardware. I don't think a
telco is going to use
I'm running Zaptel CVS from April 8, LibPRI CVS April 8, and v-1.0 CVS April 7. With
dual T400P cards with no PRI errors at all. Possibly something driver/config related?
Are you timing from your PRI? I remember getting some PRI errors when my timing
config was hosed. Could you post your
Did
you install the micro filters that came with with your ADSL modem. Usually
you get 3-4 of these. They are used to protect your analog lines from the
additional signal noise from the ADSL signal.
-sb
Radio Shack item number 279-103 for
about $15 each
-Original Message-From:
I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite
some time, to no avail. I've googled, I've tried loads of configurations, I've
rewired phone lines, and still I am not winning the battle.
Here's my config.
PRI-T400P-Asterisk-T400P-Adtran 750(L36
PROTECTED]
Subject: Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config.
Adtran 750
Bisker, Scott (7805) wrote:
I've been trying to get a Win 2000 RAS server working with my
asterisk PBX for quite some time, to no avail. I've googled, I've
tried loads of configurations, I've rewired phone
Sent: Wednesday, April 07, 2004 12:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dial-In/Out Modem Zap Channel Config.
Adtran 750
Bisker, Scott (7805) wrote:
Same as mine. Do you know off the top of your head what firwmare
you're using? Also, what RAS card do you have on your
This could possibly be related to Bug# 0001320 where Zap channels get stuck in a Rsrvd
State. I inadvertently put the bug in Zaptel since I had upgraded to Zaptel 0.9.0 the
same time I upgraded to asterisk v1-0_stable. When I rolled back to asterisk 0.7.1
with -DOLD_DSP_ROUTINES the problem
I've just started having the same problem here today. I did and upgrade over the
weekend to
Zaptel-0.9.0 and the release candidate for Asterisk-1.0 CVS 3/28/04.
I have 6 Adtran 750 FXS_KS for all channels. 1 T-1PRI and one EM_W T-1.
-sb
-Original Message-
From: [EMAIL
Title: Message
Please
post the portion of your dialplan that you are explaining. More than
likely you don't have an "r" in your dial command. That lets the
calling party hear a ring.
e.g.
Dial(SIP/1234|20|Tr)
-sb
-Original Message-From:
[EMAIL PROTECTED]
w you what I have.
Thanks for your help.
Mark
-Original
Message-From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bisker,
Scott (7805)Sent: Monday, March 22, 2004 12:13 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users]
I got
my fuses from a local supplier. Looks like the OEM is Littelfuse.
PN: 0481003.V
-sb
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Jacques
LeisySent: Friday, March 19, 2004 10:23 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott
(7805)
Sent: Tuesday, March 16, 2004 11:09 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] PRI Errors
I just had the same exact problem this morning. The only thing I've done in the last
span=7,1,0,esf,b8zs
span=8,2,0,esf,b8zs
rgrds
Quoting Bisker, Scott (7805) [EMAIL PROTECTED]:
Update on this. I had the exact same issue today. At almost exactly the
same time as yesterday. Possible telco problem? Timing issue with zaptel?
Never had this issue before updating libpri
I just had the same exact problem this morning. The only thing I've done in the last
couple of days is update update zaptel. I rolled back my zaptel to 2/11/04 from
3/8/04. And kept my libpri from 3/8/04. I never had this error before updated. I
had other issues, but not this one.
-sb
In your SIP.conf set callwaiting = no. This will work for single registrations. If
you have multiple call appearance on you phone, then it will just ring to the second
line (e.g. Cisco 7960). If you only have a single registration, then you should be
fine.
-sb
-Original Message-
Title: Pri Errors, Hanging up Owner
I had
the same problem a few weeks ago. I updated to latest zaptel and libpri,
and the problem went away. My date is 3/8/04
-sb
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Matthew
BrantonSent:
I'm just pinging the list for some quick info that I could turn up in google. Has
anyone played with doing ZapRAS over an IAX channel? i.e. call comes in T-1 to server
1. Server 1 sends call to server 2 via IAX. Server 2 picksup call with ZapRAS, runs
ppp... etc. I don't see why this
Make that could not turn up in google.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott
(7805)
Sent: Monday, March 15, 2004 4:07 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] ZapRAS over IAX anyone?
I'm just pinging the list for some
Michiel,
Are you using WinFax? or one of the Products Based on Winfax? I've seen this on all
of our WinFax Stations, but none of our standalone Fax machines.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of michiel betel
Sent: Wednesday, March 10,
and short delay before voice
starts after ring.
Bisker, Scott (7805) wrote:
I think what James is referring to is the delay once the call already
been dialed. It's not specific to Ciscos, as I'm experiencing the
same problem on my polycom phones. Must be SIP related.
The problem is that once a call
] Cisco 7960 and short delay before voice
starts after ring.
Bisker, Scott (7805) wrote:
I think what James is referring to is the delay once the call already
been dialed. It's not specific to Ciscos, as I'm experiencing the
same problem on my polycom phones. Must be SIP related.
The problem
I think what James is referring to is the delay once the call already been dialed.
It's not specific to Ciscos, as I'm experiencing the same problem on my polycom
phones. Must be SIP related.
The problem is that once a call is dialed, when the remote party picks up the phone,
the first half
Buy SmartNet support for the phone. That grants you access to software images through
their website. Try Insight. 1-800-INSIGHT. They sell all quantities.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Hermann Wecke
Sent: Thursday, February 19,
Title: Message
Works
fine here. Post your SIP and Zapata configs
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of B. J.
BomarSent: Wednesday, February 18, 2004 4:31 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Call Pick
on Cisco
Make sure you have your extensions.conf setup to dial out the T-1. Something like
this.
exten = _81NXXNXX,1,Dial(${LONGDISTANCET1}/${EXTEN:1})
exten = _81NXXNXX,2,Hangup
exten = _71NXXNXX,1,Dial(${LONGDISTANCET1}/${EXTEN:1}||d)
exten = _71NXXNXX,2,Hangup
-Original
Here's a wierd one. I'm have a problem where periodically a couple of my extensions
dont' get hungup properly. The channel bank doesn't show the channel as active, show
channels doesn't show the channel as active, but a zap show channel has the Actual
Confinfo: as an active call. This
Did you possibly have astman running on the localhost? I found that I was getting
kernel panics while using astman on an SMP machine with dual T400P cards. Did you see
the message on the console before you reset the box? Did you possibly have a serial
console connected logging console
Has anyone experienced problems with dialup through asterisk. I'm having some varied
success with dial-in and dial-out.
All my analog extensions are connected to * via Adtran 750 FXS channelbanks using
FXO_KS signalling. I have a longdistance T-1 (em_w) from sprint and a local T-1 PRI
from
Take a look at dialplan.xml on your tftp server.
DIALTEMPLATE
TEMPLATE MATCH=0 Timeout=1 User=IP/ !-- Local operator--
TEMPLATE MATCH=8,011* Timeout=6 User=IP/ !-- International
calls--
TEMPLATE MATCH=8,1.. Timeout=0 User=IP/ !-- Long Distance --
Hello All,
I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't
configured properly. Now heres the next question. 12 EM wink lines from telco. I
have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are
configured DPO. How do I signal
I tried both featd and em in zapata.conf, to no avail. I restarted in between all
changes. Is it possible to signal the DPO ports on the 750 with fxo_ls or fxo_ks?
This is the last piece to my DID puzzle. Anyone else with experience on this oddball
config?
Thanks,
-sb
-Original
Yes. Adtran FXS cards.
Did you say you were using Adtran FXS cards?
Bisker, Scott (7805) wrote:
Hello All,
I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't
configured properly. Now heres the next question. 12 EM wink lines from telco. I
have
I take it you are running RedHat 8 (or 9) since this is the most up-to-date kernel.
Did you install the kernel-sources and kernel-util rpms as well? You'll need these in
order to compile and install zaptel.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
your problem.
Regards,
Kekin
Subject: RE: [Asterisk-Users] Incoming DID call Voice Problems
Date: Mon, 26 Jan 2004 09:31:32 -0500
From: Bisker, Scott (7805) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
I have an updated question on this one. It seems that only inbound
: [Asterisk-Users] Re: Adtran 750 DID question.
On Friday, January 30, 2004 3:56 PM, Bisker, Scott (7805)
[SMTP:[EMAIL PROTECTED] wrote:
I guess asterisk is winking properly then, because the line rings
when
dialed. In zaptel.conf the lines are set to em and in zapata.conf
they are set to em_w
on the PRI work flawlessly.
Any ideas
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott
(7805)
Sent: Saturday, January 24, 2004 3:18 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Incoming DID call Voice Problems
Hello All,
I am
Hello All,
I am experiencing some intermittent problems with calls coming inbound on my DID
trunk. I have 12 DIDs that come into an Adtran 750. From there T-1 to a port on
T400P. The problem is that some calls that come in don't seem to bridge properly.
Heres what happens.
Call comes in on
Ali,
If
Zap/82 is channel 20 on Span 3, then it looks like it's hanging up before the
channel restarts as this line indicates.
== Spawn extension (inbound, 9009170,
2) exited non-zero on 'Zap/82-1'
Maybe
there is a problem with your agi script.
B
channels only restart when the PRI
An even better way to get asterisk started is to use the init scripts provided with
asterisk and the zaptel kernel modules.
cp /usr/src/asterisk/init.asterisk /etc/init.d/asterisk
cp /usr/src/zaptel/init.zaptel /etc/init.d/zaptel
Then do the proper linking, etc to get asterisk to start in your
Ariel,
You can install them from the RH9 CD. Also, make sure you use readline and not
redline.
Insert the RH9 CD
cd /mnt/cdrom/RedHat/RPMS
rpm -ivh readline*.rpm
You may need to switch CDs in order to find the correct disc.
-sb
-Original Message-
From: [EMAIL PROTECTED]
Hello All,
Here is a patch that fixes the problem when forwarding messages with vmail.cgi. Bug
submitted with patch on bugs.digium.com.
-sb
--- /usr/src/asterisk/vmail.cgi.orig2003-12-17 14:21:47.0 -0500
+++ /usr/src/asterisk/vmail.cgi 2003-12-17 15:07:36.0 -0500
@@
Has anyone on the list been able to locate and try out the 1.1.0 firmware? It was
released in November, but I have yet to get my hands on it. The Polycom site has way
more docs online, but the link to the firmware only brings up the release notes.
-sb
Could you post the console output from when you run the softphone application? Maybe
there is a problem with registration.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Hao Zhong
Sent: Friday, December 12, 2003 5:16 PM
To: [EMAIL PROTECTED]
In sip.conf do you have
type=friend
for your softphone?
If not you'll only be able to send or receive calls depending on the option you
selected.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Hao Zhong
Sent: Friday, December 12, 2003 2:29 PM
To:
I had a similar problem with my 7960 phones.
It ended up being a problem with quotes in the SIP.cnf file.
Do a sip show peers from the console to see if the 7960 is registered properly.
For a test set the following values in the cnf file
line1_name: 8005
line1_shortname: 8005
John,
I have 12 7960 phones with 6.0 with no issues. Sounds like a hardware problem to me.
-Original Message-
From: John Todd [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 30, 2003 1:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 6.0 + Asterisk question
I have
Michael,
Where in your extension definition to you dial a channel (SIP, Zap, or other)? You
are missing the dial entry.
-sb
-Original Message-
From: Lists [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 29, 2003 10:53 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sip Issue
You can get them from any cisco reseller.
If you are in the US, the part number is CP-PWR-CUBE=
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lists
Sent: Sunday, November 30, 2003 6:49 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] cisco
Marc,
This is the typical behavior for call waiting. While you are initiating a
call, people who call your number will get a busy signal until your first
call connects. Once the call connects, the number 2 caller will hear a ring
until you pickup.
If you want to disable callwaiting then put
(the SIP BYE message).
Jerry
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bisker,
Scott (7805)Sent: Thursday, November 13, 2003 6:17
PMTo: '[EMAIL PROTECTED]'Subject: RE:
[Asterisk-Users] Overhead Paging
Our
Title: Overhead Paging
Our
setup is to set the OSS device to autoanswer. The output of the soundcard
feeds into a bank of overhead speakers. If the channel is in use, then the
call gets put in a queue until the OSS device is free.
-sb
-Original Message-From: Johnson, Randy
How far is your server from the telco box? I found that with extended
distances, my reliabilty was significantly decreased. If you still have
problems, check your RJ-48X jack for connection problems.
-sb
-Original Message-
From: Eduardo Goncalves [mailto:[EMAIL PROTECTED]
Sent:
Default User Password is 123
Default Admin Password is 456
-sb
-Original Message-
From: Roman Pelikh [mailto:[EMAIL PROTECTED]
Sent: Friday, October 31, 2003 11:54 PM
To: '[EMAIL PROTECTED]'
Subject: [Asterisk-Users] Polycom Soundpoint IP600
Does anyone have the Admin password for the
Title: Polycom SoundPoint IP 500
The
SIP version of the IP500 runs the same firmware, etc as the IP600. The
config files are the same. The only difference is that the IP500 has three
lines instead of six. I believe that the model number is the same for all
IP500 phones, its just the
I have 6 750s attached to my pbx server. The 850s have a lot of
functionality you don't really need.
-sb
-Original Message-
From: TC [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 30, 2003 1:07 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie hardware question
You
Pretty much anything from Cisco or Foundry support QOS. Linux and BSD
support it as well.
-sb
-Original Message-
From: Nick Knight [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 28, 2003 6:52 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] QOS
Hello all,
Apologies as not
Jim,
What type of cabling are you using? What's terminated on the other end of
each port (Channel Bank, Telco Demarc?) How far away are you from what's
connected on cards 2 3?
This will have a lot to do with signal and noise?
-sb
-Original Message-
From: Jim Paraschou
my mistake, I was thinking a T-1 card.
-sb
-Original Message-
From: Jim Paraschou [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 28, 2003 4:57 PM
To: [EMAIL PROTECTED]
Subject: # [Asterisk-Users] TDM 400P signal problem
It is a cable 4-5 meters long that has handssets
connected
I
Just submitted a patch for this on asterisk-dev.
Quick fix add the following line above line 5022 in chan_sip.c
ast_setstate(c,AST_STATE_DOWN);
Should look like this when you are done.
} else {
5021
I've attached two SIP debugs in reference to bug #116. They are from
today's CVS build.
1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the
call. After which, SIP(2) rings for about 30 seconds then stops.
2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging
I found the best way to upgrade is install Red Carpet from www.ximian.com.
Subscribe to the RH 9.0 channel. And do a complete update. The only
drawback is that this method doesn't update the kernel. To do the kernel,
ftp the latest kernel from updates.redhat.com. rpm -ivh latest
kernel.rpm.
Hello all,
I got the following error compiling h323 support in the latest cvs. Below
the error is a diff to the file that I got to make it work. I took an
example out of sip as far as the syntax for ast_rtp_new. Not sure if it is
correct or not, but it seems to work. Please correct me if I am
Hello All,
Couple of quick (hopefully) questions.
1. I noticed in the latest h.323 cvs log that callerid is now supported.
Is there any special configuration needed to get this to work. I have tried
callerid= in h323.conf to no avail. Calls from a h.323 device show
callerid as the user
Hello All,
I'm having a weird problem when connecting up to a TA 750 from adtran. The
problem I'm seeing is that the third wire on my 66 block is behaving as the
tip (or ring) for every extension. Is this indicative of a bad BCU? The
only extension that works properly is extension Zap 2.
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