Anyone know if Asterisk offers SVP support (Spectralink protocol)
-MD-
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I have to connect a Meridian S1 to Asterisk for a slow migration to VoIP.
What is the best way to connect them?
1. Is a T1 the best solution?
2. Can I pass Caller and Callee information across the link?
If a T1 is best, I recall a standalone T1-SIP device a year ago on this
list. Does anyone
Over a year ago I saw a discussion about a standalone device which converted
a T1 in/out to SIP in/out (over 10/100 LAN). Anyone recall what this device
is?
(I'm looking for a standalone device - not a PCI card).
Thanks
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To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Meridian S1 to Asterisk via T1
Michelle,
On Fri, 7 Sep 2007, Michelle Dupuis wrote:
I have to connect a Meridian S1 to Asterisk for a slow
migration to VoIP.
What is the best way to connect them
Huff [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 08, 2007 11:34 AM
To: Michelle Dupuis
Subject: Re: [mythtv-users] Real Time Clock Alarm Broken with 2.6.22+ kernel
On 9/7/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
Craig,
I wrote an nvram-wakeup replacement call acpi-wakeup
I have a new Sangoma A102 and I'm trying to get it running in asterisk. A
look through the dmesg log shows the card is detected and the various
channels created. However, when I start asterisk I get the error below.
Any ideas?
My zapata.conf is below.
Thanks,
MD
== Registered custom
I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My
Sangoma A102D shipped with 2 T1 cables - which I assume are straight
through. Do I need to make crossover cables for this scenario?
Thanks
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I have a new asterisk system with a T1 card. It appears that running ztcfg
-vv is required in order for asterisk to start properly.
Is this correct? Are people adding this command to the asterisk startup
script?
Thanks
___
--Bandwidth and
I'm tying a Nortel option 61 to asterisk via T1. I don't want to split each
of the t1 channels out into individual lines (tied to a specific extension)
- so a trunk in and out.
Assuming PRI over T1 signaling, how would I pass the CALLED and CALLER info
across the channels so each side knows
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get sangoma A102D setup
on asterisk
On Fri, Oct 26, 2007 at 04:00:30PM -0400, Michelle Dupuis wrote:
I have a new Sangoma A102 and I'm trying to get it running in
asterisk. A look through the dmesg log shows the card
: Re: [asterisk-users] Treating T1 as trunk in/out, not individual
lines
Michelle Dupuis wrote:
I'm tying a Nortel option 61 to asterisk via T1. I don't want to split each
of the t1 channels out into individual lines (tied to a specific extension)
- so a trunk in and out.
Assuming PRI over T1
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Treating T1 as trunk in/out,
not individual lines
Michelle Dupuis wrote:
Ok..so how would the CALLED and CALLERID ID be presented to
Asterisk
when using PRI signaling.
When you are using a PRI, you'll see
I've got a few Cisco 7921 wifi phones to use with an Asterisk pilot. For
the purpose of the pilot (i.e. low investment) I want to configure the
phones from the keypad.
Each phone shows settings locked! whenever I try to edit the network
profiles. I can't seem to unlock them! Hopefully there
That did the trick! It appears that all of the config is retrieved from a
.cnf.xml file, so there wasn't much more I could do at the phone level other
than set the networking parameters.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Yehavi
We are planning a very large Asterisk deployment, using Wifi SIP phones.
We've done installs using Spectralink and the SVP to manage congestion at
the access points, but we have a client that doesn't want Spectralinks.
Anyone have experience with an alternative congestion management (AP
installation with configuration. Without ascii
config files (or a tool from the mfg to create binary config files from a
script), each soft device must manually configured.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit us
We have an Asterisk v1.2.16 box registering with an ITSP using SIP. The
registration succeeds, and is confirmed with SIP SHOW REGISTER. However,
we frequently (every few minutes) see this on our console:
REGISTER attempt 1 to [EMAIL PROTECTED]
REGISTER attempt 2 to [EMAIL PROTECTED]
Any
I've reposted with a more meaningful subject - hopefully someone will
replyWe have an Asterisk v1.2.16 box registering with an ITSP using SIP.
The registration succeeds, and is confirmed with SIP SHOW REGISTER.
However, we frequently (every few minutes) see this on our console:
REGISTER
Ast 1.4 will pass through T.38, but not terminate/originate T38. Be sure
you understand the implications for your fax termination
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cesar Benjamin
Garcia Martinez
Sent: Saturday, May 05, 2007 3:26 PM
How about forking the process when the AGI launches, and pass the PID back
to Asterisk in a variable. When the call ends (caught at the h), call
another AGI script to kill/stop that pid.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
The IAXMODEM might get you half way there...but if you want to connected it
to a windows box (which I assume is why you use the RAS acronym), you'll
have to look for remote serial port software.
-MD-
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Have a look at poptop (PPTP server) - pretty straight forward. Great if you
have Windows clients.
If you have Linux clients (or want a permanent tunnel), there are other
options.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit
We're contemplating Linksys WIP 330 phones, but we're concerned about
configuration effort. Does anyone have the file format for an XML file to
configure this phone? We got auto-provisioning to D/L a file, but the XML
file format seems to be a secret...
Thanks,
Michelle
We're looking at a large wifi phone deployment, and we're looking for wifi
phones that:
1. Are SIP compliant (Asterisk friendly)
2. Provision capable (ideally TFTP of own MAC address)
3. Industrial quality (no cheap plastic stuff).
4. Well documented (and none of the only telco's get
I can't find reference to TFTP for provisioning - does this phone support
it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcus Franke
Sent: Monday, June 25, 2007 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
: [asterisk-users] Best wifi IP phone for asterisk
On Tue, 26 Jun 2007, Hendrik Visage wrote:
On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
We're looking at a large wifi phone deployment, and we're looking
for wifi phones that:
HAve a look at the Linksys WIP 300 (or something) Can
Does anyone know if Asterisk can natively support the SVP protocol from
SpectraLink?
Thanks,
MD
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Same here. We have commercial call center clients on Unlimitel. They've
had a few outages during business hours, but Unlimitel is responsive.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dr. Michael J.
Chudobiak
Sent: Monday, July 02, 2007 5:29
- Non-Commercial Discussion
Subject: Re: [asterisk-users] Best wifi IP phone for asterisk
Michelle Dupuis wrote:
We're looking at a large wifi phone deployment, and we're looking for
wifi phones that:
1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally
TFTP of own MAC address
We've helped a lot of customers with fax for VoIP...often turns out to be
audio quality issues (eg: test fax from within office is ok, but fax from
customer fails). One solution is to optimize route (minimize latency) if
you have that control.
Moving fax line retry/resend control up a level (to
Here are another $0.02
We too have put in a lot of polycoms and aastras. I agree with a lot of
what you noted below...but there are two big strikes against aastra:
1. Firmware bugs. Even some basic functions of the 480i are
unusable/unstable due to firmware bugs. The word from support is
I am trying to set callerid from a PHP script, using one of two functions as
shown below (setid1 and setid2). The first function works great with
regular names and numbers, BUT, if I call the function with
(Test,UnknownNumber), the cid number gets set to asterisk. Why is my
passed number
Take a look at smartCID (at www.generationd.com) Does a reverse lookup for
name/location/etc. Based on phone number.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Rubenstein
Sent: Wednesday, February 07, 2007 8:30 AM
To: Asterisk-Users
We used Aastra's for a good while, but gave up on them (and switched to
Cisco). Aastra's seem cheaper up front (hardware costs), but the time
wasted chasing firmware bugs, lack of documentation, and poor support
quickly eat up any savings. (unless your needs are very basic).
MD
_
If the PSTN side is only complaining about conversations with a single phone
on the SIP side, look at the SIP phone.
Check the settings for that SIP phone/PC (VAD disabled, NIC settings,
runaway processes). Do PSTN callers here choppiness from the SIP phone
caller?
-Original Message-
on SIP
Well, the PSTN side is complaining about a random phone on the SIP side.
Yes, they do hear choppiness.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
Dupuis
Sent: Monday, February 12, 2007 2:49 PM
To: 'Asterisk Users Mailing List - Non
you how
it's set up there, but the network technicians said it is enabled.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
Dupuis
Sent: Monday, February 12, 2007 3:19 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE
I would suggest you grab the menu from the .conf file and paste it into the
new setup. (After even a little asterisk experience, they should be able to
get away from the gui).
The sound files could be copied as well. I'm guessing from your question
that you/your client may not having Linux
We use a lot of mini-itx pc's, including the pCI slot. I don't think any of
the systems have shared an irq with the PCI slot
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vincent
Delporte
Sent: Tuesday, February 13, 2007 5:29 PM
To:
@lists.digium.com
Subject: Re: [asterisk-users] Mini-ITX board + FXO PCI card?
At 10:09 11/02/2007 -0500, Michelle Dupuis Henderson wrote:
We use a lot of mini-itx pc's, including the pCI slot. I don't think
any of the systems have shared an irq with the PCI slot
Thanks for the tip. In that case
I'm looking at setting up an asterisk box dedicated to SIP-H323 conversion
(a 3rd party is currently converting the protocols for us).
1. Is it worthwhile to split this functionality onto a second server? Or
should we let the ast pbx handle the conversion? (we have a couple hundred
active
I don't think Asterisk plays a role in this (unless I'm missing your point).
A simply script to ping your server room will do. Upon failure, the script
could initiate a PPP connection outbound.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I'm working on calls coming in to an asterisk box as H.323, and going out as
SIP to a remote device (a VoiceMaster). The remote device is refusing the
calls with SIP error 406 (Not Acceptable).
I have attached the SIP debug output below. It looks like codecs overlaps -
can anyone see why the
We have a * box with sip in, and h.323 out. When the H.323 call setup is
underway, will Asterisk translate the progress/status/result codes to SIP
automatically?
Ordo we have create our own result codes in SIP headers?
Thanks,
MD
___
Have you tried SMSSEND? It's open source, available as RPM.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: Sunday, February 25, 2007 6:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sending SMSa
On
T.38 won't work over the H.323 leg of your call (even with Open H.323),
chan_h323 won't support it.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Octavarium
Sent: Tuesday, February 27, 2007 12:51 PM
To: asterisk-users@lists.digium.com
Subject:
Have you tried starting Linux with irqpoll / noapic? Sounds like a BIOS
bug..
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco
Parisotto
Sent: Tuesday, February 27, 2007 3:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] TE212P on FC6 -
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] H323-to-SIP proxy
What about the SIP leg?
- Mensaje Original -
De: Michelle Dupuis [EMAIL PROTECTED]
Para: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Enviados: martes 27 de febrero de
Isn't there a zap dummy (or something that uses the RTC) included in
Asterisk 1.40 that creates the timing source? We don't install any external
timing sources and we don't have choppyness problems on pure sip
connections...
Jason - is this on a standard PC motherboard (or a mini device like
I have a high volume asterisk 1.40 installation and I ran a SIP SHOW
CHANNELS. (see partial output below). My questions are:
1. wc-l of the output shows 4000 lines. Does this mean 2000 active calls?
(2 channels per call)
2. The latter part of the output shows unkn for Form column. Why does
My asterisk install is showing the following every 1/2 second:
chan_sip.c:2739 auto_congest: Auto-congesting SIP/my.domain.net-0e118a90
There are lots of calls going through.
1. What can I do about this?
2. Is there a way to limit the number of calls (responding to invites with
no
For an all electronic solution, use fax2mail and mail2fax (from
www.generationd.com). For a fancier all VOIP solution consider hylafax.
For analog only you can plug your fax machine in as you suggest. For a step
up, buy an ATA with T.38 capability and plug your fax machine into that.
MD
You will likely have latency issues - causing choppiness. Start with a
traceroute to validate latency.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support
Visit us at www.generationd.com
-Original Message-
From: [EMAIL
frequencies and what's left? If
there's a lot of noise, then the other party is doing a bad job encoding the
DTMF. Otherwise we can start to chase your machine causes
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit us
Can one do an in-place update to 1.4.1 from 1.4.0 ? (Just compile and
install)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Development Team
Sent: Friday, March 02, 2007 7:04 PM
To: undisclosed-recipients:
Subject: [asterisk-users] Asterisk
We installed a quad xeon 3ghz which transcoded ~100 active channels (as a
gateway). Take a look at the codec demands (in asterisk show codecs I
believe) and scale from there. This box was 60% loaded - which is all we're
comfortable with before latency goes too high.
Michelle Dupuis
Technical
We regularly install * on Fedora (clients with lots of leading edge hardware
like Fedora). No problems
I expect you will only encounter * 1.4.x errors like everyone else.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit
If something changed on its own 6 hours ago (i.e. you didn't touch
Asterisk or its config), then look beyond Asterisk. Did you do a hard
reboot yet? Memory check? Reseat PCI cards and memory? Check power
supply? New drivers/software loaded? (Did YUM run in the background?)
If everything
Asterisk isn't a simple apt-get and run type program...have a look at the
asterisk wiki for help getting started. There's a lot to configure
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josu Lazkano
Lete
Sent: Tuesday, March 27, 2007 11:20 AM
To:
Start with a codec check (sounds like the CNG tone frequencies are out of
spec)...
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Wulter
Sent: Wednesday, March 28, 2007 4:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
You don't need the cfg files (or a tftp) to boot the phones or register.
There are some sample configs lying around, but Aastra's are very poorly
documented (and their firmware still has big bugs - so don't modify from
default too much). We've setup a number of 480i's and got very frustrated
with
Look at a channel bank with MGCP/TDMoE/USB interface (there are plenty
around). We too prefer to keep fxs/fxo hardware outside of the * box.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit us at
www.generationd.com
I'd like to setup a single extension for which all INBOUND and OUTBOUND
calls are recorded to a wav file. I took a look at the wiki:
http://www.voip-info.org/wiki/view/Asterisk+record+calls
but it's not too helpful. Can someone show some sample code in out
recording?
Thanks,
MD
I want to record calls in wav format. Can someone tell me how many MB of
storage per minute each recording requires (assuming SIP / uLaw codec / full
duplex recording)
Thanks,
MD
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Pick a release and stick with it as long as you can. Only when you have to
jump, pick a new release, test the hell out of it, and then leave it alone.
Too many people try to keep on the latest release...
_
From: asterisk-users-boun...@lists.digium.com
I created a mysql table and lookup script for this. One one server were we
could not use mysql, we created an array of exchanges and compared to those.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean
Just check the version of the card (5v vs 3v) - I don't think PCI X is
compatible with the older 5v cards.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, May 27, 2009 9:20 AM
You're not alone...we never found the cause of this (rare) occurance...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Sunday, May 31, 2009 8:58 PM
To: Asterisk Users List
Subject:
, 2009 2:15 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Suddenly the voice became garbage
(likerobot)using Asterisk 1.4.19.2
Michelle Dupuis escribió:
You're not alone...we never found the cause of this (rare) occurance...
-Original Message-
From: asterisk-users-boun
Just out of curiosity, how are you planning to use it? (Reading email,
etc?)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software
Sent: Monday, June 08, 2009 7:58 AM
To: Asterisk Users List
Subject: [asterisk-users]
Yes - we typically install behind NAT. The issue will usually be your
firewall setup ...assuming you have setup your peers for NAT.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo
Martins
Sent:
I have a asterisk box with some queues for a call center and need help on
two points:
1. I have a scenario where if a queue has no agents logged in, an inbound
call should immediately failover to the failover destination for that queue.
However, this does not seem to be working in that, even if
Interesting trick!
On the down side, won't sending this tone be pointless? If the receiver is
not sure a fax is calling, then he will BEEP every caller (even voice
calls). If the receiver is sure a fax is calling, why play the tones?
MD
-Original Message-
From: [EMAIL PROTECTED]
There's a program cid_rewrite (at www.generationd.com) which includes a
blacklist database field. It also autopopulates based on 411.com reverse
lookup.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Swint
Sent: Sunday, October 08, 2006 9:45
Plantronics makes something like this...designed to go inline with handset
cable, with 2 2.5mm audio connectors for connection to PC.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Tuesday, October 10, 2006 9:19 AM
To: Asterisk
Cory,
You may wish to search the archives of this list (and more
appropriately the commercial list).
There seem to be a number of open support issues, lack of
follow-through, and unprofessional behavior on the part of VoIPSupply
support. It's always hard to separate fact from fiction on
Grab the fax2mail script from www.generationd.com and set it to convert the
tiff to pdf before sending. Works great.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: Monday, October 23, 2006 4:38 AM
To: Asterisk Users Mailing List
Take a look at smartCID (found at www.generationd.com)
You can take actions such as block/limit call times/accept based caller
number. It will also fill in the missing CID name based on database lookup
(or 411 reverse lookup).
MD
-Original Message-
From: [EMAIL PROTECTED]
Try to fa2mail script a www.generationd.com
Just set it to email to a null/sink account after fax
receipt.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
RobarSent: Friday, November 03, 2006 12:10 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
We have an AGI app which queries a SQL database, matching the phone number
and checking flag on whether to accept/refuse call.
Go to www.generationd.com for app smartCID.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Friday,
48VDC is a long time telco standard - and has become the Power over Ethernet
standard.
Keep in mind that 'electricity' isn't the measure - it's power. Power is
not synonymous with voltage.
The formula V=IR (voltage equals current time resistance) points to a higher
voltage allowing lower
Anselm:
Try using smartCID (www.generationd.com). You'll get the benefit of ranges
of numbers mapping to single ID's (good for corporate blocks), action field
for blocking/accepting calls, etc).
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I have an asterisk server with a dynamic public IP address. Once the IP
changes, remote clients suddenly have one-way audio again.
I can resolve the problem with a restart, but am thinking have adding a cron
command which does this every night. Will a reload cause asterisk to
respect the new
Check out www.generationd.com for a couple of useful scripts (fax2mail and
mail2fax). If I interpret your question properly, you looking for scripts.
If in fact you are looking for sendmail/libtiff help, have a search through
the archives.
MD
-Original Message-
From: [EMAIL PROTECTED]
Well, I'll bite and get the war going.
I'm putting in my first Sangoma card at the moment...so I have some current
experience.
The card installs great. Hardware compatibility is good (tried in a few
machines). Documentation on website is weak.
Tech support...mixed. I spent a lot of time
I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian
Option 61C. Calls either way drop with error Channel 0/23, span 1 got
hangup, cause 100. Can anyone offer insight into the cause and
solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading
matching zaptel
:[EMAIL PROTECTED] On Behalf Of
Michelle Dupuis
Sent: Tuesday, October 30, 2007 9:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Digium Vs sangoma Hradware
Well, I'll bite and get the war going.
I'm putting in my first Sangoma card
The T1 was setup as tie line, not a trunk. The Bell guy tried setting up
the line 2 ways:
1. As a trunk. This did not work because:
a) When he typed in the access code for the trunk on a phone set (and
then any numbers), the call never appeared on the Asterisk side.
b) The Bell guy said
We are connecting an asterisk box to a Nortel Option 61 via a T1 with PRI.
We have hit a problem we cannot overcome; specifically, the Nortel is asking
for the ROSE information element (IE) over the PRI connection. This causes
libpri to drop the connection, with cause 100. The Nortel cannot turn
: [asterisk-users] PRI over T1 calls dropping, cause 100
Michelle Dupuis wrote:
I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a
Meridian Option 61C. Calls either way drop with error
Channel 0/23, span 1 got
hangup, cause 100. Can anyone offer insight into the cause
Are all of your settings getting lost, or just some? We've encountered some
interesting bugs in the Aastra's...(tech support said wait for the next
firmware release, for 8 months - and yes there have been firmware releases
in-between).
If your tftp is in fact working, strip you .cfg down to the
Use the web interface of the phone to retrieve the config file that you
uploaded. Is it only partially there?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Roi Stork
Sent: Wednesday, November 07, 2007 9:27 PM
To: Asterisk Users Mailing List
read from the cfg file
and set, but it wasn't the case.
Same problem happened to your setup?
On Nov 7, 2007 6:33 PM, Michelle Dupuis [EMAIL PROTECTED] wrote:
Use the web interface of the phone to retrieve the config file that
you uploaded. Is it only partially there?
-Original
Have a look at the smartCID script on www.generationt.com
It allows you to have a database of numbers and override the name (and
number), flag numbers for screening, etc.
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman
Sent: Thursday, November 08,
We have a client with a Nortel PBX with digital phone sets. Due to T1
problems (old firmware), we are interested in trying a FXO channel bank.
Is there a channel bank (or equivalent) which emulates Meridian digital
phone sets? In order words, an FXO channel bank that's Meridian digital?
Can someone advise on how to go about finding someone QUALIFIED to make
changes to libpri?
We have a pilot stuck on hold, due to old buggy PRI software on a meridian
PBX. Upgrading the meridian software is not an option, sowe would like
to have libpri changed to compensate for the bug.
Is
channel bank?
Exists?
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; DelSp=Yes;
format=flowed
Quoting Michelle Dupuis [EMAIL PROTECTED]:
We have a client with a Nortel PBX with digital phone
Can someone explain what the facilityenable setting does in zapata.conf
I've read the wiki archive, but it's not even clear what an ISDN
facility is.
Thanks,
MD
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asterisk-users
There is a bug in the 480 firmware where if the callerid of the incoming
call is malformed (or basically the Aastra doesn't like, for example have a
# sign in the number), the phone won't ring.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Well, we can already integrate to major platforms via SMTP. The real value
is in deep integration to the most popular email platform in business:
Exchange.
I would love to see smart Exchange integration, where deleting the VM
attached email will delete the corresponding message from asterisk.
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