-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Harald Holzer
Sent: Wednesday, August 24, 2005 3:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users]
Why don't you post YOUR config files, then you might get some replies as
to what is wrong.
What you are trying to do can be done.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Huw Morgan
Sent: Tuesday, August 23, 2005 8:33 AM
To:
there?
See comments inline!
Damon Estep wrote:
I have officially engaged in a pissing contest with the local Telco
over
PRI calling name delivery.
Welcome to my world, I deal with theses guys daily! Errgiant arn't
they. We have a saying around work 'The telco is always wrong
I have officially engaged in a pissing contest with the local Telco over
PRI calling name delivery.
The telco publishes their calling name delivery over PRI feature as
being bellcore gr-1367-core compliant.
The gr-1367-core spec states that the calling name is to be included as
a facility IE in
I am using realtime mysql for extensions, sip, and voicemail.
Outbound call routing does not really perform well in realtime
extensions due to the high number of rows in the database (300k), so I
can not use it. It appears with my limited knowledge that the query
method is not robust enough for
for mysql query from dialplan
On Thu, 2005-08-18 at 09:41 -0600, Damon Estep wrote:
I am using realtime mysql for extensions, sip, and voicemail.
Outbound call routing does not really perform well in realtime
extensions due to the high number of rows in the database (300k), so
I
can not use
,
r.NPA DESC, r.NXX DESC
Query took 0.0025 sec.
I don't see how your table with 300K rows is preforming worse than
ours.
You got indexes?
To make this even better, our MySQL server is a Quad P3 500 Mhz
machine.
Works great here.
-Matthew
Damon Estep wrote:
I am using realtime mysql
Anyone know if the application command Realtime() in asterisk can do
more complex queries, like match the values in 2 columns?
Show application realtime suggests it might be limited to one parameter
queries.
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Asterisk-Users mailing list
There is a different approach to this;
Put a priority 'a' in the extension dialplan that goes to
Voicemmailmain(${EXTEN})
Users then dial there own extension from any location and press the *
key once voicemail picks up.
This method seems to emulate what most people are already used to.
If you
It seems that some options are not re-read when caching is on, for
example, changing the caller ID value in the sip table has no effect
until a reload (or expiration), so at least in some cases
rtcahcefriends
makes realtime notsorealtime.
No. It is doing exactly what it says it
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Wednesday, August 17, 2005 9:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] realtime caching
It seems that some
It was fixed a while ago, download new code. There is a bug in the
tracker on it.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Wednesday, August 17, 2005 9:23 AM
To: Asterisk Users Mailing List - Non-Commercial
Hello,
How do you guys implement LCR in Asterisk?
I have experimented with 2 ways, both seem to have issues and further
testing is taking place now.
Method1, use realtime for extensions and load your routing tables in an
outbound context. Our requirements are LCR for the ~150,000 USA
It is amazing to me at this point that there is not an official
Digium list of supported servers (including 1u models!). Clearly the number 1
issue with the Digium PRI cards is the server that they are used in.
The new cards even go as far as listing server that DO NOT
work on the
Is there a method in SIP to set the CALLING number type to
national and the calling number plan to isdn? I am dealing with an issue where
a media gateway is not sending the correct values and would like to know if SIP
has an equivalent parameter that can be set and mapped in the media
?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
Sent: Tue, 16 Aug 2005 12:57:14 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk and LCR
On Tue, Aug 16, 2005 at 10:22:01AM -0600, Damon Estep wrote:
Any input from others that have
Damon Estep wrote:
What 1u server combos work with the new quad pri cards UNDER LOAD
(more
than 75% channel use). Every user that buys a Digium PRI card should
not
have to play hit or miss with 2 or 3 servers that cost more than the
card to get it to work.
We use a Sangoma 4 port
Are you saying realtime mysql is not clever? That is exactly what it
is
supposed to do.
BTW, how do you integrate mysql with asterisk?
any link, documention, tutorials would be greatly helpful.
Search www.voip-info.org for asterisk realtime
to
libpri.
http://sangoma.com/linux/README.asterisk
Hope that helps.
Chad
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Damon
Estep
*Sent:* August 16, 2005 12:33 PM
*To:* asterisk-users
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Anthony Rodgers
Sent: Tuesday, August 16, 2005 1:21 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Called Party Identification on Polycom IP501
Greetings,
The
directly to
libpri.
http://sangoma.com/linux/README.asterisk
Hope that helps.
Chad
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Damon
Estep
*Sent:* August 16, 2005 12:33 PM
Can anyone shed some light on realtime caching?
My desired behavior is that MWI works with realtime
voicemail/sip/extensions AND updates to the database take place on the next
call to the extensions.
Right now I have rtcachefriends=yes, and MWI works, but
updates to the database for
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Tuesday, August 16, 2005 4:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] realtime caching
I have reviewed the
Try quotes and no spaces between name and number.
Callerid=first last2471
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Anthony Rodgers
Sent: Tuesday, August 16, 2005 5:31 PM
To: Asterisk-Users@lists.digium.com
Subject:
Block sip on a firewall between * and the public internet, and then
create rules for your peers IP range.
This assumes you know the IP that all peers and client use; if not just
block from regions of the world you do not need to connect to/from.
We find that most hack attempts come from one well
Damon Estep wrote:
When executing: Dial (SIP/[EMAIL PROTECTED],60
mailto:SIP/[EMAIL PROTECTED],60) I get about 15 seconds
of
ringing, the called party rings, but if not answered in the ~15
seconds
I get back SIP 480 temporarily unavailable.
If the call is answered everything
this to -dev since it seems to be going that route.
Damon
See apps/app_voicemail.c:
#define MAXMSG 100
Then recompile the app and reload the module (or restart asterisk).
--Luki
On 8/12/05, Damon Estep [EMAIL PROTECTED] wrote:
Anyone know how to override the 99 message limit in voicemail
When executing: Dial (SIP/[EMAIL PROTECTED],60)
I get about 15 seconds of ringing, the called party rings, but if not answered
in the ~15 seconds I get back SIP 480 temporarily unavailable.
If the call is answered everything is fine and the call will
continue as expected.
The call is
Anyone know how to override the 99 message limit in
voicemail? (yeah, we have a public VM that gets that many a day).
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To
So caller ID name is passed when available and nothing is passed when
not?
That worked. The following line also got rid of asterisk without
entering any custom info:
callerid=
Thank you,
Hugh
On Thu, 2005-08-11 at 03:19 +0100, Tony Hoyle wrote:
In the [default] section of
, voicemail_users, that you created:
category,
var_name,
var_val,
cat_metric,
filename,
commented
Every item mentioned in a Select query must exist in the table that is being
queried.
Rollin Weeks
On 8/10/05, Damon
Estep [EMAIL PROTECTED]
wrote:
I'm having a few issues with the MySQL realtime
I'm having a few issues with the MySQL realtime configuration in
CVS-HEAD. I tested it initially with realtime extensions (realtime_ext
= mysql,asterisk,extensions) and a realtime switch in extensions.conf
and that works fine, So I though I'd go back and test a static
configuration mapping.
Anyone out there have success getting caller id name from a
pri, through a lucent tnt, to asterisk?
What about from other media gateways?
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Asterisk-Users@lists.digium.com
How many digits is your pri provider sending in the setup message? It needs to
match your dilaplan, ie if they are sending 4 you need 4 digit extensions or
some other monkey business to translate.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Panitaxx
-extensions
[internal-extensions]
;sip users with 10 digit extensions
[egress]
;media gateway terminating local 10 digit calls
[pri-ingress]
;inbound PRI via media gateway
Regards,
Derek
- Original Message -
From: Damon
Estep
In the example below context2 is included in context3
because it is included in context1.
Is there a way to include context2 in context1, and context1
in context3, but not context2 in context3 as a result.
[Context1]
;sip users with 10 digit extensions
Include = context2
We recently upgraded a production system
to current cvs head, things are working well. We do use queues extensively. There
were two bugs in our environment that have been fixed as of 8/3/2005, one was a
segfault in voicemail if a user did not enter a password and hung up, the other
was the
I have added the following to a macro that is used for all
extensions so a user can access voicemailmain by pressing * during the
voicemail prompt
; check voicemail
exten = a,1,voicemailmain(${macro_exten})
exten = a,2,hangup
The behavior is a little weird, the * key is not
?page=Asterisk%20cmd%20Macro
On 8/2/05, Damon Estep [EMAIL PROTECTED] wrote:
I have added the following to a macro that is used for all
extensions so
a
user can access voicemailmain by pressing * during the voicemail
prompt
; check voicemail
exten = a,1,voicemailmain
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Wilson Pickett
Sent: Friday, July 15, 2005 1:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * behind NAT and local subnet
Asterisk
correctly the externip and localnet keywords in sip.conf?
Julian.
On 7/15/05, Damon Estep [EMAIL PROTECTED] wrote:
I have an * box behind a NAT router (static NAT, port ACLs set up
correctly)
Most of the SIP users are on the local subnet with the * box, they
work
fine
Take one
Does anyone have a mirror of this running?
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I have an * box behind a NAT router (static NAT, port ACLs
set up correctly)
Most of the SIP users are on the local subnet with the *
box, they work fine
Take one of the same users off of the local subnet and come
in through the NAT router and these results;
The remote user can
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Walid Azab
Sent: Tuesday, July 05, 2005 4:23
AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk
on Linksys WRT54G
Hi all,
Any one tried installing
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Deon
Sent: Tuesday, July 05, 2005 8:32 AM
To: Asterisk Users
Subject: [Asterisk-Users] How does Vonage support fax machines?
My boss is insisting we support fax, and I keep telling
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists)
Sent: Tuesday, July 05, 2005 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How does Vonage support fax machines?
Dear All,
I
am using Linux-High Availability between two Asterisk servers, everything is
fine but I do have one problem with
this, When a server fails and the other assumes the ip address and start
asterisk on server 2, the ip phone must
re-register themselves again, otherwise the
If you need a fast solution put two gotoif
statements in a row, one to check for the first condition, another to check for
the next, you can leave out the redirect If the condition is not matched so it
just goes to the next priority.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
The 1 m internet connection will be the
limiting factor in your setup, you did not state what type of internet
connection, but given the speed of 1 mbit it must be DSL (or maybe fraction
t/e1).
Is the outbound speed also 1m? Is there
data on the line also? How much? What about voice
I have an application that calls for a single greeting to be used
exclusively in a voicemail box (rather than busy/unavailable).
It is simple enough to implement in the dialplan, but is there a way to
remove the option in the voicemail menu to record the busy greeting
which only serves to confuse
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of kurt x
Sent: Friday, June 24, 2005 6:24 AM
To: Asterisk
Subject: [Asterisk-Users] Dial peer preference
Does Asterisk support preference for the dial peers.
For example:
I
will call them
when
a call comes in, but they are free to make outbound calls in the
meantime.
Julian.
Damon Estep wrote:
Is there a way for a logged in agent (hearing music on hold) to
initiate
an outbound call without logging out of the queue?
We want sales agents to be able to make
Damon Estep wrote:
I assume the bandwidth is being donated or something, but surely
someone
would be willing to donate reliable bandwidth as the knowledge
hosted on
the site (which is also donated!) is worth way more than the
bandwidth.
Sure it's the bandwidth? If the wiki is loaded, I
call * to make an outbound call .
Julian.
Damon Estep wrote:
Yes, I know.
In this case the agent is logging in from a remote phone (pots line)
and
staying logged in. If they used agentcallbacklogin they could make
outbound calls, but the long distance bill would hit their line
Anyone have any insight as to why voip-info.org has been up
and down all day, and more importantly unreliable for the last month?
I assume the bandwidth is being donated or something, but
surely someone would be willing to donate reliable bandwidth as the knowledge
hosted on the site
Is there a way for a logged in agent (hearing music on hold)
to initiate an outbound call without logging out of the queue?
We want sales agents to be able to make outcalls when there
is no callers in queue, but still be logged in to get new inbound calls if they
come in.
?
If you do a sip show peers I think you will see that your PAP2 setup
registers its port with * as being 5060 on line 1 and 5061 on line 2,
but it stills calls port 5060 on asterisk when it makes the
registration.
I think * is actually listening on the first configured port.
You might get the
Forget about MS SQL, odbc drivers that run on linux to talk to MS SQL
stink Odbc in general stinks.
You might be able to get MS SQL DTS (data transformation services) to
link to the mysql database and present the data as it were in your ms
sql database.
There is a pretty good odbc 3.51 mysql
I just ran a couple of test with CVS Head
Port=5060
Port=5061
Result = chan_sip reports listening on 5060
Port=5061
Port=5060
Result = chan_sip reports listening on 5060 (ignoring port=?)
Port=5061
Result = chan_sip STILL reports listening on 5060
Bindport=5061
Result = chan_sip reports
in mysql with
data
from MSSQL? App is running on .NET, in this case it will need to have
assess to both DBs and update them simultaneously. Sorry, I'm not a DB
admin.
I.N.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Monday
-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Listen to multiple ports
Hi,
What about using IPTABLES DNAT stuff in order to map all incoming
5061 traffic to 5060 port ? That may work.
On 6/14/05, Damon Estep [EMAIL PROTECTED] wrote:
Conclusion - asterisk only listens
Second day in a row...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Marcel van Kaam, Fonetica
Sent: Tuesday, June 14, 2005 8:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] VOIP-INFO
Sounds like to much use of the general context, remove etensions from
general that you require authentication for or use includes.
Post you extensions.conf for better help.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stojan Sljivic
What is the deal with voip-info.org, is it a commercial agreement or a
donation that has worn out its welcome? Needs more bandwidth or a faster
(load balanced) server!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent
Wow! I never learn so much! Thanks Guys
So if I understand correctly, a full T1 should be 1.5Mbps full duplex.
And
it should support 22 SIP Users at once - Right?
Bart
Probably closer to 20 depending on setup/teardown frequency. This is
only if the line is dedicated VoIP, no other
Is there any way to accomplish the following? (searched and searched and
can not find any examples)
In extensions.conf (text file) define a macro that accepts a handful of
arguments
From realtime mysql (extensions) - call the macro with arguments (where
the macro is static in the text file)
If
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: Monday, June 13, 2005 6:17 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SNOM, Asterisk and Attended transfer (bug?)
Hi,
I am using a number
It seems that the wiki pages at www.voip-info.org are not responding,
and this has happened before. Responds to ping but not http requests.
Is there a session limit on the web site? Is it too low? Maybe another
explanantion?
Anyone else notice?
___
You are aware that DSL (even SDSL) is half duplex and a T1 is full
duplex, right?
1.5m sdsl can only do 768 sustained duplex, or 1.5 out 0 in, or 0 out
1.5 in. a T1 will do 1.5 in and 1.5 out sustained.
This is due to a separate transmit and receive path on a t1 and a shared
path on sdsl.
The s
.
Excerpt from
http://www.isp-select.com/SDSL.htm
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Monday, June 13, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users
Notice below that the only forms of DSL touted by anyone as replacements
for full duplex 1.544mbps T1 lines is HDSL.
Telcos regularly use HDSL as replacements for traditional DS1 service
wher the line distances are VERY short, in most cases the HDSL circuit
requires 2 pair, in some very short
Title: Message
Race,
Are you saying that the default is
autocreatepeers=yes?
I was under the impression that the
default is no and yes must be explicitly defined.
Same holds true for insecure=, default no,
optional yes or very.
Please tell me I am not mistaken so I do
not feel
Are you in the USA?
If so call the FCC, they do not like port 5060 blocking (or any other
VoIP port blocking)
See here: http://www.google.com/search?hl=enq=fcc+fine+voip
Not the technical answer you are looking for but the RIGHT answer.
-Original Message-
From: [EMAIL PROTECTED]
Search for asterisk realtime at www.voip-info.org
Answer is yes, mysql or odbc.
Requires head, not stable.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Irakli Natsvlishvili
Sent: Monday, June 13, 2005 11:22 PM
To:
As far as performance, * caches static config, but queries realtime
configs, so scalability must be impacted, but I personally have not
approached the limits of realtime yet.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Irakli
in realtime mode, does * uses static configs at all? Is it
possible
to
operate in realtime mode and have static configs (which are build
based on
information taken from DB) as fallback solution?
I.N.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
I have an issue where when an agent answers a call from a
queue and places the caller on hold the caller hears no MOH and the agent hears
congestion.
When a call is placed on hold that is not from a queue MOH
works fine.
The hold is the SIP hold feature on the phone, not a park.
Is anyone aware of any fixes to DTMF in voicemail after CVS head
11/15/04.
I have seen a few other posts about dtmf failing in voicemail and it
seems in a least one other post the CVS date was around 11/04.
We use snom phones with cvs 11/15/04 dtmfmode=rfc2833
If there are fixes an upgrade
Do you have a firewall between * and the internet?
Have you limited the IP address ranges that have access to *
Did you determine if the other call center uses the same telco, may the
telco has an issue.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
What kind of voip phone? Is it possible the user conferenced 3 calls
inadvertently? Easy to do on some multi call appearance phones (snom in
particular)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Thursday, April 14,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Patrick May
Sent: Thursday, April 14, 2005 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Overheard conversation. Comments please
!
Has anyone ever explored what would be required to enable * to produce
NENA standard CAMA signaling for interconnection with conventional e911
services?
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Call XO www.xo.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of snacktime
Sent: Thursday, April 07, 2005 5:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Getting a good deal on a PRI
A word of caution, we ran that same setup
for a while and then bagged the TDM400P in favor of 2 Sipura SPA2000 ATAs. The TDM400P
kept locking up and the SPA2000 never has. No problems getting fax from * to
the SPA2000 via g.711 over a FastE LAN.
I am not sure if the TDM400P has gotten
What I'm still wondering about is, while you can post to that group,
whether your postings are actually propagated to this list. Did
anybody
try that?
Regards, Bruno.
___
Postings to google are not mirrored here, tried it. I think we are
Why? I'd say it's only a config issue. As long as the google group
has this mailing list as it's only feed and posting to the group
is equivalent to posting to the list everything should be fine.
How do you propose getting posts from google to here? Email?
Why? I'd say it's only a config issue. As long as the google group
has this mailing list as it's only feed and posting to the group
is equivalent to posting to the list everything should be fine.
How do you propose getting posts from google to here? Email?
Well, the group receives
Postings to google are not mirrored here, tried it. I think we are
going
to start seeing many people new to * using the google group and not
getting the benefit of the infinite wisdom here.
They will if google keeps getting the content from this list. The
problem would be that people
On Fri, 2005-04-08 at 12:01 -0600, Damon Estep wrote:
Why? I'd say it's only a config issue. As long as the google
group
has this mailing list as it's only feed and posting to the
group
is equivalent to posting to the list everything should be fine.
How do you propose
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Friday, April 08, 2005 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Google Group?
On Fri, 2005-04
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 9:07 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] unlimited iax termination
We are planning on offering unlimited
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Loretitsch
Sent: Thursday, April 07, 2005 9:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] PRI Advice...
Looking for some help any
Anyone out there (in the US) using a CLEC to do third party local number
ports? Let me be more specific;
Our inbound calls come in via inbound only PRIs from a local CLEC, our
outbound calls go via SIP termination to a wholesale VoIP carriers
softswitch.
On the inbound numbers we use the
numbers as an LSR
charge from other carriers NRC.. and as low as 5 cents MRC per Month.
I've also seen cases with no MRC per DID per month, but an NRC per
number.
-m
On Thu, 7 Apr 2005, Damon Estep wrote:
Anyone out there (in the US) using a CLEC to do third party local
number
ports
http://groups-beta.google.com/group/Asterisk-test
Stuff shows up fast! Anyone have insight on this, did I miss something?
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To
Subject: Re: [Asterisk-Users] Asterisk Google Group?
Damon Estep wrote:
http://groups-beta.google.com/group/Asterisk-test
Stuff shows up fast! Anyone have insight on this, did I miss
something?
Looks like a mirror of the mailing list...
--
Cheers,
Matt Riddell
Hi,
This is not entirely an asterisk question but I figure someone
here may know the answer to this question.
On several occassions we will lose the ability to use one of our
PRI lines well for our phone system anyway (we also sometimes
lose PRIs on some of our access
Title: Message
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William M. Sandiford
Sent: Tuesday, April 05, 2005 7:53
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] OT: CRTC
mandates 911/E911 for VoIP in Canada
For those of you out
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andre Normandin
Sent: Tuesday, April 05, 2005 4:36 AM
To: Asterisk-Users
Subject: [Asterisk-Users] WRT54GP2A-AT
Hi,
I've seen these Linksys wireless routers with an ATA already
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of James Taylor
Sent: Tuesday, April 05, 2005 9:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] VOIP 911 Mandatory in Canada
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