From: Steve Kennedy [EMAIL PROTECTED]
Date: Mon, 30 Apr 2007 19:33:43 +0100
Does anyone know of an (E)AGI or program to develop a IVR dial-plan
which will take a list of words and then do something when a unique
branch has been found.
i.e.
Say there's 3 words
demon
deacon
bishop
On a phone
I have transitioned to other DID's. I think that company is out of business.
Sellvoip is best avoided at all costs.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las
I suspect that Jed has a substance abuse problem and that he may be in
rehab. I don't know for sure of course. This kind of silence is indicative
of people being hauled back to rehab. Anyway, maybe he just makes a habit of
running off with people's money.
Thanks to all who replied to my thread a few days ago SIP devices
with packet loss tolerance. One of the suggestions that came out of
that thread was to replace routers at users' premises with ones that
support QoS.
I've used m0n0wall's QoS in the past with reasonable success, but
Bruce McAlister wrote:
Hi All,
I have an issue with the ODBC voicemail storage option with asterisk. All
appears to work fine, however, I get several sql execute warnings. I was
wondering if anyone out there could help me get to the bottom of what is
causing this and how I could possibly go
Kristian Kielhofner wrote:
After several years of using Asterisk I have always been frustrated
by the support for DNS. I have seen all kinds of strange behavior
when Asterisk is used on a system with iffy DNS servers:
Maybe that's where you need to start - by fixing the iffy DNS? :-)
- no
Well on the other side of things there are plenty of adsl equipment running
linux and qos capables and customizable firmware. Normally you can get the
source of the device with binary drivers of devices like adsl wireless or
ethernet switch.. but as long as you stay with the linux version and
I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a
little dim if they believe they can openly go about borrowing
email-addresses like this.
/Per Jessen, Zürich
--
http://www.spamchek.com/ - managed email security.
___
I have 4-5 different Nokias, none have a 2.5mm jack. Nothing that even
remotely resembles a jack.
The older ones did have 2.5 jacks
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Anyone want create a fix for our issue (I will get a price from the client on
how much he wants to spend)? Will forcing attended transfers fix this ?
- Original Message -
From: Jonathan Barratt
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, May 01,
I am using the same Asterisk server for 2 different functions. I have users on
one side and have a calling platform on one side so I put in a context under
general but then only the context for a2billing (calling card platform works)
and the other extensions won't work. Below is how I have it
Try DIDx.net, I would not say they're best but at least they willing to help
you when there is problem and they have a large pool of numbers.
-- Original message --
From: Salvatore Giudice [EMAIL PROTECTED]
I have transitioned to other DID's. I think that company is
We had the same problem as well.
We ended up blocking all REFER requests on our SIP proxy when the URI was for a
PSTN number. A bit inconvenient for customers but preferrable to losing buckets
of money.
Regards,
Grey Man
- Original Message
From: Jonathan Barratt [EMAIL PROTECTED]
Per Jessen wrote:
I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a
I got it as well.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
So I have whose autoattendant is colliding with their extensions...
Quick fix anyone?
Second someone presses say a person's extension (101) ... Autoattendant
sends them to the first context...
[companyx-main-aa]
exten = s,1,Background(companyx/companyx-main)
exten =
Hi all,
my local channel seems to be not working properly. im doing this:
exten= s,1,Dial(Local/[EMAIL PROTECTED],,Tt)
some times it rings the phone at extension 123, and sometimes it doesn`t.
When it doesnt, it actually displays a msg that it could not find that
extension.
[May 1 16:54:02]
Hi all,
I have the same problem using SIP with G729 and it's just on one direction.
But ... there is bandwidth management on the FW equipment (sonicwall) and
others clients (we are a IP centrex) works find using the same server.
A idea ?
Thomas
I am reading comments on the Wiki for meetme
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
from 2004 about how and AGI does work with non zap channels.
Is this still valid 3 years later and 1.4.4?
How do I bring people into a meetme and play a message to all of them
when they are on SIP
In article [EMAIL PROTECTED],
Jerry Geis [EMAIL PROTECTED] wrote:
I am reading comments on the Wiki for meetme
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
from 2004 about how and AGI does work with non zap channels.
Is this still valid 3 years later and 1.4.4?
Yes it is. That part of
Hi all,
my sip provider does'nt send a 183 Message when the opposite party
rings. It sends the ringing indication on the audio stream. Is there any
chance that the asterisk can analyze this audio stream (meta)
information. I saw there is a zaptel configuration entry that sound
pretty close
You need to include the context to the extensions (10x)
On 5/1/07, J. Oquendo [EMAIL PROTECTED] wrote:
So I have whose autoattendant is colliding with their extensions...
Quick fix anyone?
Second someone presses say a person's extension (101) ... Autoattendant
sends them to the first
Hi
I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've
allowed ulaw and g729. I want to change the codec for outbond calls. Please
help not able to find anything using search.
thanks
arun
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Having master and slave servers in the same switch fabric is the only
situation in which I would consider replication.
The cases that I described were with machines in separate subnets.
Replication simply doesn't work that well when there is significant
latency. Did they mention that in
Per Jessen wrote:
I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a
little dim if they believe they can openly go about borrowing
email-addresses like this.
Me too.
--
Warm Regards,
Lee
___
--Bandwidth and Colocation
On Tue, May 01, 2007 at 05:36:30AM -0400, Doug Lytle wrote:
Per Jessen wrote:
I just got spammed by X
I got it as well.
Same here. However, no point in giving those spammers extra free
publicity on the list...
--
Tzafrir Cohen
Per Jessen wrote:
I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a
little dim if they believe they can openly go about borrowing
email-addresses like this.
Ha -- I was just about to post something myself!
Yes - I got this too, and immediately suspected a cull of
Hello All,
Is it possible to have both Digium cards installed on one Server
(TDM400P and TE405P)?
I have one site which requires both connection POT and T1/E1.
How can I configure both cards?
Thanks,
Nitesh
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where are the out of office replies when they're needed?
On 4/30/07, Dovid B [EMAIL PROTECTED] wrote:
I love these :)
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 27,
That stuff is so dangerous. There are too many compliance requirements
regarding spam. Doing this kind of stuff opens them up to a lawsuit in more
than one state.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
J. Oquendo wrote:
So I have whose autoattendant is colliding with their extensions...
Quick fix anyone?
Second someone presses say a person's extension (101) ... Autoattendant
sends them to the first context...
Two things:
1 -- your include statement is missing. Asterisk doesn't even know
Nitesh Divecha wrote on 5/1/07 10:28 AM:
Is it possible to have both Digium cards installed on one Server
(TDM400P and TE405P)?
I have one site which requires both connection POT and T1/E1.
How can I configure both cards?
Should work just fine. The Zaptel drivers will pick up both.
Stephen Bosch wrote:
J. Oquendo wrote:
So I have whose autoattendant is colliding with their extensions...
Quick fix anyone?
Second someone presses say a person's extension (101) ... Autoattendant
sends them to the first context...
Two things:
1 -- your include statement is missing.
Stephen Bosch wrote:
Oh and by the way... What i did was... I added a number 5 then sent that
to its own context...
exten = 5,1,Goto(companyx-directory,s,1)
[companyx-directory]
exten = s,1,Background(companyx/companyx-directory)
exten = 1,1,Dial(SIP/companyx100,15,tr)
exten =
Dave Miller wrote:
Nitesh Divecha wrote on 5/1/07 10:28 AM:
Is it possible to have both Digium cards installed on one Server
(TDM400P and TE405P)?
I have one site which requires both connection POT and T1/E1.
How can I configure both cards?
Should work just fine. The Zaptel
Hi All
We have a private network setup (no nat) with three types of phones
connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco
7940 IP phones.
When we ring polycom to grandstream or grandstream to polycom then both
phones can send and receive voice fine and all is well.
When
Put similar allow/disallow statements in the sip or iax entry you create for
your outbound ip calls. Be aware that if you use different codecs for phones
and your termination provider, all media will have to go through asterisk
and you will incur the processing overhead of codec conversion.
I have a pc with the following characteristics:
Pentium IV 2.4Ghz HyperThreading
512 MB PC3600 Dual DDR RAM
Seagate 80GB SATA HDD
4-port ethernet 10/100 PCI Card
Netgear MA-311 802.11b Wireless Card
On this machine runs a VPN server, an Apache server and an Asterisk
Does anyone know
i sent a product suggestion to HP. It was a request to use software that
already exists in their JetDirect and Multifunction Fax machines to make
them seamlessly interoperate with a fax gateway in a way transparent to the
end user. Essentially, giving the sysadmin a choice in fax transport
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone. I'd like to have the
SIP phone ring for some arbitrary number of seconds before it is sent
off to the mobile phone. Using something like a Wait() within a Dial()
would
My problem is this
We have a location outside of our network which is done over vpn.
Everything works except for the voice quality to that location isn't
very good. To try to resolve this, I wanted to try to make all calls go
over gsm. Right now, when i say show sip channels, they all show
You should get a packet capture of both cisco-cisco and
grandstream/polycom-cisco. Compare the SDP's. The cisco phone may not be
able to understand the other vendor's devices. BTW, what version of firmware
are you running on the cisco phones?
--
The answer is about 42 handsets...
Seriously though - you don't mention traffic on the vpn server, you
don't mention traffic on the apache server you don't mention anything
about transcoding, conference rooms, or if you are using SIP or IAX.
You ask an unanswerable question so my
Hi All,
I tried to send this email this morning, but I think it has been moderated
due to size issue's, so I'll resend it again in 3 parts:
PART 1
Hi All,
Just an update, after looking a little further into this, it appears that *
tries to delete a record that does not exist before inserting
Hi All,
I tried to send this email this morning, but I think it has been moderated
due to size issue's, so I'll resend it again in 3 parts:
PART 2
Database Table Definition (taken from asterisk readme's)
CREATE FUNCTION loin (cstring) RETURNS lo AS 'oidin' LANGUAGE
internal IMMUTABLE
Will you be allowing reinvites? If the server processes media, it will
obviously support less simultaneous calls. Also, you may want to rethink the
wireless portion. Odds are you will have horrible QoS problems if you run
multiple calls or mixed traffic over wireless. BTW, what do you use VPN for?
Hi all,
I have created a menu from which the caller can select several options such as
being transfered to our phones and my mobile phone, meetme, etc. If the caller
press an invalid option i have set it to play a message like invalid choice
please try again. If the caller make three invalid
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone.
Not directly, but yes. Hint: Local channel + Wait. Something like this:
Dial(SIP/phoneLocal/[EMAIL PROTECTED])
[delayed]
exten = XX,1,Wait(10)
exten =
Antonopoulos Angelos wrote:
I have a pc with the following characteristics:
Pentium IV 2.4Ghz HyperThreading
512 MB PC3600 Dual DDR RAM
Seagate 80GB SATA HDD
4-port ethernet 10/100 PCI Card
Netgear MA-311 802.11b Wireless Card
On this machine runs a VPN server, an Apache server and an
It's probably not your codec. Do you have your asterisk box on a Voice VLAN
with priority queing configured? If you have mixed traffic on your uplink
without VLAN's and priority queuing (or possibly 802.1p), then your QoS will
suffer. Changing your codec to GSM will lower bandwidth consumption,
Hi Steve,
put a timeout in the Dial command, if the call isn't answered it
returns after the timeout has expired
e.g.:
exten = _X.,1,Dial(SIP/${EXTEN}|15)
It waits 15 secs for the call to be answered
Look at http://www.voip-info.org/wiki-Asterisk+cmd+Dial for more
informations
Regards
Hi,
I have a problem where some PRI channels get stuck in a Call mode. If I do
a zap show channel XX, it shows as PRI Flags: Call. However there is no calls
on that channel. Trying to force a hangup does not work:
[EMAIL PROTECTED] Dialer]# asterisk -r -x soft hangup zap/27-1
At work, I have 4 branch offices at which I've deployed asterisk.
Call termination/origination at each branch office is handled either
through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing
the legacy PBX at our HQ with asterisk.
Each branch office has between 3 and 20 employees,
However, even once I reloaded the extensions, its still only
using ulaw.
You didn't reload the sip config? Maybe that's your problem?
--
Andreas Sikkema
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asterisk-users mailing list
To
Steve Finkelstein wrote:
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone. I'd like to have the
SIP phone ring for some arbitrary number of seconds before it is sent
off to the mobile phone. Using something like a
I want to capture all my Asterisk traffic (including RTP) and then analyse
it.
My plan was to use tcpdump and then analyse with Wireshark. The following
works:
tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1
But I want to be a bit more selective:
tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp
Hello All,
To avoid conflicts I removed TE405P and left the TDM400P and
reconfigured the card using genzaptelconf.
When I run ztcfg -vv I saw the card and modules are loaded and also I
used ztmonitor 1 -v and I saw the gain moving up and down. I did
create trunks and outbound routes using
Test emails and out of office emails make my day.
- Original Message -
From: Wilson Pickett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, May 01, 2007 5:37 PM
Subject: Re: [asterisk-users] Test
where are
You can use real time with an agi.
- Original Message -
From: mohammad mirzaee
To: asterisk-users@lists.digium.com
Sent: Sunday, April 29, 2007 12:50 PM
Subject: [asterisk-users] Voicemail Creation
HI All;
I want to use Asterisk for just Voicemail Server and I need a
I was in the asterisk console and I typed reload. Is this not enough
to reload the sip.conf file?
Rob
Andreas Sikkema wrote:
However, even once I reloaded the extensions, its still only
using ulaw.
You didn't reload the sip config? Maybe that's your problem?
Can you get a layer 3 analyzer trace (e.g., Q.931) and post it here?
John Treble
Ottawa, Ontario, Canada
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andre Courchesne - Consultant
Sent: May 1, 2007 12:44 PM
To:
Hi Christian,
Increase a variable in the menu loop, or exactly in the t and i
extensions like this :
exten = s,1,Wait(3)
exten = s,n,Answer()
exten = s,n,Set(LoopStep=1)
exten = s,n,Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Wait(1)
exten =
You mean a PRI debug trace? right now I have some channels that are in this
state. There is not much I can do as this is a production system...
John Treble wrote:
Can you get a layer 3 analyzer trace (e.g., Q.931) and post it here?
John Treble
Ottawa, Ontario, Canada
-Original
I have run into the exact same situation and have the same question. I did
it in the dial plan manually due to time contraints but if DUNDi or ENUM or
something else is better suited I would love to know.
Also the guides and tutorial that I found did not touch on specifics for a
situation like
Erik,
Your setup is very similar to one of my own, and I started of manually
configuring it, creating IAX connections for each site and then using dial
plan to route the call. When I looked at Dundi and finally got it working, I
have one IAX connection for all sites and the connections are
Hello,
I would like to implement a few decision making process inside the dialplan
using information stored in MySQL (like LCR, etc.). I see the MYSQL()
application, but as far as I understand I have to connect to the database each
time I want to query it; this seems a CPU eater to me. Is this
Yeah that is fine. You don't need to do any more than that.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214)
Me 3.
- Original Message -
From: Salvatore Giudice [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, May 01, 2007 5:38 PM
Subject: RE: [asterisk-users] did we all get spammed by TechnoCo ?
That stuff is so
I think you want:
tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange
5060-65534
dst port port
True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a
destination port value of port. The port can be a number or a name used in
/etc/services (see tcp(4P) and udp(4P)).
Reload will reload your sip.conf file! As well as iax.conf,
extensions.conf, queues.conf, voicemail.conf, users.conf
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall
Sent: Tuesday, May 01, 2007 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial
The RTP traffic is not going to be on port 5060, that is the sip only. Check
your rtp.conf file in asterisk for the port range used for RTP traffic.
On 5/1/07, CSB [EMAIL PROTECTED] wrote:
I want to capture all my Asterisk traffic (including RTP) and then analyse
it.
My plan was to use
On Tue, May 01, 2007 at 02:01:33PM -0400, Nitesh Divecha wrote:
Hello All,
To avoid conflicts I removed TE405P and left the TDM400P and
reconfigured the card using genzaptelconf.
When I run ztcfg -vv I saw the card and modules are loaded and also I
used ztmonitor 1 -v and I saw the gain
DUndi or enum only make sense if you plan to move extentions dynamically
without having to touch you Asterisk configs or if you want to expose your
addressing to the outside world.
Personally, I would do it statically so you can avoid delays in processing
addressing especially - in the case of
DUNDi would be very well suited to this particular application. Publish
the extensions that are reachable at each location and when one site
dials an extension it gets routed to the one that says i have this.
ENUM would probably work just as well for this. I like ENUM with
PowerDNS and MYSQL.
wireshark can further filter out what you don't want,
you can also pipe the dump to grep and match only what you want
On May 1, 2007, at 11:32 AM, CSB wrote:
I want to capture all my Asterisk traffic (including RTP) and then
analyse it.
My plan was to use tcpdump and then analyse with
CSB wrote:
I want to capture all my Asterisk traffic (including RTP) and then
analyse it.
My plan was to use tcpdump and then analyse with Wireshark. The
following works:
tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1
But I want to be a bit more selective:
tcpdump -C 100 -W 10 -w /tmp/tcpdump
From: Knud Müller [EMAIL PROTECTED]
Date: Tue, 01 May 2007 15:19:17 +0200
Hi all,
my sip provider does'nt send a 183 Message when the opposite party rings.
It sends the ringing indication on the audio stream. Is there any chance
that the asterisk can analyze this audio stream (meta)
Eric ManxPower Wieling wrote:
Steve Finkelstein wrote:
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone. I'd like to have the
SIP phone ring for some arbitrary number of seconds before it is sent
off to the mobile
Well, you should be able to leave it open. However, I don't know what
would happen if MySQL times out and disconnects the connection because
it considers it stale. I don't know if you can check that error and
reconnect.
Yehavi Bourvine +972-8-9489444 wrote:
Hello,
I would like to
Would anyone care to recommend a T1 interface method for Asterisk that
would function as an (external) alternative to a PCI card like the
Digium TE120P? Like some sort of T1-SIP gateway?
Also, would anyone with experience using these products care to comment
on the practical value of the
I remember an app called 'vomit' that could allegedly reconstruct audio
files from tcpdump pcap files.
Salvatore Giudice wrote:
I think you want:
tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange
5060-65534
dst port port
True if the packet is ip/tcp, ip/udp, ip6/tcp or
On 5/1/07, Bruce Reeves [EMAIL PROTECTED] wrote:
Your setup is very similar to one of my own, and I started of manually
configuring it, creating IAX connections for each site and then using dial
plan to route the call. When I looked at Dundi and finally got it working, I
have one IAX connection
I haven't been using Asterisk for long, but I have not yet experienced
any DNS-related oddities.
Then keep using it, and you will.
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To UNSUBSCRIBE or update options
That's what I did though. So my sip.conf file no longer has any allows
in it. Instead, it should be relying on the realtime settings for that.
However, even though I told it to only use 5053, it still is using ulaw.
Rob
Salvatore Giudice wrote:
Yeah that is fine. You don't need to do any more
I posted about this problem last week and thought it was a combination
of SIP/ZAP causing issues in Asterisk. Since then I've realized it's
only the SIP channel that's hanging. When this happens a call can still
come in and hit the IVR, but no one can connect to the server from a SIP
client.
Stephen Bosch wrote:
Eric ManxPower Wieling wrote:
Steve Finkelstein wrote:
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone. I'd like to have the
SIP phone ring for some arbitrary number of seconds before it is
I see that stanaphone is not accepting new customers. Does anyone
know if they are doing ok? I have a number with them and would like
to start redirection people before it gets canceled on me if they are
having trouble
thanks
Todd
Not sure if this can be done or not, but I can't seem to find it
anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I
would like to have the caller id of the person I am dialing displayed
and not the number I just dialed. Is this possible? So, if extension
4023 is John Doe, and I
Olle E Johansson wrote:
23 apr 2007 kl. 19.55 skrev Russell Bryant:
John Todd wrote:
To morph this into a -dev thread: if this patch were to become (again)
useful and error-free, is there any objection or usefulness in adding it
to TRUNK? Personally, I think there is, if there is a method
It's amazing how simple some answer are.
Thank you kindly for your responses Edoardo and Luki. :-)
- sf
Edoardo Serra wrote:
Hi Steve,
put a timeout in the Dial command, if the call isn't answered it
returns after the timeout has expired
e.g.:
exten = _X.,1,Dial(SIP/${EXTEN}|15)
It
Yehavi Bourvine +972-8-9489444 wrote:
Hello,
I would like to implement a few decision making process inside the dialplan
using information stored in MySQL (like LCR, etc.). I see the MYSQL()
application, but as far as I understand I have to connect to the database each
time I want to
You could get yourself a cisco universal gateway or a Audiocodes Mediant
1000 Single Span T1 SIP Gateway.
With regard to the cards: In my experience, you want an echo cancellation
card if you are connected to a carrier without echo cancellers. Typically,
LEC circuits do not have echo cancellers
Ethereal will let you export an rtp stream as a .au file. That's one of the
very minor items we cover in our conference series and our VoIP 100 course.
There is a lot more fun to be had when you get into RTP sequence number
prediction and RTP stream I injection.
Salvatore Giudice wrote:
DUndi or enum only make sense if you plan to move extentions dynamically
without having to touch you Asterisk configs or if you want to expose your
addressing to the outside world.
Personally, I would do it statically so you can avoid delays in processing
addressing
How did you set it to 5053?
Can you post your sip.conf? You should remove the passwords and ip
addresses, etc.
Usually, it's just an allow and a disallow statement inserted into each
inbound and outbound channel definition.
--
Salvatore
Eric ManxPower Wieling wrote:
Stephen Bosch wrote:
Eric ManxPower Wieling wrote:
Steve Finkelstein wrote:
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone. I'd like to have the
SIP phone ring for some arbitrary
Write them and ask.
--
Salvatore Giudice
[EMAIL PROTECTED]
VoIP Security Training, LLC
http://VoIPSecurityTraining.com
848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906
-Original Message-
From:
Hmmm that's not good, I've been very happy using them as a backup line
to my packet 8 services.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Todd H
Kevin,
It seems to me that what you are really talking about is manipulating the
display features of the phone. Caller ID is unlikely to have this effect
as the phone does not consider the From: URI in the SIP header unless the
call is of an incoming nature.
The solution to this is bound to
On 5/1/07, Savoy, Kevin - Williston, ND [EMAIL PROTECTED] wrote:
Not sure if this can be done or not, but I can't seem to find it anywhere
on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to
have the caller id of the person I am dialing displayed and not the number I
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Mitch Jackson wrote:
Evening,
My latest asterisk box is having a difficult problem. It is
configured with one TE210P and TDM400P with four FXO modules. I'm
running FC6.
The TE210P only has a single PRI.
When the system boots, it is
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