Why not setup a default catch-all route that goes to either your main line (to
drive sales) or a pre-recorded message (the number you dialed is
disconnected...etc), and then setup more specific pattern matches for assigned
numbers? I've done this before for clients that have large blocks of
On 1 Jun 2011, at 22:50, Jesse Thompson wrote:
We are managing an Asterisk installation for residential VOIP service, and we
are having a problem where all inbound calls to DIDs which are assigned to us
by our wholesaler but not yet assigned to a downstream customer get caught in
a routing
Hello, this is Jesse with Webformix.
We are managing an Asterisk installation for residential VOIP service, and
we are having a problem where all inbound calls to DIDs which are assigned
to us by our wholesaler but not yet assigned to a downstream customer get
caught in a routing loop.
** For
I am using asterisk 1.4.41 and the AMI
I am trying to execute a command over AMI, specifically core show
channels concise
sometimes I get this back:
asterisk_command_show_channels() execute failed. 'Response: Follows[CR
][LF ]Privilege: Command[CR ][LF
Jerry Geis wrote:
I am using asterisk 1.4.41 and the AMI
I am trying to execute a command over AMI, specifically core show
channels concise
sometimes I get this back:
asterisk_command_show_channels() execute failed. 'Response: Follows[CR
][LF ]Privilege: Command[CR ][LF
I an running centos 5. I added this to the digium.repo file in
/etc/yum.repos.d directory.
[digium-current]
name=CentOS-$releasever - Digium - Current
baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/
enabled=1
gpgcheck=0
On 05/16/2011 08:36 AM, Jerry Geis wrote:
I an running centos 5. I added this to the digium.repo file in /etc/yum.repos.d
directory.
[digium-current]
name=CentOS-$releasever - Digium - Current
baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/
enabled=1
gpgcheck=0
On 11-05-06 02:56 PM, Watkins, Bradley wrote:
Yes, use the MinivmMWI application.
That's how I've done it in the past as well.
Leif.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Jerry Geis wrote:
Is there a way in asterisk to Activate/Clear the blinking light on
polycom phones
I've always just dumped a msg000.txt in the voicemail directory of that
phone and removed it when not needed. Under 1.4, the Polycoms act on it.
Doug
--
Ben Franklin quote:
Those who
Is there a way in asterisk to Activate/Clear the blinking light on
polycom phones
indicating VM. Either from an AGI or some way in the dialplan?
I want to be able to control this light for from my application.
I have an AGI to do something similiar to VM and want to light /clear
the light
On Fri, May 6, 2011 at 2:14 PM, Jerry Geis ge...@pagestation.com wrote:
Is there a way in asterisk to Activate/Clear the blinking light on polycom
phones
indicating VM. Either from an AGI or some way in the dialplan?
I want to be able to control this light for from my application.
I have an
immediately and
then destroy it.
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Friday, May 06, 2011 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question on ways to activate
I thought I has everything using IP addresses.
I am not making outside calls this is all internal.
I have a connection between two machines both running asterisk.
I am using 1.8.3 and I see a lot of dnsmgr_lookup's for mymachine.
I have a register line in sip.conf that is the only place
On Wednesday 06 April 2011 09:49:17 Jon Farmer wrote:
Hi
I have a call into a MeetMe conference that when I do a core show
channel returns
NativeFormats: 0x4 (ulaw)
WriteFormat: 0x1000 (g722)
ReadFormat: 0x1000 (g722)
Can someone explain what the differences between Native, Wite
When I switched to 1.8 from 1.4 I am getting this error
pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension
(default, s, 1)
http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
This page says its in 1.0 and I dont think has been removed.
Did
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Wednesday, March 02, 2011 3:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question on Asterisk 1.8
On Wed, 2011-03-02 at 16:33 -0500, Jerry Geis wrote:
When I switched to 1.8 from 1.4 I am getting this error
pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension
(default, s, 1)
http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
This
Best guess is that syntax changed from 1.4 to 1.8. Change line to
Exten = s,1,Wait(1)
Danny
Your correct. it was a syntax change. the above works.
jerry
--
_
-- Bandwidth and Colocation Provided by
Jerry Geis wrote:
Your correct. it was a syntax change. the above works.
I've always used Wait(#) in my 1.4.x dial plans.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
Hopefully this is a simple question.
How does a non-secure phone that is on a PBX connected to an asterisk over a
SIP trunk communicate with a secure phone connected to the Asterisk server?
I like to think that the secure call terminates on the Asterisk and the
non-secure call is somehow
On Mon, Feb 28, 2011 at 8:49 PM, Mitch Johnson mitch.johns...@gmail.comwrote:
How does a non-secure phone that is on a PBX connected to an asterisk over
a SIP trunk communicate with a secure phone connected to the Asterisk
server?
I think, although I'm not positive, that if either leg of the
On 2/10/11 5:54 AM, Christian Gansberger christian.gansber...@accm.at
wrote:
Hello,
Maybe try that:
In your incoming isdn context:
[isdn-incoming]
exten = s,1,Set(TIMEOUT(digits)=3)
exten = s,2,WaitExten(2)
exten = s,3,Dial(SIP/operator...)
exten = 10,1,Dial(SIP/10)
exten =
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question about EuroBRI final 2 digits
Hello,
I have an installation in Austria; ISDN service provided by Austria
Telekom. The main number of the service is 6 digits. Incoming calls may
contain 2 additional digits, which I
Hello,
Maybe try that:
In your incoming isdn context:
[isdn-incoming]
exten = s,1,Set(TIMEOUT(digits)=3)
exten = s,2,WaitExten(2)
exten = s,3,Dial(SIP/operator...)
exten = 10,1,Dial(SIP/10)
exten = 20,1,Dial(SIP/20)
So if a call comes in Asterisk waits, 2 seconds for further digits
dialed and
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian
Gansberger
Sent: Thursday, February 10, 2011 5:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question
On Thu, Feb 10, 2011 at 6:08 AM, Andrew Thomas a...@datavox.co.uk wrote:
This sounds like a job for DISA.
http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA
helps.
If OP is using Asterisk18, perhaps we should direct him to look here?
).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Beers
Sent: 10 February 2011 14:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about EuroBRI final 2 digits
2011/2/5 Roberto Piola roberto.pi...@visiant.it
In Italy, you must enable overlapdial=yes
Is this relevant for incoming calls, as OP asked ?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
In Italy, you must enable overlapdial=yes
On Thu, Feb 3, 2011 at 7:45 PM, Cassius Smith cass...@cassius.org wrote:
Hello,
I have an installation in Austria; ISDN service provided by Austria Telekom.
The main number of the service is 6 digits. Incoming calls may contain 2
additional digits,
Hello,
I have an installation in Austria; ISDN service provided by Austria Telekom.
The main number of the service is 6 digits. Incoming calls may contain 2
additional digits, which I then use to route the call to the correct
extension. Telekom sends me all the digits.
My problem is that when
Hi Siobhan,
Asterisk is all capacity to work-on but you need to find out some way of
handling conference system through WEB part , also one more thing on last
point for switching between conference
i am not much sure about it but i think it is possible if i will look into
code implementation.
Anyone else know about the holding concurrent conferences (and switching
back and forth) issue ? Is it possible?
And can you set up dynamic conferences that continue even when the initiator
leaves?
Thanks!
On Tue, Jan 4, 2011 at 7:11 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:
Hi
My company is building a VOIP application, and initially were just using a
barebones OpenSIPS implementation to host one-on-one calls; however, we want
to expand the functionality to conferencing (which, of course, OpenSIPS
doesn't handle) and was looking into Asterisk (the other option being
My company is building a VOIP application, and initially were just using a
barebones OpenSIPS implementation to host one-on-one calls; however, we want
to expand the functionality to conferencing (which, of course, OpenSIPS
doesn't handle) and was looking into Asterisk (the other option being
Currently using 1.4.X and looking to JUMP to 1.8
was reading the docs and have a question.
in 1.4 I could do: /usr/sbin/asterisk -rx meetme to see all the
current meetme's.
I dont see what this is now in 1.8?
Thanks
Jerry
--
Our SMS-gateway is not PSTN accessible.
On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese l...@lcrcomputer.net wrote:
AMARDEEP SINGH wrote:
Hello All,
Scenario:
-We use asterisk as voicemail server for our cellular network. Asterisk box
is talking to Cell switch(GSM/VOIP/PSTN gateway) through
Maybe you need to read the man page for qpage. The qpage client can
send the page to an SNPP server over TCP/IP.
Lyle
AMARDEEP SINGH wrote:
Our SMS-gateway is not PSTN accessible.
On Thu, Jul 22, 2010 at 5:04 PM, Lyle Giese l...@lcrcomputer.net
mailto:l...@lcrcomputer.net wrote:
Do you have working script?
On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese l...@lcrcomputer.net wrote:
Maybe you need to read the man page for qpage. The qpage client can
send the page to an SNPP server over TCP/IP.
Lyle
AMARDEEP SINGH wrote:
Our SMS-gateway is not PSTN accessible.
On
qpage -s snppserver.example.com -p lyle -f lyle test page
AMARDEEP SINGH wrote:
Do you have working script?
On Fri, Jul 23, 2010 at 10:14 AM, Lyle Giese l...@lcrcomputer.net
mailto:l...@lcrcomputer.net wrote:
Maybe you need to read the man page for qpage. The qpage client can
send
Hello All,
Scenario:
-We use asterisk as voicemail server for our cellular network. Asterisk box
is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip.
-Extensions in * are virtual, just for leaving and accessing voicemail.
Requirement:
Asterisk to send SMS to cell switch(running SMSC) on
AMARDEEP SINGH wrote:
Hello All,
Scenario:
-We use asterisk as voicemail server for our cellular network.
Asterisk box is talking to Cell switch(GSM/VOIP/PSTN gateway) through sip.
-Extensions in * are virtual, just for leaving and accessing voicemail.
Requirement:
Asterisk to send SMS to
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jerry Geis
Sent: Friday, June 18, 2010 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question on nortel
I am using asterisk 1.4.32 and wish to connect using SIP to a nortel
1000 switch
with the ability to have 90 calls at a one time outgoing or incoming.
the nortel reseller is asking me what to do. I dont know nortel at all.
I thought I just needed a SIP trunk and IP address of the their server
Hi,
I'm still on 1.4 and am wondering if 1.6 would fix an issue for me.
Specifically, I have been given the impression that, in contrast to 1.4
which always sends packet from the default IP (if the server has multiple
IPs), 1.6 sends packets back from the IP address that was used by the peers.
I'm doing some automated calling by putting .call files in the Outgoing
folder of Asterisk. I'm concerned this might be a stupid question, but I'm
pretty sure I've done my research well and I'm unable to come up with an
answer on my own.
I want to know: what happens to the .call files after the
Ok so a while back I found an example for having a number dial multiple
numbers and then whoever answers and confirms gets the call. (don't
recall who the example was from, but thank you!)
But Now today I've been playing with TTS and STT and came across the
BackgroundDetect command. Now If I
15 jan 2010 kl. 08.23 skrev Yuji Kondo:
I have two questions for Asterisk feature.
1. Can Asterisk support presence feature ?
Asterisk is a telephony PBX and supports presence subscriptions for extension
states - if a phone line is busy or not, over a few different SIP presence
Dear Team,
I have two questions for Asterisk feature.
1. Can Asterisk
support presence feature ?
I found following information. but it is too
old...?
**
Of Robert Lister
Sent: martedì 12 gennaio 2010 18.51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about SIP registration
On Tue, 2010-01-12 at 18:16 +0100, Aggio Alberto wrote:
Then I have configured an account as following:
[999]
type
Hi guys,
I recently faced an issue regarding SIP registration: I have a 2-NIC Linux PC,
with eth0 set to address 192.168.1.1 (NATted over public network, with address
89.X.Y.Z) and eth1 set to address 1.1.1.1. In [sip.conf] I set general option
bindaddr=0.0.0.0; IP address to
Instead of host=dynamic, use host=1.1.1.1, or
host=1.1.1.0/255.255.255.0.
Thanks,
--Warren Selby
On Jan 12, 2010, at 11:16 AM, Aggio Alberto
alberto.ag...@loquendo.com wrote:
Hi guys,
I recently faced an issue regarding SIP registration: I have a 2-NIC
Linux PC, with eth0 set to
On Tue, 2010-01-12 at 18:16 +0100, Aggio Alberto wrote:
Then I have configured an account as following:
[999]
type=friend
username=999
You don't appear to have a secret= line in there with a password
option... or did you snip it?
Can someone explain me this kind of behaviour? Is it
There is a line like in codes/Makefile
$(if $(filter
codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10)
What is filter? Where is filter?
whereis filter doesnt return anything
find . | grep filter in asterisk root directory returns nothing.
Thanks,
Jerry
On Wednesday 06 January 2010 13:45:55 Jerry Geis wrote:
There is a line like in codes/Makefile
$(if $(filter
codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10)
What is filter? Where is filter?
whereis filter doesnt return anything
find . | grep filter in asterisk root
It's a Makefile command. See:
http://www.gnu.org/software/automake/manual/make/Text-Functions.html#index-filter-554
great - thanks
is there no method by the configure command to --disable-FEATURE???
the help says its there but doesnt seem to do anything for me.
example: ./configure
Jerry Geis wrote:
is there no method by the configure command to --disable-FEATURE???
There is not. The Asterisk configure script is used for platform
specific settings, locating libraries and header files and the like. It
is not used (directly) for controlling which portions of Asterisk are
Hi,I want to know how to do to work PLC of Asterisk.
Anyone plz help me.
PLC (Packet Loss Concealment) is included in Asterisk,I read at voip-info.org
or release note.
And I see in codecs.conf, genelicplc setting.
So I put codecs.conf in '/etc/asterisk' ,and wrote genericplc = true.
And I
11 dec 2009 kl. 17.18 skrev Jerry Geis:
Where in the code does something like:
register = user[:secret[:authuse...@host[:port][/extension]
from sip.conf 1) get parsed 2) actually register.
I tried looking in channels/chan_sip.c and don't see where that happens.
Can
Hi,
I was able to implement T122p one port PRI and was able to call out, but I
am planning to use TE412p (includes echo cancellation) 4 port digital card
(PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI
connections) with proper hardware like dual core quadcore processor
The calls itselves doesn't take a lot of CPU resources, even more considering
you're willing to use hardware echo cancelling. The real CPU hogs are apps like
MeetMe() and AGI scripts. Those are no worse than audiotranscoding thought.
You also should design the system in such way there's as few
Hi!
Having two TE410P with heavy load in a Pentium4 3,2GHz system running
Asterisk 1.2 was no problem. It did only IVR and bridging with no
transcoding though.
Chris
2009/12/14 das sandesh sandesh...@gmail.com:
Hi,
I was able to implement T122p one port PRI and was able to call out, but I
am
Hello,
I would like to connect 2 asterisk boxes together, so this is my scenario:
Asterisk Main: it is connected to many sip providers and its main purpose as
a call termination forwarder.
Asterisk B: its connected to E1, and its purpose to terminate calls. It
will receive SIP
Thanks Victor and Vinícius for the information.
I will not be doing any transcoding but using some AGI scripts, I will
update the status once I configure and start using them.
Thanks
Sandesh
On Mon, Dec 14, 2009 at 1:59 PM, Christian Victor
christ...@victormedia.dewrote:
Hi!
Having two
I have been looking for a way from the dialplan to inquire if there are
any members in a queue.
So what I want to do is if no users are members of a queue then I can
send the call to a given extention.
I have the queue setup all that is working. Just need to be able to send
the call to a
On Dec 13, 2009, at 7:20 PM, Jerry Geis wrote:
I have been looking for a way from the dialplan to inquire if there are
any members in a queue.
So what I want to do is if no users are members of a queue then I can
send the call to a given extention.
I have the queue setup all that is
Sent: Sunday, December 13, 2009 4:20:40 PM
Subject: [asterisk-users] question on queues
I have been looking for a way from the dialplan to inquire if there are
any members in a queue.
So what I want to do is if no users are members of a queue then I can
send the call to a given extention.
I
Where in the code does something like:
register = user[:secret[:authuse...@host[:port][/extension]
from sip.conf 1) get parsed 2) actually register.
I tried looking in channels/chan_sip.c and don't see where that happens.
Can someone point me the right file and or function.
Sent: 02 December 2009 01:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Question about g729
All calls.
Landy Landy wrote:
You only need to purchase 10 licenses, if all 10 clients
will be making calls at the same
Sorry for the repetition.
I didn't see the other responses.
-Original Message-
From: Thomas Kenyon dig...@sanguinarius.co.uk
Sent: 02 December 2009 07:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Question
- Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Question about g729
All calls.
Landy Landy wrote:
You only need to purchase 10 licenses, if all 10 clients
will be making calls at the same time.
Ok. Does this apply only for outbound calls
Alex Balashov wrote:
My understanding is that Asterisk will not pass through calls in codecs
for which it does not have support and/or licenses; it simply does not
advertise them in the SDP negotiation.
'support' - yes, 'licenses' - no. Asterisk supports passthrough,
recording and playback
Hello.
I am currently testing an asterisk server using the default codecs, I have
allow=all, and noticed everytime I test it in a wireless lan the latency
rockets off the roof to over 1000ms. I would like to test g729 since it uses
less bandwidth but, read somewhere I have to buy a license per
it better or correct me if
Im wrong.
Dan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: 01 December 2009 23:24
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Question about
You only need to purchase 10 licenses, if all 10 clients
will be making calls at the same time.
Ok. Does this apply only for outbound calls using a voip provider and/or
applies to calls within the lan?
___
-- Bandwidth and Colocation
All calls.
Landy Landy wrote:
You only need to purchase 10 licenses, if all 10 clients
will be making calls at the same time.
Ok. Does this apply only for outbound calls using a voip provider and/or
applies to calls within the lan?
On Tuesday 01 December 2009 19:10:08 Alex Balashov wrote:
All calls.
Landy Landy wrote:
You only need to purchase 10 licenses, if all 10 clients
will be making calls at the same time.
Ok. Does this apply only for outbound calls using a voip provider and/or
applies to calls within the
Tilghman Lesher wrote:
On Tuesday 01 December 2009 19:10:08 Alex Balashov wrote:
All calls.
Landy Landy wrote:
You only need to purchase 10 licenses, if all 10 clients
will be making calls at the same time.
Ok. Does this apply only for outbound calls using a voip provider and/or
applies to
Hi all,
Refer to http://www.voip-info.org/wiki/view/Asterisk+config+features.conf.
It shows we can use variable BLINDTRANSFER to call back the one who
transfer the call. However, in my tests below. The result is not as
expected.
case 1:
A calls B (dial(sip/B||Tt)
B answers and connects to A
B
AsteriskNow use CentOS 5 and it comes preinstalled with dahdi and
asterisk with the freepbx GUI interface and it seems to be missing all
the dev packages
Martin
On 2009-11-17, at 02:19, Olivier wrote:
2009/11/17 Martin Roy m...@mac.com
I was previously using an old computer running
I was previously using an old computer running Asterisk 1.2 with
zaptel. Once the CPU fried I switch to a new computer and I chose
AsteriskNow 1.5 running in 64bits to simplify the installation
process. I manage to find my way with configuring dahdi instead of
zaptel and to switch all my
2009/11/17 Martin Roy m...@mac.com
I was previously using an old computer running Asterisk 1.2 with
zaptel. Once the CPU fried I switch to a new computer and I chose
AsteriskNow 1.5 running in 64bits to simplify the installation
process. I manage to find my way with configuring dahdi instead
OK,
Now I am responding to myself, because I have figured it out (finally).
It turns out it's a feature of asterisk (at least the older versions).
This is where I found my answer:
https://issues.asterisk.org/view.php?id=9678
So the solution for me was to simply rearrange my sip.conf so my
Ok I am replying to myself, because I still don't have this figured
out,, but I think I have more info.
On Nov 5, 2009, at 8:57 PM, Martin Joseph wrote:
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now, with
On Nov 6, 2009, at 5:14 AM, John A. Sullivan III wrote:
On Thu, 2009-11-05 at 20:57 -0800, Martin Joseph wrote:
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now, with pretty good results.
Recently I added a
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now, with pretty good results.
Recently I added a callerid service to my landline (qwest).
I am using the audiocodes MP114 2fxo/2fxs gateway, which is an
outstanding
Hi,
I have a queue and 3 agents in the queue like below
SIP/1001
SIP/1002
SIP/1003
When I dial the queue number, the agent start to ring. How can I get
the instance ringing agent as I want to pause the agent
(pausequeuemember) after the queue timeout? Any application or
variable can use to
: [asterisk-users] question on SIP and call manager
Here are two ways to address this
1. Dial(SIP/CCMMAINSIP/CCMSLAVE) - this tries both at once
2. exten = s,1,Dial(SIP/CCMMAIN,10,KkTt)
Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt)
CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds
Customer has 2 call manager systems and I am using asterisk to place
calls through the CCM.
One for the main use - CCMMAIN and another for disaster CCMSLAVE.
Can asterisk be setup in such a way that calls first try to use CCMMAIN
and if thats not available use CCMSLAVE.
Example if I place a
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, October 15, 2009 4:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question on SIP and call manager
Customer has 2 call manager
Here are two ways to address this
1. Dial(SIP/CCMMAINSIP/CCMSLAVE) - this tries both at once
2. exten = s,1,Dial(SIP/CCMMAIN,10,KkTt)
Exten = s,n,Dial(SIP/CCMSLAVE,10,KkTt)
CCMSLAVE only gets called if no one answers CCMMAIN in 10 seconds (2-3
rings)
Danny thats good to know for
Running asterisk 1.4.26.2
help pri
pri debug span Enables PRI debugging on a span
pri intense debug span Enables REALLY INTENSE PRI debugging
pri no debug span Disables PRI debugging on a span
pri set debug file Sends PRI debug output to the specified file
pri intense debug span Enables REALLY INTENSE PRI debugging
add span keyword
or use a tabulator that will do that for you
Martin
On Wed, Sep 30, 2009 at 10:08 AM, Jerry Geis ge...@pagestation.com wrote:
Running asterisk 1.4.26.2
help pri
pri debug span Enables PRI debugging on
Discussion
Subject: [asterisk-users] question on pri intense debug
Running asterisk 1.4.26.2
help pri
pri debug span Enables PRI debugging on a span
pri intense debug span Enables REALLY INTENSE PRI debugging
pri no debug span Disables PRI debugging on a span
pri set
Jerry Geis wrote:
Running asterisk 1.4.26.2
help pri
pri debug span Enables PRI debugging on a span
pri intense debug span Enables REALLY INTENSE PRI debugging
pri no debug span Disables PRI debugging on a span
pri set debug file Sends PRI debug output to
pri intense debug span span number
Just pointing out that was not clear from the HELP command.
I thought span was the span number
not span span number
Thanks for the direction.
Jerry
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On Wednesday 30 September 2009 11:54:11 Jerry Geis wrote:
pri intense debug span span number
Just pointing out that was not clear from the HELP command.
I thought span was the span number
not span span number
Thanks for the direction.
At the list level, we only provide the keywords. If
Discussion
Subject: [asterisk-users] question on pri intense debug
Running asterisk 1.4.26.2
help pri
pri debug span Enables PRI debugging on a span
pri intense debug span Enables REALLY INTENSE PRI debugging
pri no debug span Disables PRI debugging on a span
pri set debug
Hi
I am in the process of move a company from pstn to an asterisk setup.
They had 2 pstn lines - only really needed a max of 2 previously.
Now I have installed a tdm410 to handle the cross over from pabx to voip
handset. this has been done, the tdm is now just used to provide a
backup pstn
It's been my experience that when asterisk does a dns lookup, for externhost
or to do a SIP register, it blocks the whole server. Not sure if 1.6 has
that problem or just 1.4 though as my internet has been stable while im
awake these days
On Sun, Aug 30, 2009 at 5:54 PM, Alex Samad
On Sun, Aug 30, 2009 at 06:49:06PM -0700, Kyle Kienapfel wrote:
It's been my experience that when asterisk does a dns lookup, for externhost
or to do a SIP register, it blocks the whole server. Not sure if 1.6 has
that problem or just 1.4 though as my internet has been stable while im
awake
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