Thank you! I updated the tutorial as well.
l.
2009/5/25 Atis Lezdins a...@iq-labs.net
On Mon, May 25, 2009 at 7:42 PM, Lenz Emilitri lenz.lo...@gmail.com
wrote:
Hi everyone,
after doing the same thing multiple times and struggling to remember how
it
was done, I have prepared a small
Hi all,
I download asterisk-addon 1.6.1 but the VoIP phone failed to
register to the system with the message below.
[May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317
realtime_mysql: MySQL RealTime: Invalid database specified: asterisk
[May 26 15:45:11] WARNING[29665]:
Is there a method to fetch the ${EXTEN} of the channel that has been
hung up when exten h is started?
The nearest thing I can think of is to set another variable to the
extension and pick that up. Would that be a reliable method though?
___
--
Hi All,
We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge
calls, as follows:
ISDN Provider --- Span 1(pri_cpe) --- Span 2(pri_net) Phone
System
The company that looks after our internal phone system can no longer dial in
using their data modem in order to maintain
Hi
I am a premium voice service provider giving some services on IVR to a Telco X
. As my premises is some 10 kms away from that telco , i have taken a PRI
connection (30 DID with 1 hunting/pilot number) from telco Y When a customer
of Telco X dials my short code @Rs.6/- per minute his call
Sure - you just need to figure out what number is being dialed, make
sure that number rings on the incoming PRI, and make sure the phone
system expects that call to come in the standard PRI trunk group and
not some dedicated analog craft port.
--
Sent from mobile device
On May 26, 2009, at
On 26 May 2009, at 11:48, Sriram wrote:
Now the problem arises during billing , many customers of Telco X /
Telco Z / Telco Y somehow get to know the pilot number of telco Y
and they directly dial in.
How exactly? You might have it accidently listed somewhere. Worth just
looking on
On 5/26/2009 10:57, Thomas Kenyon wrote:
Is there a method to fetch the ${EXTEN} of the channel that has been
hung up when exten h is started?
The nearest thing I can think of is to set another variable to the
extension and pick that up. Would that be a reliable method though?
Which is
Hi, I'm in the process for setting up an asterisk server for four
organisations sharing a SIP trunk. In order to split the costs according
to usage, it would be nice to log all incoming, outgoing and missed
calls.
Is there a simple way of doing this, preferrably in a database? Perhaps
someone has
Hi,
I'm trying to understand an issue I'm seeing between two Asterisk
servers. I think it has to do with Domain definitions.
Server A), has extension 5550 defined. Has a sip client 2000 defined,
and has guest-invites enabled.
Server B), Dials to server A for any 5550 dialled. Has sip
Hi All;
I discover that most of the voice cutting complain are coming from the Internet
bandwidth when we are connecting two remote offices togethor via Asterisk or
any other IP PBX.
Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So
we can resolve the problem of
As part of a project to move a clients Cisco phones to SIP to work with an
Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk
setup. Now, I've gotten the firmware files from the site, the latest version is
8.4 which contains the following files:
apps41.8-4-3-16.sbn
A lot of the ADSL CPE (customer premise equipment) deployed has basic
QoS capabilities in a pre-set kind of way, but if you want to do your
own DiffServ tagging the standard practice is to do Layer 2 Ethernet
bridging to a more intelligent box behind the ADSL CPE.
bilal ghayyad wrote:
Hi
m0n0wall and pfsense both do traffic shaping, which forcibly allocates
bandwidth for your VoIP traffic.
Michael
On Tue, 26 May 2009 04:32:59 -0700 (PDT), bilal ghayyad wrote:
Hi All;
I discover that most of the voice cutting complain are coming from the
Internet bandwidth when we are
On Tue, 26 May 2009, bilal ghayyad wrote:
Hi All;
I discover that most of the voice cutting complain are coming from the
Internet bandwidth when we are connecting two remote offices togethor
via Asterisk or any other IP PBX.
Anyone has an idea on a ADSL router that work as ADSL +
As does ZeroShell (www.zeroshell.net/eng).
Bruce Komito
WPTI Telecom
(775) 236-5815
On Tue, 26 May 2009, Michael Graves wrote:
m0n0wall and pfsense both do traffic shaping, which forcibly allocates
bandwidth for your VoIP traffic.
Michael
On Tue, 26 May 2009 04:32:59 -0700 (PDT), bilal
Hello,
Is it possible to hangup an active call by simply sending a DTMF code to
Asterisk for example # code.
If yes, What function to use in the dialplan.
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
I set a variable CalledID to ${EXTEN} before dial it. So in h extension I
can use ${CalledID}.
2009/5/26 Thomas Kenyon dig...@sanguinarius.co.uk
On 5/26/2009 10:57, Thomas Kenyon wrote:
Is there a method to fetch the ${EXTEN} of the channel that has been
hung up when exten h is started?
Sorry. I don't get many opportunities to test this system as its live. Here
are the results:
-- Executing [...@dlpn_dialplan1:1] Dial(SIP/19722-b650fb80, DAHDI/1)
in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called 1
-- Channel 0/1, span 1 got hangup, cause 90
If you do Dial(tech/line,,Hh), either side can hang up the call with *.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of abdelkader
Sent: Tuesday, May 26, 2009 7:46 AM
To: asterisk-users@lists.digium.com
Subject:
Darrin,
The files you are using are consistent with SIP for Cisco Call Manager.
Anything other than Callmanager will essentially be a hack. I am not sure how
proprietary the Avaya system is in regards to registration and open-SIP
support. Asterisk and any iteration of it will support it, but
Based on this link -
http://www.trixbox.org/forums/vendor-forums-certified/sangoma/a101dx-hangup-
cause-code-90-outbound-calls
I'd check my polarity settings in dahdi.conf. Maybe signaling?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Install nv_faxdetect. This will make asterisk not attempt to process the
modem call for a specified period of time.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Morgan
Sent: Tuesday, May 26, 2009 5:06
On Saturday 23 May 2009 11:03:13 sean darcy wrote:
I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 over this weekend.
I'm getting:
[May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit:
Serious Network Trouble; __sip_xmit returns error for pkt data
[May 23 10:56:33]
My thoughts exactly. I've tried National2, 4ess and now ni1
ni1 just worked on Asterisk 1.4.22. (failover box I downgraded). So I'm
swapping back to 1.4.24 to test that now.
On 26/5/09 3:34 PM, Danny Nicholas da...@debsinc.com wrote:
Based on this link -
Now that I've slogged through everyone else's reply and got to the original
post, here's an idea. You seem to have the dialplan part worked out; why
not do a simple HTML interface to do the Berkley maint using asterisk -rx to
do the CLI reads/pokes? With asterisk -rx you can automate 90+ percent
It is my experience that /e/i/dahdi doesn't always work correctly (opensuse
11.0). For whatever reason, it doesn't do the required modprobe to get the
dadhi module activated.
Try doing modprobe wctdm
Then
Dahdi_cfg -vv
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Ok I've solved the problem. I do not think it was as switchtype issue after
all as it is now working with a national2 configuration.
I need to sort out some of the changes and I'll post back for reference.
However it appears to be some form of parsing order issue between all the
locations that
Ok, ignore what I said below. I've got it working now, thanks a million for
this link:
http://www.greenwireit.com/blog/2009/04/reflash-your-cisco-7940-7941-7960-or-7961-phone-to-sip/.
However, now I'm wondering about the dialplan.xml, can it handle regular
expressions like 9[2-9]..?
Hello,
I have 8 DID: 7 from a provider1 and 1 from provider2.
Each time a customer calls one of the DID, the system plays a message.
The problem is that the message is played normally for all the DIDs from the
provider1 and is not played (not heard) for the DID from provider2.
My question is:
Hello.
I am looking for details of the maximum allowed/usable/effective wire/cable
length of the connection from a FXS port of Digium analog cards to the
analog telephone handset.
To clarify my intention, I need to have an analog telephone connection to my
asterisk box that is 3000 meters
Thanks for taking the time to answer.
I've played with the server a lot in the past few days, and I am not sure
what did it, but for futur reference this is my best guess: I think I had
32-bit code or RPMs installed on a 64-bit machine (specifically: HP-hardware
specific RPMs for hardware
On 5/26/2009 14:08, Marco Sambo wrote:
I set a variable CalledID to ${EXTEN} before dial it. So in h extension
I can use ${CalledID}.
Thanks for the response.
In that case if there is an intervening call that is shorter, then the
$calledID will be wrong.
I found a better approach than using
Ahh I see.
In response to your other question about the auto-provisioning of Cisco phones,
I wrote some scripts that work against an active directory and setup the phones
automagically. I'll send the link your way if you'd like.
-Dave
-Original Message-
From:
The best a native cat5 can run is 100 meters. Unless you like paying your
telco huge bucks, you should go for some kind of SIP connection to your box.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
asterisk-us...@rogg.is
Did not mean to infer they don't perform wonderfully with Asterisk. By hack
I meant that Cisco does not offer any official support for them on Asterisk.
Cory J. Andrews
Director New Market Initiatives
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
asterisk-us...@rogg.is wrote:
Hello.
I am looking for details of the maximum allowed/usable/effective
wire/cable length of the connection from a FXS port of Digium analog
cards to the analog telephone handset.
To clarify my intention, I need to have an analog telephone connection
I could be wrong but I don't think the cat5 limit of 100 meters applies to any
analog signaling over that copper. I believe it only applies to Ethernet
signaling.
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
On Mon, May 25, 2009 at 10:27:22AM -0400, Mike wrote:
I did run make install, probably 3-4 times before I ended up asking that
question in the mailing list.
Here is the required output: to the first one, could not find module
dahdi.
To the second, it found dahdi in
On Tuesday 26 May 2009 02:52:18 Rilawich Ango wrote:
Hi all,
I download asterisk-addon 1.6.1 but the VoIP phone failed to
register to the system with the message below.
[May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317
realtime_mysql: MySQL RealTime: Invalid database specified:
There is indeed... well i was about to say there was, but it turns out the
one i've got is an fxo adapter, have a look and see if sangoma have any fxs
adapters in the series, it seems to be called the usbfxo u100
2009/5/26 Diogo Saad diogos...@gmail.com
What is the easiest way to connect my
what I want to do is to answers to mobile calls using a regular phone.
Is a usb fxs all I need? Does this u100 have smooth integration with
Asterisk ?
On Tue, May 26, 2009 at 11:55 AM, Geraint Lee gera...@gmail.com wrote:
There is indeed... well i was about to say there was, but it turns out
On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote:
I run my analog telco over cat5, but that's in-house and definitely not 3km.
That sounds really far for current loop stuff.
I was doing that too. I asked this same question a few years ago and
the answer was 100-200
On Tue, 26 May 2009, Steve Howes wrote:
On 26 May 2009, at 16:39, Jeff LaCoursiere wrote:
YMMV
I think thats the problem :D sorry couldn't resist..
I did kind of mean that tounge-in-cheek :):)
j
___
-- Bandwidth and Colocation Provided by
On 26 May 2009, at 16:39, Jeff LaCoursiere wrote:
YMMV
I think thats the problem :D sorry couldn't resist..
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Tue, 26 May 2009, randulo wrote:
On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote:
I run my analog telco over cat5, but that's in-house and definitely not 3km.
That sounds really far for current loop stuff.
I was doing that too. I asked this same question a few
That is a pretty long run.
The type of analog phone can be an issue. How LITTLE loop current will
it operate on? Most need more than 20 Ma to signal properly, and the
voltage output of the ATA needs to be known
Type of signaling? DTMF? pulse?
Interconnection cable wire size and capacitance
sean darcy wrote:
Maybe I've not explained this correctly. I know, or can look up, the 40+
local exchanges that are local. I can parse the dial EXTEN to determine
the exchange. I can check the exchange against a DB. I want to determine
which exchanges are local. I do not want to store an
Cory,
Precisely what do you mean by 'Anything other than Callmanager will essentially
be a hack'?
I've got nearly 100 Cisco 79x1s in production using Asterisk against the SIP
image. They're not 'hacked', they're set up properly against the Cisco provided
SIP image and are rock-solid stable. I
On Tue, May 26, 2009 at 05:39:46PM +0200, randulo wrote:
On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote:
I run my analog telco over cat5, but that's in-house and definitely not
3km. That sounds really far for current loop stuff.
I was doing that too. I asked this
On Tue, 26 May 2009, Diogo Saad wrote:
What is the easiest way to connect my black phone to a PC running
asterisk?
I don't need multiple extensions, I've got just 1 phone. Is there any
USB FXS adapter?
An Ethernet based ATA would be more versatile. I like Digium's
discontinued IAXy. Dead
Sigh, lets repeal Ohm's law.
;-)
In practice the controlling rules are:
Murphy's Law: If anything can go wrong it will.
O'Toole's corollary to Murphy's law: And, it will produce the worst
possible results.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
What is the easiest way to connect my black phone to a PC running
asterisk?
I don't need multiple extensions, I've got just 1 phone. Is there any USB
FXS adapter?
Thanks
--
Diogo Saad
___
-- Bandwidth and Colocation Provided by
Hi,
In my sip.conf, I've got :
[general](+)
;
register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129
register=trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129
When I'm using the TCP line instead of the other, I've got :
randulo escribió:
On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote:
I run my analog telco over cat5, but that's in-house and definitely not 3km.
That sounds really far for current loop stuff.
I was doing that too. I asked this same question a few years ago and
On Tue, May 26, 2009 at 10:09 AM, asterisk-us...@rogg.is wrote:
I am looking for details of the maximum allowed/usable/effective
wire/cable length of the connection from a FXS port of Digium analog cards
to the analog telephone handset.
To clarify my intention, I need to have an analog
Also be wary of the loop you get.
Depending on the Telco you are dealing with, and the type of loop you get,
Alarm circuit, etc. they may , and have the right to, put in a low pass
circuit to limit bandwidth to 15 Hz. That keeps people from using cheap
alarm circuits for voice. It is not
Hi,
I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer.
My extensions.conf:
[globals]
__TRANSFER_CONTEXT = transfer
[common]
exten = 123,1,Playback(demo-congrats)
exten = 123,n,Hangup()
exten = _0X.,1,Dial(SIP/${ext...@pstn-gw,60)
exten = _0X.,n,Hangup()
exten =
How about a low cost ATA? Just plug the ATA into the network, configure it
- along with a SIP definition within sip.conf and you're ready to go.
Lee
From: Diogo Saad [mailto:diogos...@gmail.com]
Sent: Tuesday, May 26, 2009 10:41 AM
To: asterisk-users@lists.digium.com
Subject:
Using an ATA, Do I still need a softphone or it´s embedded in the hardware?
On Tue, May 26, 2009 at 12:09 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Tue, 26 May 2009, Diogo Saad wrote:
What is the easiest way to connect my black phone to a PC running
asterisk?
I don't need
Diogo Saad wrote:
Using an ATA, Do I still need a softphone or it´s embedded in the
hardware?
plain old walmart phone plugs in the ata (with or without callerid,
adsi, cordless, etc)
On Tue, May 26, 2009 at 12:09 PM, Steve Edwards asterisk.org
http://asterisk.org@sedwards.com
You are correct.
Telcos normally supply dial tone to business and residence for miles if
there is no DSL
Loading coils are used to offset the capacitance of cables, and precise
spacing of these is required, and are engineered for different types of
cable.
John Novack
David Gibbons wrote:
I
Please do!
Cory J. Andrews
Director New Market Initiatives
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
candr...@sayersmedia.com
Have I exceeded your expectations? Please share your experience with my boss,
Miguel Molina wrote:
randulo escribió:
On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote:
I run my analog telco over cat5, but that's in-house and definitely not
3km. That sounds really far for current loop stuff.
I was doing that too. I asked this same
On Tue, 26 May 2009, abdelkader wrote:
I have 8 DID: 7 from a provider1 and 1 from provider2.
Each time a customer calls one of the DID, the system plays a message.
The problem is that the message is played normally for all the DIDs from the
provider1 and is not played (not heard) for the
On Tue, 26 May 2009, Danny Nicholas wrote:
I run my analog telco over cat5, but that's in-house and definitely not 3Km.
Of course - and that is just fine. If you were running ethernet
signalling over that CAT5 than your 100m limit would apply. If you were
running gigabit over that same
Assuming you mean the firewall in front of the client, you don't need to open
any ports as long as the VPN client is tunneling all traffic to and from the
Asterisk server.
I set NAT=yes in the config file for the extensions behind a VPN.
-Dave
From: asterisk-users-boun...@lists.digium.com
Excellent analysis of the real world. Start with this, and work out the
issues, or go to VOIP.
Cary Fitch
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wilton Helm
Sent: Tuesday, May 26, 2009 11:33 AM
To: 'Asterisk
how do I configure my SIP account information? I mean, sip proxy and etc.
On Tue, May 26, 2009 at 1:19 PM, Jon Pounder j...@inline.net wrote:
Diogo Saad wrote:
Using an ATA, Do I still need a softphone or it´s embedded in the
hardware?
plain old walmart phone plugs in the ata (with or
There are a lot of factors that impact this. First, CAT 5, while usable is
overkill. Cat 3 (otherwise known as I/O wire) works equally well for voice
grade lines. That being said, for that long a run, a heavier gauge wire
would be better. I believe telcos use 18 - 22 guage (Cat 5 and Cat 3
Hello All,
We are setting up call center of 10 agents and expecting its growth till 30
agents. Mainly calling is within UK. Please suggest some good service for UK
dialing with London DID.
Regards,
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345
Wilton Helm wrote:
one thing I missed mentioning about fxs devices - the linksys/sipura
ones actually allow you to set line characteristics on the slic inside
it. you can vary from the 600ohm default, and tweak gains a bit. Some
mix of a capacitive line or different resistance may help. never
You are exactly right. Cat 5 had no advantage over cheaper wire for voice,
and the length limitations are meaningless. Consider that Cat 5 is
typically use with signals that extent to 30 MHz or beyond. A voice grade
analog circuit must go to 4 KHz (1/10,000 as much). At 4 KHz, the wire
Diogo Saad wrote:
how do I configure my SIP account information? I mean, sip proxy and etc.
you need just a couple pieces of information
server (put this in any setting that says proxy or host etc, all set the
same)
account (the extension in asterisk, put anywhere that sounds like a
Wilton Helm wrote:
You are exactly right. Cat 5 had no advantage over cheaper wire for
voice, and the length limitations are meaningless. Consider that Cat 5
is typically use with signals that extent to 30 MHz or beyond. A voice
grade analog circuit must go to 4 KHz (1/10,000 as much). At
I run my analog telco over cat5, but that's in-house and definitely not 3Km.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Tuesday, May 26, 2009 10:28 AM
To: Asterisk Users Mailing List
On Tue, May 26, 2009 at 12:44:26PM -0400, Jon Pounder wrote:
Wilton Helm wrote:
one thing I missed mentioning about fxs devices - the linksys/sipura
ones actually allow you to set line characteristics on the slic inside
it. you can vary from the 600ohm default, and tweak gains a bit. Some
Tzafrir Cohen wrote:
On Tue, May 26, 2009 at 05:39:46PM +0200, randulo wrote:
On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas da...@debsinc.com wrote:
I run my analog telco over cat5, but that's in-house and definitely not
3km. That sounds really far for current loop stuff.
Thanks for all.
But what all gave me was a software need to be installed on PC, but I am
looking for a router (ADSL router) that can does this, because usually the ADSL
router is the default gateway where all the traffic goes out and in.
Any ADSL router device can do this?
About Draytek, as
For long distances, a wireless point-to-point might be more economical
than trenching.
e.g: Carlson Wideband CDMA Spread Spectrum Phone Line Extender
http://www.oksolar.com/communications/phone_line_ext.htm
___
-- Bandwidth and Colocation Provided by
Hi all,
I have a question. I have a VPN and I want to use a SIP softphone on my
notebook using with asterisk. But I have some problem with firewall and
port.
Someone knows which ports I should open on my firewall??? I can't connect
...
Thanks all.
Marco
On Tue, 26 May 2009, Danny Nicholas wrote:
The best a native cat5 can run is 100 meters. Unless you like paying your
telco huge bucks, you should go for some kind of SIP connection to your box.
He was asking about an analog telco connection - not an ethernet drop.
j
_
From:
A lot of ISP adsl modems aren't capable. But most should be. Just log in to it
give it a private IP for your lan(192.X.X.X or whatever your using) then have
all your computers use that local IP as their gateway address.
If you have an ADSL modem which doesn't then simple get a router (hell a
Hi Folks,
I'm trying to use my mobile as a trunk via bluetooth - calls done in a
softphone go thru GSM network and calls destinated to my mobile are answered
at the softphone.
I have asterisk configured to do so but I'm facing an issue - Audio is
audible but it’s not intelligible. I feel like
I have been trying out several stun servers with Asterisk 1.6.0.9 and
1.6.1.0 and I keep getting the following message:
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
[May
Jon Morgan wrote:
Hi All,
We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge
calls, as follows:
ISDN Provider --- Span 1(pri_cpe) --- Span 2(pri_net) Phone
System
The company that looks after our internal phone system can no longer dial in
using their data
I've had good luck using a sangoma S518 ADSL card in a linux box. the
logging capabilities are supurb (cought my provider not providing what they
said they were and great for troubleshooting as it logs line speed and
dropouts to the second). support is also top notch. once installed it
looks to
While I was in the console looking for something else, this appeared when I
called in on my cell.
[May 26 12:17:26] NOTICE[3364]: chan_sip.c:17229 handle_request_invite: Sending
fake auth rejection for user xxx xxx xx
sip:xxx...@xxx.xxx.xx.xxx;tag=as04e93fb9
What does this mean?
Hi,
these are my first steps with DAHDI and I finally managed to get
asterisk to load chan_dahdi (after I found out, that I need libpri).
But how do I tell chan_dahdi on which isdn numbers it should react? I
haven't found a parameter like incomingmsn for chan_capi in the
documentation.
On Wed, May 13, 2009 at 11:53 AM, Barry L. Kline blkl...@attglobal.net wrote:
If I insert a Monitor() prior to dialing the outbound call, I get no
audio in the recording and the caller hears no audio. Occasionally it
works (perhaps 1 out of 5 times) but most of the time the caller can't
hear
You define context= for the channels in dahdi.conf
and then in extensions.conf you define those numbers in that
particular context name
eg:
dahdi.conf
context=incoming
channel = 1-15,17-31
extensions.conf
[incoming]
exten = _X.,1,Answer
exten = _X.,2,Echo
and it will react to all numbers
[May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor:
stun failed
No matter which STUN server I point to I get those messages. Am I
missing some other setting?
Hey Carlos,
That just means the stun request failed, there are several reasons for
that, I won't even try
Hi,
Digging on this case :
2009/5/26 Olivier oza-4...@myamail.com
Hi,
In my sip.conf, I've got :
[general](+)
;
register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129
Sorry if this is a repost - I never saw a copy of this go out last week.
Can anyone tell me if there is a way to vary the output levels (dB) of the
tones generated in indications.conf? I generate a few custom tones and
sometimes people tell me they are a little too loud.
Thanks
Lee
On Tue, 2009-05-26 at 10:26 -0400, John Novack wrote:
That is a pretty long run.
The type of analog phone can be an issue. How LITTLE loop current will
it operate on? Most need more than 20 Ma to signal properly, and the
voltage output of the ATA needs to be known
Type of signaling? DTMF?
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David Backeberg wrote:
5) exten = s,n,Dial(${OUTGOING_PRI}/${ANSWERINGSVC},15,r);
6) exten = s,n,Goto(s-${DIALSTATUS},1);
What is the 6 for?
What is the goto supposed to do?
Hi David.
The '6' is in case I get a CHANUNAVAIL or other error back
Hi,
In this thread
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/223399/focus=223401,
one conclusion was that an easy way to set 2 different trunks with
different binding ports was to use TCP and UDP transport.
serverA udp:5060- serverB
|
Customer has a Verizon Business SIP trunk, I'm still used to PRI T1
myself for local service. The fax machines are having some issues (I
can use analog phone to call out fine) and I'm checking on modem
passthrough with Verizon, but wonder if any else is using Verizon
Business for SIP trunk and
Thomas Kenyon dig...@sanguinarius.co.uk writes:
In that case if there is an intervening call that is shorter, then the
$calledID will be wrong.
That isn't how Asterisk variables work. They aren't global to all calls,
they are local to the call you happen to be in. So no, an intervening
call
Modem and/or analog passthrough over SIP trunk, not on a LAN? I
wouldn't bother - it doesn't even work very well on an uncontended LAN
due to excessive jitter, let alone over the Internet or semi-private
Layer 2 cloud product.
T.38 or bust. The other's fax mileage is measured in gallons per
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