Hi,
I'm trying to configure some Quality Of Service among an Asterisk server
with RedHat3 and some IP phones on my LAN.
I read about 802.1p (level 2) QoS, using 3 bits of VLAN tag.
Two questions:
- do I need to use tagged links (trunks) end-to-end? In other words, do all
ports on all switches from
From The CLI with iax debug (IP address faked)
asterisk1*CLI>
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 0ms SCall: 22774 DCall: 0 [99.99.99.212:1720]
CALLING NUMBER : 1055
CALLING NAME: 1055
FORMAT : 256
CAPABILITY
The card is telling:
> CAPI INFO 0x34a2: No circuit / channel available
so the other channel must be in use by something else.
Maybe another device on the ISDN line?
Armin
On Sat, 28 Jan 2006, Ralf Mueller wrote:
> Hi,
>
> I like to forward an incoming call on an ISDN line to my mobile phone.
> I've been running 1.6.4.0064 for the last few weeks..
> I've had no problems with it, I haven't done a whole lot of speaker
> phone with it yet though.. Once my IP4000 reboots It'll be running it as
> well so that will be a good test.
Which bootrom version are you using?
-Ron
__
On Fri, January 27, 2006 17:23, Ian Cowley said:
> Iax.conf
>
> [general]
> ;bindport = 4569 ; Port to bind to (IAX is 4569)
> bindport = 5036 ; Port to bind to (IAX is 4569)
> bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
> disallow=all
> allow=g729 ; 4
To which context of the dial-plan does asterisk tries to
match incoming calls when acting as a sip client?
To be more specific:
In extensions.conf… Under which context should I place
“exten => 648064,1,Dial(TECH/peer)”
for an entry like this “register => 648064:[EMAIL PROTECTED]/648064”
I've been running 1.6.4.0064 for the last few weeks..
I've had no problems with it, I haven't done a whole lot of speaker
phone with it yet though.. Once my IP4000 reboots It'll be running it as
well so that will be a good test.
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EM
I have an analogue trunk to an AT&T Definity.
It has a DISA context defined.
From a Definity handset call the analogue port extension 1008 and wait
for dial tone from asterisk. It takes between 3&4 rings.
Likewise from Asterisk SIP handset takes
nearly 10 secs to ring.
Is this configurable?
Ian
Hi,
I like to forward an incoming call on an ISDN line to my mobile phone.
Since ISDN offers two channels, I thought that this should work, but Asterisk
tells me, that there is no channel available.
There is no one else using this line, so guess I made a mistake in the
configuration or it might
Iax.conf
[general]
;bindport = 4569 ; Port to bind to (IAX is 4569)
bindport = 5036 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=g729 ; 4 simultaneous allowed
allow ilbc ; prefered for iax2
allow=gsm ;
On 1/27/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> Sean Cook wrote:
>
> > Is there an implementation for shared line support in asterisk? I know
> > that with hint I can "watch" line status... I just want to be able to
> > pick up on an extension when ringing or jumping in on a call by punch
I have installed a Digium card TE210P and unicall for use MFC/R2. I think
that it´s all right but I can´t make and receive calls. I´m using asterisk 2.1
with the patch made by José P. Leitão and the follow libs:
libsupertone-0.0.2 libunicall-0.0.3 libmfcr2-0.0.3 zaptel
2.1
My number is 34
-Original Message-
From: Ian Cowley
Sent: 27 January 2006 15:10
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] External IAX2 phone defined as internal
behaving as from PSTN
Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg 10.
hello,
can someone help me with ser redirect to asterisk.
any help appreciated.
Thanks,
AA
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We use APIC on all servers, so interrupt sharing is not an issue :)
On Jan 26, 2006, at 3:02 PM, Damon Estep wrote:
And in some (many) cases it will do so while sharing an interrupt
with a
NIC and disk controller!
We run sangoma a104 cards in Dell SC1425 1U servers with great success
under h
Ram,
On my AAH the stock dial plan requires a 9
first. For kicks, try dialing 919197543700 and see what you get.
-MC
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ram
Sent: Friday, January 27, 2006
6:14 AM
To: Asterisk
Users Mailing List - Non-Commerci
Hi,
I've ordered a few IP501s from PC Connection, basically since we have an
account with them. I like the phones for what they do, and now would like
establish a relationship with a reseller that can give us maintenance and
access to the most current firmware.
What are some good resellers out th
On Fri, January 27, 2006 16:09, Ian Cowley said:
> Have [EMAIL PROTECTED] 1.2.1
> The server is on an internal network eg 10.10.10.10
> It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
> 50.50.50.50
>
> The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
> extensi
On Fri, January 27, 2006 15:13, ram said:
> Hi all
>
> I have installed AAH 2.2 in my P4 PC
>
> following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp
>
> and made as per the guide says
>
> and downloaded SJ Phone, and registered user
>
> and when i try to dial the 19197543700
>
>
Aha, I see it's 4.1, cool. So I just have to do a straight upgrade to 5.0
and I have this new toy to play with, correct?
-Original Message-
From: Sven Fischer (support) [mailto:[EMAIL PROTECTED]
Sent: Friday, January 27, 2006 5:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discuss
Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension 1055.
Outbound calls to 1055 work perfectly.
Inbound calls
Can anyone shed some light on "rules" that might make the task of
parsing the country code and city codes from a dialed number in the
CDRs?
I know that there is almost never a case where a concatenated country
and city code could overlap with another country code, but what about
city codes and loc
It does use the same kernel for everything. It's a specially modified
kernel for the VPS support. I guess the only way to see if ztdummy works in
the VPS is to try it.
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: <[EMAIL PROTEC
Bartosz Piec wrote:
Ronald Wiplinger wrote:
exten => 600,1,Dial(${PHONE_LOCAL},60,tr)
Type this:
exten => 600,1,Dial(${PHONE_LOCAL},60,tTwWr)
dial at 600 and see if this helps. If so, change all commands in that
way (tT is for transfer, wW is for recording).
You must also have sox install
there is no error message coming up on the pbx for in-bound calls (there is only debugging messages for outbound calls). thanks in advance for any hint or suggestion. Ama I just post my configuration file here for sip phone: extensions.conf-
Hi all
I have installed AAH 2.2 in my P4 PC
following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp
and made as per the guide says
and downloaded SJ Phone, and registered user
and when i try to dial the 19197543700
i get message that, all circuits are busy now, please t
On 1/27/06, Chris Mason (Lists) <[EMAIL PROTECTED]> wrote:
> Have any providers started to offer T.38 yet? I am anxious to find a
> solution for faxing.
>
commpartners does offer it. I haven't personally used it yet, but I
know they offer the service.
--
Bird's The Word Technologies, Inc.
http://
Hi Guys,
We are using Grandstream BT-102 phones internally to talk directly to
our SIP provider's SIP server. Each of our phones is configured with a
CLI provided by our SIP provider.
I have a couple of spare phones and about 5 spare CLI's, so I decided to
set up AsteriskAtHome to see what
On Friday 27 January 2006 13:29, Manuel Dominguez wrote:
> Hello all,
>
> I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal
> analogue lines. The same number is assigned to these lines. These lines are
> connected to 2 spa3k registered to my asterisk box.
>
> When calls arriv
Ronald Wiplinger wrote:
exten => 600,1,Dial(${PHONE_LOCAL},60,tr)
Type this:
exten => 600,1,Dial(${PHONE_LOCAL},60,tTwWr)
dial at 600 and see if this helps. If so, change all commands in that
way (tT is for transfer, wW is for recording).
You must also have sox installed for calls recordin
Hi I basically allow=all and NAT=no for all the phones. but still can't see why I can't receive calls (i.e. in-bound) but I can make outbound calls. also there is no debuging on pbx for sip (unless it's outbound call). do you have anymore advice? thanks AmaMd Sani Johari <[EMAIL PROTECTED]>
Whatever happened to Google? why don't people use that?
Tha actual limit according to Google/wiki is/was 255 for zap channels:
http://voip-info.org/tiki-index.php?page=Asterisk+dimensioning
However, in that same post someone corrected it that it is no longer limited.
On 1/27/06, Andrew Nowrot <[EM
Have any providers started to offer T.38 yet? I am anxious to find a
solution for faxing.
--
Chris Mason
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
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[EMAIL PROTECTED] wrote:
> Hi,
>
> We have a task to reduce voice call bandwidth. IP+UDP+RTP are using 40 bytes
> per
> packet and for voice GSM FR 33 bytes. We are trying to reduce this bandwidth
> accommodating multiple GSM frames in one packet. If we want to use per packet
> 10 GSM frames how
Henry Margies wrote:
> Hi all,
>
> When I do outgoing calls via my FXO card (TDM400, analog line), they get
> always marked "ANSWERED" in my CDR. I guess it is not that easy for fxo
> to determine if there is actually a call or just ringing.
>
> But anyway, is there a way to get this working righ
> > There is no doubt that given a particular scenario, anything won't work
> > properly. This is not necessarily a problem with the SPA3000 or the TDM
> > cards, this is much more of a phone line issue. Granted, those devices don't
> > handle line issues as well as some other devices (such as the
Ronald Wiplinger wrote:
does still not do the trick!
Show your Dial command from extensions.conf file.
--
Best regards,
Bartosz Piec
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Phil
I have very good experience
with the vegasteam ATA’s devices.(you might also want to look @ sipura
ATA’s, since vegastream is doing an oem on there boxes)
They support modem until
v.90 speeds and faxes for g3.
They are expensive, and
again, work great and configure very easy
[EMAIL PROTECTED] wrote:
Hi,
I'm currently in the process of building Asterisk for our new office and
have hit a snag. We need two internal Analog lines for a modem and fax
machine. Am I right in thinking I can use two ATA's, one on each piece
of equipment which will then talk to Asterisk
it means that your sender is capable of sending t38, but asterisk (without at
a minimum the t38-patches for passthrough) is not capable of handling this.
if you have reinvite for this channel allowed and your sender can send the
fax over g711 asterisk will send a reinvite and the fax has a chanc
> > >>Hi I'm looking for a pinout for the above. Note this has
> > >>what i'd call
> > >>RJ45 sockets (or someone smart can correct me). I need to
> > >>plug into BT (rj13?).
> > >
> > >
> > > Are you sure the TDM400 has RJ45 sockets? The pair I've got here have RJ12
> > > sockets.
> > >
> > > I
Hi.
Use massdeployment for putting the licenses on to your phones.
There is a setting called "license_url" you can use like the firmware update
URL, the macro {mac} will be replaced by the MAC address of the phone. So if
you provide the setting like this:
license_url: http://yourwebserver/{mac
Sean Cook wrote:
Ok... I am having a serious brain fart this evening. IIRC, the next sip
draft addresses shared lines and I thought I remembered something on the
list about support for it in the near future.
'the next sip draft'? There are probably 150+ IETF drafts circulating
regarding SIP
Could someone please outline the differences between:
allowed_not_screened: Presentation Allowed, Not Screened
allowed_passed_screen : Presentation Allowed, Passed Screen
allowed_failed_screen : Presentation Allowed, Failed Screen
allowed : Presentation Allowed, Network
What caller id method is used in spain? Is it before or after the ring.
If you can set the ISDN termination box for UK caller id then the ID is sent
before the first ring.
on the sipura thats ETSI FSK with PR(UK)
Chris
- Original Message -
From: "Manuel Dominguez" <[EMAIL PROTECTED]>
Hi asterisk and ser users,
Is there a solution to monitor asterisk and ser with
snmp ?
Regards
Harry
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exce
I didn't find that exact message in the RFC's, but I did find something
similar in RFC 3407 (http://www.rfc-archive.org/getrfc.php?rfc=3407),
a=cdsc: 4 image udptl t38
Which means that the sender is capable of sending T.38 fax over UDP.
I wouldn't worry about it unless you were trying to re
hello all,
i have a * 1.2.1, in a lab, only for test,
with 4fxo "clone - md3200 - intel537", connect to pstn.
All work well, but, 1 once day 2 of this cards,
stop make call, and receiv call thought.
i kill the asterisk, remove modules, wcfxo and zaptel,
mount the modules again, a
Hi Phil,
if you want to use ATAs take a look at grandstream site...they are
better than digium but you could use a card, TDM400 is excellent for
analog lines and devices.
Giorgio Incantalupo
[EMAIL PROTECTED] wrote:
Hi,
I'm currently in the process of building Asterisk for our new office
Since I passed from version 1.0 to the 1.2.3. I have Pb with the
callerid. If somebody call with presentation of the number all is well.
If somebody make call in masked number, i couldn't send a callerid to
the phone.
It is in a call center and i use the callerid to present the name of the
n
Hello all,
I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal
analogue lines. The same number is assigned to these lines. These lines are
connected to 2 spa3k registered to my asterisk box.
When calls arrive, TR1 try to pass call to the first spa. If spa not takes
the call
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jean-Michel Hiver
> Sent: Friday, January 27, 2006 10:49 AM
>
> If you're connecting Asterisk boxes between each other, it
> would make sense to use IAX as it's Asterisk's 'native' protocol.
> ...
Hi all,
When I do outgoing calls via my FXO card (TDM400, analog line), they get
always marked "ANSWERED" in my CDR. I guess it is not that easy for fxo
to determine if there is actually a call or just ringing.
But anyway, is there a way to get this working right?
Thanks in advance,
Henry
___
HiIn my environment I have to connect 6 * boxes with each other so IAX is probably the best solutionThanksCheers
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Hi,
I'm currently in the process of building
Asterisk for our new office and have hit a snag. We need two internal
Analog lines for a modem and fax machine. Am I right in thinking
I can use two ATA's, one on each piece of equipment which will then talk
to Asterisk and route via our ISDN30?
If
Hi,
I'm using asterisk 1.2.1.
Is there anybody out there who knows what this warning means?
*WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer:
image 5004 udptl t38*
Google does not help at all.
TIA
Giorgio Incantalupo
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> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
> Sent: 27 January 2006 08:21
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] paging agi
>
> Hi
>
> Some petty notes notes regarding the perl:
>
HiI had same problems yesterday but ts fine now,DanOn 27/01/06, Mark Adams <[EMAIL PROTECTED]
> wrote:
I would expect a reply in about 4-5 days …
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
Angelito Manansala
Sent: Thursday, January 26, 2006
8:44 PM
I need to connect two (or more) asterisk boxes. They will exchange a
lot calls. What is the best approach? Which protocol should I use IAX
or SIP or what? I never did that so first I want to ask people who
have some experience.
If you're connecting Asterisk boxes between each other, it would
Hi, It does sound like a typical case of urban legend, where "Zap is limited
to 256 channels" becomes "Asterisk is limited to 256 channels". Asterisk!= Zap.I've never said that Asterisk is limited to 256 channels. I only asked a question. That is the main reason of this list isn't it?
But leave the
Ronald Ramos wrote:
Hi,
Has anyone implemented astpp? I'm configuring one right now and I have
a problem on the pricelist.
I followed the steps here
http://www.astpp.org/index.php?n=ASTPP.Installation and created tables
using http://www.astpp.org/index.php?n=ASTPP.Structure, but i didn't
see
I've installed the last released asterisk 1.2.2 on my own HLFS system
with a 2.6.14.3 kernel. I've also a 2 FXO/ 1 FXS digium card on it.
Every thing is working correctly.
For ODBC, I'm using UnixODBC with pgsql. The voice messages are
correctly written to the database and also their number is co
This is not a problem of the ISDN line (or chan_capi), Asterisk is just
not doing anything after
-- Executing GotoIfTime("CAPI/ISDNL1/5912211-0","20:01-7:59|mon-sun|*|*?9")
in new stack
and without further commands (like Ringing(), Answer(), ...) the ISDN line
timed out and disconnects.
So e
Hi all,
I'm running Asterisk SVN-trunk-r8643M and face following problem:
I'm trying to get incoming call from a provider and calls ended with a
404 error. On the INVITE I get "Found no matching peer or user for address>:5060" and then "Looking for in
(domain xxx.xxx.xxx.xxx)". My question i
Hello,
I'm trying to catch channel hangup in DeadAgi script. For example, A
calls to DeadAgi script which connects (Dial) to B. After Dial command
exits I need to identify where hangup came from: A or B. CHANNEL
STATUS returns 6 (Line is Up) regardless of who hungup.
In CLI "show channels" states
Andrew Nowrot a écrit :
Hi,
Where are you pulling this number from? (other than the obvious
"traditional" 2^8)?
That is not my imagination ;).
Actually I talked with a guy who was one of the designers of Asterisk.
He told me about this limitation but I don't know if he was talking
Andrew Nowrot a écrit :
Hi,
Yeah, I think it was all about thew zap channels
But what opportunities I have when I need to connect two or more
Asterisk boxes. IAX, SIP or what?
What is most efficient.
Your question doesn't make any sense.
Tell us what you are trying to do and you might get
Hi,
We have a task to reduce voice call bandwidth. IP+UDP+RTP are using 40 bytes per
packet and for voice GSM FR 33 bytes. We are trying to reduce this bandwidth
accommodating multiple GSM frames in one packet. If we want to use per packet
10 GSM frames how to do this using asterisk? Assume the si
Hi, Where are you pulling this number from? (other than the obvious "traditional" 2^8)?
That is not my imagination ;).Actually I talked with a guy who was one of the designers of Asterisk. He told me about this limitation but I don't know if he was talking about Zap channels only or in general. I w
Hello Sam,
use host=IP_ADDRESS when defining user in sip.conf
regards,
Umair Bari
On 1/26/06, Sam Tam <[EMAIL PROTECTED]> wrote:
Do anyone know how to setup asterisk to authenticate the user through IPrather than username and password?
I know most carriers will do that but smaller end user pr
I would like to add that I did have at one point
problems figuring out 4.0 and there were no problems
downgrading. Also I made a special email account
@mydomain for SNOM liscence's. This helps if at a
later dat you need to re-enter it again.
Regards,
Dovid
--- Christian Stredicke <[EMAIL PROTECTED
Hello list,
I've got a problem provisioning my snom 360's in the office (about 20 of them).
I'm using DHCP option 66/67 to set the provisioning URL but the phone
won't connect to it to retrieve it's configuration.
We are using a Cisco Catalyst Epress 500 to power the phones (poe), however if
i p
Hi,Yeah, I think it was all about thew zap channelsBut what opportunities I have when I need to connect two or more Asterisk boxes. IAX, SIP or what?What is most efficient.CheersAndrew
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Hi,
Has anyone implemented astpp? I'm configuring one right now and I have a
problem on the pricelist.
I followed the steps here
http://www.astpp.org/index.php?n=ASTPP.Installation and created tables
using http://www.astpp.org/index.php?n=ASTPP.Structure, but i didn't see
there a query on crea
/etc/init.d/asterisk stop
Stopping Asterisk PBX: .
censys:/usr/src/asterisk-8632# cd ..
censys:/usr/src# asterisk -vc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk SVN-trunk-r8620, Copyright (C) 1999 - 2006 Digium, Inc. and
ot
Hi
Some petty notes notes regarding the perl:
On Thu, Jan 26, 2006 at 11:23:27PM -0800, Jeremy wrote:
1. you didn't use strict and -w. Debugging will be a whole lot tougher
2. Consider using the nagging -T (taint mode), to explicitly know when
you trust the input.
3. Consider the latency this
Andrew Nowrot a écrit :
Hi,
Does anyone know what is the amount of max concurrent calls that can
be made in one Asterisk box?
I heard that it is 256 and it doesn't depend on how good your machine
is. It is the program constraint.
I wasn't aware of such limit and I seriously doubt it. Where
> Hi! For reasons that I won't bore people with, I'd like to disable
> echo cancellation on-the-fly, depending on which DID a call came in
> on. I've seen things like spandsp disable EC for faxes, so I know
> it's possible. Any idea where to start looking? (I assume I'll have
> to make a helper
Do anyone know how to setup asterisk to authenticate the user through IP
rather than username and password?
I know most carriers will do that but smaller end user providers will not
do.
Sam
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There is no such thing as a hard limit in asterisk. (Except for zap
channels, those are limited to 256 iirc).
With iax you can go higher, but the limit might be lower than 256 if you
are doing a lot of transcoding.
The limit depends on what exactly the server has to do with your call,
and ho
On Thu, Jan 26, 2006 at 03:10:09PM -0600, Mike Hammett wrote:
> I'm running a VPS and I need to pass the device drivers from the
> host OS to the VPS. What files do I need to pass through for
> ztdummy to work? I'm assuming they're in /dev/zap, but I'm not
> sure which ones are needed.
ztdumm
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