Hello asterisk-users,
My SIP phones are in pickupgroup, and if some of them ringing from
other phone can pick up with *8 as usual. But I want to know if this
happen. I've tried the a extension, but seems not working.
Any other idea?
--
Best regards,
Gergo
Hi,
One of the solutions would be to overwrite standard *8 behaviour with
your custom macro that will 1) pickup a call as usual b) send
notification via AMI or whatever else you want. This can be done with
[applicationmap] in features.conf - see
I agree with the training course, it takes extensive resources.
But people that have been on in the ground floor should get a dCAP. I
specifically said I was thread jacking, so possibly frowned upon, it
is still on-topic.
Finally, last I knew, you could go stand-by for the dCAP exam and not
Hi,
I am trying to establish a call between two users (A and B) but because
I use Asterisk only to provide services, the request has to pass by the
same Asterisk twice.
Here what I am expecting to do :
User A Equipment1 Asterisk1
Equipment1
Hi Robb,
Have a look in your features.conf file and see what keys you have
enabled for transfers.
Regards,
Greyman.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register
Hi,
I try to get anonymous calling working on ZAP. But I am unsuccessful on
PRI as well as on BRI.
I tried all parameters from the application SetCallerPres(). Nothing
worked.
I even traced with my ISP and they told me that I am not sending any
parameter to hide the callerid.
I found on the
Hi,
Whatever the verbosity level (even 0), my Asterisk console is full of
Really destroying SIP dialog messages.
Is there a way to get rid of those ?
If not, do you think it deserves to marked as a bug ?
Regards
___
-- Bandwidth and Colocation Provided
Hello,
When managing a stable system, which verbosity level do you adopt ?
Leaving a higher level helps to catch root cause, if for any reason, things
go wrong.
Leaving a lower level saves resources if you need (have) to backup logs.
What are current best practices ?
Do you change verbosity
On Wednesday 17 September 2008 09:18:58 pm hugolivude wrote:
I think it's better to find out what is listening on port 4520.
CentOS 5
Asterisk 1.4.20
Presumably my other Asterisk server is listening on 4520.
The problem here is that I can change the port, and it will work...
until I
I managed to achieve that on a PRI line with the following:
1. On zapata.conf, for the PRI line channels, add
facilityenable=yes
usecallerid=yes
usecallingpres=yes
I do not known if these are all strictly required for anonymous calling,
but it works for me.
2. On your extensions.conf, just
And do You have usecallingpres=yes in your zapata.conf ?
Hi,I try to get anonymous calling working on ZAP. But I am unsuccessful onPRI
as well as on BRI.
I tried all parameters from the application SetCallerPres(). Nothingworked.
I even traced with my ISP and they told me that I am not
On Thu, Sep 18, 2008 at 05:31:08AM -0500, Anthony Messina wrote:
On Wednesday 17 September 2008 09:18:58 pm hugolivude wrote:
I think it's better to find out what is listening on port 4520.
CentOS 5
Asterisk 1.4.20
Presumably my other Asterisk server is listening on 4520.
The
Yes,
thats why I do not get it. Also on BRI I know that it worked on my
customers old PBX so I really exclude the carrier.
Loic
On Thu, 2008-09-18 at 12:38 +0200, Igor Zamocky wrote:
And do You have usecallingpres=yes in your zapata.conf ?
Hi,I try to get anonymous calling working on
Hi all,
I want to configure my asterisk for sending and receiving faxes. I see in my
sip.conf that i have to enable the t.38 capability. I have done that but the
rxfax and txfax applications are not installed. They are not listed in
applications when i do make menuselect. i have searched in
Dear All,
Pl. any one can give me help. B'coz I have to implicitly work
for Outgoing call from PSTN Agent. I have also may to call out side the
office from exten = s,n,Dial(Zap/4/111,60) on testing purpose. But,
how to dial number via PSTN agent's phone like zero or nine dialing. I
Its a good question
I have lots of disk space so leave it high, I would rather have the
detail if I need it
It probably would seem sensible to revisit stable systems after a year
and lower the verbosity, but then since I can afford the space I am not
too fussed.
Cheers Duncan
Olivier wrote:
Hi Guys, we need an urgent help with Pre-paid Billing.
We are using Asterisk at work with our own prepaid billing system. We
calculate max number of minutes user is allowed to talk based on his
balance and destination. We then used Dial command with S(x) parameter
to create a call.
However, this
On Thursday 18 September 2008 13:34:06 Rizwan Hisham wrote:
How can i install these applications. Are there anyother components
required to make my asterisk a fax-passthru system.
http://sourceforge.net/projects/agx-ast-addons
t.
--
knowledgeTools® ... managing complexity.
On Sat, 2008-09-13 at 01:13 +0100, robb wrote:
I'm trying to get a simens IP pjone working so I can transfer calls
using the recall key when I run sip debug I get the below text on
screen, but I don't get dialtone returned, any advice would be greatly
appriciated
I don't claim to know
On Wed, 17 Sep 2008, Jared Smith wrote:
On Wed, 2008-09-17 at 19:58 +0200, Remco Barendse wrote:
Why doesn't Asterisk allow both usernamepass as well as setting an ip
adress on a sip.extension?
It does. To enforce ACLs on a SIP user or peer or friend, simply use
permit and deny statements
2008/9/18 Duncan Turnbull [EMAIL PROTECTED]
Its a good question
I have lots of disk space so leave it high, I would rather have the
detail if I need it
It probably would seem sensible to revisit stable systems after a year
and lower the verbosity, but then since I can afford the space I am
Another question :
exten = 999,n,Log(DEBUG,local_ssrc:
${CHANNEL(rtpqos,audio,local_ssrc)})
Are those ${CHANNEL( (rtpqos,audio,local_ssrc)} values available today in an
Asterisk version or is it something describing what should be coded ?
Regards
Olivier schrieb:
Another question :
exten = 999,n,Log(DEBUG,local_ssrc:
${CHANNEL(rtpqos,audio,local_ssrc)})
Are those ${CHANNEL( (rtpqos,audio,local_ssrc)} values available today in an
Asterisk version
Yes. 1.4 and 1.6. But only for SIP channels obviously.
chan_sip.c:
Dae,
Activate debug full:
asterisk -vr
in other console do:
tail -vf /var/log/asterisk/full
Try to put call and send us more details about your logs
Regards,
Luis Morales
On Thu, Sep 18, 2008 at 11:49 PM, Dae Yeung Um [EMAIL PROTECTED] wrote:
In fact I see 1101 in the
Do as Luis says, however, I feel that as long you keep getting 1101
Unicall won't work. AFAIK The only IDLE bit pattern recognized by
libmfcr2 as IDLE is 10XX, as long you have 11 in the first 2 bits
(AB), libmfcr2 will report the lines as blocked.
On Thu, Sep 18, 2008 at 8:16 AM, Luis Morales
You need some outside process to keep call state, probably using the
Manager API and/or AGI. The outside process can listen to periodic call
setup events at a relatively low polling interval and make appropriate
adjustments to the user's credit in the database, which will then allow
you to
Isn't 'don't allow multiple calls' acceptable solution?
At least, it's the simplest one :)
I can imagine solution with multiple calls allowed, but it needs some external
synchronous processing. With every call you should start process, that will
decrement user's balance based on dialled
Hello
I got:
[Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call called -
'g1/6055151'
[Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: unicall_call caller id -
'1102'
[Sep 17 19:24:50] DEBUG[4934] chan_unicall.c: no UC_CATEGORY specified for
chan UniCall/1-1, using default
2008/9/18 Philipp Kempgen [EMAIL PROTECTED]
Olivier schrieb:
Another question :
exten = 999,n,Log(DEBUG,local_ssrc:
${CHANNEL(rtpqos,audio,local_ssrc)})
Are those ${CHANNEL( (rtpqos,audio,local_ssrc)} values available today in
an
Asterisk version
Yes. 1.4 and 1.6. But only for
Igor Zamocky wrote:
Isn't 'don't allow multiple calls' acceptable solution?
At least, it's the simplest one :)
I can imagine solution with multiple calls allowed, but it needs some external
synchronous processing. With every call you should start process, that will
decrement user's balance
So here is the deal. I have an Asterisk server here at work that I
have recently taken over and the boss is wanting the server to do a
lot of things that it didn't do before. I have already configured much
of what he wanted including a voice messaging line where anyone can
call in and
Remco Barendse schrieb:
Suprising that this feature isn't used much, i would suspect that many
asterisk installations (including mine) have very simple (short) extension
numbers which makes brute forcing them rather easy.
Extension numbers and SIP account basically have nothing to do with
Hi,
How can I create a web page allowing people to listen (with their own PC) a
couple of .wav/a-law files stored on a Linux server ?
Chances are users would access this web page from Internet Explorer but if I
could make it available to other browsers, that would be better.
I googled a bit and
Ok,
in your E1 setup:
1-15: to outgoing calls
16-30: for incomming calls
?
Now for make calls your telephone company must be provide MFC-R2
signaling. In your case the logs files show an invalid signal on make
call.
Regards,
Luis Morales
On Fri, Sep 19, 2008 at 10:10 AM, Dae Yeung Um
Thanks guys for inputs...not allowing multiple call is not an option -
essentional thats the problem we try to solve :)
Since we have our own CDR module, we can avoid external process. What
are the evens to listen for?
Other ideas will also be appreciated.
On Thu, Sep 18, 2008 at 8:23 PM, Alex
Take a look at the Asterisk Manager API documentation on voip-info.org
and experiment empirically by connecting and watching what transpires.
On Thu, September 18, 2008 11:14 am, Jim Boykin wrote:
Thanks guys for inputs...not allowing multiple call is not an option -
essentional thats the
How do you feel about converting them to RIFF/MSPCM WAV format and encoding
them into MP3?
On Thu, September 18, 2008 11:01 am, Olivier wrote:
Hi,
How can I create a web page allowing people to listen (with their own PC)
a
couple of .wav/a-law files stored on a Linux server ?
Chances are
On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote:
On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote:
Anybody knows how to get a Coupon Code for the discount on the Asterisk
training classes??? I am interested on taking that upcoming Asterisk
Advance course,
On Thursday 18 September 2008 05:16:21 Olivier wrote:
Whatever the verbosity level (even 0), my Asterisk console is full of
Really destroying SIP dialog messages.
Is there a way to get rid of those ?
Turn off debugging: core set debug 0 (and don't specify -d on your command
line).
If not,
On Thursday 18 September 2008 09:54:39 Steve Anness wrote:
So here is the deal. I have an Asterisk server here at work that I
have recently taken over and the boss is wanting the server to do a
lot of things that it didn't do before. I have already configured much
of what he wanted including a
On Thu, 18 Sep 2008, Olivier wrote:
Hi,
How can I create a web page allowing people to listen (with their own PC) a
couple of .wav/a-law files stored on a Linux server ?
Chances are users would access this web page from Internet Explorer but if I
could make it available to other browsers,
Hi,
It has IOS is 12.3(14)T7 - I'm wondering if this router can be made SIP aware.
Apparently, this current
firmware/programming is not, one way audio problems.
Is there a version that support VoIP directly for this router?
Thanks, Bart___
--
Tilghman Lesher wrote:
On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote:
On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote:
Anybody knows how to get a Coupon Code for the discount on the Asterisk
training classes??? I am interested on taking that upcoming
Barton
I think this will help you out
http://articles.techrepublic.com.com...1-6136216.html
http://articles.techrepublic.com.com/5100-1035_11-6136216.html
Cory J. Andrews
Director New Market Initiatives
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
Gordon Henderson schrieb:
If the web server is running php, then this will work:
?
$action = $HTTP_GET_VARS[action] ;
$file = $HTTP_GET_VARS[file] ;
$caller = $HTTP_GET_VARS[caller] ;
if (empty ($action) || empty ($file))
die (Something went wrong) ;
// Open
2008/9/18 Alex Balashov [EMAIL PROTECTED]
How do you feel about converting them to RIFF/MSPCM WAV format and encoding
them into MP3?
Why not ?
I don't know why I came to stick with A-law (as this is the codec used
elsewhere and audio prompts will be recorded using hardphone) but thinking
Another idea can be have the customers to opt-in for auto-refill if they
want to use multiple call feature. Usually this does not have be a high
number, just autorefill the account if the balance goes down $1.
Jai
www.didforsale.com
*Buy DID at low cost http://www.didforsale.com;
On Thu, Sep 18,
Apparently I mis-interpreted was the original poster was wanting. Good
thing. I'm glad he has a solid answer.
But, this does bring up the my issue of yore, and I'd be curious how
people have handled this. Key items:
* It's a distributed server farm. There are N asterisk servers serving
2008/9/18 Gordon Henderson
[EMAIL PROTECTED][EMAIL PROTECTED]
On Thu, 18 Sep 2008, Olivier wrote:
Hi,
How can I create a web page allowing people to listen (with their own PC)
a
couple of .wav/a-law files stored on a Linux server ?
Chances are users would access this web page from
I have an Ubuntu 8 server running Asterisk 1.4 with three interface cards in
it:
* Wildcard TDM400P
* Wildcard TDM410P
* Wildcard TE122
I'm using zaptel 1.4.11, and the difficulty I'm running into is that
with EVERY reboot, the order in which the hardware appears changes. This
makes ztscan cough
Barton Fisher schrieb:
It has IOS is 12.3(14)T7 - I'm wondering if this router can be made SIP
aware. Apparently, this current
firmware/programming is not, one way audio problems.
Is there a version that support VoIP directly for this router?
Do you have firewall feature set? Then you
All channels 1~15, 17~31 is supposed to be double way. To place and receive
calls.
The line is supposed be E1-MFC/R2 and works perfect with a Panasonic PBX,
actually
Exists any variant of MFC/R2? And how can I configure it to get working?
Your help will be very appreciated!
Thank you!
On Thu, Sep 18, 2008 at 01:09:38PM -0400, Jason T. Nelson wrote:
I have an Ubuntu 8 server running Asterisk 1.4 with three interface cards in
it:
* Wildcard TDM400P
* Wildcard TDM410P
* Wildcard TE122
I'm using zaptel 1.4.11, and the difficulty I'm running into is that
with EVERY
I'm not sure but on E1 setup you can have only one way (in or out). In
my case i have 15 in and 15 out.
Told me more about your hardware:
- E1 cards
- How did you do to connect E1 interface to E1 asterisk's card ?
- You can receive calls ?
Please send us zapata.conf and unicall.conf
Regards,
In our last exciting episode, Tzafrir Cohen ([EMAIL PROTECTED]) said:
Those cards use each a different driver. Write those driver, in your
preffered order, in /etc/modules .
Ah, I should have mentioned I did that (snippit from /etc/modules below)
zaptel
wcte12xp
wctdm
Having this doesn't seem
On Thu, 18 Sep 2008, Philipp Kempgen wrote:
Gordon Henderson schrieb:
If the web server is running php, then this will work:
?
$action = $HTTP_GET_VARS[action] ;
$file = $HTTP_GET_VARS[file] ;
$caller = $HTTP_GET_VARS[caller] ;
if (empty ($action) || empty ($file))
On Thu, Sep 18, 2008 at 12:20 PM, Barton Fisher [EMAIL PROTECTED] wrote:
Hi,
It has IOS is 12.3(14)T7 - I'm wondering if this router can be made SIP
aware. Apparently, this current
firmware/programming is not, one way audio problems.
Is there a version that support VoIP directly for this
Olivier wrote:
Hi,
How can I create a web page allowing people to listen (with their own
PC) a couple of .wav/a-law files stored on a Linux server ?
Chances are users would access this web page from Internet Explorer
but if I could make it available to other browsers, that would be
On Thu, Sep 18, 2008 at 12:26 PM, Eric ManxPower Wieling
[EMAIL PROTECTED] wrote:
Tilghman Lesher wrote:
On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote:
On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote:
Anybody knows how to get a Coupon Code for the discount
On Thu, Sep 18, 2008 at 01:50:42PM -0400, Jason T. Nelson wrote:
In our last exciting episode, Tzafrir Cohen ([EMAIL PROTECTED]) said:
Those cards use each a different driver. Write those driver, in your
preffered order, in /etc/modules .
Ah, I should have mentioned I did that (snippit
Steve Totaro wrote:
On Thu, Sep 18, 2008 at 12:26 PM, Eric ManxPower Wieling
[EMAIL PROTECTED] wrote:
Tilghman Lesher wrote:
On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote:
On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote:
Anybody
The line is supposed be E1-MFC/R2 and works perfect with a Panasonic PBX,
actually
Exists any variant of MFC/R2? And how can I configure it to get working?
As I said, no matter which variant you try, the AB bits MUST be in 10
to be able to make calls with Unicall/libmfcr2. I have never seen
Folks,
I have an odd problem (at least, it's odd to me).
System language is spanish (es) and when users check their voicemail,
if they don't delete it it goes into the Old directory.
That's all well and good, but those users with messages in their Old
directory try to get into voicemail and
On Thu, 2008-09-18 at 14:22 -0400, Steve Totaro wrote:
Are there braindumps out there, or TroyTec (http://www.troytec.com) 13
page cheat papers that will allow you to hold the highly coveted dCAP?
Not that I'm aware of.
dCAP is useless if not based on real world experience. That is how I
On Thursday 18 September 2008 14:19:29 David A. Bandel wrote:
System language is spanish (es) and when users check their voicemail,
if they don't delete it it goes into the Old directory.
That's all well and good, but those users with messages in their Old
directory try to get into voicemail
Kristian Kielhofner schrieb:
IMNSHO, the less SIP aware the better...
I have to disable SIP inspection on every IOS/PIX device I come
across. Fix the one-way audio problems on your proxy, registrar, etc
(in the case, Asterisk).
Most SIP ALGs are broken.
Interesting. I have my
On Thu, Sep 18, 2008 at 4:18 PM, Stefan Gofferje
[EMAIL PROTECTED] wrote:
Interesting. I have my Asterisk with RFC-1918 IPs behid a NATting PIX
and the FIXUP SIP of the PIX makes it very easy for me to use my * as
server for external clients as well as as client for SIP providers.
The PIX
Just in case anyone is having DNS SRV timeouts with their Polycom
phones, the following Polycom KB article should help:
http://knowledgebase.polycom.com/kb/search.do?cmd=displayKCdocType=kcexternalId=12856sliceId=SAL_PUBLIC_1_2dialogID=7620671stateId=1
We have set
Hello,
We had the same problem in the past and the last idea I had was to remove
first the modules and load them (using /etc/rc.local) in the right order.
Like:
rmmod wcte11xp
rmmod wctdm
modprobe wcte11xp
modprobe wctdm
modprobe zaptel
Maybe not the best way to do the job but it works for us.
2008/9/18 Tilghman Lesher [EMAIL PROTECTED]
On Thursday 18 September 2008 05:16:21 Olivier wrote:
Whatever the verbosity level (even 0), my Asterisk console is full of
Really destroying SIP dialog messages.
Is there a way to get rid of those ?
Turn off debugging: core set debug 0 (and
On Thu, Sep 18, 2008 at 3:22 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Thu, 2008-09-18 at 14:22 -0400, Steve Totaro wrote:
Are there braindumps out there, or TroyTec (http://www.troytec.com) 13
page cheat papers that will allow you to hold the highly coveted dCAP?
Not that I'm aware of.
Depending on what e-mail server software you use, it may be easier to direct
the voicemail to a specific e-mail address and have your e-mail software
rewrite the subject, and then forward it on to your boss.
On Thu, Sep 18, 2008 at 11:07 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Thursday
That is how I do it as well but don't forget /usr/sbin/asterisk or you
will just have a bunch of loaded modules.
I never bother with init scripts or /etc/modules. rc.local all the
way and I once challenged the list to give me a reason why that is NOT
a good way. No replies...
Thanks,
Steve
We have done something similar using the category option with the voicemail.
Our emails look like this:
--
TO : Big Boss
ID : 2
CAT. : EMERGENCY
BOX : 100
FROM : Emergency Line 5552221212
DUR : 0:20
DATE : Wednesday, October 10, 2007 at 01:28:27 PM
--
Internal
- Olivier [EMAIL PROTECTED] wrote:
A somehow related question, is broadcasting streaming music as music
on hold, submitted to any licencing fee ?
I got here late.
The only way you can legally use music as music on hold is if you either pay,
or are not subject to pay, performance royalty
Steve Totaro schrieb:
I never bother with init scripts or /etc/modules. rc.local all the
way and I once challenged the list to give me a reason why that is NOT
a good way. No replies...
http://lists.digium.com/pipermail/asterisk-users/2008-April/210491.html
I've had the same experience. I probably have 20-30 customers with
multiple SIP phones behind PIX running 6.3(5) (which has been out almost
3 years) and I have no issues at all. You can even have two phones
behind a PIX being PAT'd to a single external IP with reinvite enabled
in * and you
Philipp Kempgen wrote:
Steve Totaro schrieb:
I never bother with init scripts or /etc/modules. rc.local all the
way and I once challenged the list to give me a reason why that is NOT
a good way. No replies...
http://lists.digium.com/pipermail/asterisk-users/2008-April/210491.html
It's a Digium TE121P with Echo Cancellation
Zapata.conf
# Span 1: WCT1/0 Wildcard TE121 Card 0 HDB3/CCS/CRC4 RED RECOVERING
span=1,1,0,ccs,hdb3
cas=1-15:1101
dchan=16
cas=17-31:1101
# Span 2: WCTDM/0 Wildcard AEX800 Board 1 (MASTER)
fxsks=32
fxsks=33
fxsks=34
fxsks=35
# channel 36, WCTDM, no
Hi Dae,
In zaptel.conf change ccs for cas and comment dchan line, for example:
span=1,1,0,cas,hdb3
cas=1-15:1101
#dchan=16
cas=17-31:1101
--
Humberto Figuera - Using Linux 2.6.22
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA
I got my dCAP by turning up to the exam at Astricon in Madrid a couple years
ago without doing any training. It may have changed since then but I found
that the practical exam would be difficult if not impossible to pass without
knowing what you were doing - either through real world experience
Hi all,
When one uses the follow-me logic to forward calls to lots of phone
devices do subsequent calls get routed to the VM (or whatever the 10x
is)?
In other words, if I want my office, house and cell phones to ring
whenever a call comes in and I answer it on my cell, does the next call
that
On Thu, Sep 18, 2008 at 2:52 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Thursday 18 September 2008 14:19:29 David A. Bandel wrote:
System language is spanish (es) and when users check their voicemail,
if they don't delete it it goes into the Old directory.
That's all well and good, but
If you use iax, the console will tell you what codec is being used.
But for sip, nothing is shown. With sip debug on, I get:
Capabilities: us - 0x130e (gsm|ulaw|alaw|g729|speex|g722), peer -
audio=0x100e (gsm|ulaw|alaw|g722)/video=0x0 (nothing)/text=0x0
(nothing), combined - 0x100e
Sean,
Try 'sip show channels' or 'sip show channel channelid' for the drill down. I
believe the codec in use will be displayed with either command.
Dave
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of sean darcy [EMAIL
PROTECTED]
Sent: Thursday,
is this a feature in asterisk?
On Mon, Sep 15, 2008 at 3:20 AM, Patrick Maartense
[EMAIL PROTECTED]wrote:
Ice is the feature you're looking for I think
If two clients support ice, a direct link between them will be made
--
*From:* [EMAIL PROTECTED]
David Gibbons wrote:
Sean,
Try 'sip show channels' or 'sip show channel channelid' for the drill down.
I believe the codec in use will be displayed with either command.
Dave
Thanks that worked. Now how do I get it show the codec when I'm not at
the CLI?
sean
sean darcy wrote:
David Gibbons wrote:
Sean,
Try 'sip show channels' or 'sip show channel channelid' for the drill
down. I believe the codec in use will be displayed with either command.
Dave
Thanks that worked. Now how do I get it show the codec when I'm not at
the CLI?
Show it
Greetings,
I am running kernel 2.6.26.5, Asterisk 1.6.0rc2 and DAHDI 2.0.0rc4/rc2
and cannot get the DAHDI drivers to detect my Digium T100P:
01:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
The message when loading the wct1xxp module is:
t1xxp: probe
On Wed, Sep 17, 2008 at 1:10 PM, logan [EMAIL PROTECTED] wrote:
Thanks a lot Nhadie. I appreciate your help.
Could you also suggest some brands or models of the FXO+FXS card that
are seamlessly compatible to Asterisk? Also what hardphone I should go
for as there are so many in the market?
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