Agree that the 603 is wrong. It hasn't caused me issues but I see where it
could. And it goes against what I have been teaching in my classes, which is
irritating ;-)
In Asterisk, it's only used when we have no other hangup cause - and is
propably an indication that there is a code path that
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich
Inviato: giovedì 28 gennaio 2010 21:41
A: asterisk-users@lists.digium.com
Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short
I would very much
I don't know about 4xx, but 503 would be more benign for general/
miscellaneous errors than 603.
--
Sent from mobile device
On Jan 29, 2010, at 3:54 AM, Olle E. Johansson o...@edvina.net wrote:
Agree that the 603 is wrong. It hasn't caused me issues but I see
where it could. And it goes
29 jan 2010 kl. 10.25 skrev Alex Balashov:
I don't know about 4xx, but 503 would be more benign for general/
miscellaneous errors than 603.
503 indicates that there's a problem with the server, so that's not a good
replacement.
We're sending this when there's a failed call, in most cases
Hello to all.
I have installed asterisk-1.6.2.1 + asterisk-addons-1.6.2.0 (for chan_mobile) +
bluez-4.60.
Bluetooth Dongle: Canyon CN-BTU4 (0a12:0001 Cambridge Silicon Radio, Ltd
Bluetooth Dongle (HCI mode))
Device Descriptor:
Hi,
In the aftermath of Digium's and Counterpath's Bria for Asterisk
announcement, we're happy to chat with Todd Carothers, Counterpath
Product Manager today at 1 PM EST.
For more info, http://vuc.me
Join us on IRC #vuc on Freenode.net or use the web client at http://vuc.me/irc
Call in
After an upgrade to asterisk 1.6.2.1 I'm unable to make outgoing calls via
Vitelity. I get lots of these on my asterisk console:
[Jan 27 08:58:41] WARNING[25653]: chan_sip.c:3581 __sip_xmit: sip_xmit of
0x834ae08 (len 927) to 64.2.142.18:0 returned -1:
Address family not supported by
Hi,
How can I disable comfort noise on Asterisk?
Szabolcs Szasz
--
_
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To UNSUBSCRIBE or update options visit:
Szasz Szabolcs wrote:
How can I disable comfort noise on Asterisk?
Asterisk does not have a comfort noise generator, so there is nothing to
disable. You'll have to be more specific about what you are trying to
accomplish.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Hello list !
Having troubles with multiple registrations to one and the same ITSP.
This sip.conf :
register = user1:pass...@sip.itsp
register = user2:pass...@sip.itsp
; outgoing conversations
[user1-out]
type=peer
host=sip.ITSP
username=user1
secret=secret1
fromuser=user1
dtmfmode=rfc2833
I don't know what it showed at that time, I didn't know to look for that. I'll
try to get the customer's permission to re-create the symptoms this weekend and
post back with the lsdahdi output.
Thank you,
Noah Engelberth
Direct Link Computer Systems
-
Message: 12
Date: Thu, 28 Jan 2010
Apparently it is a telco issue. As I stated, when I do the dial from my
Polycom 501 handset, the landline call indicates ringing at 0:01 or 0:02
on the timer; the cell call doesn't indicate ringing until 0:04 or 0:05.
Manual calls excluding Asterisk from the process produced suitably similar
On Jan 29, 2010, at 5:59 AM, Olle E. Johansson wrote:
29 jan 2010 kl. 10.25 skrev Alex Balashov:
I don't know about 4xx, but 503 would be more benign for general/
miscellaneous errors than 603.
503 indicates that there's a problem with the server, so that's not a good
replacement.
Hi Noah,
IMHO replace the FXS module with new one or working module. That could be
proved where is the root cause. Mostly, the trouble issue is caused by FXS
modules in my experience.
Good Luck,
Johnson
On Fri, Jan 29, 2010 at 10:10 PM, Noah I. Engelberth
n...@directlinkcomputers.com wrote:
Something else that is flaky, missing or otherwise irritatingly broken
in the piece of shit that is 'Asterisk'.
[Cary Fitch]
It is an open source project. When can we count on your contribution of a
comfort noise generator that will not be a piece of s--t?
Can you have that by Monday?
CF
On Fri, 2010-01-29 at 07:29 -0600, Kevin P. Fleming wrote:
Szasz Szabolcs wrote:
How can I disable comfort noise on Asterisk?
Asterisk does not have a comfort noise generator, so there is nothing to
disable. You'll have to be more specific about what you are trying to
accomplish.
--
Just read RFC 3389; Guess the solution is going to be to sell Asterisk to
someone, make us pay for it and make everyone run a g.711 codec? If you
don't like it, don't use it!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On 29 Jan 2010, at 14:53, listu...@spamomania.co.uk wrote:
Something else that is flaky, missing or otherwise irritatingly broken
in the piece of shit that is 'Asterisk'.
It's open source. Fix it yourself, no one else is going to fix it for
you with that attitude.
--
On Fri, Jan 29, 2010 at 5:59 AM, Olle E. Johansson o...@edvina.net wrote:
29 jan 2010 kl. 10.25 skrev Alex Balashov:
I don't know about 4xx, but 503 would be more benign for general/
miscellaneous errors than 603.
503 indicates that there's a problem with the server, so that's not a good
I have an Asterisk 1.4.2 system installed at our office, and have a few
'on the road' sales people that want to make calls from their cell
phones in transit, but they are complaining that people returning calls
that they make from their cell phones are simply just using the CID that
is coming
On 1/29/2010 10:13 AM, Myles Wakeham wrote:
Basically I wanted to see if I could get them to call our phone number
on Asterisk, enter some special extension and/or enter a passcode, and
then enter the phone number that they wanted to call in which our phone
system would route the call to
Try this
- exten = 393,1,noop(forward this call)
- exten = 393,n,authenticate(7277,a)
- exten = 393,n,Read(custno,customerphone,11,skip,5,1)
- exten = 393,n,Dial(DAHDI/1,w${custno},30,mKkg)
- exten = 393,n,Playback(vm-goodbye)
- exten = 393,n,Hangup
This will require the caller to enter a
Hi,
I`m having a problem I cannot explain. When dialing 555-555- (for
example), I get a ringing sound until the person answers. When I have my
Polycom forwarded to 555-555-, I do not get the ringing, but it dials
fine and eventually when the person answers everything works fine.
Hello Noah,
Just shifting your TDM2400 and system to other computer, it might be figured
out your issue. As my opinion, the issue is involved with computer reset and
TDM2400 epld programming. Would figuring out the issue completely, it is
better to replace the TDM2400 card with Digium's.
Garry
Kristian Kielhofner wrote:
I don't want to ruin your plans for tonight (RFC3261 is a lot of
fun) but how about 403:
21.4.4 403 Forbidden
The server understood the request, but is refusing to fulfill it.
Authorization will not help, and the request SHOULD NOT be repeated.
Well,
Here are snippets from my OP that I tested and the DISA that Jeremy
suggested;
; authenticate and dial
exten = 393,1,noop(forward this call)
exten = 393,n,authenticate(7277,a)
exten = 393,n,Read(custno,enter-phone-number10,7,skip,5,1)
exten = 393,n,Dial(DAHDI/1/w${custno},30,mKkg)
exten =
Please post CLI output from the 2 calls with the number xxx'ed out.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Friday, January 29, 2010 9:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
On Fri, Jan 29, 2010 at 10:31 AM, Kevin P. Fleming kpflem...@digium.com wrote:
Well, that's the problem, and it's the reason why 603 is so commonly
used. This is a situation where the current request has failed, but
there is no indication that repeating the request will also fail. 403
means
On Fri, 2010-01-29 at 15:09 +0100, jonas kellens wrote:
Hello list !
Having troubles with multiple registrations to one and the same ITSP.
This sip.conf :
register = user1:pass...@sip.itsp
register = user2:pass...@sip.itsp
; outgoing conversations
[user1-out]
type=peer
Hi William,
I appreciate your answer, though can you make things more clear for me:
1- i am not using extensions when registering PBX boxes in IAX files.
2- is inbounx context in the call sender PBX (pbx1) and outbound context is
in the call receiver (or dialer) PBX (pbx2)?
3- i am using two
Is it possible to use the smsq command in asterisk to
send SMS messages to a aggregator.
So If I have an IP address, password and port for my connection
can I use smsq to send SMS messages? I dont see how to set that up?
I am looking: http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms
Thanks,
Hi there! First mail on the list :)
1.- is it possible to use an spa2102 to make and revice calls from a
normal phone? I mean, I know I can use it to connect an analog to an
asterisk server, but I want to know if it can be used to connect
asterisk to the analog phoneline.
2.- I'm trying to
When setting type=friend for the incoming calls :
; outgoing conversations
[user1-out]
type=peer
host=sip.ITSP
username=user1
secret=secret1
fromuser=user1
; incoming conversations
[user1]
type=friend
host=sip.ITSP
context=user1incoming
; outgoing conversations
[user2-out]
type=peer
You are using contexts..
Look @ destination pbx, you should see something like this:
Rejected connect attempt from ip of source pbx, request 'ext@incoming
conext' does not exist
If you didn't put a context under the peer, it uses the default one in the
iax.conf file which is normally
I have quite a number of users complaining that when they are using handsfree
to talk over a landline, the other end can't hear them. It's like the person
is speaking 5 feet away and can barely hear their voice. However between
internal SIP calls, it's fine.
What could be the problem?
To get back to the original poster's possible situation, i've seen this
with my first IP phone, which was a cisco 7912 (SIP image). With that
phone, asterisk sometimes gave me this same error. I'm quite sure i've
asked the very same question here back then (probably i was a bit more
specific
Just if it is helps someone, based on information at the blog:
http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.htmlI've
summarized the following steps:
*Step 1:*
Configure audiocodes to have registration account with asterisk, this can be
done easily with Protocol
Damn, where were you 6 months ago? ;)
Daniel - Asterisk wrote:
Just if it is helps someone, based on information at the blog:
http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.html
I've summarized the following steps:
*Step 1:*
Configure audiocodes to have
You don't state this, but the assumption would be that your external calls
are DAHDI based, so you might need to tweak txgain in dahdi.conf.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee
Sent: Friday, January 29,
Den 28-01-2010 20:15, Danny Nicholas skrev:
Here's one solution:
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4,Background(emergencyin${IMAT})
Where you would record /var/lib/asterisk/sound/emergencyin100
HI:
I had this problem before with TDM2400P but with fxo modules and VPMADT032
(echo canceller),there was no audio at all.but then i unpulgged the ehco
canceller module (VPMADT032) from the TDM2400P board and started
the server and then i didnt face this issue any more.
In your case first
This might help
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4(keepup),Background(emergencyin${IMAT})
- exten = _911,5,wait(10)
- exten = _911,6,Goto(keepup)
This would repeat the message every 10 seconds...
--
Hi,
I've uploaded a new patch at
https://issues.asterisk.org/view.php?id=16732which adds two new AMI
commands, called DeviceStateSet and
DeviceStateGet.
These commands let you update Custom device states, and read all
devicestates from AMI.
It would be very nice if someone could help me test
It was a pending draft I forgot to send.. sorry.
On Fri, Jan 29, 2010 at 1:23 PM, Matt Collins mcoll...@ccdservice.netwrote:
Damn, where were you 6 months ago? ;)
Daniel - Asterisk wrote:
Just if it is helps someone, based on information at the blog:
Hello,
I have Asterisk 1.6.1.12 with
FAX For Asterisk Components:
Applications: 1.6.1.5_1.1.6
Digium FAX Driver: 1.6.1.5_1.1.6 (optimized for core2_32)
If I use call file with spool
Channel: SIP/IP/DEst No
MaxRetries: 0
I've been running Asterisk with a standard PRI for regular telecoms.
This is also connected to our Nortel PBX for 'ordinary users'. The
system has been working nicely (including Cisco 7970 phones that are
connecting via SIP).
But now I'm going 'on net' with broker lines (for a trading room
Den 29-01-2010 19:38, Danny Nicholas skrev:
This might help
- exten = _911,1,Set(IMAT=EXTEN)
- exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
- exten = _911,3,Dial(DAHDI/1,w911)
- exten = _911,4(keepup),Background(emergencyin${IMAT})
- exten = _911,5,wait(10)
- exten =
The idea behind the OP was that the caller was a man down who couldn't
speak to 911, just dial the number. You could always change wait to
waitexten and make an exten to break the loop if you were able to talk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi:
i did set the rtp ports in rtp.conf to rtpstart=5000 , rtpend=31000 ,and i
used canreinvite=no and the problem still exists ,however i did the rtp debug
and here is the output :
= Spawn extension (direct, 9613070741, 2) exited non-zero on
'SIP/03070741-083b9da0'
-- Executing
Leif Neland wrote:
2: Often callers are answered with an automated message This is 911,
please hold, just to give pranksters/misdiallers a chance to hang up
before disturbing the operator. Unless 911 records the incoming call
right from the start, they will never hear the im-at message.
Hi All,
I tried using some music on hold (music) files, when I test it with normal
SIP phone its clear and good, but when I call from my cell phone or POTS
line it sounds a bit scratchy/static and not clear at all, is there any
software that i need to install in the asterisk system to make this
Mpg123 works well for us. You have to get your files into mp3 format, but
LAME does this simply.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of das sandesh
Sent: Friday, January 29, 2010 4:26 PM
To: Asterisk Users Mailing
listu...@spamomania.co.uk wrote:
On Thu, 2010-01-28 at 23:11 -0600, Karl Fife wrote:
Appears completely resolved!
No more home-spun patches!
Thanks!
-K
It's *not* fixed here:
DAHDI Version: 2.2.1 Echo Canceller: MG2
But as is depressingly the 'norm' for Asterisk it comes back to
Sandesh-
I tried using some music on hold (music) files, when I test it with normal
SIP phone its clear and good, but when I call from my cell phone or POTS
line it sounds a bit scratchy/static and not clear at all, is there any
software that i need to install in the asterisk system to make
Calling from my home using Asterisk 1.6.2.1 to an office extension
(Asterisk 1.6.1.13) the callerid is not honored:
Home:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@internal:2] NoOp(DAHDI/1-1, Context:
I have a strange problem with CallerID that only affects some phones.
The problem is that whenever I receive a call the Callerid Name is
correct but the Callerid number is always my own extension. It does not
matter if the call is internal or external. So far only Aastra phones
and
On Fri, 29 Jan 2010, Danny Nicholas wrote:
Mpg123 works well for us. You have to get your files into mp3 format,
but LAME does this simply.
Why would you want to compress files when you will have to decompress them
again every single time the are used? I'd rather use the CPU cycles to
Hello,
I have been trying to setup asterisk 1.6.1.1 to receive fax. Whenever
a SIP peer (zoiper soft phones) tries to send a fax message asterisk
responds by sending a 488 Not acceptable here and the sending fails.
I tried changing a few sip settings like canreinvite and codec
preferences, but it
On 30/01/10 11:48 AM, sean darcy wrote:
Sigh.
OK you don't like asterisk - sorry. Obviously some other software works
better for you. I'm glad.
Don't worry, he/she's trolling, second post like that for the day :)
Obviously has an issue with something, but rather than try and get it
sorted
Kosa wrote:
Hi there! First mail on the list :)
1.- is it possible to use an spa2102 to make and revice calls from a normal
phone? I mean, I know I can use it to connect an analog to an asterisk
server, but I want to know if it can be used to connect asterisk to the
analog phoneline.
On Fri, 29 Jan 2010, Kosa wrote:
1.- is it possible to use an spa2102 to make and revice calls from a
normal phone? I mean, I know I can use it to connect an analog to an
asterisk server, but I want to know if it can be used to connect
asterisk to the analog phoneline.
The 2102 is an FXS
H all...
I have an Astribank (8FXS/16FXO), IBM X3200 M2, Asterisk-1.6.2.1,
dahdi-linux-complete-2.2.1, libpri-1.4.10.2, centos-5.4 final.
My problem is, every time i unplug the astribank power supply, and
reconnect it, astribank cannot work again (lsusb result is 11x0)...
but, after reinstall
On Fri, Jan 29, 2010 at 1:14 PM, ad...@3a.hu wrote:
To get back to the original poster's possible situation, i've seen this
with my first IP phone, which was a cisco 7912 (SIP image). With that
phone, asterisk sometimes gave me this same error. I'm quite sure i've
asked the very same
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