Hello..
How is this possible?? I have 65 active calls .. but
making new calls and registering isn't possible
anymore
the CLI command restart now didn't even work .. had to
kill the process before it worked
again...
myasterisk*CLI show
channels Channel
(Context Extension Pri )
Hello.. I have configured asterisk to send CDR's to an ODBC datasource
on IAX calls I can find the IP address of the caller in the 'channel'
field
For example: IAX2/username@ipaddr:4569-458
On SIP calls I never see the IP address of the caller
For example: SIP/username-9d51
So on
Hi,
Asterisk will work, but in your situation I think it's better use a SIP
proxy for that (SER for example http://www.iptel.org/ser) which is
really meant for this purpose
Niels.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vikram
Rangnekar
Sent
on the simultaneous calls asterisk behaves well on SMP sytems..
The problem could be in the many user-logon's per second you have (user
logons are handled in a single thread only) With about 1000 to 1500
IAX/SIP users online per box (dual xeon 3.2 ghz) we start expiriencing
problems over here
Try setting
defaultip=192.168.44.23
Too
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of barney
Sent: Monday, May 23, 2005 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Getting a Cisco gateway to
Hi .. When I use the GoTo statement in realtime to goto a priority only
... E.g. Goto(3) then there's no problem
But, If I try to jump to another context ... E.g.
Goto(othercontext,${EXTEN},3) then it doesn't work
If I process the same statement in extensions.conf things go well
Are there
?
is this ONLY a cpu problem or is there more I have to take a look at??
Memory speed etc?
all ideas apprecieated.
Thx. Niels
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Hello All!
I just got my IAXy.. Configured it.. Got it Up and Running
Calls OUT have no problems (that means from IAXy - Asterisk -
ZAP/SIPclient/IAXclient)
Calls IN do have problems (that means from ZAP/SIPclient/IAXclient -
Asterisk - IAXy)
On those incoming calls on my IAXy I hear the
Hello
I was wandering
If I let 2 asterisk boxes (let's name them ast01 and ast02) connect to
one SQL realtime iaxfriends/sipfriends database
What happens if I register my client to ast01, The ast01 box will update
the client's record in the iaxfriends database (ipaddr/port/regseconds)
Let's
Hello all!
Has anyone expirienced the following:?
With an IAXclient softphone (like diax/iaxcomm/etc) Dialing to the PSTN
(zap) or a SIP device has no problems .. but when I make calls between 2
softphones I have weird problems
in about 4 out of 10 IAX-2-IAX softphone calls I get a big
I see I am using a quite old version of DIAX (I am more an iaxcomm user
where I DO use the newest version of :-)
I will make some tests with the newest version of DIAX today or tomorrow
and get back on this!
Regards, Niels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
other possibility to store the
SIP/IAX callers IP address on every call?
Thanks
Niels
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Hi all... I have a slight problem with getting speex running
I Downloaded Speex sources (v. 1.0.4 stable version) and did make; make
install sucessfully
Then I re-maked the asterisk sources and clearly saw a speex.so module
being built (so the makefile for sure detects that there is a speex lib
After updating to the latest CVS
Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The
'sipfriends' table is obsolete, update your config to use sipusers and
sippeers, though they can point to the same table.
== Binding sipusers to mysql/asterisk/sip
== Binding sippeers to
Hello
On every Incoming SIP and IAX call I see the following in asterisk
debug:
Accepting AUTHENTICATED call from 10.10.10.10, requested format = gsm,
requested prefs = (), actual format = g729, my prefs =
(g729|gsm|g723|g726|ulaw|alaw) priority = mine
The problem is that the codec
I don't want that... because
- for outbound calls I want priority to be g729 first
- for inbound calls I want no priority at all (e.g. the calling asterisk
to decide which codec we will use)
The last doesn't happen..
This by the way DID happen correctly with previous versions of asterisk
it appears in.)
make: *** [app_addon_sql_mysql.o] Error 1
Can anyone help me out?
Niels
PS.. Below the full output of the make
[EMAIL PROTECTED] asterisk-addons]# make
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls
*.c`
app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE
: `AST_LIST_REMOVE' undeclared (first use in
this function)
app_addon_sql_mysql.c:164: (Each undeclared identifier is reported only
once
app_addon_sql_mysql.c:164: for each function it appears in.)
make: *** [app_addon_sql_mysql.o] Error 1
Can anyone help me out?
Niels
PS.. Below the full output
You missed it :-)
2 possibilities
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MeetMeCount
or
http://www.voip-info.org/wiki-Asterisk+cmd+GetGroupCount
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
Sent: dinsdag 21 december 2004
wrong or is this behaviour by
design?
Regards,
Niels
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Because else people will complain that they can't register two
softphones anymore with same user/pass (because only one of the two
softphones can receive the incoming calls) :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Ivar
Helbekkmo
Sent:
SER I am not happy with it.. didn't manage to get these below things
functioning:
It's too strict with authentication (user has to set specific
domain/realm) have problems with several types of hardphones authing to
SER
You can't make config changes without having to restart SER
Can't
I am running it statefull because else I would have to open port 5060 on
my cisco AS5400's to the world, and that's too insecure
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas
Sikkema
Sent: Tuesday, December 14, 2004 8:07 AM
To: Asterisk Users
Hmmm that's bad...
This is the last issue I have which makes that I can't get rid of the
SER proxy in front of asterisk.. Want to get rid of it
Are there any plans to change this design?? (that multiple UA's can
register to one peer?)
Niels
-Original Message-
From: [EMAIL PROTECTED
And that sounds
logically because you have session transport tcp ... And asterisk
doesn't support that... Use session transport udp
Regards,
Niels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jan Baggen
Sent: Tuesday, November 30, 2004 2:36 PM
To: [EMAIL
Hi,
In constant search for optimization, a friend told us about his experience
with Gentoo Linux-distro. He claimed that he doubled the performance of his
server by changing to Gentoo from Debian.
Does anyone have any experience with running Asterisk on a Gentoo linux?
/Niels
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:-)
Thanks in advance
/Niels - Denmark
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Hello
In some SIP invite messages I see the below codec negotiation string, I
am wandering what the 101 telephone-event/8000 means Which codec
is that?
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
Hello
Has anyone ever got a quintum A800 or A400 with SIP firmware on it
succesfully talking to asterisk's SIP stack?
I tried it.. But get many call leg does not exist errors
Seems like quintum's sip implementation is not the most compatible
one..??
Anyone experienced with this?
There is already one chipmaker who thought that
IAX was important or competitive advantage enough
to embed it into their chip.
Which?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
on Asterisk Mailing Lists
Sent: Tuesday, November 02,
would have this all on IAX this would be
unmanageable, we would need 50 linux boxes.
Conclusion.. IAX Is a good performer behind NAT and perfect for small
setups but to work in an enterprise, Much work has to be done.
Niels.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I'd say some learning on high
availability Linux/clustering etc
is in order.
I know all about it, but 50 boxes is just too much.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Tuesday, November 02, 2004 1:00 AM
To: Asterisk
Check these url's
http://www.voip-info.org/wiki-Asterisk+cmd+CheckGroup
http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup
http://www.voip-info.org/wiki-Asterisk+cmd+GetGroupCount
Niels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Rozman
I would like that too.. The easiest for me would be to be able to DISABLE codec
translations of any kind
Regards
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Friday, October 29, 2004 12:19 PM
To: Asterisk Users Mailing List
.
NiElS
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host.. If you limit this to
lets say about 25 to 50 connection attempts per second per host I would
say you're pretty safe and your asterisk box can't really get overloaded
with malicious packets. this burst limit depends on your config as you
might get much traffic from certain IP's ofcourse
Niels
variable which represents the
originating IP
Or have I missed something?
Regards
Niels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Tuesday, October 26, 2004 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
In my sip.conf or iax.conf I like to set a user or friend to the
following context
[User0001]
context=outbound0001
[User0002]
context=outbound0002
Then I extensions I would like to capture this with only one context,
doing something with the last 4 digits
I mean something like this:
Hello
I have chan_h323.so compiled.. And got is up and running
I can place calls now from my cisco AS5350 to asterisk and back
Only..
In h323.conf it doesn't seem to 'see' the my user .. It's just always
using the default context
If I dial from 192.168.1.50 (my cisco) to 192.168.1.100 (my
Ofcourse I tried that :-)
In this case an h323 call to asterisk doesn't work anymore (at all)..
The asterisk debug window is then complaining that it can't find a
default context at the moment I set up a h323 call
Thx
Niels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Ohh.. do I know the originating IP address then once arrived in
extensions.conf ?
I checked this page http://www.voip-info.org/wiki-Asterisk+Variables but
can't seem to find any predefined variable which represents the
originating IP
Or have I missed something?
Regards
Niels
-Original
Hello,
On an AS5350 this works so I expect this to work too your 3620
Regards,
Niels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon
Lawrence
Sent: Tuesday, October 26, 2004 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I missed something?
Regards
Niels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Tuesday, October 26, 2004 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] H323.conf question
Work
represents the
originating IP
Or have I missed something?
Regards
Niels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Tuesday, October 26, 2004 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk
represents the
originating IP
Or have I missed something?
Regards
Niels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Tuesday, October 26, 2004 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
I never tried it, but may I understand why you want to use MGCP?
I use SIP for the both way traffic between my cisco's and Asterisk boxes
which works well. SIP is also much better implemented in Asterisk
Niels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
Hello
I compiled the new 1.02 over 1.01
My old asterisk 1.01 was compiled (on redhat 9.0) by downloading the src
tarball from ftp.asterisk.com/pub/asterisk
I did this the exact same way now, downloaded the 1.02 tarball, unpacked
it, killed all asterisk 1.01 processes, issued a 'make' and
I did the trick, Wonderfull!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: dinsdag 26 oktober 2004 23:21
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk 1.0.2
rm all the .so's and
Hello
I have this in my
sip.conf ... it's a cisco which I authenticate on it's IP adress
this works..
asterisk authenticates sip calls from this IP as user 1234567 and uses the right
context,
only, the CALLERID
is the ip adress e.g. 19216801 instead of the callerid which i try to set
No it's the same... it still uses the IP address as callerid
I tried the following ones:
callerid=1234567
callerid=1234567
callerid=1234567
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Monday, October 25, 2004 2:55
I just do a -- asterisk -rx reload
This picks up the changes in sip.conf for sure :-)
If I comment out the defaultip line then asterisk still uses the IP
address as callerid :-(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Monday,
PS..
If I send NO callerid at all from my cisco, asterisk translates the
callerid to the ip address where the call originates from (in this case
my cisco).. When I DO send a callerid from my cisco it uses THAT
callerid
But still I don't get it to overrule the callerid by setting the
callerid=
My UA is not a phone it's a cisco AS5350 gateway
When I comment out the host= line calling doesn't work anymore (asterisk
uses the default context then which doesn't allow calling at all :-)
If I set the defaultip= line then (and keep commenting out the host=
line) it works again.. But the
No my problem is not solved...
Because I still want Asterisk to overrule the callerid :-(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Monday, October 25, 2004 4:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Thanks for yor answers!
-Now I've got it running just
fine:-)
NRB
I've got Zaphfc working running Asterisk v.
0.7.2
Then I have tried with Asterisk V. 1.0 and the
latest from CVS - with no succes.
Has anybody got zaphfc working with newer version
than 0.7.2 ?
NR
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