From inside a Perl script, Asterisk 1.4, I am trying to get this information
$xipaddress = $AGI-get_full_variable('CHANNEL(recvip)');
or using pan get_variable
But I get nothing. How do I read the IP address of origin from an AGI Perl
script?
I cannot update the version, for this is an old system
I have an Asterisk 13 that only processes app Transfer with PJSIP, to the
tune of 60 per second. No voice calls.
After like 2 hours, I can no longer get into Asterisk. This command,
asterisk -r, fails, and also asterisk -rx core show channels, etc. I am
returned to the bash prompt. I checked the
I checked
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
But I cannot find a way to insert CALLERID(ani2), which I can read, but
when I try to set it for a new call, I get a runtime error.
This information, known as isup-oli comes embedded in the From header,like
Two years ago we added logic to parse the isup-oli parameters, that arrive
as part of the FROM Sip header. We need to finish the job and allow setting
of this parameter for outbound calling, both in traditional SIP channel and
PJSIP, which I believe will replace all instances of the old SIP
Does the channel chan_motif and res_xmpp still work?
I heard that Google had blocked this technology.
Philip
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I keep getting this error
[Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:614
__allocate_taskprocessor: Unable to start taskprocessor listener for
taskprocessor bbe08c34-9d1c-4e5f-8ae0-0cc75289caca
[Nov 8 22:51:31] ERROR[8192]: taskprocessor.c:245
default_listener_shutdown: pthread_join(): Cannot
The amount of threads went through the roof
ls /proc/15373/task | wc -l
682
in version SVN-branch-12-r427618M
it used to be 18 in Asterisk SVN-branch-11-r412226M
How can I trace this? There are no calls open, on a disconnected system
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Is this normal to create one thread per peer in Asterisk 12, chan_sip
regular, not pjsip?
What happens is I have 659 peers, and I get 682 tasks on
ls /proc/15373/task | wc -l
If this is normal then of course I can only get a few instances before my
box collapses.
Is it any different in pjsip?
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I am using Asterisk 12 svn, from today, and after a few thousand
calls, I run out of ports.
This happens eith PJSIOP and regular old SIP. I think it is RTP related.
Any idea how can I troblshoot this. It happened teh same with Asterisk 11.
On the other end there is a freeswitch. My guess is that
I need to respond with 380 Alternative Service. Is there a way to do
this in PJSIP? Please note that I am not picking up the call. For
instance, the Transfer app closes the call if you did not answer it
first. There is a bug open about this. I want to stay with PJSIP, for
I found that it scales
I have a multihomed machine. How can I assign multiple IPs to and
endpoint, not all of them, just two, for instance, out of many?
Suppose the machine as 30 IPs, but my asterisk needs listen on two,
and one single endpoint needs to be associated with those two IPs. I
tried to add a second bind line
I am having the issue described in this question:
http://lists.digium.com/pipermail/asterisk-users/2005-May/099075.html
Does anybody has an insight? I guess Asterisk is trying to match the
combination IP:Port, but in H223 this changes call by call. There is
no way to add insecure=port like in
I had some confusion here. The endpoint needs a transport in order to
carry calls out. But the transports are also used by the application
PJSIP at large, in order to listen for incoming connections. In order
to just receive calls, I think you only need a transport, but no need
to assign that
I get tons of these messages
chan_sip.c:10088 process_sdp: Declining non-primary audio stream:
audio 30660 RTP/AVP 4 101 13
What does it mean and does it show a problem like one-way audio?
Thanks for your help.
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I have been seeing errors saying the Asterisk cannot establish an RTP
connection, so I did this:
lsof -i -n -P | grep asterisk | wc -l
10483
but I have only
Asterisk 11 has 1 open calls
Asterisk 12 has 21 open calls
Asterisk 14 has 19 open calls
Asterisk 15 has 22 open calls
Asterisk 16 has 15
In channel PJSIP I use this format
Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss)
what would be the equivalent of this format in old SIP?
I tried
Dial(SIP/peer/${EXTEN}@ip.add.re.ss)
but it does not work. I just cannot embed the IP address in the peer's
definition, but I need to use some other
I found that PJSIP allows only one asterisk per box. I tried to start
several asterisks with the parameter -C and PJSIP only worked on the
first process. In the other processes, the command pjsip reload was
absent. Each pjsip transport in the second and subsequent processes
was bound to a
I am trying to launch Asterisk on a different directory with the parameter 'C
asterisk -vvgc -C /etc/asterisk1/asterisk.conf
Parsing '/etc/asterisk1/extconfig.conf': Found
Resetting translation matrix
UUID system initiated
Parsing /etc/asterisk1/asterisk.conf
== Parsing
I want to compile Asterisk always for the native architecture of the
machine, and I find that it is never available. It says
Depends on: native_arch(E)
Can use: N/A
Conflicts with: N/A
Support Level: core
This is Fedora 20
gcc (GCC) 4.8.3 20140624 (Red Hat 4.8.3-1)
many thanks
Philip
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I cannot wait for the regular bug-patch process to play out. I am
considering hiring a developer to fix bug 24015, and of course submit the
patch for the bug. The issue is simple, the app Transfer does not transfer
when using PJSIP.. I called Digium and they said that they do not do this
kind of
I tried to do what I with regular SIP to Transfer a call via 302
Redirect. In asterisk 12 we need to add the Tech, or not, but in any
case, there is no transfer done. The call is closed.
Here is a trace. How do I do this?
[Jul 9 21:39:29] DEBUG[47716][C-0002]: pbx.c:4869
Dear friends
After spending few days converting my app to PJSIP, today I had to roll back
the upgrade because in the SDP, the Owner section is wrong, or I
misconfigured something
This is what my client said:
That OK message from 1.1.1.1 can not be parsed by our switch due to
address
Dear Friends
I keep getting this warning
[Jul 2 19:19:11] WARNING[16033][C-0441]: chan_pjsip.c:645
chan_pjsip_write: Can't send 10 type frames with PJSIP
But I could not find an explanation by googling. Any idea?
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I could not find a way to set a max on the calls allowed through a
PJSIP endpoint.
In case we decide to add it, the we need another reason for the call
to fail in the Dial application, something like limit reached
Am I missing this capability?
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In a PJSIP endpoint, how do I set all no-named settings so they get
inherited from another place and I don't need to mention them again
and again for all my endpoints?
In regular sip you could specify those options and they remained valid
if not redefined by a peer. A case would be the codecs
I did what we use to dim that is add a line to pjsip.conf like
#include /etc/asterisk/pjpeers.conf
but the file is not loaded. Am I doing something wrong this
functionality is disabled?
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I am migrating my app to Asterisk12 and pjsip, but I cannot find
chan_local, what happened?
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Dear friends
This is my simple dialplan
[demopjsip]
exten = _X.,1,Dial(PJSIP/${EXTEN}@10.10.10.2)
exten = _X.,n,Hangup()
I need to dial out via an IP address, not using an endpoint, as shown above.
It fails with
Executing [1957408@demopjsip:3] Dial(PJSIP/federico-0002,
A few months ago I started using and had to abandon PJSIP because my
dialplan could not read the inbound signalling IP address, which I can
read now in Asterisk11 using CHANNEL(recvip). My app relies on this
information. The
question is, is it possible now access the signalling IP of an
incoming
Okay, I see that there is no way to shorten the inter-digit delay when
we dial out DTMF. The reason I need this is because when dialing a
two-stage call, the PDD can be shortened dramatically while still
working fine, and that opens whole new world of business. But many
people confused
...@lists.digium.com
You can reach the person managing the list at
asterisk-users-ow...@lists.digium.com
When replying, please edit your Subject line so it is more specific
than Re: Contents of asterisk-users digest...
Today's Topics:
1. Shorten time between DTMF (CDR)
2. Re: Shorten time
I already shortened the DTMF duration, but I need to change the time
elapsing between them.
The first thing I achieved by changing a parameter in asterisk.conf,
but how do I conquer the second goal?
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I already shortened the DTMF duration, but I need to change the time
elapsing between them.
The first thing I achieved by changing a parameter in asterisk.conf,
but how do I conquer the second goal?
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When a client send me an INVITE with this type of caller ID
From: eurus
sip:null@XX.XX.XX.XXX;tag=3430296121-3809549020-352327076-1077499159
Asterisk 14 sends back
SIP/2.0 500 Server error occurred (1/SL)
My client says
Yes, I know the null is there but this not illegal and perfectly
Is there a way to add an X-Header to hangup(34), which translates to a 503?
I tried adding it before the hangup but it never gets transmitted
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I fund the issue and it was in my own code. I apologize.
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Dear friends
I filed a bug
https://issues.asterisk.org/jira/browse/ASTERISK-23656
but I am wondering if somebody can figure a workaround. I am stuck
trying to deliver an application.
The case is this: A Record is executed and an immediate Playback
follows. Asterisk returns an error, saying that
I wonder if anybody know how to hire Alice or some professional
voice-artist. I need to record 12 messages for a customer.
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I have had the issue for years. The problem is that Asterisk
developers are removed from the business. We desperately need simple
way to eliminate transcoding when unnecessary. Transcoding brings a
server to its knees. It is a very simple new setting in sip.conf
prioritize_matching_codecs=yes
I
Digium is 100% lost in the map. If they would come up with a Paid
version of Asterisk, one that would use the .NET framework in Windows,
something simple to install, they could go public on the product.
Linux has a very steep learning curve. A Windows application that
would do exactly the same
I am using Asterisk 11.6.0 built by root @ linux-t784 on a x86_64
The issue is a huge UDP handle leak, presumably coming fro ooh323
With 45 calls open calls (ooh323 to SIP), I have
netstat -anp | grep asterisk | wc -l
6669
lsof -p 6785 -i -n -P | grep UDP | wc -l
6567
The machine needs to be
I am trying to identify the module (*.so) that contains the Asterisk
Management Interface, so as to set noload=XXX.so in modules.conf. Any
idea?
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New to
I need to log out from a Google Voice account, before I use a
different account, otherwerwise Gvoice will block calling capability.
How do I do that? I cannot figure this out. There should be function
or application to cleanly log out
Philip
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Right now,there is no way know to capture the Media IP. The channel
variable does not know about it. It requires adding anew variable to
CHANNEL(), and also it entails to force every channel to update that
variable. New channels like PJSIP do not even update the known
variables like
is the most widely used
low-cost technology for telecommunications.
I need Digium to store this IP in the CDR. I will be honest with the
government and let them know that my tool is incapable of saving lives
or safeguarding our national security because nobody thought about
this.
PD: I am not paying
The CHANNEL() function has no idea about the media IP, and also
SIP_HEADER(), since the media IP is not known until the call has been
established and a reinvite has been received and dispatched. I am
using of course, directmedia=yes and directrtpsetup=yes. Hence my
question to the group.
--
I am not proxying the media, but never the less I am forced to store
the source media IP in my CDR, for regulatory reasons. Asterisk gets
that information when the reinvite comes, but how do I store it?
If I don't figure this out my next email will be from Federal Prison.
Kindly help me stay away
In Asterisk 12, how should I call the function CDR_PROP
set(CDR_PROP(disable)=true)
or simply CDR_PROP(disable)
I am getting two records per call attempt, and I cannot figure out how
to go back to get only one record. So far I am using this technique,
but it changes nothing. My calls always
I have one single endpoint
[inhouse](endpoint-basic)
type=endpoint
and one section like this
[indentify]
endpoint=inhouse
match=X.Z.Y.X
But when I call in from IP X.Z.Y.X, it does not match my peer. It
tries to send the call the default context, and the number is not
found.
I know that I
The Wiky needs to be updated
https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip#Configuringres_pjsip-IDENTIFY%28res_pjsip_endpoint_identifier_ip%29
This is the example shown:
[6001]
endpoint=6001
match=203.0.113.1
It should be:
[6001]
type=identify
endpoint=6001
match=203.0.113.1
I am stuck in channel PJSIP trying to see the real flow of SIP
messages, what in regular sip
we used to type sip set debug on
Also, is there an automated way to convert sip.conf options to pjsip.conf?
Philip
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I cannot find in Asterisk 12, the channel variable ${CHANNEL(recvip)},
so if I use PJSIP, for scalability, how do I read what the signalling
IP where the inbound call is coming from and what is the inbound
codec?
You would think that the new channel would set those variables up, isn't it?
Philip
I am stumped
In features.conf,I programmed this
[applicationmap]
Answer0 = 0,self/both,Macro,nway_start
But do I pass an argument or parameter to my macro? I tried
Answer0 = 0,self/both,Macro,nway_start^0
Answer0 = 0,self/both,Macro,nway_start,0
but the usuar variable ${ARG1} is empty in my
I need to block any audio before there is a connect, in SIP. How do I tell
the DIAL application to behave like that?
Yours
Philip
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Is it me or Google just blocked Asterisk's chan_motif? I get violation of
terms of service audio message whenever I send a call.
Philip
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Dear friends
I need to loopback the audio on my channel. Did anybody on the development
team thought about a function or app that would do that? If it is not
clear, I mean that whatever audio I get, I send back.
Philip
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In safe_asterisk, there is a section where the script executes some
startup scripts, located in /etc/asterisk/startup.d
However, when you restart asterisk with core restart now or you go
ahead and kill the asterisk process, these scripts that are so
important do not get executed.
The question is:
I think I found another bug, but please let me know if there is a
workaround, since my bugs never get fixed.
In safe_asterisk, there is a section where the script executes some
startup scripts, located in /etc/asterisk/startup.d
However, when you restart asterisk with core restart now or you go
I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI-command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has
to get the offset on the result and I will try.
Yours
Philip
On Fri, Jun 29, 2012 at 6:27 AM, Thorsten Göllner t...@ovm-group.com wrote:
Am 29.06.2012 11:38, schrieb CDR:
I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI-command. I
I need to send SMS from Asterisk to an SMPP server. Is there a SMPP
channel or any other know way to send SMS via Asterisk?
I don't care if its is paid software.
Philip
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My customer needs to set a forwarding based on number of rings,i.e.,
if the phone rings 5 times (user-selectable), then try another number.
Is there a way to do such a thing with Asterisk? I could not find way
to do it based on the documentation of the Dial function. The protocol
is SIP only,
I need to add a semi-colon to a variable, but no matter how I quote
it, the parser ignores it and considers the semi-colon as the
beginning of a comment.
Si how do I concatenate the content of a variable to a semi-colon? I
tried surrounding it with double quotes, single quotes, using a
backslash
In Trunk, or earlier, is it possible to execute an AGI or any piece of
the Diaplan when a new peer registers successfully?
Venefax
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In 1.8, somebody left a message that shows up like this
Remotely bridging SIP/Client.XX.XX.XX.125-00010456 and SIP/XX.XXX.XX.X-00010457
It could be also Local Bridging
The point is that this message should not print in the console unless
the verbose level reaches some level. Never at level zero.
I am using 1.8. I need to change timerb to 6500, that is, if there is
no response of some sort in 6.5 seconds, consider the call failed and
try another route. It does not matter what do I set for the other
timers:
T1min=100
timert1=100
Timerb=6500
The command sip show settings always shows Timer
This expression that worked fine in 1.6.2 is returning an error:
exten =_X.,n,Set(i=$[${i} + 1])
It needs to add 1 to the value if i. Did I miss something?
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I encourage the developers to check this out
http://forums.asterisk.org/viewtopic.php?f=1t=77692p=161590#p161590
I am calling from behind a NAT, and there is no way to force Asterisk
to stay in the path. If the codec is the same as the outbound leg, it
always does Remote bridging, but of course,
This is turning into a political issue such as the one in Washington
and the impending default on US debt. The point is that a minor change
in the code would have a dramatic effect on security, and carry a
lower impact on CPU that using Iptables. The simplicity of the change
cannot understated.
Only way to cope with hackers would be that Digium comes to its senses
and accepts to disable any response to a REGISTER whose username is
unknown. I cannot think of a good reason why Digium finds this
proposal unacceptable, given the onslaught of hacking that we are
seeing in the industry. It
I beg to differ. Digium is hiding from the real world and somebody is
going take the software and run with it. My customers lost in excess
of $50.000 and cut my pay in half, because of hackers. The hackers
figured out how to scan every asterisk for weak passwords or open
ports, and bang them real
I need to keep out all connection from 5 countries, which originate
most of the Denial of Service attacks. The entries are
around 9000 if used as xx.xx.0.0/16. I heard that there is a smarter
way to do this by using User Tables in iptables, that will keep the
speed equal to LOG(x). I already tried
Is it my imagination of the application senddtmf does not work inside a
macro? Should I open a bug case or this is by design, and if so, what are
the grounds for that decision? I called myself and no matter what I do, I
cannot hear the tones, but if I place them inside the D(XXX) option of the
Is there an option in sip.conf for 1.6.2 that would add this to the INVITE?
Supported: ms-early-media
In my invites I see:
Supported: replaces, timer
But I have not seen any option that would add the ms-early-media option.
Here’s a link to the RFC3960 Describing the benefits of early media vs
Every time I start Asterisk or do a simple reload I see this message:
Cannot open maximum file descriptor 32767 at boot? No such file or
directory
Does anybody have some idea of what can it be? It did not happen in version
1.4.
Philip
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Every time I start Asterisk or do a simple reload I see this message:
“Cannot open maximum file descriptor 32767 at boot? No such file or
directory”.
It only works if I set 1024 in asterisk.conf maxfiles
However, my
sysctl fs.file-max
fs.file-max = 65535
and my ulimits are
ulimit -a
core file
Tilghman Lesher wrote
Not available in anything other than trunk (to be 1.8). It depends upon a
new
feature, so it's not something you can easily backport. After dialling, the
SIP code is available in ${HASH(SIP_CAUSE,channel-name)}
In a real dialplan, how do I get a variable with channel-name?
I need to access number received after a I dial a SIP or H323 call?
suppose I get one of these:
*404 Not found
**486 Busy here
**408 Request Timeout
**480 Temporarily unavailable
**480 Temporarily unavailable
**403 Forbidden (+) **
410 Gone
**301 Moved Permanently
**410 Gone **
404 Not Found (=)
I have an Asterisk with multiple IP's, on the same subnet. When a call comes
in, I need to send it back out via SIP, but need that only one IP is used as
originating IP for all calls.
For example
machines has
192.168.50.3
192.168.50.4
192.168.50.5
but when I originate the second leg of a
My dear friend Matt Riddell insists that the Manager only can dial 5 calls
per seconds, which I find ridiculous. Is there a way to prove him wrong and
have him lift the limit that has been plaguing the life of us users of
SineDialer and SmoothTorrque
Philip
--
I have a business problem that is killing me. I do SIP2SIP, only. I place a
call after receiving the incoming request, and I need to send a Hangup(Code)
to the caller, based on the result of the outbound leg. How can I do that in
Asterisk? Is that even possible at all?
I can use Hangup(code), but
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