Re: [asterisk-users] Mailing List Future
Frank Vanoni wrote: On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote: To that end, we’ve decided to discontinue the mailing lists effective February 1st, 2024. That's a very sad news! :-( Agree. Yet another giant step backward. Interesting that they will continue to send e-mails when postings to the (UGH) forum happen though. John Novack -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local calls not possible when Internet connection down
Are the phones and the server in the same subnet? You might making note of the IPs and just simply try pinging everything with the uplink disconnected. Also, if you are using domain names for registration, it is possible a dns server must be reachable. If you are using database for any of your call processing, an unreachable dns server can also be the cause of trouble. For some reason, even if you are using IP addressing, Mysql will try to resolve a connection and can hang (there is a mysql parameter to not resolve addresses). On Wed, Nov 8, 2023 at 8:46 AM Marek Greško wrote: > Hello, > > it did not seem the call hung. It seemed it never started. There was no > dialplan execution on the asterisk side. It looked like phones were > unregistered. Same shows the log posted previously. > > Marek > > > > > > Sent with Proton Mail secure email. > > --- Original Message --- > On Wednesday, November 8th, 2023 at 1:21, John Harragin < > jharra...@mw.k12.ny.us> wrote: > > > > Marek, > > > > See if calls hang in the system if you encounter another outage > > core show channels > > > > ...if so, > > core set verbose 3 > > and see what instructions subsequent calls hang on. > > > > > > > > On Mon, Nov 6, 2023 at 4:44 PM Marek Greško marek.gre...@protonmail.com > wrote: > > > > > Hello, > > > > > > sure I have local DNS server and public resolving should not be needed > for phone registrations. Running pjsip show endpojnt show the endpoints as > not in use. > > > > > > When looking into logs I see only res_pjsip_outbound_registration.c: > No response > > > received from sip provider. Nothing else. > > > > > > In phone log I see: > > > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > > > lid=0, par=0, par2=(nil)) > > > > > > The phone is Cisco SPA525G2. > > > > > > Thanks. > > > > > > Marek > > > > > > --- Original Message --- > > > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp > jc...@sangoma.com wrote: > > > > > > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško > marek.gre...@protonmail.com wrote: > > > > > > > It looks like all phones get unregistered, but I am not aware of the > cause. Why are get not registered when there is a connectivity between them > and asterisk? > > > > > > Are the REGISTER requests reaching Asterisk (do they show up in a > packet capture, do they show up in "pjsip set logger on")? It needs to be > further isolated. How are the phones configured to reach Asterisk? If using > a hostname, are they still able to resolve it? > > > > > > -- > > > Joshua C. Colp > > > Asterisk Project Lead > > > Sangoma Technologies > > > Check us out at www.sangoma.com and www.asterisk.org > > > > > > -- > > > _ > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > > > > > New to Asterisk? Start here: > > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > > > New to Asterisk? Start here: > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local calls not possible when Internet connection down
Marek, See if calls hang in the system if you encounter another outage core show channels ...if so, core set verbose 3 and see what instructions subsequent calls hang on. On Mon, Nov 6, 2023 at 4:44 PM Marek Greško wrote: > > Hello, > > sure I have local DNS server and public resolving should not be needed for > phone registrations. Running pjsip show endpojnt show the endpoints as not in > use. > > When looking into logs I see only res_pjsip_outbound_registration.c: No > response > received from sip provider. Nothing else. > > In phone log I see: > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > lid=0, par=0, par2=(nil)) > > The phone is Cisco SPA525G2. > > Thanks. > > Marek > > > > --- Original Message --- > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp > wrote: > > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško > wrote: >> >> It looks like all phones get unregistered, but I am not aware of the cause. >> Why are get not registered when there is a connectivity between them and >> asterisk? > > > Are the REGISTER requests reaching Asterisk (do they show up in a packet > capture, do they show up in "pjsip set logger on")? It needs to be further > isolated. How are the phones configured to reach Asterisk? If using a > hostname, are they still able to resolve it? > > -- > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Deleting voicemail by program
Here is something I wrote years ago. I expect you can adjust it for your
needs
# cat remove_blank_vmail
#!/bin/bash
# remove_blank_vmail takes arguments as voicemail boxes and removes
messages with audio files shorter then MINSIZE (in bytes)
#--
# Description:
# Author: John Harragin Monroe-Woodbury CSD
# Created at: Thu Nov 6 12:27:35 EST 2008
#
# Copyright: None. Modify and use however you like...
#
#--
# Configure section:
BASEDIR=/var/spool/asterisk/voicemail/default/ # default
context
MINSIZE=12000 # 1.5
seconds
#--subroutines:
ProcessDir () {
lastfile=""
delcnt=0
for file in $(ls -A ${msgdir}/msg*.txt 2>/dev/null); do # the
redirect supresses msg when dir empty
if [ $(stat --format=%s ${file/.txt/.wav}) -lt ${MINSIZE} ]; then
rm ${file/.txt/.*}
let delcnt++
fi
lastfile=${file}
done
if [ $delcnt -gt 0 ]; then echo "$delcnt short messages deleted from
${msgdir}"; fi
partfilename=${lastfile/*\/msg/} # get number
from file name
highest=${partfilename/.txt/}
while [[ $highest = 0* ]]; do highest=${highest#0}; done # bash does
not like leading zeros
if [ ${#highest} -eq 0 ]; then highest=0; fi # ...or
blanks for math
realcount=0
for ((x=0;x<=${highest};x+=1)); do
chkname=msg$(printf "%04d" $x) # build name
- pad with zeros...
if [ -e ${msgdir}/${chkname}.txt ]; then
if [ $realcount -ne $x ];then
newname=msg$(printf "%04d" $realcount)
for idivfile in $(ls -A ${msgdir}/${chkname}.*); do
mv ${idivfile} ${msgdir}/${newname}.${idivfile/*\/*./}
done
fi
let realcount++
fi
done
}
#--main:
for ext in "$@"; do
if [ -d ${BASEDIR}${ext} ];then
for msgdir in $(ls -d ${BASEDIR}${ext}/*); do
ProcessDir ${msgdir}
done
else
echo "${BASEDIR}${ext} is not a valid directory"
fi
echo "Processed extension $ext"
done
On Mon, Oct 9, 2023 at 3:06 PM Mike Diehl wrote:
> Hi all,
>
> I need to be able to delete a voicemail message using a program.
>
> Is is sufficient to simply delete the .wav and .txt files in the spool
> directory?
> Or do I need to also renumber the remaining files?
>
> For example, let say a given mailbox has 20 messages in it and I want to
> delete message number 5. Can I just delete the 2 files and expect that
> asterisk will renumber them? Or do I need to?
>
> Also, is the answer the same when I migrate to storing voicemails in a
> database?
>
> Thanks in advance.
>
> Mike
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Segmentation fault
>Setting "batch=yes" in your /etc/asterisk/cdr.conf file may fix this by moving the database interaction to separate threads away from the call. I can certainly try this, but I believe the maria-odbc-driver is what is running as a single thread. Its been a few months since I played with this, I think there may also be a compiled in parameter in the unixodbc regarding threading that's default value was changed in recent years. The only asterisk issue is why it would hang at SET(CDR_PROP(disable)=1)? Perhaps there is an interaction where odbc_adaptive always checks an aspect of the tables it interacts with (and needlessly when disabling cdr). I started looking at the source - but got pulled onto another project before I was able to determine anything, and have not revisited the issue. So for me this isn't current but just me throwing in my recollections that may pertain to Federico's issue. On Thu, Aug 17, 2023 at 5:37 PM C. Maj wrote: > On 8/17/23 12:44, John Harragin wrote: > > You should be able to define multiple data sources. However I'm having my > > own issues. I have my dialplan accessing one maria database which is > hosted > > locally on the asterisk server then logging cdr with odbc adaptive which > > connects to maria on a remote machine. This works fine except when the > > remote server is out of reach the calls zombie and eventually lead to a > > fault. > > Setting "batch=yes" in your /etc/asterisk/cdr.conf file may fix this by > moving the database interaction to separate threads away from the call. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmentation fault
You should be able to define multiple data sources. However I'm having my own issues. I have my dialplan accessing one maria database which is hosted locally on the asterisk server then logging cdr with odbc adaptive which connects to maria on a remote machine. This works fine except when the remote server is out of reach the calls zombie and eventually lead to a fault. The cli imply that the calls hang on SET(CDR_PROP(disable)=1) which is ironic since instructing the system not to connect to the cdr server appears to cause the problem and there may be a defect in the adaptive_odbc implementation where it somehow still communicates with the server during the cdr_func(disable). My experimentation suggests that the maria_odbc driver utilizes a single thread (there are odbc directives to adjust this - but it seems they have been superseded). Once a cdr hangs, new calls hang on odbc calls to the local database. In my case the remote server is 5' away and in the same switch that Asterisk is plugged into, so it hasn't been critical to fix this - but it may provide some clues to your situation. On Thu, Aug 17, 2023 at 11:04 AM Federico wrote: > I tested this issue with version 13 and version 18. > > In res_odbc.conf, if I add a second, new data source like > > > > [asterisk] > > enabled=yes > > dsn=asterisk > > sanitysql => select 1 > > isolation => read_committed > > username=root > > ;password= > > pre-connect => yes > > forcecommit => yes > > connect_timeout => 10 > > negative_connection_cache => 0 > > max_connections =>500 > > > > my odbc.ini > > [cdr] > > Description = MySQL ODBC Driver Testing > > Driver = maria > > Socket = /var/run/mysqld/mysqld.sock > > User = root > > Password = > > Database = public > > Option = 3 > > > > > > I get, immediately, segmentation fault. > > With only one, it works fine. > > Is this by design? > > > > Philip > > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem getting dahdi-linux to work with kernel 6.1.0-10
Hi. I have run into a problem compiling dahdi-linux in kernel 6.1.0-10. Apparently there was a change, so I found a patch to fix stdbool.h but now I have an implicit declaration of pci_alloc_consistent in drivers/dahdi/wct4xxp/base.c I don't see any other references to that name anywhere. I am using version from git 5c840cf43838e0690873e73409491c392333b3b8 . So, the question, how to fix, so I can get the tompile to work? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voip.ms ( was Re: Problems solved )
aster...@phreaknet.org wrote: voip.ms is also the only major VoIP provider that supports IAX2, so if you do anything else you'll probably have to use SIP. voip.ms works so well and is certainly affordable, so why would anyone want to use anything else? Even my really cheap friends use it! I have used it as my PSTN provider for more than 10 years, with only one hacking issue with voip.ms, which they fixed fairly quickly. I see no reason to change to a protocol that ( it seems ) every thief in the world is banging away on 24/7!! JMO John Novack On 5/27/2023 10:23 AM, Steve Matzura wrote: -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Broken link in LICENSE file
https://github.com/asterisk/asterisk/blob/master/LICENSE#L48 broken (PS I hope I never find a bug to report, because I don't use Github... embrace, extend, extinguish is still alive and well) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk simply stops call processing
I've been having a related problem. I have Asterisk with some call
processing accessing Maria (hosted on the phone server, running Ubuntu) via
func_odbc. That same odbc driver is used to write cdr records on a
different server. I had never noticed a problem (and no threading attribute
defined) until after I did a system update several months ago.
Now if the ethernet cable is disconnected to the cdr server, call
processing then hangs when func_odbc trys to access the locally hosted
(same machine as asterisk) call process database. The zombied channels then
accumulate.
In my research, I read that the default threading value was changed in
unixodbc to assume that threading would be handled by the individual
odbc-drivers - rather than the odbc framework. Also, I read that unixodbc
has to be compiled with a threading directive set to yes for the
odbcinst.ini key-value to have any effect.
Anyway I am suspecting that the ubuntu unixodbc package is now compiled
without threading enabled.
This is happening on a production machine, so I am somewhat limited in when
& how much experimentation I can do. One thing I'd like to try is to
redefine the maria driver as maodbc-cdr in odbcinst.ini and see if it
exists in it's own thread?
root@phone:~# cat /etc/odbc.ini
[cdr-bmaria]
Driver = maodbc
DATABASE= cdr
DESCRIPTION = MariaDB ODBC to remote-cdr-database
SERVER = 192.168.1.11
UID = cdr-reporter
PASSWORD= secret
PORT= 3306
[call-process-maria]
Driver = maodbc
DATABASE= phone
DESCRIPTION = MariaDB ODBC local (to self)
SERVER = 192.168.2.22
UID = dialplan-user
PASSWORD= secret
PORT= 3306
root@phone:~# cat /etc/odbcinst.ini
[maodbc]
Driver64= /usr/local/lib64/mariadb/libmaodbc.so
Description = MariaDB ODBC Connector
Threading = 2
!! The proposed addition: - also changing the cdr-maria
conection key to Driver=maodbc-cdr
[maodbc-cdr]
Driver64= /usr/local/lib64/mariadb/libmaodbc.so
Description = MariaDB ODBC Connector
Anthony,
...anyway, enough about my problems. Have you put a:
Verbose(0, Your built out sql statement)
...before your ODBC application in both contexts to see if you just have
maybe an undefined variable creating a syntax error in your sql?
John
Here is a bit about odbc threads:
https://stackoverflow.com/questions/4207458/using-unixodbc-in-a-multithreaded-concurrent-setting
On Tue, Feb 28, 2023 at 9:02 AM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
> On Wednesday 22 February 2023 at 15:29:38, John Harragin wrote:
>
> > If there are multiple connections that the utilize the same driver, try
> > putting:
> >
> > Threading = 2
> >
> > in the appropriate driver section of
> > /etc/odbcinst.ini
>
> I'll give that a go, however I doubt that it is the problem, since I see
> the
> correct result from the ODBC query recorded in the assignment verbose log
> output, therefore the query is done and the result has been used by the
> time
> Asterisk freezes.
>
> > ...this would be a possibility if the problem is intermittent.
>
> It's actually extremely repeatable - I have not seen call processing
> proiceed
> beyond this stage once so far.
>
> > Also can you successfully execute the same SQL from the cli?
>
> Yes, and as I say, they query is working fine and Asterisk is correctly
> using
> the returned value in the assignment.
>
> The further detail which I think I added in a later post is that this is
> actually in a context which gets called using a Gosub() from two different
> places in the dialplan.
>
> From one, it works fine; from the other, it gets stuck. Completely
> consistent.
>
> > By the way, what driver is asterisk using?
>
> You mean ODBC? That's connected to MariaDB.
>
> > On Mon, Feb 20, 2023 at 11:12 PM Antony Stone wrote:
> > > Hi.
> > >
> > > I have a strange problem and I'm looking for suggestions on how to
> > > investigate it.
> > >
> > > I have a dialplan which is processing a call, and Asterisk simply stops
> > > doing anything for that call.
> > >
> > > I have verbose and debug logging turned on.
> > >
> > > There are two steps at a particular point in the dialplan:
> > > Set(UserCredit=${ODBC_GENERIC(select Credit('${DDI}'))})
> > >
> > > Verbose(6,Current credit level for user ${DDI} is ${UserCredit}
> > > pence)
> > >
> > >
> > > Everything gets processed up to and including the first line - the
> > > verbose log file shows me:
> > >
> > > pbx.c:2946 in pbx_extension_helper: Executing
> > > [0044509903@Dia
Re: [asterisk-users] Asterisk simply stops call processing
If there are multiple connections that the utilize the same driver, try
putting:
Threading = 2
in the appropriate driver section of
/etc/odbcinst.ini
...this would be a possibility if the problem is intermittent.
Also can you successfully execute the same SQL from the cli?
By the way, what driver is asterisk using?
On Mon, Feb 20, 2023 at 11:12 PM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
> Hi.
>
> I have a strange problem and I'm looking for suggestions on how to
> investigate
> it.
>
> I have a dialplan which is processing a call, and Asterisk simply stops
> doing
> anything for that call.
>
> I have verbose and debug logging turned on.
>
> There are two steps at a particular point in the dialplan:
>
>
> Set(UserCredit=${ODBC_GENERIC(select Credit('${DDI}'))})
>
> Verbose(6,Current credit level for user ${DDI} is ${UserCredit}
> pence)
>
>
> Everything gets processed up to and including the first line - the verbose
> log
> file shows me:
>
> pbx.c:2946 in pbx_extension_helper: Executing [0044509903@DialBleg:46]
>
> Set("SIP/TrunkTwo-1184", "UserCredit=999") in new stack
>
> (Phone number obscured here for anonymity).
>
> Then, that is it. Nothing further happens with call processing (until one
> of
> the parties hangs up) and the second dialplan command above never appears
> in
> the verbose log file. I have several other Verbose(6,.) commands
> preceding
> this, and they all output into the log file as expected.
>
> If another call arrives on the same server, Asterisk quite happily starts
> processing it and records what it's doing in the log files.
>
>
> Can anyone suggest how I can investigate what Asterisk is doing at the
> point
> where it "gets stuck", and how to find out why it simply stops processing
> the
> call and doesn't continue with the dialplan commands?
>
>
> Thanks,
>
>
> Antony.
>
> --
> "The future is already here. It's just not evenly distributed yet."
>
> - William Gibson
>
>Please reply to the
> list;
> please *don't* CC
> me.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] github - mlan
If you clone one of their repo's you can see their email address in the commit log... On Tue, 7 Feb 2023 at 16:56, Jeff LaCoursiere wrote: > Hi all, > > Curious if the github user "mlan" is on this list? Could you please > contact me off list if so, I was hoping to reference your work in a talk > at Astricon next week, and... I don't know how to contact github users lol. > > Cheers, > > -- > Jeff LaCoursiere > StratusTalk, Inc. > 703 496 4990 x108 > 815 546 6599 cell > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Global variables in global variables
You have posted the same message several times in the last few days!!
I would assume no one has an answer to your question, at least on this list.
It seems most have migrated to another (UGH!) venue, so the few that are left
can't help.
JMO
John Novack
Antony Stone wrote:
Hi.
I have a very old dialplan (ie: a dialplan for a very old version of Asterisk)
which I've just transferred to Asterisk 16.28.0
The [globals] section of that dialplan includes:
Kphones=SIP/KC470IP/KSnom870
Sphones=SIP/SYealinkT38G/SGC610IP
Allphones=${Kphones}&${Sphones}
In the old system, this results in ${Allphones} containing:
SIP/KC470IP/KSnom870/SYealinkT38G/SGC610IP
I can use this in a dial() command.
On the new system, the variable ${Allphones} ends up containing:
${Kphones}&${Sphones}
(ie: the unexpanded variable names, not the content of those previously-
defined variables.)
This fairly obviously does not work in a dial() command.
a) is this a deliberate backward incompatiblity at some stage in the
development of Asterisk?
b) if not, is this a known bug?
c) is there some other way I'm supposed to be doing this now, to be able to
define a global variable including the value of another global variable?
d) if not, is there some workaround?
Thanks,
Antony.
--
Dog is my Co-Pilot
--
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Re: [asterisk-users] cannot load res_geolocation.so
I have had a similar problem. I think geolocation introduced some additional prerequisites run: /usr/src/asterisk-X/contrib/scripts/install_prereq test then recompile asterisk That script installs a bunch of crap you don't need, but running it in test mode rather than install might help you determine what specific additional packages are. I'm running Ubuntu. For me the troubles showed up about 18.7.0 and that script did get me going, but messed up my minimalist installation. On Tue, Dec 6, 2022 at 8:35 AM Joshua C. Colp wrote: > > On Mon, Dec 5, 2022 at 4:31 PM Nick Olsen wrote: >> >> Hello, >> >> On a fresh install of 18.9 Cert2 (Or the latest 19 if I recall the previous >> version I tried. >> >> PJSIP fails to load properly. It seems that the new res_geolocation module >> fails to load. But I can't seem to figure out why. And being that it's a >> fairly new module (So it seems) google-fo isn't being very helpful. This is >> running on Debian 11 and a freshly compiled install with only "make samples" >> done to create the config files. Any help would be appreciated! >> >> newasterisk*CLI> module load res_pjsip.so >> Unable to load module res_pjsip.so >> Command 'module load res_pjsip.so ' failed. >> [Dec 5 15:26:18] ERROR[2420]: loader.c:283 module_load_error: res_pjsip >> loaded before dependency res_geolocation! >> >> newasterisk*CLI> module load res_geolocation.so >> Unable to load module res_geolocation.so >> Command 'module load res_geolocation.so ' failed. >> [Dec 5 15:26:28] WARNING[2420]: config_options.c:1102 >> xmldoc_update_config_type: Cannot update type 'location' in module >> 'res_geolocation' because it has no existing documentation! >> [Dec 5 15:26:28] ERROR[2420]: res_geolocation/geoloc_config.c:672 >> geoloc_config_load: Failed to register geoloc location object with sorcery > > > This would mean that the documentation isn't in the core-en_US.xml file, > normally located in the /var/lib/asterisk/documentation directory. I just > built 18.9-cert3 and it is definitely there for me: > > jcolp@kappa:~/development/asterisk/public [certified/18.9-cert3| …2⚑ 4]> > grep "geolocation" /var/lib/asterisk/documentation/core-en_US.xml > res_geolocation > > res_geolocation > > > > xpointer="xpointer(/docs/configInfo[@name='res_geolocation']/configFile[@name='geolocation.conf']/configObject[@name='location']/configOption[@name='format'])"/> > xpointer="xpointer(/docs/configInfo[@name='res_geolocation']/configFile[@name='geolocation.conf']/configObject[@name='location']/configOption[@name='location_info'])"/> > xpointer="xpointer(/docs/configInfo[@name='res_geolocation']/configFile[@name='geolocation.conf']/configObject[@name='location']/configOption[@name='confidence'])"/> > xpointer="xpointer(/docs/configInfo[@name='res_geolocation']/configFile[@name='geolocation.conf']/configObject[@name='location']/configOption[@name='location_source'])"/> > xpointer="xpointer(/docs/configInfo[@name='res_geolocation']/configFile[@name='geolocation.conf']/configObject[@name='location']/configOption[@name='method'])"/> > Get or Set a field in a geolocation profile > This geolocation profile will be applied to all calls received > This geolocation profile will be applied to all calls received > > And the module loads fine: > > *CLI> module show like geolocation > Module Description Use > Count Status Support Level > res_geolocation.so res_geolocation Module for Asterisk 2 > Running core > res_pjsip_geolocation.so res_pjsip_geolocation Module for Asteris 0 > Running core > 2 modules loaded > > Did you build Asterisk putting things in other directory locations? Is there > an old core-en_US.xml file somewhere? > > -- > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgraded from asterisk 18.14.0 to 20.0.0 and inbound registration(?) is now failing
I had similar issues. It looks like modules related to pjsip (geolocation?) introduced new prerequisites. There is a script in the source that prepares for an asterisk build. Try running that, then recompile asterisk and see if that fixes things. John On Fri, Dec 2, 2022 at 3:36 PM Justin Piszcz wrote: > > Hello, > > I have been using asterisk for the past decade and never had an issue with > upgrades until now. Recently, in November I upgraded from 18.14.0 to 20.0.0 > and afterwards my SPA3102 can no longer register with asterisk. I have not > made any asterisk or SPA3102 configuration changes in ~1-2 years. > > asterisk versions: (old -> new) > 18.14.0~dfsg+~cs6.12.40431414-1+b1 > 20.0.0~dfsg+~cs6.12.40431414-2 > > An example of the log from the SPA3102 under asterisk (succeeds) 18 vs. > asterisk 20 (fails), kindly inquiring what I may have missed that is causing > these failures? > > asterisk18_sip_success.txt (inbound call success) from the SPA3102 (with > asterisk 18 installed) > Dec 1 17:34:55 system1 local3 fs: 11707:11782:65536 > Dec 1 17:34:55 system1 local3 fls: af:1:0:0 > Dec 1 17:34:55 system1 local3 fbr: 0:3000:3000:03d6a:0008:0007:5.1.10(GW) > Dec 1 17:34:55 system1 local3 fhs: 01:0:0001:upg:app:0:3.3.6(GW) > Dec 1 17:34:55 system1 local3 fhs: 02:0:0002:upg:app:1:3.3.6(GW) > Dec 1 17:34:55 system1 local3 fhs: 03:0:0003:upg:app:2:3.3.6(GW) > Dec 1 17:34:55 system1 local3 fhs: 04:0:0004:upg:app:0:5.1.10(GW) > Dec 1 17:34:55 system1 local3 fhs: 05:0:0005:upg:app:1:5.1.10(GW) > Dec 1 17:34:55 system1 local3 fhs: 06:0:0006:upg:app:2:5.1.10(GW) > Dec 1 17:34:56 system1 local3 fu: 0:3d91, 0003 0001 > Dec 1 17:35:19 system1 local2 FXO: Start CNDD > Dec 1 17:35:21 system1 local2 FXO: CNDD name=11234567890, number=1234567890 > Dec 1 17:35:21 system1 local2 FXO: Stop CNDD > Dec 1 17:35:21 system1 local3 FXO: CNDD Name=11234567890 Phone=1234567890 > Dec 1 17:35:22 system1 local2 AUD: Stop PSTN Tone > Dec 1 17:35:22 system1 local2 AUD: Stop PSTN Tone > Dec 1 17:35:22 system1 local2 Calling: 1...@system1.int.com:0 > Dec 1 17:35:22 system1 local2 [1:0]AUD ALLOC CALL (port=16458) > Dec 1 17:35:22 system1 local2 [1:0]RTP Rx Up > Dec 1 17:35:22 system1 local2 CC: pc(0)=18 not in codec list > Dec 1 17:35:22 system1 local2 [0:0]AUD ALLOC CALL (port=16460) > Dec 1 17:35:22 system1 local2 [0:0]RTP Rx Up > Dec 1 17:35:22 system1 local2 CC: Ringback > Dec 1 17:35:22 system1 local2 [1:0]RTP Rx Dn > Dec 1 17:35:22 system1 local2 AUD: Play PSTN Tone 9 > Dec 1 17:35:23 system1 local3 IDBG: sc-0 > Dec 1 17:35:23 system1 local3 IDBG: rs:10 > Dec 1 17:35:26 system1 local3 IDBG: sc-0 > Dec 1 17:35:26 system1 local3 IDBG: rs:8 > Dec 1 17:35:32 system1 local2 AUD: Stop PSTN Tone > Dec 1 17:35:32 system1 local3 FXO: On Hook > Dec 1 17:35:32 system1 local2 AUD: Stop PSTN Tone > Dec 1 17:35:32 system1 local2 FXO: Stop CNDD > Dec 1 17:35:32 system1 local3 [0]FM Alert Stop RxTx (c=002550b0;a=0) > Dec 1 17:35:32 system1 local2 [1:0]AUD Rel Call > Dec 1 17:35:32 system1 local3 [0]FM Alert Stop RxTx (c=0024e5e8;a=0) > Dec 1 17:35:32 system1 local2 [0:0]AUD Rel Call > Dec 1 17:35:32 system1 local2 CC: Ended > > > asterisk20_sip_error.txt (inbound call failure) from the SPA3102 (with > asterisk 20 installed) > Dec 1 17:23:21 system1 local2 FXO: Start CNDD > Dec 1 17:23:23 system1 local2 FXO: CNDD name=11234567890, number=1234567890 > Dec 1 17:23:23 system1 local2 FXO: Stop CNDD > Dec 1 17:23:23 system1 local3 FXO: CNDD Name=11234567890 Phone=1234567890 > Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone > Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone > Dec 1 17:23:24 system1 local2 Calling: 1...@system1.int.com:0 > Dec 1 17:23:24 system1 local2 [1:0]AUD ALLOC CALL (port=16418) > Dec 1 17:23:24 system1 local2 [1:0]RTP Rx Up > Dec 1 17:23:24 system1 local2 [1]SIP:ICMP Error -1 (a01:5060, 2) > Dec 1 17:23:24 system1 local3 RSE_DEBUG: getting alternate from > domain:system1.int.com > Dec 1 17:23:24 system1 local3 [0]FM Alert Stop RxTx (c=002550b0;a=0) > Dec 1 17:23:24 system1 local2 [1:0]AUD Rel Call > Dec 1 17:23:24 system1 local2 CC: Failed w/ Calling > Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone > Dec 1 17:23:24 system1 local2 AUD: Stop PSTN Tone > Dec 1 17:23:39 system1 local2 AUD: Stop PSTN Tone > Dec 1 17:23:39 system1 local3 FXO: On Hook > Dec 1 17:23:39 system1 local2 AUD: Stop PSTN Tone > Dec 1 17:23:39 system1 local2 FXO: Stop CNDD > > Regards, > Justin > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community for
[asterisk-users] menuselecting res_corosync
Trivial issue. I have a script to rebuild asterisk with the following line: menuselect/menuselect --disable MENUSELECT_MOH --disable CORE-SOUNDS-EN-GSM --enable CORE-SOUNDS-EN-WAV --enable app_macro --enable codec_opus --enable chan_phone --enable chan_sip --enable chan_sip --enable chan_sip --enable chan_oss --enable chan_phone --enable format_ogg_speex --enable res_corosync --enable format_vox --disable-category ME NUSELECT_AGIS --disable-category MENUSELECT_CORE_SOUNDS --disable-category MENUSELECT_MOH Everything compiles as desired except res_corosync. I then can run: make menuselect ...and manually select res_corosync, and the module compiles without incident. Does the order of enables matter? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk IP PBX VoIP Servers Hacked by Hackers
I am using freepbx latest 16 version -- am I subject to this problem? I am not using elastics, but I installed on a Debian bullseye server, so this is of definite concern to me. Thanks. On Mon, 18 Jul 2022 06:45:41 -0400, Joshua C. Colp wrote: > > [1 ] > [1.1 ] > On Mon, Jul 18, 2022 at 7:43 AM Turritopsis Dohrnii Teo En Ming < > c...@teo-en-ming.com> wrote: > > > > > Dear Joshua Colp, > > > > Noted with thanks. So the vulnerability is not related to the Asterisk > > open source project at all? > > > > It is not. The vulnerability mentioned is regarding FreePBX and Elastix, > which do use Asterisk but the vulnerability has nothing to do with Asterisk > itself. > > -- > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > [1.2 ] > [2 ] > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to detect which confbridge user is talking or muted
Hi. Is there a way in confbridge where I can enquire if a channel is muted, or if the channel is talking? There seems to be nothing except ami events, but I would just like to check a channel to see if he is talking or muted at a particular time and display that information on the console. I have been using meetme and there you can just display the list of users and you get that information. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] a couple of problems with confbridge
OK, thanks, that is what I was hoping for. On Fri, 01 Jul 2022 12:02:46 -0400, Dan Cropp wrote: > > I believe the answer #2 depends on the user options for each participant. > > If all participants have user options with wait for marked set to true there > will be no conference/recording until at least one marked user joins. > If any participants have user options with wait for marked set to false, when > they join the conference bridge it is actually going. Thus, if the bridge > options had the record enabled it would start recording. > If only marked user joins first, it's met the criteria and will conference > and start recording. > > Dan > > -Original Message- > From: asterisk-users On Behalf Of > John Covici > Sent: Tuesday, June 28, 2022 6:28 PM > To: asterisk-users@lists.digium.com > Subject: [External] [asterisk-users] a couple of problems with confbridge > > Hi. I have been using meetme for years, but I wanted to try > confbridge as meetme is going away soon.I am having a few > problems/questions doing this. > > 1. When I list the confbridge users in a bridge, I only get the caller id > number -- I have a number of contacts in contact manager and I am using > superfecta, but the name does not appear. I do need the name to see who is > on there. > > 2. I will be using a conference with a marked user -- and I would like to > record the conference -- when does the recording start -- when the first user > comes on or when the marked user joins? > > 3. In the sample file it says you cannot have more than one user profile on > a bridge, but I need two, one for the marked user and another one for regular > users -- how do I work around this? > > Thanks in advance for any suggestions. > > > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > cov...@ccs.covici.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a couple of problems with confbridge
On Tue, 28 Jun 2022 19:54:11 -0400, Joshua C. Colp wrote: > > [1 ] > On Tue, Jun 28, 2022 at 8:28 PM John Covici wrote: > > > Hi. I have been using meetme for years, but I wanted to try > > confbridge as meetme is going away soon.I am having a few > > problems/questions doing this. > > > > 1. When I list the confbridge users in a bridge, I only get the > > caller id number -- I have a number of contacts in contact manager and > > I am using superfecta, but the name does not appear. I do need the > > name to see who is on there. > > > > You'll need to be specific on how you are listing. The AMI action provides > all of the information. ... I was using the confbridge list command from the console and that only gives the number -- any way to fix or is there some other way I could get this information on the console? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] a couple of problems with confbridge
Hi. I have been using meetme for years, but I wanted to try confbridge as meetme is going away soon.I am having a few problems/questions doing this. 1. When I list the confbridge users in a bridge, I only get the caller id number -- I have a number of contacts in contact manager and I am using superfecta, but the name does not appear. I do need the name to see who is on there. 2. I will be using a conference with a marked user -- and I would like to record the conference -- when does the recording start -- when the first user comes on or when the marked user joins? 3. In the sample file it says you cannot have more than one user profile on a bridge, but I need two, one for the marked user and another one for regular users -- how do I work around this? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GET DATA on AGI
I thought one of the arguments to the read command was the terminator, is that the command you have in your agi? On Sun, 27 Feb 2022 12:26:50 -0500, Tom Ray wrote: > > [1 ] > [1.1 ] > I believe that # in the default terminator for GET DATA and I don’t think > that can be disabled. But I’m not a 100% as I’ve always used # as the > terminator. > > > > From: asterisk-users On Behalf Of > Dovid Bender > Sent: Sunday, February 27, 2022 11:01 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [asterisk-users] GET DATA on AGI > > > > Hi, > > > When using GET DATA in an AGI it seems that the # key ends the input. So if > say I want the user to input 123#456 the system will return 123. I did not > see this in the documentation. Is this a bug, lack of documentation or do I > have a bug in my AGI? > > > > TIA. > > > > Dovid > > > > [1.2 ] > [2 ] > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] a few confbridge questions
Hi. I am using meetme application and I am interested in switching to confbridge, but there seems to be no way to do certain things in the dialplan with confbridge. How would I get the number of users in a particular conference? I want the leader to only start the recording when there are sufficient participants, which I will give him in an ivr. How would I increase or decrease the volume for a particular user in a conference? I can do these things using meetme, so I don't want to lose functionality when going to confbridge. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] strange sound on conference call
Hi. I am having a problem with a conference call on my server which a vps in the cloud. I am using chan_sip and meetme. What I get is a bit of a staticy or robotic sound, but it goes away if the user lowers the volume a bit which we can do with *4 in meetme. So, is the problem with the chan_sip, meetme or something else entirely? Nothing relevant in the logs. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to escape the & in BackGround
I have been using system commands in my dialplan for years and the &
goes through and puts the process in background like it should,
asterisk does not do anything, so you are left with what the shell
does.
On Thu, 27 Jan 2022 17:48:46 -0500,
Dovid Bender wrote:
>
> [1 ]
> [1.1 ]
> I tried tinyURL and that did not work. I got an error of:
> file.c:789 ast_openstream_full: File https://tinyurl.com/bdfye5ts9 does not
> exist in any format (URL changed to hide aws key). I tried adding
> \;foo=wav. but that did not work either.
>
>
> On Thu, Jan 27, 2022 at 3:32 PM Kingsley Tart wrote:
>
> > Does asterisk follow HTTP redirects? If so can you use something like
> > tinyurl.com to produce an alternative URL?
> >
> > Or, base64 encode the URL, and then set a variable with
> > Set(url=${BASE64_DECODE(${encodedURL})) ?
> >
> > Cheers,
> > Kingsley.
> >
> > On Wed, 2022-01-26 at 16:56 -0500, Dovid Bender wrote:
> > > I tried but it seems it does not.
> > >
> > >
> > > On Tue, Jan 18, 2022 at 2:57 PM John Runyon
> > > wrote:
> > > > ${SPRINTF(%c,38)}
> > > > or
> > > > %26
> > > >
> > > > should work, I think.
> > > >
> > > > On Sun, 16 Jan 2022 at 13:21, Dovid Bender
> > > > wrote:
> > > > > Hi,
> > > > >
> > > > > I am trying to play a sound file from AWS S3. The URL is
> > > > > something like this http://example.org?foo=bar=b. The issue
> > > > > seems to be that as soon as Asterisk see's the & it assumes there
> > > > > is a new file and the a=b is not sent along. I tried doing \& but
> > > > > that did not work. Does anyone know a way of telling Asterisk
> > > > > that & is part of the URL and to pass it along as a string?
> >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > Check out the new Asterisk community forum at:
> > https://community.asterisk.org/
> >
> > New to Asterisk? Start here:
> > https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> [1.2 ]
> [2 ]
> --
> _
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>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
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>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici wb2una
cov...@ccs.covici.com
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Re: [asterisk-users] How to escape the & in BackGround
${SPRINTF(%c,38)}
or
%26
should work, I think.
On Sun, 16 Jan 2022 at 13:21, Dovid Bender wrote:
> Hi,
>
> I am trying to play a sound file from AWS S3. The URL is something like
> this http://example.org?foo=bar=b. The issue seems to be that as soon
> as Asterisk see's the & it assumes there is a new file and the a=b is not
> sent along. I tried doing \& but that did not work. Does anyone know a way
> of telling Asterisk that & is part of the URL and to pass it along as a
> string?
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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Re: [asterisk-users] asterisk and maybe a freepbx question
OK, that tells me something, I will disable pjsit for now, learn about it and try again. On Sun, 09 Jan 2022 06:39:55 -0500, John Harragin wrote: > > [1 ] > [1.1 ] > You can also set up multiple physical or vlan(ed) interfaces and bind sip > to one and pjsip to the other - then you have to set up the appropriate > interface routing too for both inbound and outbound packets which takes a > good understanding of your network topology and the locations of your > respective devices. You might be able to do it with multiple addresses on > your interface too (although I haven't tried it). > > All of the packets have to be presented to the appropriate channel > otherwise get discarded. You can't set it up so if a packet is from a > device not registered with pjsip, it gets passed to chan_sip to try. > > For me, I had both channel types running on production machines while I > migrated to pjsip or when not being able to figure out how to set up some > property in pjsip that you had running in sip. Each time I've had to do > this, eventually I was able get it all running within pjsip. I also already > had multiple vlans configured for my servers (with voip exclusive to one). > > The short story is that it is easier to learn how to get things working > within pjsip than learning the tricky networking setup. > > > On Sun, Jan 9, 2022 at 2:49 AM Duncan Turnbull > wrote: > > > > > > > > > > > > On 9/01/2022, at 7:11 PM, John Covici wrote: > > > > > > On Sat, 08 Jan 2022 19:17:57 -0500, > > > Antony Stone wrote: > > >> > > >>> On Sunday 09 January 2022 at 00:50:27, John Covici wrote: > > >>> > > >>> Hi. I am using asterisk 18.3 and freepbx. > > >> > > >> Hm, which version of FreePBX uses Asterisk 18.3? > > >> > > >>> How can both sip and pjsip be listening at port 5060 at the same time > > >> > > >> They can't. > > >> > > >> One might be on TCP and the other on UDP, but you can't have them both > > >> listening on the same port with the same protocol. > > > > In freepbx you enable chan sip or pjsip or both and set what ports they use > > > > The choices are either in advanced settings or sip settings > > > > Disable pjsip and reset the chan_sip port to 5060 or use pjsip. With them > > both enabled sometimes odd things happen but it will still work. You will > > get lots of error messages though > > > > > > >> > > >>> for instance I get: > > >>> > > >>> [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: > > >>> > > SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity=" > > >>> > > Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="20 > > >>> 25076022",LocalAddress="IPV4/UDP/166.84.7.53/5060 > > ",RemoteAddress="IPV4/UDP/ > > >>> 45.134.144.118/5823 > > ",ACLName="registrar_attempt_without_configured_aors" > > >> > > >> What makes you think chan_sip and pjsip are both listening on UDP 5060? > > >> > > >>> I would like pjsit not to listen,till I figure out how to configure > > >>> the thing, so my logs don't fill up with messages. > > >>> > > >>> Thanks in advance for any suggestions. > > >> > > >> As far as I recall using FreePBX, there is a selector for the SIP > > protocol to > > >> tell it whether you want it to use pjsip or chan_sip. I don't think it > > even > > >> supports using both at the same time, so simply make sure that is set > > to > > >> chan_sip and you should be fine. > > >> > > >> On the other hand, why do you need to learn "how to configure the > > thing" if > > >> you're using FreePBX? Part of the whole point is that it does the > > fiddly > > >> techie sutff in the background for you, and you just need to use the > > personnel- > > >> department-friendly web GUI. > > > > > > This is what I thought as well, I just generated one trunk using the > > > old chan_sip and expected nothing from pjsit, yet I get all kinds of > > > errors like > > > [2022-01-08 17:08:59] WARNING[487628] res_pjsip_registrar.c: Endpoint > > > 'anonymous' (45.134.144.118:5823) has no configured AORs > > > > > > so I am very confused a
Re: [asterisk-users] asterisk and maybe a freepbx question
On Sat, 08 Jan 2022 19:17:57 -0500, Antony Stone wrote: > > On Sunday 09 January 2022 at 00:50:27, John Covici wrote: > > > Hi. I am using asterisk 18.3 and freepbx. > > Hm, which version of FreePBX uses Asterisk 18.3? > > > How can both sip and pjsip be listening at port 5060 at the same time > > They can't. > > One might be on TCP and the other on UDP, but you can't have them both > listening on the same port with the same protocol. > > > for instance I get: > > > > [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: > > SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity=" > > Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="20 > > 25076022",LocalAddress="IPV4/UDP/166.84.7.53/5060",RemoteAddress="IPV4/UDP/ > > 45.134.144.118/5823",ACLName="registrar_attempt_without_configured_aors" > > What makes you think chan_sip and pjsip are both listening on UDP 5060? > > > I would like pjsit not to listen,till I figure out how to configure > > the thing, so my logs don't fill up with messages. > > > > Thanks in advance for any suggestions. > > As far as I recall using FreePBX, there is a selector for the SIP protocol to > tell it whether you want it to use pjsip or chan_sip. I don't think it even > supports using both at the same time, so simply make sure that is set to > chan_sip and you should be fine. > > On the other hand, why do you need to learn "how to configure the thing" if > you're using FreePBX? Part of the whole point is that it does the fiddly > techie sutff in the background for you, and you just need to use the > personnel- > department-friendly web GUI. This is what I thought as well, I just generated one trunk using the old chan_sip and expected nothing from pjsit, yet I get all kinds of errors like [2022-01-08 17:08:59] WARNING[487628] res_pjsip_registrar.c: Endpoint 'anonymous' (45.134.144.118:5823) has no configured AORs so I am very confused as to why this is happening. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and maybe a freepbx question
Hi. I am using asterisk 18.3 and freepbx. How can both sip and pjsip be listening at port 5060 at the same time, for instance I get: [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity="Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="2025076022",LocalAddress="IPV4/UDP/166.84.7.53/5060",RemoteAddress="IPV4/UDP/45.134.144.118/5823",ACLName="registrar_attempt_without_configured_aors" I would like pjsit not to listen,till I figure out how to configure the thing, so my logs don't fill up with messages. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 18.7.1 - can't load res_fax, can't stop app_fax
I always empty /usr/lib/asterisk/modules if I am going to do an install with a different version, or better to do it always. On Wed, 03 Nov 2021 09:14:37 -0400, Kingsley Tart wrote: > > > Is the app_fax.so module still in /usr/lib/asterisk/modules? If so - > > if you remove it do things work. > > Is app_fax.so explicitly being loaded in modules.conf? > > Thanks. > > I was already waiting for it to finish recompiling after Doug's > suggestion but yes, app_fax.so was still in there and removing it then > let me remove the noload => res_fax.so line from modules.conf and > everything started fine. > > At the end of the re-compile it was nice to see it point this out > actually: > > --8<-- > WARNING WARNING WARNING > > Your Asterisk modules directory, located at > /usr/lib/asterisk/modules > contains modules that were not installed by this > version of Asterisk. Please ensure that these > modules are compatible with this version before > attempting to run Asterisk. > > app_fax.so > > WARNING WARNING WARNING > --8<-- > > > No, modules.conf didn't mention app_fax. > > Thanks. All sorted. Now to work on the next one ;) > > -- > Cheers, > Kingsley. > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco Multiplatform 8865 configuration file
Does anyone have example config file for this phone with essential elements defined. I have a bunch of 7960s that I am provisioning with tftp - but now have to get 8865s going. This is for the multi-platform phone image - not the standard callmanager image. The only sample I've found so far is a gigantic bit of xml in cisco's provisioning manual. Cisco IP Phone 8800 Series and Cisco IP Conference Phone 8832 Multiplatform Phones Provisioning Guide I'm hoping I can maintain more minimal configurations. John (sorry if there are multiples. I just changed my account) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STIR/SHAKEN
Sebastian, There are many reasons why someone would want the DIDs provided by one provider and outbound calls to go out via 1,2 3, or more providers. In one of my installs I have a situation where local calls are placed via a local telco switch but LD calls go out via a voip provider. The Local telco has the DID but the LD does not so I have to verify the DIDs with the Voip provider(s). Another case may be for least cost routing. There are other reasons but you can see that it is not always as simple as using the same provider for DID and origination. Thanks, John On 3/11/21 3:34 PM, Sebastian Nielsen wrote: I reallt don’t understand why people simply use the same operator to terminate your calls, which also provide DIDs for you. Then you don’t need to touch this at all, your carrier will do all the STIR/SHAKEN handling for you, you are just a PBX customer. And then the operator then simply limits your account to only present your DID as outgoing number. Seems to be a unneccesary complicated solution just to have your numbers at company 1 and have your call termination at company 2. So fricking unneccessary. What I know there is requirements of number portability, so as long as company 2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from company 1 to company 2 – then company 2 owns your DIDs. Best regards, Sebastian Nielsen *Från:* asterisk-users-boun...@lists.digium.com *För *Alexander Perkins *Skickat:* den 12 mars 2021 01:23 *Till:* asterisk-users@lists.digium.com *Ämne:* Re: [asterisk-users] STIR/SHAKEN Hi Jeff. What exactly do you mean by the 'inbound piece'? I've spent quite a lot of time with the folks at TILTX understanding the framework; but I am not exactly sure what you mean by the 'inbound piece. Greg/Doug, like many folks here, we use LCR. So, the terminating carrier is not necessarily the one that issued us the telephone numbers. So, they will not sign it or simply cannot sign it. Remember that a very limited number of companies can actually sign the calls; the rest have to buy it from these 'Service Providers'. And there is another situation - the company you purchase your numbers from and the company you place your calls through may be different and both may not be able to sign your calls. Again, a very limited number of service providers that can actually sign your calls. So what do you do in that scenario? You have to find a Service Provider that can: 1. Verify you own that telephone number(s). 2. Sign your calls. 3. Provide you with the technical means to do so. So, that's that... I hope this makes sense. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium or Sangoma? What happened to Digium cards
Sangoma purchased Digium. You can find Sangoma cards at https://www.sangoma.com/telephony-cards/ On Tue, Jan 12, 2021 at 2:29 PM bilal ghayyad wrote: > Hello All; > > We were using Digium cards, now I am not able to reach for digium website > that contains the telephony cards and Asterisk website currently is taking > us for Sangoma, so what happened in Digium cards? > > Regards > Bilal > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I found an excellent guide: Configure Asterisk VoIP IP PBX SIP Server with Cisco IP Phones
The 7940 has 2 line buttons, the 7960 has 6 buttons There are other versions similar, including the 7970 with a color screen Not really sure you will have access to the SIP firmware without some relationship with Cisco. Better check before poring money down a rat hole! Also, you will need a TFTP server working on your Asterisk box Good luck JN Turritopsis Dohrnii Teo En Ming wrote: I am planning to get a cheap and used Cisco 7960 IP phone for testing. The excellent guide I found says that the configuration and setup is extremely well documented for this model of Cisco IP phone. I also read that I can google search for SIP firmware and download them? Is Cisco 7960 better and more advanced than Cisco 7940? On 2020-12-18 22:41, John Novack wrote: When purchasing these phones, make sure they are SIP, as these were available with several different firmware loads You may end up with one used with Call Manager, and struggle to get the SIP firmware for it. In addition there were several versions of SIP firmware. None of the code will be available to you without proper credentials from Cisco John Novack Turritopsis Dohrnii Teo En Ming wrote: Subject: I found an excellent guide: Configure Asterisk VoIP IP PBX SIP Server with Cisco IP Phones Good day from Singapore, Today 18 December 2020 Friday, I found an excellent guide on configuring Asterisk VoIP IP PBX SIP server with Cisco IP phones. I think the author explains very well and very clearly. The guide is certainly very detailed and well written. Title of Guide: Configure Asterisk with Cisco IP Phones Author: Tyler Winfield Link: http://docshare02.docshare.tips/files/6706/67061980.pdf Original website: http://minded.ca/ (no longer accessible) I am going to buy a cheap and used Cisco 7940 or 7960 IP phone for about SGD$20 and configure it to work with my FreePBX 15 and Asterisk 16 PBX appliance by following this guide. I will provide feedback and my own custom guide after I have done so. I just want to share this very excellent guide. Another excellent website is https://www.voip-info.org/ On the front page, it says: "Welcome to VOIP-info: a reference guide to all things VOIP." Sharing is caring. Thank you. Mr. Turritopsis Dohrnii Teo En Ming, 42 years old as of 18 December 2020 Friday, is a TARGETED INDIVIDUAL (TI) living in Singapore. He is presently an IT consultant with a System Integrator (SI)/computer firm in Singapore. He is an IT enthusiast. -- Dog is my Co-Pilot -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I found an excellent guide: Configure Asterisk VoIP IP PBX SIP Server with Cisco IP Phones
When purchasing these phones, make sure they are SIP, as these were available with several different firmware loads You may end up with one used with Call Manager, and struggle to get the SIP firmware for it. In addition there were several versions of SIP firmware. None of the code will be available to you without proper credentials from Cisco John Novack Turritopsis Dohrnii Teo En Ming wrote: Subject: I found an excellent guide: Configure Asterisk VoIP IP PBX SIP Server with Cisco IP Phones Good day from Singapore, Today 18 December 2020 Friday, I found an excellent guide on configuring Asterisk VoIP IP PBX SIP server with Cisco IP phones. I think the author explains very well and very clearly. The guide is certainly very detailed and well written. Title of Guide: Configure Asterisk with Cisco IP Phones Author: Tyler Winfield Link: http://docshare02.docshare.tips/files/6706/67061980.pdf Original website: http://minded.ca/ (no longer accessible) I am going to buy a cheap and used Cisco 7940 or 7960 IP phone for about SGD$20 and configure it to work with my FreePBX 15 and Asterisk 16 PBX appliance by following this guide. I will provide feedback and my own custom guide after I have done so. I just want to share this very excellent guide. Another excellent website is https://www.voip-info.org/ On the front page, it says: "Welcome to VOIP-info: a reference guide to all things VOIP." Sharing is caring. Thank you. Mr. Turritopsis Dohrnii Teo En Ming, 42 years old as of 18 December 2020 Friday, is a TARGETED INDIVIDUAL (TI) living in Singapore. He is presently an IT consultant with a System Integrator (SI)/computer firm in Singapore. He is an IT enthusiast. -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Legacy TDM400
AFAIK it requires DAHDI version 2 For unknown reasons, this and many other card drivers were removed in DAHDI version 3 Suggest you compile from source, rather than any repository, selecting the last DAHDI version 2 and at least Asterisk 13, though it is EOL or nearly, it still is a good version to work with For learning there isn't any good reason to have the latest of anything I have a working version of Asterisk 13 with DAHDI and a 4 port T1 card on CentOS 6, and support a buddy with a TDM 400 or 410 - no issues YMMV John Novack Roy Kidder wrote: Hello all, It's been quite some number of years since I played around with Asterisk and I'm just now getting back into it. I think the last version I worked with was 1.8. I have a legacy Digium TDM400 PCI card and am wondering if that will still work on newer versions of Asterisk. My initial attempts on Debian 10 and the Debian repository version of Asterisk didn't get me very far. Any pointers would be appreciated. -Roy -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to DIY/Setup An Open Source IP PBX Appliance/Server?
JMO AstLinux installed on an HP Thin Client is a good choice for someone with limited knowledge of Linux who wants a less steep learning curve. YMMV John Novack Turritopsis Dohrnii Teo En Ming wrote: Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server? Good day from Singapore, After reading recent reviews, I gather that Asterisk is the gold standard when it comes to open source VoIP systems and it is the most famous open source PBX out there. Article: Compare the Top 10 Best Open Source PBX Software of 2020 Link: https://www.voipreview.org/business-voip/best-open-source-pbx-software Article: Top 10 Free Open Source PBX Software Solutions Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/ The following is an excerpt from Wikipedia: "Asterisk is a core component in many commercial products and open-source projects. Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software with an open-source distribution model. AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software to realize all telephony functions. AstLinux is a "Network Appliance for Communications" open-source software distribution.[15] FreePBX, an open-source graphical user interface, bundles Asterisk as the core of its FreePBX Distro[16] LinuxMCE bundles Asterisk to provide telephony; there is also an embedded version of Asterisk for OpenWrt routers. PBX in a Flash/Incredible PBX and trixbox are software PBXes based on Asterisk. Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer PBX, fax, instant messaging and email functions, respectively, before switching to 3CX. Issabel is an open-source Unified Communications software which uses Asterisk for telephony functions. It was forked from the open-source versions of Elastix when 3CX acquired it. *astTECS uses Asterisk in its VoIP and mobile gateways." Link: https://en.wikipedia.org/wiki/Asterisk_(PBX) I would like to DIY/setup an IP PBX appliance/server using free open source projects. Which free open source project, mentioned in the list and links above, would you recommend to DIY my IP PBX appliance/server? Should I buy a desktop computer or get one of those appliances listed in the link below to serve as my IP PBX appliance/server? Link: https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celeron-j1800-dual-core-mini-pc-firewall-soft-router-with-ddr3l-msata-ssd-4-gigabit-lan-rj45-com-port-i449270007-s1196780479.html?spm=a2o42.searchlist.list.89.100857d22PjCYx=1 Please also refer me to very good, detailed and well explained guides/tutorials/manuals on setting up open source IP PBX appliances/servers. Lastly, please recommend a cheap and affordable IP phone (suggest brand and model) to go along with my DIY open source IP PBX appliance/server. Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020 Tuesday, is a TARGETED INDIVIDUAL (TI) living in Singapore. Thank you very much. -BEGIN EMAIL SIGNATURE- The Gospel for all Targeted Individuals (TIs): [The New York Times] Microwave Weapons Are Prime Suspect in Ills of U.S. Embassy Workers Link: https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and Australia (25 Dec 2019 to 9 Jan 2020): [1] https://tdtemcerts.wordpress.com/ [2] https://tdtemcerts.blogspot.sg/ [3] https://www.scribd.com/user/270125049/Teo-En-Ming -END EMAIL SIGNATURE- -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is anyone using autohints=yes with Queue hints and PJSIP?
Hello, I'm working on converting my 18.0.1 test system from SIP to PJSIP and I've run into something odd. I have a queue defined named acme-test that has two agents in it, PJSIP/7001acme and PJSIP/7002acme. I have autohints=yes in my acme-intern context, I have not defined hints for either of the extensions I'm subscribed to: PAUSE_acme-test_7001 and acme-test_avail My phone is subscribed to both extensions, I'm changing the port number used by the phone to register to swap between chan_sip and chan_pjsip. If I use chan_sip the subscription for both of these extensions work, if I switch to chan_pjsip I get the below errors and neither extension shows it's BLF status on the phone. [Nov 19 14:38:01] NOTICE[17172]: res_pjsip_exten_state.c:418 new_subscribe: Endpoint '7001acme' state subscription failed: Extension 'PAUSE_acme-test_7001' does not exist in context 'acme-intern' or has no associated hint [Nov 19 14:38:01] NOTICE[17172]: res_pjsip_exten_state.c:418 new_subscribe: Endpoint '7001acme' state subscription failed: Extension 'acme-test_avail' does not exist in context 'acme-intern' or has no associated hint I know there's a lot of stuff in play here, but I'm not sure if it's a problem with autohints=yes or the queue's hints themselves. -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx VPN SIP Client (SIP/2.0 401 Unauthorized)
On 06/11/2020 14:28, basti wrote: > Hello, > i try to connect my SIP Client (linphone) via VPN to FreePBX. > The routing looks OK. I can ping the Endpoints and traffic is routing. > I can also Register my Sip Client. > > debpbx*CLI> pjsip list contacts > > Contact: > > == > > > Contact: 731/sip:731@192.168.30.132:5060 163a967d99 > Avail 15.722 > Contact: 734/sip:734@10.8.0.143:5060 1b1aa8cbac > Avail 62.180 > > So far so good. When I try to an other extension I get a timeout. > tcpdump: > > root@debpbx:/etc/asterisk# tcpdump -ni enp0s15 host 10.8.0.143 and not > port 80 > tcpdump: verbose output suppressed, use -v or -vv for full protocol > decode > listening on enp0s15, link-type EN10MB (Ethernet), capture size 262144 > bytes > 13:03:04.086687 IP 10.8.0.143.5060 > 192.168.30.28.5060: SIP: INVITE > sip:7...@asterisk.kes SIP/2.0 > 13:03:04.087364 IP 192.168.30.28.5060 > 10.8.0.143.5060: SIP: SIP/2.0 > 401 Unauthorized > 13:03:04.126101 IP 10.8.0.143.5060 > 192.168.30.28.5060: SIP: ACK > sip:7...@asterisk.kes SIP/2.0 > 13:03:09.054643 IP 10.8.0.143.5060 > 192.168.30.28.5060: SIP > 13:03:14.112561 IP 192.168.30.28.5060 > 10.8.0.143.5060: SIP: OPTIONS > sip:734@10.8.0.143:5060 SIP/2.0 > 13:03:14.162609 IP 10.8.0.143.5060 > 192.168.30.28.5060: SIP: SIP/2.0 > 200 Ok > 13:03:19.057752 IP 10.8.0.143.5060 > 192.168.30.28.5060: SIP > 13:03:29.060765 IP 10.8.0.143.5060 > 192.168.30.28.5060: SIP > 13:03:44.672509 IP 10.8.0.143.5060 > 192.168.30.28.5060: SIP > > I think the SIP/2.0 401 Unauthorized is the problem. > I also had add the VPN IP range to the local_net but that does not > solve the problem. > > root@debpbx:/etc/asterisk# grep -ri 10.8.0 > sip_general_additional.conf:localnet=10.8.0.0/24 > pjsip.transports.conf:local_net=10.8.0.0/24 > > Your tcpdump doesn't show the full data of the invite and the 401 response. You'd probably be better of logging the sip messages from asterisk console with something like: pjsip set logger host 10.8.0.143 It's quite normal to have an initial 401 response to the first unauthorized INVITE. The 401 should contain an authentication header. The 401 response should be followed up by a second INVITE containing an authorization header. Maybe credentials are not setup correctly on the sip client. John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound
t;>>>>>>> >>>>>>>> Does anyone know how this can be achieved? >>>>>>>> >>>>>>> >>>>>>> If termination.com is only on 1.1.1.1 and pstn.com is only on >>>>>>> 2.2.2.2, create 2 transports, one specifically bound to 1.1.1.1, >>>>>>> transport-1.1.1.1 for instance, and another to 2.2.2.2: >>>>>>> transport-2.2.2.2. The names aren't important as long as you can tell >>>>>>> the >>>>>>> difference. Then explicitly configure endpoint termination.com's >>>>>>> "transport" parameter to "transport-1.1.1.1" and pstn.com's >>>>>>> "transport" parameter to "transport-2.2.2.2". In your dialplan, you >>>>>>> can >>>>>>> see which endpoint the call came in on, and route it out the same >>>>>>> endpoint. >>>>>>> >>>>>>> If both providers are available from both interfaces, you can create >>>>>>> 2 endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1, >>>>>>> termination.com-2.2.2.2 and pstn.com-2.2.2.2; Then configure each with >>>>>>> the >>>>>>> same transports as above. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> >>>>>>>> Thanks in advance for your help, >>>>>>>> >>>>>>>> -- >>>>>>>> David Cunningham, Voisonics Limited >>>>>>>> http://voisonics.com/ >>>>>>>> USA: +1 213 221 1092 >>>>>>>> New Zealand: +64 (0)28 2558 3782 >>>>>>>> -- >>>>>>>> >>>>>>>> _ >>>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com >>>>>>>> -- >>>>>>>> >>>>>>>> Check out the new Asterisk community forum at: >>>>>>>> https://community.asterisk.org/ >>>>>>>> >>>>>>>> New to Asterisk? Start here: >>>>>>>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>>>>>>> >>>>>>>> asterisk-users mailing list >>>>>>>> To UNSUBSCRIBE or update options visit: >>>>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> George Joseph >>>>>>> Asterisk Software Developer >>>>>>> direct/fax +1 256 428 6012 >>>>>>> Check us out at www.sangoma.com and www.asterisk.org >>>>>>> [image: image.png] >>>>>>> -- >>>>>>> _ >>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com >>>>>>> -- >>>>>>> >>>>>>> Check out the new Asterisk community forum at: >>>>>>> https://community.asterisk.org/ >>>>>>> >>>>>>> New to Asterisk? Start here: >>>>>>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>>>>>> >>>>>>> asterisk-users mailing list >>>>>>> To UNSUBSCRIBE or update options visit: >>>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> David Cunningham, Voisonics Limited >>>>>> http://voisonics.com/ >>>>>> USA: +1 213 221 1092 >>>>>> New Zealand: +64 (0)28 2558 3782 >>>>>> -- >>>>>> _ >>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>>>> >>>>>> Check out the new Asterisk community forum at: >>>>>> https://community.asterisk.org/ >>>>>> >>>&g
[asterisk-users] Directory Application
Hello all, Anyone know an easy way to have the Directory Application<https://wiki.asterisk.org/wiki/display/AST/Directory+Application> lookup all the voicemail contexts in the system. Like a global option John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI vs. Dialplan Originate
You could do the old school method and create and move a .call file from
your dialplan.
exten => writefile,1,NoOP()
same => n,Set(CALLFILE=/var/spool/asterisk/tmp/${FileName}-${ARG1}.call)
same => n,Set(FILE(${CALLFILE},,,al,u)=Channel: SIP/bob)
same => n,Set(FILE(${CALLFILE},,,al,u)=WaitTime: 20)
same => n,Set(FILE(${CALLFILE},,,al,u)=Context: alice)
same => n,Set(FILE(${CALLFILE},,,al,u)=Extension: s)
same => n,Set(FILE(${CALLFILE},,,al,u)=Priority: 1)
same => n,Set(FILE(${CALLFILE},,,al,u)=SetVar: John=AWESOME
same => n,Set(FILE(${CALLFILE},,,al,u)=Archive: Yes); we want to keep the
call file for debugging
same => n,system(mv ${CALLFILE} /var/spool/asterisk/outgoing/); Move the
file into the spool
On Tue, Sep 22, 2020 at 4:47 AM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
> On Tuesday 22 September 2020 at 13:27:27, Joshua C. Colp wrote:
>
> > On Tue, Sep 22, 2020 at 7:37 AM Antony Stone wrote:
> > > Hi.
> > >
> > > (Asterisk 16.2.1)
> > >
> > > I'm using AMI Originate to initiate calls, and I'm passing some
> > > additional data in to the dialplan context using the Variable:
> > > parameter. Works fine.
> > >
> > >
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+ManagerAction_Orig
> > > inate
> > >
> > > Now I need to do the same thing but from another context in my
> dialplan,
> > > so I was expecting to use the Originate() dialplan command, but how do
> I
> > > pass the additional data? I don't see any Variable: equivalent
> parameter.
> > >
> > >
> > >
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Origin
> > > ate
> >
> > The Originate dialplan application itself has no inherent built in
> ability
> > to set variables.
>
> Oh :( Not as direct an equivalent as I'd expected, then.
>
> Any suggestions for how to get round this - my first idea is to get
> something
> into the Asterisk internal DB in the context which does the Originate(),
> and
> then read the data out in the called context.
>
> Maybe I need to set some inherited variables and then use Dial() instead
> of
> Originate...
>
> Ideas welcome :)
>
>
> Antony.
>
> --
> It is also possible that putting the birds in a laboratory setting
> inadvertently renders them relatively incompetent.
>
> - Daniel C Dennett
>
>Please reply to the
> list;
> please *don't* CC
> me.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
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asterisk-users mailing list
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[asterisk-users] Confbridge
To all:
No matter what I try, I cannot get the system to wait for the admin to join. It
just dumps users into the bridge directly.
I do not have a pin for users, does that matter?
What am I missing?
Another issue the absolute timeout is not working ? ... have recordings that
last for over 24 hours... and this should not happen...
All calls should hangup after 4 ?
Any ideas ?
Any help is much appreciated.
Thanks
This is my dialplan.
exten => s,1,Wait(1)
exten => s,n,Answer
exten => s,n,Set(TIMEOUT(absolute)=14400)
exten => s,n,NoOp(${CALLERID(name)})
exten => s,n,NoOp(${CALLERID(num)})
exten => s,n,NoOp()
exten => s,n,Playback(church) ; "Please hold while..."
exten => s,n,Set(CONFBRIDGE(user,announce_join_leave)=no)
exten => s,n,Set(CONFBRIDGE(user,startmuted)=yes)
exten => s,n,Set(CONFBRIDGE(user,template)=church)
exten => s,n,Set(CONFBRIDGE(user,marked)=no)
exten => s,n,Set(CONFBRIDGE(user,wait_marked)=yes)
exten => s,n,Set(CONFBRIDGE(user,end_marked)=yes)
exten => s,n,ConfBridge(xaccel)
exten => s,n,hangup
confbridge.conf
[general]
[church]
type=user
startmuted=yes
announce_join_leave=no
announce_user_count=no
wait_marked=yes
end_marked=yes
music_on_hold_when_empty=no
quiet=yes
;
[xaccel]
type=bridge
record_conference=yes
;
Then calling in I see this
Conference Bridge Name Users Marked Locked Muted
== == == =
xaccel 1 0 No No
John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax: 201.806.2604
Cell: 973.390.1090
www.xaccel.net<http://www.xaccel.net/>
CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender
by reply e-mail and destroy all copies of the e-mail.
--
_
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Re: [asterisk-users] Problem with OPTIONS requests.
I've got this setup in a test context. [test] exten => s,hint,SIP/7124 exten => s,1,NoOP(Options to $EXTEN) same => n,Hangup() exten => _x.,hint,SIP/7124 exten => _X.,1,NoOP(Options to $EXTEN) same => n,Hangup() exten => Anonymous,hint,SIP/7124 exten => Anonymous,1,NoOP(Options to $EXTEN) same => n,Hangup() I added hints to see if that would make a difference and it hasn't. I also made a 'Anonymous' peer to see if that would help without any luck. On Thu, Jul 16, 2020 at 6:11 PM Joel Serrano wrote: > Hey John, > > In one installation I have, we use several monitoring tools (nagios based > and custom scripts based) and we have the following: > > ; Reply OK to SIP:OPTIONS > [public] > exten => s,1,Wait(1) > same => n,Hangup > : For Nagios > exten => nagios,1,Wait(1) > same => n,Hangup > > NOTES: > > 1- We have context=public in sip.conf, if you have anything else, you must > update the dialplan above accordingly. > 2- The second 'nagios' extension, is because the scripts need to send a > user, so we have it preconfigured to "nagios", if it's from Kamailio it > won't send a user and thus it will match in the s,1 exten. Feel free to > remove this one. > > Give it a try and let me know how it goes. > > Alternatively, you may also be able to configure your SBC > (kamailio/opensips? if so check dispatcher docs for *_reply_codes modparam) > to accept a 404 reply to a SIP:OPTIONS as a valid response. > > > Hope it helps. > > Cheers, > Joel. > > > On Thu, Jul 16, 2020 at 5:04 PM John Kiniston > wrote: > >> I'm implementing a SBC with my Asterisk PBX but the keeps disabling the >> trunk group I've configured and I think it may be because Asterisk is >> returning a 4r04 to the OPTIONS. >> >> I've created a test context and have put in a wildcard pattern match to >> try and catch those options but it doesn't seem to work. >> >> Is there a way to have asterisk respond with an 200 OK instead of a 404? >> >> -- >> A human being should be able to change a diaper, plan an invasion, >> butcher a hog, conn a ship, design a building, write a sonnet, balance >> accounts, build a wall, set a bone, comfort the dying, take orders, give >> orders, cooperate, act alone, solve equations, analyze a new problem, pitch >> manure, program a computer, cook a tasty meal, fight efficiently, die >> gallantly. Specialization is for insects. >> ---Heinlein >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with OPTIONS requests.
I'm implementing a SBC with my Asterisk PBX but the keeps disabling the trunk group I've configured and I think it may be because Asterisk is returning a 4r04 to the OPTIONS. I've created a test context and have put in a wildcard pattern match to try and catch those options but it doesn't seem to work. Is there a way to have asterisk respond with an 200 OK instead of a 404? -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ICE error
Hello all, Running Asterisk 16.10.1 Does anyone know what this means? rtp_recvfrom: PJ ICE Rx error status code: 70004 'Invalid value or argument (PJ_EINVAL) How can I find what value it doesn't like ? I switched to a few different stun servers and I still get the same error. Calls still go through Any help is much appreciated. Thanks John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken
On Mon, 13 Jul 2020 15:44:12 -0400, Matthew Fredrickson wrote: > > On Mon, Jul 13, 2020 at 2:34 PM Saint Michael wrote: > >> > >> There is a big confusion here about Stir Shaken. It is NOT a provider > >> issue. Un fact, all providers are whasing their hands and modifying their > >> swihtches to pass-through the Signature. They cannot sign the call because > >> then the become the responsible party for the call before the FCC, and > >> liable for any illegal call. Every owner of a PBX that sends calls to the > >> network, except if you use a trunk for the likes of Vonage, needs to sign > >> their calls. So if you send calls with any kind of dialer and use DIDs, > >> real or "borrowed", you need to get the signature service urgently or your > >> business will stop terminating calls. You cannot self-sign, you cannot get > >> around it, the calls will either go to straight to voicemail or fail. Even > >> worse, the carries wil play a fake voicemail and charge you a fee, > >> something that some already a are doing when they detect robocallig. > > > > Don't even think about Transnexus, because they use 302 Redirect with a > > header, and no version of Asterisk supports it. I am the only game in the > > world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is > > literally true. If you need to sign your calls to get through, with > > Asterisk, you need to connect to my service. I am an approved Service > > Provider from the FCC. If you keep thinking this is not happening, it is, > > and your business will disappear overnight. > > The issue is that Vicidial, for example, does not provide res_odbc and > > func_odbc, so you need to solve that first with Vicidial. Then you can > > apply the code I provided earlier and your calls with have a legal, binding > > signature. The carriers verify each signature and discard the ones that > > fail the cryptography test. > > Sounds like you're trying to sell/direct people towards a service that > you've created. Feel free to do so on the -biz list but the -users > list isn't the right place for that sort of thing. But the question is, are his statements correct that we need some service -- not necessarily his -- to sign the call before sending it to our normal carrier, or will the normal carrier -- whoever -- sign the call if they know the number? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redis in place of astdb
Dovid, You could use func_odb + a ODBC Redis driver to keep from having to
shell out.
On Wed, Jul 8, 2020 at 4:37 AM Dovid Bender wrote:
> Hi,
>
> Does anyone know of any projects that would allow you to use Redis in
> place of AstDB? By in place of I don't mean for what Asterisk needs but to
> store values. For instance for CNAM currently we need to use an AGI to
> connect to redis to pull CNAM. So in place of:
> Set(CALLERID(name)=${DB(CNAM/${CALLERID(num)})}
> it would be done with redis for example:
> Set(CALLERID(name)=${REDIS1(CNAM_${CALLERID(num)})}
>
> If not can the devs here give me a pointer to where to look?
>
> TIA.
>
> Regards,
>
> Dovid
> --
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--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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[asterisk-users] includes with time and timezone.
Hello, I cannot find much on examples but I did find one in Russian that shows this to use + or - the time difference from GMT. I have been testing and it does not work. 1st question do includes work with timezone include => day,08:00-17:00,mon-fri,*,*,[+5] Not sure on the formatting, is it correct ? ... I tried without the brackets... that also doesn't work. If not supported in includes What is the formatting for timezone in gotoiftime. GotoIfTime(times,weekdays,mdays,months,[timezone]?[labeliftrue:[labeliffalse]]) Any helps is much appreciated. Thanks John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempting to get BLF working with linphone
On 12/06/2020 16:19, Olivier wrote: It seems a new Linphone 4.2 is to be published next week ! Hopefully, ... 1. its call history is useless to me, it works very poorly with sip proxying (i.e. asterisk), the design is clunky (no simple list of all calls) 2. it has no simple busy light support, only the complicated presence stuff that asterisk doesn't really support well and needs manual intervention. 3. the Qt interface is "prettier" than the old gtk2 interface but way less complete for telephone functions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk hints can be in multiple states; most sip NOTIFY dialogs only send one state
On 10/06/2020 15:40, Joshua C. Colp wrote: You wouldn't be able to access such information from ast_sip_presence_exten_state_to_str, that function is strictly for taking in instructions/data and producing the output. The user of it would need to pass in a value to turn on this new behavior. From that level the ast_sip_exten_state_data structure can optionally have a subscription, which itself has the endpoint that was used to establish the subscription. Ok, I'll look at that when I get around to moving to chan_pjsip. I'm very slow at changing working configurations. :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk hints can be in multiple states; most sip NOTIFY dialogs only send one state
Asterisk can know that one of the attached phones is both "ringing" and "on the phone". However the sip NOTIFY it sends out to interested parties can only communicate one state, for example with pidf+xml it can either send "Ringing" or "On the phone" and so it sends "Ringing". This makes the "busy lights" less than useful, if a call makes multiple phones ring you can't tell, looking at the busy lights, which ones are busy, and so less likely to answer. In the chan_sip configuration there is an option "notifyringing": notifyringing *notifyringing* enables or disables notifications for the RINGING state when an extension is already INUSE. Only affects subscriptions using the *dialog-info* event package. Option can be configured in the general section only. It cannot be set per-peer. As the doc says this only applies to dialog-info style NOTIFY, not the pidf+xml format my phones use. Here is a patch that makes notifyringing work for pidf+xml. Generalising it for other formats is left as an exercise for the reader. Of course chan_sip is obsolete. How might this be done for chan_pjsip? Parts of the code are similar, but the layering is vastly different. How could the ast_sip_presence_exten_state_to_str function in res/res_pjsip/presence_xml.c get at the pjsip configuration? Description: make "notifyringing" work with pidf+xml If sip config specifies notifyringing=no and an extension is in a call then we send out "On a call" instead of "Ringing" so people can see who is not going to pick the call up. Author: John Hughes Last-Update: 2020-06-09 --- asterisk-13.14.1~dfsg.orig/channels/chan_sip.c +++ asterisk-13.14.1~dfsg/channels/chan_sip.c @@ -14966,7 +14966,10 @@ static void state_notify_build_xml(struc statestring = (sip_cfg.notifyringing) ? "early" : "confirmed"; local_state = NOTIFY_INUSE; pidfstate = "busy"; - pidfnote = "Ringing"; + if (subscribed == PIDF_XML && !sip_cfg.notifyringing) + pidfnote = "On the phone"; + else + pidfnote = "Ringing"; break; case AST_EXTENSION_RINGING: statestring = "early"; -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cdr_mysql: Cannot connect to database server - SSL error: SSL_CTX_set_default_verify_paths failed
On Mon, 8 Jun 2020 at 05:18, Markus wrote: > Hi list! > > I'm getting this error frequently: > > ERROR[25193][C-0004f387]: cdr_mysql.c:203 mysql_log: Cannot connect to > database server localhost: (2026) SSL connection error: > SSL_CTX_set_default_verify_paths failed > "SSL_CTX_set_default_verify_paths() specifies that the default locations from which CA certificates are loaded should be used." ( https://www.openssl.org/docs/man1.1.0/man3/SSL_CTX_set_default_verify_paths.html ) "The SSL_CTX_set_default_verify_paths failed error occurs if paths to any of the certificate files are invalid (either missing or have incorrect permissions)." ( https://stackoverflow.com/questions/27031318/what-does-this-error-mean-in-mariadb-ssl-error-2026-hy000-ssl-connection-e/45083610 ) Sounds to me like your CApath / CAfile being used by MySQL doesn't exist, and so it's unable to verify the certificate. You can try some of the answers in that stack overflow thread... or just don't use SSL to connect to a local MySQL instance. Unless you've done some weird configuration, only the same user as one of the processes, or root, has the ability to eavesdrop; and both could eavesdrop by attaching to the process (i.e. with a debugger) even with SSL enabled. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempting to get BLF working with linphone
On 26/05/2020 15:33, Olivier wrote: Hi John, 1. Could you get any further, in your quest for working BLF with linphone ? The patches to get linphone-3.12 BLF working with Asterisk are here: http://perso.calvaedi.com/~john/linphone-3/ They're pretty damnned trivial: 1. add the "Accept" header to the SUBSCRIBE message so asterisk doesn't reject it. 2. don't trash the SIP dialog if the SUBSCRIBE refresh is rejected because of a stale nonce. 3. If asterisk says the user is on the phone set the status to on the phone. All except the 3rd one are compatible with linphone-4. Implementing the same feature with linphone-4 is left as an exercise for the reader. 2. Have you tried with a different Linphone version (4.12 is pending on Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ? Version 4 of linphone is, frankly, rubbish. I have managed to hack it to the point where presence shows green for connected contact and grey for disconnected. However this requires setting the "send subscribe" flag in the linphone contacts db and linphone 4 has no UI for setting this flag, you have to do it using sqlite3 directly (or setting up your contacts in linphone 3). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP
Hello, Anyone know how to set the "To:" in an invite for PJSIP to custom settings. I got the "from" to be the way I need it. From: I have tried a lot of changes to get to this but nothing works. I am getting this From: sip:109643...@xaccel.net;tag=42e4a9cb-59af-4d40-a21f-00261afbd3be To: sip:34.221.174.202 I need to put "TEST" http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir-Shaken for asterisk
Nice, Do you have the code up on GitHub? I'd love to see it.
What's the source of the data? Something API driven I hope?
Have you thought about implementing your project via curl instead of
func_odbc?
On Wed, May 27, 2020, 8:52 PM Saint Michael wrote:
> In a few weeks, no SIP call is going to terminate unless they are signed
> properly, as mandated by law. We are in the business of Stir-Shaken,
> signing calls, as an FCC-approved provider. A big differentiator between
> our service and the rest: we are the only ones who don't need to receive
> the calls in our servers to sign them. We do this over a MySQL call,
> easily connectable to Asterisk via res_odbc, so you never have to send us
> your calls. This is a sample of how we do this so you may test now:
> mysql -u anonymous -h 208.73.232.47 -e "call
> strshk.stir_shaken_signature('7274433019','1957408')".
> If your caller-ID is a valid US number and not a wireless number (that is
> a NO-NO for the FCC), we sign the call as 'C', if you use your own DIDs,
> something we can verify as legit, then we sign as 'B', and if you use our
> DID as caller ID, we sign as 'A', full attestation.
> Please email to venefax at g mail if you have any questions. Do not think
> you can do business as usual. The wild west of VOIP is coming to an end.
> But we can keep you in business if you follow the rules.
> --
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Re: [asterisk-users] Dialplan - using multiple AND or OR in set is it possible ?
Use the ARRAY version of Set. same = n,ExecIf($["A" = "B"]?Set(ARRAY(C,D)=1,2)) On Tue, Apr 21, 2020 at 3:56 AM Administrator wrote: > Hello, > > we want to use something like > > same = n,ExecIf($["A" = "B"]?Set(C=1) & Set(D=2) & ...) > > Problem is that result gives C=1) & Set(D=2) & ... > > Is there a possibility to use multiple AND or OR in such a way ? > > -- > Daniel > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?
On 14/05/2020 16:41, Joshua C. Colp wrote: On Thu, May 14, 2020 at 11:31 AM John Hughes <mailto:j...@calva.com>> wrote: On 14/05/2020 08:10, John Hughes wrote: I am having a problem with one of my callers who is using either g729 or alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up. Why? Last minute thought: Is it possible that the caller is sending g729 in RTP even though the SIP negotiation clearly chooses alaw? Maybe I need some RTP debugging. And in fact that is exactly what's happening. And when I look at the RTP debugging I see the data from the remote is: Got RTP packet from xx.xx.xx.xx:50644 (type 18, seq 001338, ts 610458, len 20) AAArgh! Type 18 is g729. Why on earth is the remote sending me g729 when I clearly said the only thing I could do was alaw. Is this legal? Is the other side broken? It shouldn't be sending it, but as well we should be ignoring it. I believe we do ignore in modern versions, I can't speak for your old one. As for why... I don't really have an answer. Ok, so maybe upgrading my asterisk would be a good idea, but I don't think it'll fix this problem, they sent me 6 g729 packets before the communication was cut, I'm pretty sure they've just ignored the results of the negotiation. I hope I can get them to fix their system... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?
On 14/05/2020 08:10, John Hughes wrote: I am having a problem with one of my callers who is using either g729 or alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up. Why? Last minute thought: Is it possible that the caller is sending g729 in RTP even though the SIP negotiation clearly chooses alaw? Maybe I need some RTP debugging. And in fact that is exactly what's happening. <--- SIP read from UDP:SUPPLIER:5060 ---> INVITEsip:LOCAL@ASTERISK:5060 SIP/2.0 Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9 From:;tag=gK02498cb1 To: Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: P-Asserted-Identity: Supported: timer,100rel,precondition Session-Expires: 1800 Min-SE: 90 Content-Length: 282 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 176880 320591 IN IP4 SUPPLIER s=SIP Media Capabilities c=IN IP4 213.41.124.6 t=0 0 m=audio 8526 RTP/AVP 18 8 101 *a=rtpmap:18 G729/8000* a=fmtp:18 annexb=no *a=rtpmap:8 PCMA/8000* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 <-> So he says he wants g729 or alaw Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw|gsm), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (*alaw*) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) And asterisk calculates that the common codecs are just alaw, So asterisk says: "let's do alaw": <--- Reliably Transmitting (no NAT) to SUPPLIER:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER From:;tag=gK02498cb1 To:;tag=as4502927f Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 227409966 227409966 IN IP4 ASTERISK s=Asterisk PBX 13.14.1~dfsg-2+deb9u4 c=IN IP4 ASTERISK t=0 0 m=audio 13948 RTP/AVP 8 101 *a=rtpmap:8 PCMA/8000* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <> And when I look at the RTP debugging I see the data from the remote is: Got RTP packet from xx.xx.xx.xx:50644 (type 18, seq 001338, ts 610458, len 20) AAArgh! Type 18 is g729. Why on earth is the remote sending me g729 when I clearly said the only thing I could do was alaw. Is this legal? Is the other side broken? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?
On 14/05/2020 16:08, Michael L. Young wrote: I am having a problem with one of my callers who is using either g729 or alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up. Why? Last minute thought: Is it possible that the caller is sending g729 in RTP even though the SIP negotiation clearly chooses alaw? Maybe I need some RTP debugging. Asterisk 13.14.1 on Debian, using chan_sip. Hi John, Maybe a newer version of Asterisk would help? The latest release for 13 is version 13.33. The version you are on was released 3 years ago. Well, like I said I'm on Debian, using the packaged version. If I want to upgrade I'll have to compile it myself, or upgrade to Debian buster to get 16.2.1 Here is an issue which looks like what you describe and was fixed in 13.16 https://issues.asterisk.org/jira/browse/ASTERISK-26143 That doesn't seem to be the same problem. My problem is that the other end is sending g729, which my asterlsk can't do at all, and tells the remote it can't do. I'm shocked that the remote is trying to send me stuff using a codec that I didn't say I could handle. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?
I am having a problem with one of my callers who is using either g729 or alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up. Why? Last minute thought: Is it possible that the caller is sending g729 in RTP even though the SIP negotiation clearly chooses alaw? Maybe I need some RTP debugging. Asterisk 13.14.1 on Debian, using chan_sip. Here's the trace: <--- SIP read from UDP:SUPPLIER:5060 ---> INVITEsip:LOCAL@ASTERISK:5060 SIP/2.0 Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9 From:;tag=gK02498cb1 To: Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: P-Asserted-Identity: Supported: timer,100rel,precondition Session-Expires: 1800 Min-SE: 90 Content-Length: 282 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 176880 320591 IN IP4 SUPPLIER s=SIP Media Capabilities c=IN IP4 213.41.124.6 t=0 0 m=audio 8526 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 <-> --- (17 headers 13 lines) --- Sending to SUPPLIER:5060 (no NAT) Sending to SUPPLIER:5060 (no NAT) Using INVITE request as basis request - 205665777_90679951@SUPPLIER Found peer 'supplier' for 'REMOTE' from SUPPLIER:5060 Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|ulaw|gsm), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 213.41.124.6:8526 Looking for LOCAL in supplier-in (domain ASTERISK) sip_route_dump: route/path hop: So, all looking good here, we've worked out that the combined capabilities are (alaw) <--- Transmitting (no NAT) to SUPPLIER:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER From:;tag=gK02498cb1 To: Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <> <--- Transmitting (no NAT) to SUPPLIER:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER From:;tag=gK02498cb1 To:;tag=as4502927f Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <> Audio is at 13948 Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to SUPPLIER:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER From:;tag=gK02498cb1 To:;tag=as4502927f Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 INVITE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 227409966 227409966 IN IP4 ASTERISK s=Asterisk PBX 13.14.1~dfsg-2+deb9u4 c=IN IP4 ASTERISK t=0 0 m=audio 13948 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <> And that's good to, we've sent the OK for the INVITE saying that we want alaw. <--- SIP read from UDP:SUPPLIER:5060 ---> ACKsip:LOCAL@ASTERISK:5060 SIP/2.0 Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5bc037285f864da9 From:;tag=gK02498cb1 To:;tag=as4502927f Call-ID: 205665777_90679951@SUPPLIER CSeq: 539098 ACK Max-Forwards: 70 Content-Length: 0 <-> --- (8 headers 0 lines) --- [May 13 13:46:58] WARNING[7245][C-31da]: channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw) What's this nonsense! Why is set_format trying to use g729! Scheduling destruction of SIP dialog '205665777_90679951@SUPPLIER' in 32000 ms (Method: ACK) set_destination: Parsing for address/port
Re: [asterisk-users] Length of dial string
Or you could just increase MAX_EXTENSION and recompile. On Fri, 01 May 2020 06:25:36 -0400, Paddy Grice wrote: > > [1 ] > [1.1 ] > Hi Dovid > > Yes was one of the options but as the required list is dynamic becomes very > messy - and all combinations problem - where as "call all workers job xxx" > is what is needed so the ability to call 20+ numbers is what is needed - agi > does a database search for all jobx workers and constructs a dialstring with > SIP, DAHDI and Local devices. > > Can someone tell me where to find maximum string length for the dial data in > the DIAL command > > Paddy > > _ > > From: Dovid Bender [mailto:do...@telecurve.com] > Sent: 01 May 2020 10:26 > To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [asterisk-users] Length of dial string > > > Paddy, > > Why not use local extensions? You can do something like this. > Exten => > s,1,Dial(Local/set1@call_all/set2@call_all/set3@call_all) > > [call_all] > Exten => set1,1,Dial(SIP/100/101/102/103/104/105 > Exten => set1,1,Dial(SIP/106/107/108/109/110/111 > Exten => set1,1,Dial(SIP/112/113/114/1015/116/117 > > > On Fri, May 1, 2020 at 3:22 AM Paddy Grice wrote: > > > Hi all > > as per the new release notice for 13.33.0 received today - can anyone advise > me the max limit of the string to the Dial Command - see > * [ASTERISK-27946 > https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - > dial (API): Storage of dialed target uses AST_MAX_EXTENSION > when it shouldn't > > I have been fighting with this issue for months trying to find a solution I > need to call 20+ devices at the same time so dial strings are very long I > cant really use a queue(ringall) which was my original idea as the customer > needs different groups for virtually every call some of which are simple sip > devices and others have to be local devices (Internal and External CLIs). > > Paddy Grice > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > [1.2 ] > [2 ] > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio problem with asterisk and meetme conference
On Thu, 26 Mar 2020 09:18:24 -0400, Doug Lytle wrote: > > >>> Can I adjust the talk or listen volume for another user? > > I've never used the volume controls, but it would appear. > > https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration > > Doug According to this document, there is no way for me to change the volume(s) for another user, whereas meetme allows me to do this by specifying the conference number and user number. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio problem with asterisk and meetme conference
On Thu, 26 Mar 2020 06:54:37 -0400, Doug Lytle wrote: > > >>> I never moved to confbridge because they don't have an option for > >>> controlling the volume of other > >>> participants audio > > I have menu options in my confbridge configs that has increase and decrease > conference volume. > > I'd still configure a small confbridge and test if you still have the issue, > since meetme is no longer being developed. Can I adjust the talk or listen volume for another user? If I could do that I would switch, but otherwise I have to stay with meetme. And I wonder if its a meetme issue? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio problem with asterisk and meetme conference
On Wed, 25 Mar 2020 12:42:00 -0400, Doug Lytle wrote: > > > >>> he problem is that there is some sort of distortion in the audio > > Has been been going on for a while or is this a new setup? Do you have a > timing source? > > I bit the bullet around a year ago and moved to CONFBRIDGE; it wasn't as > horrible as I thought it would be to setup. Well, this has been going on for quite a while, my timing source is internal according to asterisk.conf. I never moved to confbridge because they don't have an option for controlling the volume of other participants audio, meetme has this feature which I use frequently. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] audio problem with asterisk and meetme conference
Hi. I have a problem with my audio in meetme conference under asterisk 13 using Debian buster compiled from source. The problem is that there is some sort of distortion in the audio -- a workaround is always to lower the listen volume (*4). I see nothing in the log and so I wonder what is happening. I have dahdi loaded so I can record the conferences. Thanks in advance for any suggestions and let me know if you need any more information. I know 13 is old, I am working on upgrading. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attempting to get BLF working with linphone
On 23/03/2020 18:51, Joshua C. Colp wrote: On Mon, Mar 23, 2020 at 2:45 PM John Hughes <mailto:j...@calva.com>> wrote: Why is asterisk giving an error 500? I can find no reason, there is nothing in any log. The sequence number is from the past. The first SUBSCRIBE is sequence number 22 (check the CSeq header). The second is 20. The third is 21. It appears as though this is from the past, so it receives a 500. Ok, I've had some back and forth with the linphone developers and they contend that although the sequence number on the 2nd and 3rd SUBSCRIBE messages start a new sequence this is legal as it is a new conversation -- the "tag=" on the From has changed. Are they right? (Notice that the tag= from asterisk also changes). <--- SIP read from UDP:10.27.128.3:5060 <http://10.27.128.3:5060> ---> SUBSCRIBE sip:jacques@10.27.128.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;rport From: ;*tag=iGH81k5xf* To: ;tag=as3c7de68c CSeq: 22 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) Authorization: Digest realm="asterisk", nonce="188b095b", algorithm=MD5, username="john", uri="sip:jacques@10.27.128.1:5060", response="bdbc7cbac4453fd643050bf28996a68e" <-> --- (14 headers 0 lines) --- Found peer 'john' for 'john' from 10.27.128.3:5060 <http://10.27.128.3:5060> <--- Transmitting (no NAT) to 10.27.128.3:5060 <http://10.27.128.3:5060> ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;received=10.27.128.3;rport=5060 From: ;*tag=iGH81k5xf* To: ;tag=as3c7de68c Call-ID: SQOclJgm4O CSeq: 22 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3144c0a9", stale=true Content-Length: 0 <> Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.27.128.3:5060 <http://10.27.128.3:5060> ---> SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;rport From: ;*tag=c3Wvuu2XH <= new conversation* To: sip:jacq...@masked.masked.com CSeq: *20 SUBSCRIBE <=== sequence restarts* Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) <-> --- (13 headers 0 lines) --- Sending to 10.27.128.3:5060 <http://10.27.128.3:5060> (no NAT) Creating new subscription Sending to 10.27.128.3:5060 <http://10.27.128.3:5060> (no NAT) sip_route_dump: route/path hop: Found peer 'john' for 'john' from 10.27.128.3:5060 <http://10.27.128.3:5060> <--- Transmitting (no NAT) to 10.27.128.3:5060 <http://10.27.128.3:5060> ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;received=10.27.128.3;rport=5060 From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com;tag=as007ffc64 Call-ID: SQOclJgm4O CSeq: 20 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4224acfb" Content-Length: 0 <> Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.27.128.3:5060 <http://10.27.128.3:5060> ---> SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;rport From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com CSeq: 21 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) Authorization: Digest realm="asterisk", nonce="4224acfb", algorithm=MD5, username="john", uri="sip:jacq...@masked.masked.com", response="eb30a9801e78d2cb2c58c61200c50cb1" <-> --- (14 headers 0 lines) --- <--- Transmitting (no NAT) to 10.27.128.3:5060 <http://10.27.128.3:5060> ---> *SIP/2.0 500 Server error* Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;received=10.27.128.3;rport=5060 From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com;tag=as3c7de68c Call-ID: SQOclJgm4O CSeq: 21 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow:
Re: [asterisk-users] Attempting to get BLF working with linphone
On 23/03/2020 18:51, Joshua C. Colp wrote: On Mon, Mar 23, 2020 at 2:45 PM John Hughes <mailto:j...@calva.com>> wrote: Why is asterisk giving an error 500? I can find no reason, there is nothing in any log. The sequence number is from the past. The first SUBSCRIBE is sequence number 22 (check the CSeq header). The second is 20. The third is 21. It appears as though this is from the past, so it receives a 500. Why does asterisk think the error 500 is going to be acked? It doesn't. The message is for something else, it refers to sequence number 103. Ok, thanks, that's clear and obvious. Now I have to go beat up(*) the linphone people. ((*) in the nicest possible way of course). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attempting to get BLF working with linphone
So I've got a bit further with my project to get BLF working between asterisk and linphone. Initially asterisk was rejecting linphone's SUBSCRIBE messages because they didn't have an Accept: header. I've fixed that and now the initial SUBSCRIBE messages work and I see all my online contacts in green. But after a few minutes linphone attempts to renew the subscriptions and asterisk is not happy at all: <--- SIP read from UDP:10.27.128.3:5060 ---> SUBSCRIBE sip:jacques@10.27.128.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;rport From: ;tag=iGH81k5xf To: ;tag=as3c7de68c CSeq: 22 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) Authorization: Digest realm="asterisk", nonce="188b095b", algorithm=MD5, username="john", uri="sip:jacques@10.27.128.1:5060", response="bdbc7cbac4453fd643050bf28996a68e" <-----> --- (14 headers 0 lines) --- Found peer 'john' for 'john' from 10.27.128.3:5060 <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;received=10.27.128.3;rport=5060 From: ;tag=iGH81k5xf To: ;tag=as3c7de68c Call-ID: SQOclJgm4O CSeq: 22 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3144c0a9", stale=true Content-Length: 0 <> Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.27.128.3:5060 ---> SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;rport From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com CSeq: 20 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) <-> --- (13 headers 0 lines) --- Sending to 10.27.128.3:5060 (no NAT) Creating new subscription Sending to 10.27.128.3:5060 (no NAT) sip_route_dump: route/path hop: Found peer 'john' for 'john' from 10.27.128.3:5060 <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;received=10.27.128.3;rport=5060 From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com;tag=as007ffc64 Call-ID: SQOclJgm4O CSeq: 20 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4224acfb" Content-Length: 0 <> Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:10.27.128.3:5060 ---> SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;rport From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com CSeq: 21 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: ;+sip.instance="" User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) Authorization: Digest realm="asterisk", nonce="4224acfb", algorithm=MD5, username="john", uri="sip:jacq...@masked.masked.com", response="eb30a9801e78d2cb2c58c61200c50cb1" <-> --- (14 headers 0 lines) --- <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> *SIP/2.0 500 Server error* Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;received=10.27.128.3;rport=5060 From: ;tag=c3Wvuu2XH To: sip:jacq...@masked.masked.com;tag=as3c7de68c Call-ID: SQOclJgm4O CSeq: 21 SUBSCRIBE Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <> [Mar 23 18:23:09] WARNING[2128]: chan_sip.c:4071 retrans_pkt: Retransmission timeout reached on transmission SQOclJgm4O for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response Why is asterisk giving an error 500? I can find no reason, there is nothing in any log. Why does asterisk think the error 500 is going to be acked? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check ou
Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH
On 23/03/2020 11:29, Joshua C. Colp wrote: On Mon, Mar 23, 2020 at 7:15 AM John Hughes <mailto:j...@calva.com>> wrote: Hi, in these dark days of COVID-19 lockdown I'm using linphone to connect to my office asterisk system for working from home. It's going pretty well but the presence/BLF functions don't appear to work. In the linphone logs and asterisk debug I find that asterisk is rejecting linphone's PUBLISH message: Asterisk has no support for receiving/storing/using such a PUBLISH message. Asterisk instead generates state itself based on whether something is on the phone, busy, etc. This is received using a SUBSCRIBE and NOTIFY. Aha! Thanks a bunch. Now I just have to fix linphone's broken SUBSCRIBE... Mar 23 11:48:37] WARNING[2128]: chan_sip.c:28198 handle_request_subscribe: SUBSCRIBE failure:*no Accept header*: pvt: stateid: -1, laststate: 0, dialogver: 0, subscribecont: '', subscribeuri: '' -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH
Hi, in these dark days of COVID-19 lockdown I'm using linphone to connect to my office asterisk system for working from home. It's going pretty well but the presence/BLF functions don't appear to work. In the linphone logs and asterisk debug I find that asterisk is rejecting linphone's PUBLISH message: <--- SIP read from UDP:10.27.128.3:5060 ---> PUBLISH sip:j...@xxx.xxx.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;rport From: ;tag=ZtFgBTxUL To: sip:j...@xxx.xxx.com CSeq: 20 PUBLISH Call-ID: SMHLUSLJD6 Max-Forwards: 70 Supported: replaces, outbound Event: presence Accept: application/pidf+xml Content-Length: 511 Content-Type: application/pidf+xml Expires: 3600 User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:pidfonline="http://www.linphone.org/xsds/pidfonline.xsd; entity="sip:j...@xxx.xxx.com" xmlns="urn:ietf:params:xml:ns:pidf"> open sip:j...@xxx.xxx.com 2020-03-23T09:40:43Z <-> --- (14 headers 3 lines) --- Sending to 10.27.128.3:5060 (no NAT) <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.GRd5yC7Wo;received=10.27.128.3;rport=5060 From: ;tag=ZtFgBTxUL To: sip:j...@xxx.xxx.com;tag=as674d428f Call-ID: SMHLUSLJD6 CSeq: 20 PUBLISH Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 I can find nothing in the asterisk logs that says *why* it doesn't like the publish. Help? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] congested/busy on trunk?
look for 'mytrunk' as thats the trunk its dialing On Wed, Mar 18, 2020 at 02:41:51PM -0300, Joshua C. Colp wrote: > On Wed, Mar 18, 2020 at 2:37 PM John Roman wrote: > > > ive enabled logging. aside from a realm error i see on my endpoint, im > > still not sure whats up > > > Did you selectively enable logging? I don't see any SIP request for the > trunk. If you did enable it for everything, then I'd suggest checking > "pjsip show endpoints" and seeing the status of the trunk. > > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- -- -- -- j...@dev1ce.com https://dev1ce.com/john.gpg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] congested/busy on trunk?
lice-0002'
status is 'CONGESTION'
<--- Transmitting SIP response (548 bytes) to
TCP:[2605:e000:130a:fb:de1:71fc:e257:6f4e]:37879 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;rport=37879;received=2605:e000:130a:fb:de1:71fc:e257:6f4e;branch=z9hG4bKd67297ab9853f9168cf8c8533868ea6b373339
Call-ID:
26b4820ac727b0f23fea131b7c5cd450@2605:e000:130a:fb:de1:71fc:e257:6f4e
From: "demo-alice"
;tag=3166828162
To:
;tag=f1b212ab-9b55-4d13-9055-f49ce55f214e
CSeq: 8613 INVITE
Server: Asterisk PBX GIT-master-0cde95ec89
Reason: Q.850;cause=34
Content-Length: 0
<--- Received SIP request (489 bytes) from
TCP:[2605:e000:130a:fb:de1:71fc:e257:6f4e]:37879 --->
ACK sip:13107950...@dunkel.dev1ce.com:5060;transport=tcp
SIP/2.0
Call-ID:
26b4820ac727b0f23fea131b7c5cd450@2605:e000:130a:fb:de1:71fc:e257:6f4e
Max-Forwards: 70
From: "demo-alice"
;tag=3166828162
To:
;tag=f1b212ab-9b55-4d13-9055-f49ce55f214e
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;branch=z9hG4bKd67297ab9853f9168cf8c8533868ea6b373339;rport
CSeq: 8613 ACK
Content-Length: 0
On Tue, Mar 17, 2020 at 10:48:13AM -0300, Joshua C. Colp wrote:
> On Sat, Mar 14, 2020 at 2:02 PM John Roman wrote:
>
> > greetings asterisk users :)
> > ive just deployed version 17 and migrated as best I can to pjsip. I can
> > receive calls, and get to my mailbox prompt, however placing calls seems
> > impossible with the following error on dial:
> >
> > Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel
> > (pid = 517890)
> > dunkel*CLI>
> > dunkel*CLI>
> > == Setting global variable 'SIPDOMAIN' to 'ringythingy.dev1ce.com'
> > -- Executing [blah@anveo_sip:1] Dial("PJSIP/demo-alice-0005",
> > "PJSIP/blah@mytrunk") in new stack
> > -- Called PJSIP/blah@mytrunk
> > -- PJSIP/mytrunk-0006 is ringing
> > -- PJSIP/mytrunk-0006 is ringing
> > -- PJSIP/mytrunk-0006 is making progress passing it to
> > PJSIP/demo-alice-0005
> >> 0x7ff39839e360 -- Strict RTP learning after remote address set
> > to: 72.9.156.128:52642
> >> 0x7ff3983994c0 -- Strict RTP learning after remote address set
> > to: [2605:e000:130a:fb:517d:7894:9482:c2bd]:54006
> > -- PJSIP/mytrunk-0006 is making progress passing it to
> > PJSIP/demo-alice-0005
> > == Everyone is busy/congested at this time (1:1/0/0)
> > -- Auto fallthrough, channel 'PJSIP/demo-alice-0005' status is
> > 'BUSY'
> >
> > Any idea what im doing wrong? Thanks :)
> >
>
> The remote side eventually terminated the call. You'd need to grab a SIP
> trace (pjsip set logger on) and provide/look at the actual traffic to see
> what is going on.
>
> Based on your version string I also don't believe you are on Asterisk 17,
> you appear to be on master which will become Asterisk 18.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
-- -- --
j...@dev1ce.com
https://dev1ce.com/john.gpg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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[asterisk-users] congested/busy on trunk?
greetings asterisk users :)
ive just deployed version 17 and migrated as best I can to pjsip. I can
receive calls, and get to my mailbox prompt, however placing calls seems
impossible with the following error on dial:
Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid =
517890)
dunkel*CLI>
dunkel*CLI>
== Setting global variable 'SIPDOMAIN' to 'ringythingy.dev1ce.com'
-- Executing [blah@anveo_sip:1] Dial("PJSIP/demo-alice-0005",
"PJSIP/blah@mytrunk") in new stack
-- Called PJSIP/blah@mytrunk
-- PJSIP/mytrunk-0006 is ringing
-- PJSIP/mytrunk-0006 is ringing
-- PJSIP/mytrunk-0006 is making progress passing it to
PJSIP/demo-alice-0005
> 0x7ff39839e360 -- Strict RTP learning after remote address set to:
72.9.156.128:52642
> 0x7ff3983994c0 -- Strict RTP learning after remote address set to:
[2605:e000:130a:fb:517d:7894:9482:c2bd]:54006
-- PJSIP/mytrunk-0006 is making progress passing it to
PJSIP/demo-alice-0005
== Everyone is busy/congested at this time (1:1/0/0)
-- Auto fallthrough, channel 'PJSIP/demo-alice-0005' status is 'BUSY'
Any idea what im doing wrong? Thanks :)
--
-- -- --
j...@dev1ce.com
https://dev1ce.com/john.gpg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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Re: [asterisk-users] Help with FUNC_MATH
My Apologies Dovid, I think I misunderstood your request.
You don't have the time you need to convert in the format of date string,
Instead you have your users entering via DTMF when they want something to
happen?
On Thu, Feb 13, 2020 at 11:08 AM Dovid Bender wrote:
> John,
>
> From looking at the wiki won't STRFIME just give me what I need based on
> the unix time that I put in? What I am actually looking to do is convert
> over from 12 hour format to 24 (unless strftime does just that and I don't
> kow what am I am doing?).
>
>
>
> On Thu, Feb 13, 2020 at 12:03 PM John Kiniston
> wrote:
>
>> Try using the STRFIME function instead of doing this by hand.
>>
>> https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME
>>
>> *%H*
>>
>> The hour as a decimal number using a 24-hour clock (range 00 to 23).
>>
>> *%I*
>>
>> The hour as a decimal number using a 12-hour clock (range 01 to 12).
>>
>> On Thu, Feb 13, 2020 at 3:49 AM Dovid Bender wrote:
>>
>>> Hi,
>>>
>>> I have some dialplan code that is trying to convert 12 hour time with
>>> AM/PM to 24 hour format. The code has something like this:
>>> Exten =>
>>> 2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>>>
>>> Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press
>>> option 2 they are selecting PM. If the time is from 1PM to 11PM then I want
>>> to add 12 to the number (so if it's 1 make it 13 etc.). When I run the
>>> above the logs show the result as false yet if the user sets HOUR_SELECTED
>>> to 12 then after this line of dialplan code it gets switched to 24. What am
>>> I doing wrong here?
>>>
>>> The exact DP code is:
>>> Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
>>> same =>n,
>>> ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>>> same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})
>>>
>>> And the output of the logs is:
>>> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
>>> [2@am_pm_select:1] NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK
>>> HOUR_SELECTED is 12") in new stack
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
>>> 'HOUR_SELECTED' is '12'
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
>>> MATH(12<12) result is 'FALSE'
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
>>> 'HOUR_SELECTED' is '12'
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
>>> MATH(12+12,int) result is '24'
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
>>> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
>>> [2@am_pm_select:2] ExecIf("SIP/204.145.219.31-81c6",
>>> "FALSE?Set(HOUR_SELECTED=24)") in new stack
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
>>> 'HOUR_SELECTED' is '24'
>>> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
>>> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
>>> [2@am_pm_select:3] NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK
>>> HOUR_SELECTED IS 24") in new stack
>>>
>>>
>>> TIA.
>>>
>>> Dovid
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> --
>> A human being should be able to change a diaper, plan an invasion,
>> butcher a hog, conn a ship, design a building, write a sonnet, balance
>> accounts, build a wall, set a bone, comfort the dying, take orders, give
>> orders, cooperate, act alone, solve equations, analyze a new problem, pitch
>> manure, program a computer, cook a tasty meal, fight efficiently, die
>> gallantly. Specialization is for insects.
>> ---Heinlein
>> --
>>
Re: [asterisk-users] Help with FUNC_MATH
Try using the STRFIME function instead of doing this by hand.
https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME
*%H*
The hour as a decimal number using a 24-hour clock (range 00 to 23).
*%I*
The hour as a decimal number using a 12-hour clock (range 01 to 12).
On Thu, Feb 13, 2020 at 3:49 AM Dovid Bender wrote:
> Hi,
>
> I have some dialplan code that is trying to convert 12 hour time with
> AM/PM to 24 hour format. The code has something like this:
> Exten =>
> 2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
>
> Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press
> option 2 they are selecting PM. If the time is from 1PM to 11PM then I want
> to add 12 to the number (so if it's 1 make it 13 etc.). When I run the
> above the logs show the result as false yet if the user sets HOUR_SELECTED
> to 12 then after this line of dialplan code it gets switched to 24. What am
> I doing wrong here?
>
> The exact DP code is:
> Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
> same =>n,
> ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
> same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})
>
> And the output of the logs is:
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:1] NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK
> HOUR_SELECTED is 12") in new stack
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '12'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
> MATH(12<12) result is 'FALSE'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '12'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function
> MATH(12+12,int) result is '24'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:2] ExecIf("SIP/204.145.219.31-81c6",
> "FALSE?Set(HOUR_SELECTED=24)") in new stack
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of
> 'HOUR_SELECTED' is '24'
> [Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
> [Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing
> [2@am_pm_select:3] NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK
> HOUR_SELECTED IS 24") in new stack
>
>
> TIA.
>
> Dovid
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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[asterisk-users] Modems
Guys, I have a customer that heavily uses modems, the problem they don't work reliably with some of the carriers I have used like Level3. This is somewhat expected due to the limits in VoIP so I need a better solution. If I set up an asterisk system on customer premise with an FXS card in it and have calls sent to another asterisk box with a PRI can I get this to be more reliable and better connect speeds.? Any way to detect a modem call and turn off the echo canceller? What if I use a lossless codec between the two systems, or would it be better to just run PCM to passthrough to the PRI. Any ideas would be helpful. Thanks John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Site to site VPN problems
Ira, What version of Asterisk are you using, and what channel driver? There has to be a better way than to create hundreds of peer entries. On Thu, Dec 12, 2019 at 12:26 PM Ira wrote: > Hello Jan, > > Tuesday, December 3, 2019, 8:49:28 PM, you wrote: > > Jan> The next thing to look at is firewall rules. > > > So it wasn't the firewall. I eventually fixed it by creating 512 entries > for Twilio so no matter what IP they sent it from it had a peer to match. > They seem to randomly use one of their range of IPs when sending me calls. > I had allowed calls from the suggested. > > MyDomain.pstn.twilio.com > > Which worked for about 1 out of 10 or 20 calls, as soon as I added all 512 > possible Twilio IPs it all started working fine. Before this I had only 10 > Peers and now I have 520 which is really annoying as now the sip show inuse > and sip show peers commands are essentially useless because there is to > much data. > > Anyway, thanks so much for trying to help. I found a hint somewhere on the > web and then learned how to do templates which at least makes sip.conf > reasonable with all those entries. > > And if you know of a way to make one peer accept a range of IPs, Id love > to know that. > > -- Ira > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Site to site VPN problems
You only need to add the Signaling IPs as peers. The Media IPs only need to be whitelisted in your firewall. Furthermore, you can actually trim the list down to only one region by following their instructions here https://www.twilio.com/docs/sip-trunking#OriginationURI-region On Thu, 12 Dec 2019 at 13:26, Ira wrote: > Hello Jan, > > Tuesday, December 3, 2019, 8:49:28 PM, you wrote: > > Jan> The next thing to look at is firewall rules. > > > So it wasn't the firewall. I eventually fixed it by creating 512 entries > for Twilio so no matter what IP they sent it from it had a peer to match. > They seem to randomly use one of their range of IPs when sending me calls. > I had allowed calls from the suggested. > > MyDomain.pstn.twilio.com > > Which worked for about 1 out of 10 or 20 calls, as soon as I added all 512 > possible Twilio IPs it all started working fine. Before this I had only 10 > Peers and now I have 520 which is really annoying as now the sip show inuse > and sip show peers commands are essentially useless because there is to > much data. > > Anyway, thanks so much for trying to help. I found a hint somewhere on the > web and then learned how to do templates which at least makes sip.conf > reasonable with all those entries. > > And if you know of a way to make one peer accept a range of IPs, Id love > to know that. > > -- Ira > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, John Runyon Simply NUC 512-766-0401 x1110 495 Round Rock West Dr, Round Rock, TX 78681 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR extract call numbers on interval on unique callers
https://dev.mysql.com/doc/refman/8.0/en/group-by-functions.html#function_count-distinct Use something like count(distinct src) instead of count(*) On Tue, Nov 12, 2019, 07:35 Andre Gronwald wrote: > hi, > > we want to extract the information when the most callers are entering > our phone system based on an interval of 15 minutes. this is quite > simple (although not perfect) with > select calldate, count(*) as anzahl from cdr where calldate > > '2019-10-12' group by unix_timestamp(calldate) DIV 900 having ; > > Unfortunately we have lots of callers who calls multiple times when they > are forwarded to a queue instead of being answered by a human > immediately. But to know when we need more people I want to count same > caller-ids within an interval as one call. > > Any ideas how to do this? > > kind regards, > andre > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up ODBC for cdr logging into MariaDB
I think you are missing a package, you need the odbc driver from mariadb, downloaded from their git repository -- if you build this using the default installation on a Debian type system, you would get /usr/local/lib64/libmaodbc.so as the driver file. On Fri, 11 Oct 2019 22:12:08 -0400, Fourhundred Thecat wrote: > > Hello, > > I am trying to set up cdr logging into MariaDB through ODBC. > > I have installed unixodbc unixodbc-dev and now I am struggling with > configuring /etc/odbcinst.ini > > All the examples online use non-existent libraries, ie: > > [MySQL] > Description = MySQL ODBC MyODBC Driver > Driver = /usr/lib/x86_64-linux-gnu/odbc/libmaodbc.so > Setup = /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so > FileUsage = 1 > > I have these odbc related libraries on my system. Which of those do I > have to use for `Driver =` ? > > /usr/lib/x86_64-linux-gnu/libodbc.so > /usr/lib/x86_64-linux-gnu/libodbccr.so > /usr/lib/x86_64-linux-gnu/libodbcinst.so > > /usr/lib/x86_64-linux-gnu/odbc/libesoobS.so > /usr/lib/x86_64-linux-gnu/odbc/libmimerS.so > /usr/lib/x86_64-linux-gnu/odbc/libnn.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcdrvcfg1S.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcdrvcfg2S.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcminiS.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcnnS.so > /usr/lib/x86_64-linux-gnu/odbc/libodbcpsqlS.so > /usr/lib/x86_64-linux-gnu/odbc/libodbctxtS.so > /usr/lib/x86_64-linux-gnu/odbc/liboplodbcS.so > /usr/lib/x86_64-linux-gnu/odbc/liboraodbcS.so > /usr/lib/x86_64-linux-gnu/odbc/libsapdbS.so > /usr/lib/x86_64-linux-gnu/odbc/libtdsS.so > > I have tries many possible permutations, but none worked. > > thanks, > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with new install with asterisk 15.7.4
hmmm, is asterisk 16 long term support? I thought only the od numbered releases were long term support. On Mon, 07 Oct 2019 08:02:51 -0400, George Joseph wrote: > > [1 ] > [2 ] > Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday. :) > You should use Asterisk 16. > > On Mon, Oct 7, 2019 at 5:58 AM George Joseph wrote: > > On Fri, Oct 4, 2019 at 1:19 PM John Covici wrote: > > Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 > system and I am running into the following problem. I need to install > meetme (I know its old), and I have dahdi installed and the configure > script answers yes to all the edahdi questions, but the app_meetme > says depends on dahdi (e). I did not install libpri as I have no > hardware of that type. > > The (E) means "external" not "error". Does the app_meetme entry in > menuselect have "[ ]" before it or "XXX"? > If "[ ]" you should be able to select it and build. > > > I installed dahdi from git and have the kernel sources and it > installed without errors. > > How can I fix? > > Thanks in advance for any suggestions. > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > cov...@ccs.covici.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: >https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > George Joseph > Digium - A Sangoma Company | Software Developer | Software Engineering > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct/fax: +1 256 428 6012 > Check us out at: https://digium.com · https://sangoma.com > > * > > -- > George Joseph > Digium - A Sangoma Company | Software Developer | Software Engineering > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > direct/fax: +1 256 428 6012 > Check us out at: https://digium.com · https://sangoma.com > > * -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with new install with asterisk 15.7.4
Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 system and I am running into the following problem. I need to install meetme (I know its old), and I have dahdi installed and the configure script answers yes to all the edahdi questions, but the app_meetme says depends on dahdi (e). I did not install libpri as I have no hardware of that type. I installed dahdi from git and have the kernel sources and it installed without errors. How can I fix? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone ever experienced a crash where Asterisk debug output a line with all nulls
Agree with Michael, this sounds like an OS crash to me. Given that you're not seeing anything in the logs, it's likely that by the time the crash occurs, the kernel is unable to write to disk, so the only real way to get info is to be looking at the console when it crashes. See also: https://help.ubuntu.com/community/DebuggingSystemCrash The initial slowness is likely due either to caching on later restarts, or to fsck. On Wed, 14 Aug 2019 at 10:23, Michael Maier wrote: > On 14.08.19 at 16:26 Dan Cropp wrote: > > We have a customer where their VM running Asterisk appears to have > crashed. Fortunately, we had some debugging enabled. > > The asterisk messages file has this... (in notepad+ the blank line in > the middle is all [NUL][NUL] [NUL][NUL]) > > > > [08/12 15:30:55.880] VERBOSE[6920] app_mixmonitor.c: Begin MixMonitor > Recording CBRec/IS__a37ae004-c780-4c7f-88a9-a04402f0ab4e-e70f > > [08/12 15:30:55.881] VERBOSE[6921] bridge_channel.c: Channel > CBRec/IS__a37ae004-c780-4c7f-88a9-a04402f0ab4e-e70f joined 'softmix' > base-bridge <23340bca-6823-4c70-a395-e3b092aeb671> > > > > > > > > > > > > [08/12 15:33:02.887] Asterisk 16.2.1 built by root @ sw-genesis-build4 > on a x86_64 running Linux on 2019-04-04 13:41:15 UTC > > Did I get it correctly: the last line in the log before asterisk starts > again after the reboot is the [nul]-line? If so, are you probably using > ext4 or maybe reiserfs? I know of this problem if the machine crashes, open > files can end up > like described. AFAIK it's a file system bug! > > > We also had debugging enabled and things were output to our debug file > for 17 more seconds > > The blank line in my e-mail is once again a line with all [NUL}... (in > notepad+ the blank line in the middle is all [NUL][NUL] [NUL][NUL]) > > > > [08/12 15:31:12.776] DEBUG[6781] audiohook.c: Read factory > 0x7f079389bff8 and write factory 0x7f079389ca38 both fail to provide 160 > samples > > [08/12 15:31:12.777] DEBUG[6709] audiohook.c: Failed to get 160 samples > from read factory 0x7f07937066d8 > > [08/12 15:31:12.777] DEBUG[6709] audiohook.c: Read factory > 0x7f07937066d8 and write factory 0x7f0793707118 both fail to provide 160 > samples > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > [08/12 > 15:33:02.915] Asterisk 16.2.1 built by root @ sw-genesis-build4 on a x86_64 > running Linux on 2019-04-04 13:41:15 UTC > > [Aug 12 15:33:02] DEBUG[1385] config.c: Parsing > /etc/iss/asterisk/logger.conf > > > > > > > > I believe this was bad enough that Ubuntu actually crashed, but there is > nothing in the syslog indicating anything until 15:32:42 where it appears > Linux is starting up. > > > > > > After this situation happens, every time Asterisk starts up, it was > taking significantly longer to load. Normally 1-2 seconds, became 26-28 > seconds. > > [08/12 15:33:03.240] NOTICE[1385] loader.c: 286 modules will be loaded. > > [08/12 15:33:23.844] VERBOSE[1385] loader.c: Loading extconfig. > > > > > > Loading the modules is taking 20 seconds after this incident occurred. > Looking at the debug logs, I see the modules loading loader.c PASS. There > all seem to load fine, just much slower than it was previously. > > Maybe because the machine is performing a file system check on some other > partitions in parallel and it's slowed down therefore? > > > Regards > Michael > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, John Runyon Simply NUC 512-766-0401 x1110 495 Round Rock West Dr, Round Rock, TX 78681 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lightweight ODBC DB
On Thu, 1 Aug 2019 at 17:07, Doug Lytle wrote: > On 8/1/19 5:08 PM, Dovid Bender wrote: > > Glenn, > > I can't use MySQL as each node currently has MySQL however there is a lot > of data that is stored locally on each box. I may have to take this route > if I can't find something else but that would mean syncing all sorts of > data that does not need to be synced. > > > If I recall correctly, you can exclude databases. > > Doug > https://dev.mysql.com/doc/refman/5.7/en/replication-solutions-partitioning.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] svnview.digium.com down?
Works for me from Comcast! John Novack Doug Lytle wrote: I'm currently reviewing the Digium wiki on migrating from chan_sip to res_pjip and I'm trying to access the script that is provided to help with conversion. https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip It would appear that said server hosting the script is no responding or the link is no longer valid. Doug -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Better audio in than just 8k
Jerry, What if you specify a higher bitrate to mpg123? You are limiting it to 8k with the 'r' option. I convert my source audio files with sox to 16khz signed linear for wideband hold music. sox -c1 hold.wav -r 16000 -c 1 -e signed-integer -r 16k hold.raw Then I rename the .raw file to a .sln16 and place it in the moh directory. On Thu, Jul 11, 2019 at 11:21 AM Jerry Geis wrote: > >>Maybe streaming will be helpful, > > >> > https://www.agix.com.au/streaming-internet-music-for-asterisk-10-on-hold-music/ > > >>Doug > > I had played with that also - but it still down converts to 8k: > application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 -@ > http://media.on.net/radio/155.m3u > > I was looking for a way to get something like an opus codec quality - not > just 8k. > > Jerry > >> -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unsolved: Re: solved: how to create a working certificate for using TLS?
On Fri, 5 Jul 2019 at 14:28, hw wrote: > I thought about that and checked the configuration I've been using to > create the certificate, and I can't see anywhere that it would expire > earlier than after 3650 days. Is there another way to check this? > openssl verify -CAfile ca.crt server.crt Which certificate is the one that can not be verified: the one I > created or the one used by the SIP provider? How can I find out > which certificate the error message is referring to? > What is the error message? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking Asterisk SIP Guru
Joshua, Thanks for looking into this, and sorry for not being more detailed. Running asterisk 16.4.0 I was able to get in touch with an AIphone tech and it turns out that these issues are known bug on their side. I will be more detailed next time Thanks John Bittner Xaccel -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua C. Colp Sent: Thursday, June 27, 2019 10:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Looking Asterisk SIP Guru On Thu, Jun 27, 2019, at 11:28 AM, John T. Bittner wrote: > > Hello, > > > I am looking for a consultant that know asterisk in and out including > how to troubleshoot sip and rtp. > > I have a device that this acting very strange and I need to prove it’s > the device code and not an issue with my setup. > > > Very simple setup, all local no nat… Grandstream video phone and a > AIphone IX-MX7 door station. > > > PJSIP … doorstation to grandstream 3370 works perfectly. Early video > works as well. > > PJSIP … grandtream to doorstation I get a error from the doorstation I > get You didn't provide the IP addresses of things involved, so anyone looking at the packet captures has to look in and decipher what is what which may be why noone on here has responded as of yet. The user agent of Asterisk is also changed so that confused things some for me to until I double checked the SDP and saw it's Asterisk. Asterisk is sending a re-invite to 192.168.1.10 as an attempt to make both the audio and video streams bidirectional. The device at 192.168.1.10 is rejecting this with a 400 Bad Request. It should respond either with a 200 OK with an SDP answer of the state of the streams, or it should respond with a 488 Not Acceptable. Both of these would keep the call up and the appropriate stream would probably flow although I haven't tested this particular usage. You also didn't specify an Asterisk version from what I can see, and stream behavior between 13 and 16 differs (as 16 understands streams) which could contribute to the behavior. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BEGIN-ANTISPAM-VOTING-LINKS -- Teach Canit xAntispam if this mail is spam: Spam: http://mx1.xantispam.net/canit/b.php?c=s=020tOGuhK=13b91acb500d=xaccel-net Not spam: http://mx1.xantispam.net/canit/b.php?c=n=020tOGuhK=13b91acb500d=xaccel-net Forget vote: http://mx1.xantispam.net/canit/b.php?c=f=020tOGuhK=13b91acb500d=xaccel-net -- END-ANTISPAM-VOTING-LINKS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking Asterisk SIP Guru
Hello, I am looking for a consultant that know asterisk in and out including how to troubleshoot sip and rtp. I have a device that this acting very strange and I need to prove it’s the device code and not an issue with my setup. Very simple setup, all local no nat… Grandstream video phone and a AIphone IX-MX7 door station. PJSIP … doorstation to grandstream 3370 works perfectly. Early video works as well. PJSIP … grandtream to doorstation I get a error from the doorstation I get SIP/2.0 400 Bad Request To: ;tag=ec09c0b4zps4.0.0 From: "108";tag=bcaee3d1-d6d1-4354-aee1-885d4b89182d Via: SIP/2.0/UDP 192.168.1.154:5060;branch=z9hG4bKPj038d15bb-c53b-4677-a471-4fa44a21599b;rport Call-ID: 08caa86d-8fc8-4ed1-bec2-6828acb9e017 CSeq: 17397 INVITE Content-Length: 0 x-reinvitekind: mediadirectionchange Tried a few things, I still don’t understand why I am getting this, I cannot find it coming from the asterisk system or the Grandstream in my traces. So Switch the Aiphone to use chan_sip on port 5099 just to test. Again SIP … doorstation to PJSIP grandstream 3370 works perfectly. Early video works as well. PJSIP … granstream to SIP doorstation works somewhat, I get early video but no audio. If I answer the doorstation before the early video pops up, I get the window in the doorstation that allows me to put a call on hold. When I do, and take back off hold, I get audio. If I wait for early video on the doorstation and then answer it, the door station never comes up with the menus to put a call on hold. So no audio. Anyone have any ideas or willing to do some consulting work please let me know asap. FYI some captures are attached. Thanks John Bittner CTO 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. 0¿ ª0NEiJ@@LÀ¨ À¨ÄÄ}ª!SIP/2.0 400 Bad Request To: ;tag=ec09c0b4zps4.0.0 From: "108";tag=bcaee3d1-d6d1-4354-aee1-885d4b89182d Via: SIP/2.0/UDP 192.168.1.154:5060;branch=z9hG4bKPj038d15bb-c53b-4677-a471-4fa44a21599b;rport Call-ID: 08caa86d-8fc8-4ed1-bec2-6828acb9e017 CSeq: 17397 INVITE Content-Length: 0 x-reinvitekind: mediadirectionchange -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hacking
Just to jump in on this, this just started happening to our system a couple days ago. (To the tune of 3GB of webserver access logs yesterday) Our server gives them a 403 for /yealink/ (and a 404 for everything else) - given that they're still trying to bruteforce it, it looks like I'm gonna be changing it to give them a 404. Looks like someone's making a big effort to find provisioning files though. On Mon, Jun 17, 2019, 13:35 John Kiniston wrote: > > > On Sun, Jun 16, 2019 at 3:37 PM John T. Bittner wrote: > >> Anyone know how someone can hack an asterisk box and register with every >> single account on the box. >> >> This box only has 3 accounts, with very complex passwords. Have VoIP >> blacklist setup and fail2ban… >> > > I've seen this happen when web-based provisioning is used, I have seen > attempts to download configuration files off of my provisioning server > increase in frequency over the last two years. > > The 'Hacker' will do a get on /polycom /cisco /yealink /aastra /mitel etc, > If they get a valid response they will start enumerating mac addresses > > /polycom/0004F2018101.cfg > /polycom/0004F2018102.cfg > ... > /polycom/0004F2018109.cfg > > Then they will use any credentials gained in the download attack to place > calls, registering as needed. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hacking
On Sun, Jun 16, 2019 at 3:37 PM John T. Bittner wrote: > Anyone know how someone can hack an asterisk box and register with every > single account on the box. > > This box only has 3 accounts, with very complex passwords. Have VoIP > blacklist setup and fail2ban… > I've seen this happen when web-based provisioning is used, I have seen attempts to download configuration files off of my provisioning server increase in frequency over the last two years. The 'Hacker' will do a get on /polycom /cisco /yealink /aastra /mitel etc, If they get a valid response they will start enumerating mac addresses /polycom/0004F2018101.cfg /polycom/0004F2018102.cfg ... /polycom/0004F2018109.cfg Then they will use any credentials gained in the download attack to place calls, registering as needed. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hacking
I took a look for that, Mysql running but blocked in the firewall. I do have a web gui but its hides the passwords + has a single login for admin with complex password. Even if they hacked the web site, they have no way of getting the passwords my configs are static in the asterisk folder. SSH is blocked. Logs do not show any http access, secure or any other fingerprints. I am going to honeypot this box to see if I can capture there invites. John Xaccel From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender Sent: Sunday, June 16, 2019 6:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hacking John, There are a lot of factors at play for instance are you using a gui that has a known vlun? Is there mysql running on the box with a simple password? Perhaps they didnt hack your PBX but they comprised a SIP phone and once they had the credentials they made calls? Do you have a provisioning system? We have seen all of the above. Most of the compromises we are seeing these days is either via a Provisioning server or phones that are accessible on the internet with weak passwords Regards, Dovid From: j...@xaccel.net<mailto:j...@xaccel.net> Sent: June 16, 2019 18:37 To: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com> Reply-to: asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com> Subject: [asterisk-users] Hacking Anyone know how someone can hack an asterisk box and register with every single account on the box. This box only has 3 accounts, with very complex passwords. Have VoIP blacklist setup and fail2ban… The hackers were able to make 2 calls to Cuba before my alerting system texted me. I am running asterisk 16.3 with PJSIP. This is my only box open to the outside world, a requirement for this one customer. Looked into my logs… can't find anything out of the ordinary. Any ideas ? Contact: == Contact: 12120001001/sip:12120001001@5.79.64.23<mailto:12120001001@5.79.64.23>:9227 ee80678930 NonQual nan Contact: 848842405/sip: 848842405@5.79.64.23<mailto:848842405@5.79.64.23>:9227 031ed703ba NonQual nan Contact: 848842405/sip: 848842405@5.79.64.23<mailto:848842405@5.79.64.23>:9227 031ed703ba NonQual nan Contact: ghbhhm/sip:ghbhhm@5.79.64.23<mailto:ghbhhm@5.79.64.23>:9227 959fc8fbf4 NonQual nan Contact: ghbhhm/sip:ghbhhm@5.79.64.23<mailto:ghbhhm@5.79.64.23>:9227 959fc8fbf4 NonQual nan Contact: ghbhhm/sip:ghbhhm@5.79.64.23<mailto:ghbhhm@5.79.64.23>:9228 d7bf838918 NonQual nan Contact: ghbhhm/sip:ghbhhm@5.79.64.23<mailto:ghbhhm@5.79.64.23>:9228 d7bf838918 NonQual nan Any helps is much appreciated. John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. Teach Canit xAntispam if this mail is spam: Spam<http://mx1.xantispam.net/canit/b.php?c=s=020pz0aHc=a5b99ef03d9e=xaccel-net> Not spam<http://mx1.xantispam.net/canit/b.php?c=n=020pz0aHc=a5b99ef03d9e=xaccel-net> Forget previous vote<http://mx1.xantispam.net/canit/b.php?c=f=020pz0aHc=a5b99ef03d9e=xaccel-net> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hacking
Anyone know how someone can hack an asterisk box and register with every single account on the box. This box only has 3 accounts, with very complex passwords. Have VoIP blacklist setup and fail2ban... The hackers were able to make 2 calls to Cuba before my alerting system texted me. I am running asterisk 16.3 with PJSIP. This is my only box open to the outside world, a requirement for this one customer. Looked into my logs... can't find anything out of the ordinary. Any ideas ? Contact: == Contact: 12120001001/sip:12120001001@5.79.64.23:9227ee80678930 NonQual nan Contact: 848842405/sip: 848842405@5.79.64.23:9227 031ed703ba NonQual nan Contact: 848842405/sip: 848842405@5.79.64.23:9227 031ed703ba NonQual nan Contact: ghbhhm/sip:ghbhhm@5.79.64.23:9227 959fc8fbf4 NonQual nan Contact: ghbhhm/sip:ghbhhm@5.79.64.23:9227 959fc8fbf4 NonQual nan Contact: ghbhhm/sip:ghbhhm@5.79.64.23:9228 d7bf838918 NonQual nan Contact: ghbhhm/sip:ghbhhm@5.79.64.23:9228 d7bf838918 NonQual nan Any helps is much appreciated. John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fail2ban for asterisk 16 PJSIP
Hopefully, this helps someone else. This seems to be working for me. # Fail2Ban configuration file [INCLUDES] #before = common.conf [Definition] failregex = NOTICE.* .*: Request \'REGISTER\' from '.*' failed for ':.*' .* - No matching endpoint found NOTICE.* .*: Request \'REGISTER\' from '.*' failed for ':.*' .* - Failed to authenticate NOTICE.* .*: Request \'REGISTER\' from '.*' failed for ':.*' .* - Error to authenticate NOTICE.* .*: Request \'INVITE\' from '.*' failed for ':.*' .* John Bittner Xaccel From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John T. Bittner Sent: Thursday, June 6, 2019 3:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fail2ban for asterisk 16 PJSIP Hello Anyone have a working copy of Fail2ban asterisk filter asterisk.conf for Asterisk 16 running PJSIP. I have tried 10 different filters but none of them show any matches when testing with fail2ban-regex I see date template hits but no matches My log [2019-06-06 15:37:20] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 50670137772977-30593645157868@192.168.1.8<mailto:50670137772977-30593645157868@192.168.1.8>) - Failed to authenticate [2019-06-06 15:37:52] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '"as100" ' failed for '188.214.128.172:5076' (callid: 03e7f9d2dcdf4252506c440137e822b7) - No matching endpoint found [2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352844365933467-383842003849650@192.168.1.8<mailto:352844365933467-383842003849650@192.168.1.8>) - Failed to authenticate [2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352844365933467-383842003849650@192.168.1.8<mailto:352844365933467-383842003849650@192.168.1.8>) - Failed to authenticate [2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352844365933467-383842003849650@192.168.1.8<mailto:352844365933467-383842003849650@192.168.1.8>) - Failed to authenticate [2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352844365933467-383842003849650@192.168.1.8<mailto:352844365933467-383842003849650@192.168.1.8>) - Failed to authenticate [2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352413680053562-322991201237060@192.168.1.8<mailto:352413680053562-322991201237060@192.168.1.8>) - Failed to authenticate [2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352413680053562-322991201237060@192.168.1.8<mailto:352413680053562-322991201237060@192.168.1.8>) - Failed to authenticate [2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352413680053562-322991201237060@192.168.1.8<mailto:352413680053562-322991201237060@192.168.1.8>) - Failed to authenticate [2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352413680053562-322991201237060@192.168.1.8<mailto:352413680053562-322991201237060@192.168.1.8>) - Failed to authenticate [2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 211973110361898-30014604441241@192.168.1.8<mailto:211973110361898-30014604441241@192.168.1.8>) - Failed to authenticate [2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 211973110361898-30014604441241@192.168.1.8<mailto:211973110361898-30014604441241@192.168.1.8>) - Failed to authenticate [2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 211973110361898-30014604441241@192.168.1.8<mailto:211973110361898-30014604441241@192.168.1.8>) - Failed to authenticate [2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 211973110361898-30014604441241@192.168.1.8<mailto:211973110361898-30014604441241@192.168.1.8>) - Failed to authenticate [2019-06-06 15:39:17] NOTICE[18081]
Re: [asterisk-users] error compiling dahdi for recent kernels
Too bad.
LOTS of users will still want to continue to use these cards, example the OP!
Good news it probably suppresses prices on used cards!
John Novack
Malcolm Davenport wrote:
Howdy,
That is correct.
The list of supported cards is in the README file (not the -complete package
README, but the dahdi-linux README)
Cheers
On Thu, Jun 6, 2019 at 2:52 PM John Novack SCII_U mailto:jnov...@comcast.net>> wrote:
Doesn't DAHDI 3.0 remove support for a bunch of older cards, including the
TDM400 and 410?
John Novack
Greg Woods wrote:
On Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport mailto:malco...@sangoma.com>> wrote:
Howdy,
There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz.
Try that.
I noticed that was there, but I didn't try it originally because it's
obviously a beta version. However, I did download it and try it. It does
compile, but doesn't work correctly. For one thing, it thinks my Digium card is
an Ethernet controller:
# lspci | grep Digium
07:01.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog
card (rev 11)
Attempting to start the dahdi service results in:
Short version:Jun 06 13:11:38 worldsys.gregandeva.net
<http://worldsys.gregandeva.net> sh[1026]: using
'/etc/dahdi/assigned-spans.conf'
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
sh[1026]: DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22)
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
sh[1026]: Selected signaling not supported
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
sh[1026]: Possible causes:
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
sh[1026]: FXO signaling is being used on a FXO interface (use a FXS signaling
variant)
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
sh[1026]: RBS signaling is being used on a E1 CCS span
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
sh[1026]: Signaling is being assigned to channel 16 of an E1 CAS span
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
systemd[1]: dahdi.service: Main process exited, code=exited, status=1/FAILURE
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
systemd[1]: dahdi.service: Failed with result 'exit-code'.
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
systemd[1]: Failed to start The DAHDI drivers allow you to use your linux computer to
accept incoming data and voice interfaces.
(The assigned-spans.conf file has nothing in it but comments)
Long version:
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable
to specify channel 1: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
asterisk[1333]: [Jun 6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to
open channel 1: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
asterisk[1333]: [Jun 6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels:
Unable to register channel '1'
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi:
Channel '1' failure ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable
to specify channel 2: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
asterisk[1333]: [Jun 6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to
open channel 2: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
asterisk[1333]: [Jun 6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels:
Unable to register channel '2'
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi:
Channel '2' failure ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable
to specify channel 3: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net>
asterisk[1333]: [Jun 6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to
open channel 3: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net&
Re: [asterisk-users] error compiling dahdi for recent kernels
Doesn't DAHDI 3.0 remove support for a bunch of older cards, including the TDM400 and 410? John Novack Greg Woods wrote: On Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport mailto:malco...@sangoma.com>> wrote: Howdy, There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz. Try that. I noticed that was there, but I didn't try it originally because it's obviously a beta version. However, I did download it and try it. It does compile, but doesn't work correctly. For one thing, it thinks my Digium card is an Ethernet controller: # lspci | grep Digium 07:01.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) Attempting to start the dahdi service results in: Short version:Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> sh[1026]: using '/etc/dahdi/assigned-spans.conf' Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> sh[1026]: DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> sh[1026]: Selected signaling not supported Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> sh[1026]: Possible causes: Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> sh[1026]: FXO signaling is being used on a FXO interface (use a FXS signaling variant) Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> sh[1026]: RBS signaling is being used on a E1 CCS span Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> sh[1026]: Signaling is being assigned to channel 16 of an E1 CAS span Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> systemd[1]: dahdi.service: Main process exited, code=exited, status=1/FAILURE Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> systemd[1]: dahdi.service: Failed with result 'exit-code'. Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> systemd[1]: Failed to start The DAHDI drivers allow you to use your linux computer to accept incoming data and voice interfaces. (The assigned-spans.conf file has nothing in it but comments) Long version: Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 1: Invalid argument Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 1: Invalid argument Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register channel '1' Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '1' failure ignored: ignore_failed_channels. Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 2: Invalid argument Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 2: Invalid argument Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register channel '2' Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '2' failure ignored: ignore_failed_channels. Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 3: Invalid argument Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 3: Invalid argument Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register channel '3' Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '3' failure ignored: ignore_failed_channels. Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun 6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 4: Invalid argument Jun 06 13:11:42 worldsys.gregandeva.net <ht
[asterisk-users] Fail2ban for asterisk 16 PJSIP
Hello Anyone have a working copy of Fail2ban asterisk filter asterisk.conf for Asterisk 16 running PJSIP. I have tried 10 different filters but none of them show any matches when testing with fail2ban-regex I see date template hits but no matches My log [2019-06-06 15:37:20] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 50670137772977-30593645157868@192.168.1.8) - Failed to authenticate [2019-06-06 15:37:52] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '"as100" ' failed for '188.214.128.172:5076' (callid: 03e7f9d2dcdf4252506c440137e822b7) - No matching endpoint found [2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352844365933467-383842003849650@192.168.1.8) - Failed to authenticate [2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352844365933467-383842003849650@192.168.1.8) - Failed to authenticate [2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352844365933467-383842003849650@192.168.1.8) - Failed to authenticate [2019-06-06 15:37:58] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352844365933467-383842003849650@192.168.1.8) - Failed to authenticate [2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352413680053562-322991201237060@192.168.1.8) - Failed to authenticate [2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352413680053562-322991201237060@192.168.1.8) - Failed to authenticate [2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352413680053562-322991201237060@192.168.1.8) - Failed to authenticate [2019-06-06 15:38:36] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 352413680053562-322991201237060@192.168.1.8) - Failed to authenticate [2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 211973110361898-30014604441241@192.168.1.8) - Failed to authenticate [2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 211973110361898-30014604441241@192.168.1.8) - Failed to authenticate [2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 211973110361898-30014604441241@192.168.1.8) - Failed to authenticate [2019-06-06 15:39:14] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" ' failed for '71.127.239.22:65476' (callid: 211973110361898-30014604441241@192.168.1.8) - Failed to authenticate [2019-06-06 15:39:17] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '"as100" ' failed for '188.214.128.172:5071' (callid: 8e12f1560bfe2c3ed5be895108727c46) - No matching endpoint found Any help is much appreciated. Thanks John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Account code PJSIP
Hopefully, this may help someone in the future. If I set this before I dial out... it works. I have always in the past set this on hangup... that does not work anymore. John Xaccel From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John T. Bittner Sent: Wednesday, May 1, 2019 12:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Account code PJSIP Does anyone know how to set accountcode into the asterisk CDR when using PJSIP? I have tried Set(CHANNEL(accountcode)=XX) and a few other older ways... nothing works. If I add accountcode into the pjsip endpoint config it works... but I need to set it via dialplan. Any help is much appreciated. Testing on asterisk 16.3.0 John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. Teach Canit xAntispam if this mail is spam: Spam<http://mx1.xantispam.net/canit/b.php?c=s=0206R70lG=6ea45c5ca3d8=xaccel-net> Not spam<http://mx1.xantispam.net/canit/b.php?c=n=0206R70lG=6ea45c5ca3d8=xaccel-net> Forget previous vote<http://mx1.xantispam.net/canit/b.php?c=f=0206R70lG=6ea45c5ca3d8=xaccel-net> -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Account code PJSIP
Does anyone know how to set accountcode into the asterisk CDR when using PJSIP? I have tried Set(CHANNEL(accountcode)=XX) and a few other older ways... nothing works. If I add accountcode into the pjsip endpoint config it works... but I need to set it via dialplan. Any help is much appreciated. Testing on asterisk 16.3.0 John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use of the intended recipient(s) and may contain confidential and privileged information which should not be shared or forwarded. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply e-mail and destroy all copies of the e-mail. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
