Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7
On Tuesday 10 Jan 2017, Olivier wrote:
> Hello,
>
> I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes.
>
> I followed this:
> cd /usr/src
> wget ... asterisk-13.13.1.tar.gz
> tar zxf asterisk-13.13.1.tar.gz
> cd asterisk-13.13.1
> ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/usr"
> ./configure ${ASTERISK_CONFIGURE} --with-pjproject-bundled
> make menuselect (shows res-srtp is available)
> make
>
> latest make command fails with (see [1]:
>
>[GENERATE] libasteriskpj.exports
>[LD] libasteriskpj.o -> libasteriskpj.so.2
> /bin/ld: cannot find -lsrtp-x86_64-unknown-linux-gnu
You probably need to install a package called something like "srtp-devel" or
"libsrtp-devel". (I'm not certain of the exact package name on a Fedora-like
system, as I'm coming from Ubuntuland ;) ) Re-run the configure script after
installing it.
Asterisk always used (pre-10!) to be distributed with a script called
"install_prereq" which was meant to install all the necessary packages for a
full install with everything, at least on a on Fedora-like or Debian-like
distro; not sure if this is still included anymore, as it seemed a bit flaky.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7
2017-01-10 5:30 GMT+01:00 Anton Teyhrib :
> Maybe libsrtp package is missing on your machine?
>
libsrtp-devel was installed and I could check it was installed with
menuselect (Resources Modules/res_srtp)
> 10 янв. 2017 г. 5:16 AM пользователь "Olivier"
> написал:
>
>> Hello,
>>
>> I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes.
>>
>> I followed this:
>> cd /usr/src
>> wget ... asterisk-13.13.1.tar.gz
>> tar zxf asterisk-13.13.1.tar.gz
>> cd asterisk-13.13.1
>> ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/usr"
>> ./configure ${ASTERISK_CONFIGURE} --with-pjproject-bundled
>> make menuselect (shows res-srtp is available)
>> make
>>
>> latest make command fails with (see [1]:
>>
>>[GENERATE] libasteriskpj.exports
>>[LD] libasteriskpj.o -> libasteriskpj.so.2
>> /bin/ld: cannot find -lsrtp-x86_64-unknown-linux-gnu
>>
>>
>> So I also build a simple VM box in which I also install CentOS and
>> Asterisk 13.3.1 but I could run the above instructions successfully.
>>
>>
>> Suggestions ?
>>
>> Best regards
>>
>> [1] Original error message was localised.
>>
>> --
>> _
>> -- 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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> _
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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
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7
Historically, I didn't use "install_prereq" but I also used it yesterday.
As make fails with "[LD] libasteriskpj.o -> libasteriskpj.so.2" which is
the first of its "kind", I still wonder
if issue comes from libsrtp somehow, or from a missing option in
./configure (such as --prefix).
2017-01-10 10:31 GMT+01:00 A J Stiles :
> On Tuesday 10 Jan 2017, Olivier wrote:
> > Hello,
> >
> > I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes.
> >
> > I followed this:
> > cd /usr/src
> > wget ... asterisk-13.13.1.tar.gz
> > tar zxf asterisk-13.13.1.tar.gz
> > cd asterisk-13.13.1
> > ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/usr"
> > ./configure ${ASTERISK_CONFIGURE} --with-pjproject-bundled
> > make menuselect (shows res-srtp is available)
> > make
> >
> > latest make command fails with (see [1]:
> >
> >[GENERATE] libasteriskpj.exports
> >[LD] libasteriskpj.o -> libasteriskpj.so.2
> > /bin/ld: cannot find -lsrtp-x86_64-unknown-linux-gnu
>
> You probably need to install a package called something like "srtp-devel"
> or
> "libsrtp-devel". (I'm not certain of the exact package name on a
> Fedora-like
> system, as I'm coming from Ubuntuland ;) ) Re-run the configure script
> after
> installing it.
>
> Asterisk always used (pre-10!) to be distributed with a script called
> "install_prereq" which was meant to install all the necessary packages for
> a
> full install with everything, at least on a on Fedora-like or Debian-like
> distro; not sure if this is still included anymore, as it seemed a bit
> flaky.
>
> --
> AJS
>
> Note: Originating address only accepts e-mail from list! If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
> --
> _
> -- 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] Can't comile bundled PJSIP on CentOS 7
is there libsrtp.so.* in /usr/lib64 ?
did ./configure script show that libsrtp is present?
2017-01-10 14:58 GMT+05:00 Olivier :
> Historically, I didn't use "install_prereq" but I also used it yesterday.
>
> As make fails with "[LD] libasteriskpj.o -> libasteriskpj.so.2" which is
> the first of its "kind", I still wonder
> if issue comes from libsrtp somehow, or from a missing option in
> ./configure (such as --prefix).
>
> 2017-01-10 10:31 GMT+01:00 A J Stiles :
>
>> On Tuesday 10 Jan 2017, Olivier wrote:
>> > Hello,
>> >
>> > I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes.
>> >
>> > I followed this:
>> > cd /usr/src
>> > wget ... asterisk-13.13.1.tar.gz
>> > tar zxf asterisk-13.13.1.tar.gz
>> > cd asterisk-13.13.1
>> > ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/usr"
>> > ./configure ${ASTERISK_CONFIGURE} --with-pjproject-bundled
>> > make menuselect (shows res-srtp is available)
>> > make
>> >
>> > latest make command fails with (see [1]:
>> >
>> >[GENERATE] libasteriskpj.exports
>> >[LD] libasteriskpj.o -> libasteriskpj.so.2
>> > /bin/ld: cannot find -lsrtp-x86_64-unknown-linux-gnu
>>
>> You probably need to install a package called something like "srtp-devel"
>> or
>> "libsrtp-devel". (I'm not certain of the exact package name on a
>> Fedora-like
>> system, as I'm coming from Ubuntuland ;) ) Re-run the configure script
>> after
>> installing it.
>>
>> Asterisk always used (pre-10!) to be distributed with a script called
>> "install_prereq" which was meant to install all the necessary packages
>> for a
>> full install with everything, at least on a on Fedora-like or Debian-like
>> distro; not sure if this is still included anymore, as it seemed a bit
>> flaky.
>>
>> --
>> AJS
>>
>> Note: Originating address only accepts e-mail from list! If replying
>> off-
>> list, change address to asterisk1list at earthshod dot co dot uk .
>>
>> --
>> _
>> -- 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
>
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Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7
* THIS IS NOT WHERE YOUR REPLY BELONGS * On Tuesday 10 Jan 2017, Olivier wrote: > Historically, I didn't use "install_prereq" but I also used it yesterday. > > As make fails with "[LD] libasteriskpj.o -> libasteriskpj.so.2" which is > the first of its "kind", I still wonder > if issue comes from libsrtp somehow, or from a missing option in > ./configure (such as --prefix). You said earlier that installing Asterisk and CentOS worked with a VM box that you set up. Were they the same architecture? Were you using all the same versions of everything? Have you still got the image? Fire it up and see what's different between the two. (You can find out your kernel version, and whether you are running 32- or 64-bit, with `uname -a`. I think the command to list all installed packages on an RPM-based system is `rpm -q -a`, but it was awhile ago since I used one.) -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] Can't comile bundled PJSIP on CentOS 7
On Mon, Jan 9, 2017 at 5:15 PM, Olivier wrote:
> Hello,
>
> I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes.
>
> I followed this:
> cd /usr/src
> wget ... asterisk-13.13.1.tar.gz
> tar zxf asterisk-13.13.1.tar.gz
> cd asterisk-13.13.1
> ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/usr"
> ./configure ${ASTERISK_CONFIGURE} --with-pjproject-bundled
> make menuselect (shows res-srtp is available)
> make
>
> latest make command fails with (see [1]:
>
>[GENERATE] libasteriskpj.exports
>[LD] libasteriskpj.o -> libasteriskpj.so.2
> /bin/ld: cannot find -lsrtp-x86_64-unknown-linux-gnu
>
>
> So I also build a simple VM box in which I also install CentOS and
> Asterisk 13.3.1 but I could run the above instructions successfully.
>
>
> Suggestions ?
>
Could you have had installed pjproject into the system locations
previously? Specifically, are there any libpj* files in /usr/lib64 or pj*
files or directories in /usr/include? If so, try uninstalling it before
building --with-pjproject-bundled. There may be a conflict.
The newest versions of pjproject require libsrtp 2 which doesn't ship with
most distributions so pjproject attempts to build it's own internal
version. Because we handle srtp in Asterisk itself not pjproject, we made
a change to the bundled version of pjproject to not build that internal
libsrtp. If you look in third-party/pjproject/patches/config_site.h you
should see "#define PJMEDIA_HAS_SRTP 0". It's possible that when the
bundled version is being built it's picking up something from the system
directories it shouldn't and looking for the internal version of libsrtp
which it didn't build.
If removing the system pjproject fixes your issue, let me know and I'll
open a ticket to look further at the conflict.
>
> Best regards
>
> [1] Original error message was localised.
>
> --
> _
> -- 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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--
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] PJSIP status check at DB level (Realtime)
On Tue, Jan 10, 2017, at 10:11 AM, Ahmed Munir wrote: > Hi, > > I would like to know how to check PJSIP status for endpoints at DB level > (realtime) just like in chan_sip? Like on chan_sip when sip extension > gets > register regseconds, ipaddr, fullcontact & many more parameters get > updated > at DB level. > > The version of asterisk I'm using is 13.8 cert 4 Registered devices are added as contacts to the "ps_contacts" table. This is because there can be multiple of them. You would need to look in that table and make the association (based on AOR). -- Joshua Colp Digium, Inc. | 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
[asterisk-users] PJSIP status check at DB level (Realtime)
Hi, I would like to know how to check PJSIP status for endpoints at DB level (realtime) just like in chan_sip? Like on chan_sip when sip extension gets register regseconds, ipaddr, fullcontact & many more parameters get updated at DB level. The version of asterisk I'm using is 13.8 cert 4 -- Regards, Ahmed Munir Chohan -- _ -- 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] Issue with handling of 480 DND
Response inline.
On Fri, Jan 6, 2017 at 12:47 PM, Markus Weiler
wrote:
> Nobody any idea?
>
> It would be really helpful,
>
> Markus
>
>
>
>
> Am 06.01.2017 um 12:07 schrieb Markus Weiler:
>
>> Hi List,
>>
>> we're calling a sip phone from our Asterisk Server, and try to add logic
>> depending on the dialstatus
>>
>> Stripped down example;
>>
>> exten = 494X,n,Dial(SIP/4120089,15,w)
>> exten = 494X,n,Goto(98-${DIALSTATUS},1)
>> exten = 494X,n,Hangup()
>>
>>
>> .
>> exten = 98-BUSY,1,NoOp(Busy)
>> exten = 98-BUSY,n,ExecIf($["${Voicemail}" =
>> "1"]?Playback(/home/4120/mitarbeiter/ab))
>>
>> exten = 98-NOANSWER,1,NoOp(noanswer)
>> exten = 98-NOANSWER,n,ExecIf($["${Voicemail}" =
>> "1"]?Playback(/home/4120/mitarbeiter/ab))
>>
>>
>> Íf the phone call times out, the call is sent to 98-NOANSWER and then
>> answered as expected.
>> If the User presses DND on his phone the call is sent to 98-BUSY which
>> is identical but then the call is hung up. This behaviour is
>> unexpected/unwanted.
>>
>> We tried to figure out what the difference is and think it's how
>> Asterisk handles the "480 Do Not Disturb" from the phone
>> (xxx.xxx.xxx.xxx).
>> It is passed to our main incoming server (zzz.zzz.zzz.zzz) as "181 call
>> is being forwarded".
>>
>> Is this a bug or a feature? :-) How could we handle this correctly?
>>
>
I believe that this is a consequence of the fact that when chan_sip
receives a "480 Do Not Disturb" it also queues a redirecting_update frame
on the channel. My guess is that the redirecting update is probably
triggering the "181 Call is being forward as well". If you add the 'I'
flag to your dial, I believe it should suppress bridging of the redirection
information and I would think that would also cause the 181 not to be
sent. If it's not that flag, I'd check the documentation for app_dial, as
I'm pretty sure there's a flag that should suppress that redirection from
from being bridged to the calling channel.
Hope that helps.
--
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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[asterisk-users] Custom INFO for Advice Of Charge
Hello, Does anyone know how to send a SIP INFO with custom data on a specified channel from Asterisk? The intention is to provide an Advice Of Charge. Asterisk has it's own AOC function but it only seems to support Snom's format, not the more general XML format. Searching came up with SIPSendCustomInfo but apparently it sends on all active SIP channels, and is only available with TEST_FRAMEWORK. Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -- _ -- 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
