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[asterisk-users] PJSIP - sessions-timers support not working on 13.X

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goseeped at gmail.com
Guest

Posted: Tue Apr 28, 2015 12:22 pm Post subject: [asterisk-users] PJSIP - sessions-timers support not working

Hi guys i was trying to get working sessions-timer over PJSIP channel i was trying to see what is supported or not about this features on the new pjsip channel since chan_sip was kind of flexible on this , at the moment since wiki says pjsip support 4 modes of operation (forced, no, required, yes) but if i try to change any of the timers parameters (timers, timers_min_se or timers_sess_expiries) the pjsip channel doesn’t load the endpoint or even not load the well the channel this happens on 13.1, 13.2 and 13.3. should i enable something different of normal variables on pjsip.conf ? Thanks Javier Riveros. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

jcolp at digium.com
Guest

Posted: Tue Apr 28, 2015 12:29 pm Post subject: [asterisk-users] PJSIP - sessions-timers support not working

Gosmac wrote:

Quote:

Hi guys i was trying to get working sessions-timer over PJSIP channel
i was trying to see what is supported or not about this features on
the new pjsip channel since chan_sip was kind of flexible on this ,
at the moment since wiki says pjsip support 4 modes of operation
(forced, no, required, yes) but if i try to change any of the timers
parameters (timers, timers_min_se or timers_sess_expiries) the pjsip
channel doesn’t load the endpoint or even not load the well the
channel this happens on 13.1, 13.2 and 13.3.

What is the exact configuration of the endpoint, and what is output on
the CLI? As well - you have one of the parameters incorrect above. It's
timers_sess_expires, not timers_sess_expiries. If that is incorrect in
your configuration this would be considered invalid and thus it would
not load.

Cheers,

--
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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

goseeped at gmail.com
Guest

Posted: Wed Apr 29, 2015 1:20 pm Post subject: [asterisk-users] PJSIP - sessions-timers support not working

Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn’t a "typo” error of timers parameters, i have an error on global tag and can’t load the timers

I was getting this :

[Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error: No category context for line 631 of /etc/asterisk/pjsip.conf

after fix global issue

[105]
type=aor
max_contacts=1
remove_existing=yes

[105]
type=auth
auth_type=userpass
password=XXXXXXXX
username=105

[105]
type=endpoint
disallow=all
allow=ulaw
allow=alaw
context=video-test
auth=105
aors=105
direct_media=no
force_rport=yes
rewrite_contact=yes
transport=transport-udp-nat
media_encryption=no
ice_support=no
timers_min_se=90 ; Minimum session timers expiration period (default:; "90")
timers=required ; Session timers for SIP packets (default: "yes")
timers_sess_expires=3600 ; Maximum session timer expiration period

now get things working and i could see how this behave.

Thanks
Regards

Quote:

On Apr 29, 2015, at 12:30 PM, asterisk-users-request@lists.digium.com wrote:

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Today's Topics:

1. PJSIP - sessions-timers support not working on 13.X (Gosmac)
2. Re: PJSIP - sessions-timers support not working on 13.X
(Joshua Colp)
3. Re: adding area code (Chad Wallace)
4. Re: adding area code (Motty Cruz)
5. Asterisk 13/PJSIP + registration (Jeremy Kister)
6. Asterisk 1.8.32.3 chan_sip deadlock (Ishfaq Malik)

----------------------------------------------------------------------

Message: 1
Date: Tue, 28 Apr 2015 12:52:22 -0430
From: Gosmac <goseeped@gmail.com>
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] PJSIP - sessions-timers support not working
on 13.X
Message-ID: <CA1EBA47-70CF-42B0-9E7C-4D48E9C9C49B@gmail.com>
Content-Type: text/plain; charset=utf-8

Hi guys i was trying to get working sessions-timer over PJSIP channel i was trying to see what is supported or not about this features on the new pjsip channel since chan_sip was kind of flexible on this , at the moment since wiki says pjsip support 4 modes of operation (forced, no, required, yes) but if i try to change any of the timers parameters (timers, timers_min_se or timers_sess_expiries) the pjsip channel doesn?t load the endpoint or even not load the well the channel this happens on 13.1, 13.2 and 13.3.

should i enable something different of normal variables on pjsip.conf ?

Thanks

Javier Riveros.

------------------------------

Message: 2
Date: Tue, 28 Apr 2015 14:29:06 -0300
From: Joshua Colp <jcolp@digium.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] PJSIP - sessions-timers support not
working on 13.X
Message-ID: <553FC362.10306@digium.com>
Content-Type: text/plain; charset=UTF-8; format=flowed

Gosmac wrote:

Quote:

Hi guys i was trying to get working sessions-timer over PJSIP channel
i was trying to see what is supported or not about this features on
the new pjsip channel since chan_sip was kind of flexible on this ,
at the moment since wiki says pjsip support 4 modes of operation
(forced, no, required, yes) but if i try to change any of the timers
parameters (timers, timers_min_se or timers_sess_expiries) the pjsip
channel doesn?t load the endpoint or even not load the well the
channel this happens on 13.1, 13.2 and 13.3.

What is the exact configuration of the endpoint, and what is output on
the CLI? As well - you have one of the parameters incorrect above. It's
timers_sess_expires, not timers_sess_expiries. If that is incorrect in
your configuration this would be considered invalid and thus it would
not load.

Cheers,

--
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

------------------------------

Message: 3
Date: Tue, 28 Apr 2015 11:54:11 -0700
From: Chad Wallace <cwallace@lodgingcompany.com>
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] adding area code
Message-ID: <20150428115411.71697421@ws78.int.tlc>
Content-Type: text/plain; charset=US-ASCII

On Tue, 28 Apr 2015 07:21:12 -0700
Motty Cruz <motty.cruz@gmail.com> wrote:

Quote:

here is what I did and worked for me:

exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444)

exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80)

I find it hard to believe this is working.

First, you don't have a leading underscore on your patterns. Your
users aren't literally dialing the N's and X's are they?

Second, what's with the plus in the extension? You want your users to
dial that?

Third, that's two different extensions, one with priority 1 and one
with priority 2. The first one will set a variable and hangup, and the
second.... there's no priority 1 for that extension... I've never tried
that... I'm assuming it just won't work.

--

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0

------------------------------

Message: 4
Date: Tue, 28 Apr 2015 12:27:10 -0700
From: Motty Cruz <motty.cruz@gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] adding area code
Message-ID: <553FDF0E.7000404@gmail.com>
Content-Type: text/plain; charset="windows-1252"; Format="flowed"

I apologize, I coppied the wrong code,
here is the code I am using:

; Adding Area code and striping 9 for local numbers
exten => _9XXXXXXX,n,Set(CALLERID(all)= <3817383444>)
exten => _9XXXXXXX,n,Dial(SIP/intelepeer/1381${EXTEN:1},80)

Thanks,
motty

On 04/28/2015 11:54 AM, Chad Wallace wrote:

Quote:

On Tue, 28 Apr 2015 07:21:12 -0700
Motty Cruz <motty.cruz@gmail.com> wrote:

Quote:

here is what I did and worked for me:

exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444)

exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80)

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------------------------------

Message: 5
Date: Tue, 28 Apr 2015 16:01:38 -0400
From: Jeremy Kister <asterisk-03@jeremykister.com>
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk 13/PJSIP + registration
Message-ID: <553FE722.6000906@jeremykister.com>
Content-Type: text/plain; charset=utf-8; format=flowed

Using Asterisk 13.3.2 on CentOS7 and pjproject 2.4, I can't make
asterisk try to send a register.

I have configured my pjsip.conf similar to
https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples#res_pjsipConfigurationExamples-ASIPtrunktoyourserviceprovider,includingoutboundregistration

my pjsip.conf: http://pastebin.com/raw.php?i=EA0PEcrb

using tcpdump, I never even see a packet sent from asterisk trying to
register.

on the asterisk console:
asterisk13*CLI> pjsip show registrations
No objects found.

asterisk13*CLI> pjsip show contacts

Contact: <Aor/ContactUri...................................>
<Status....> <RTT(ms)..>

=========================================================================================

Contact: provider1/sip:1XXXNNNYYYY@sip.provider1.com
Unknown nan

asterisk13*CLI> pjsip list aors

Aor: <Aor..............................................>
<MaxContact>

=========================================================================================

Aor: provider1 0

FYI, I can modify pjsip.conf to add configuration for a softphone to
register to asterisk - that works fine.

Can someone give me a clue on how to make this outbound registration
happen ?

------------------------------

Message: 6
Date: Wed, 29 Apr 2015 14:42:41 +0100
From: Ishfaq Malik <ish@pack-net.co.uk>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] Asterisk 1.8.32.3 chan_sip deadlock
Message-ID:
<CAHE6+j0ao6LD85D2cZFDYyZQURYEabLRsx_BX3zrGY-1TgQ_Fg@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

Hello asterisk-users,

We've been having intermittent issues with chan_sip - it stops responding
to cli requests, trying to reload chan_sip from cli doesn't seem to have
any effect, initiated calls carry on for a short period, but no new SIP
requests are processed ('sip show channels' hangs forever, server stops
responding to SIP OPTIONS, or any other SIP messages). We have updated the
build from 1.8.23.1 to the latest asterisk 1.8 (1.8.32.3), however the
problem still persists. We have gathered debugging information from 'core
show locks' and from gdb, attached to this message (with phone numbers and
extension and context names obscured). We are running realtime under CentOS
6.6, built from source and packaged using rpmbuild, with the following
menuselect options (debugging version):
menuselect/menuselect --disable BUILD_NATIVE --enable DEBUG_THREADS
--enable DONT_OPTIMIZE --disable CORE-SOUNDS-EN-GSM --disable-category
MENUSELECT_EXTRA_SOUNDS --disable MOH-OPSOUND-WAV --enable-category
MENUSELECT_ADDONS --disable format_mp3 --disable cdr_tds --disable cel_tds
--disable cdr_pgsql --disable cel_pgsql --disable res_config_pgsql
menuselect.makeopts

under kernel 2.6.32-504.el6.x86_64, and linked against the following
library versions:

/usr/lib64/libssl.so.10: symbolic link to `libssl.so.1.0.1e'
/usr/lib64/libcrypto.so.10: symbolic link to `libcrypto.so.1.0.1e'
/lib64/libc.so.6: symbolic link to `libc-2.12.so'
/usr/lib64/libxml2.so.2: symbolic link to `libxml2.so.2.7.6'
/lib64/libz.so.1: symbolic link to `libz.so.1.2.3'
/lib64/libm.so.6: symbolic link to `libm-2.12.so'
/lib64/libdl.so.2: symbolic link to `libdl-2.12.so'
/lib64/libpthread.so.0: symbolic link to `libpthread-2.12.so'
/lib64/libtinfo.so.5: symbolic link to `libtinfo.so.5.7'
/lib64/libresolv.so.2: symbolic link to `libresolv-2.12.so'
/lib64/libgssapi_krb5.so.2: symbolic link to `libgssapi_krb5.so.2.2'
/lib64/libkrb5.so.3: symbolic link to `libkrb5.so.3.3'
/lib64/libcom_err.so.2: symbolic link to `libcom_err.so.2.1'
/lib64/libk5crypto.so.3: symbolic link to `libk5crypto.so.3.1'
/lib64/libkrb5support.so.0: symbolic link to `libkrb5support.so.0.1'
/lib64/libkeyutils.so.1: symbolic link to `libkeyutils.so.1.3'

We'd appreciate any possible assistance, as we're having problems working
out what exactly triggers the deadlock and we have not been able to find
the correct sequence of steps to reproduce the issue yet, other than
waiting for it to lock up at an arbitrary time with the debugging code in
place. It does seem to happen at least once a day, however.

What is the best way of getting the core show locks output for people to
see as it appears to be too big to mail?

Ish

--

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: ish@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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End of asterisk-users Digest, Vol 129, Issue 33
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goseeped at gmail.com
Guest

Posted: Wed Apr 29, 2015 5:31 pm Post subject: [asterisk-users] PJSIP - sessions-timers support not working

Ok , digging more into this i could see that (timers=no) and (timers=forced) not work asterisk not pay attention to this options when is reloaded cli not say anything and when the pjsip show endpoint <endpoint> it show always timers=yes when (timers=no) and (timers=forced) to that endpoint.

I wonder to force asterisk to refresh the session in some cases when is needed .

pjsip is able to refresh the session ?

Cheers

Quote:

On Apr 29, 2015, at 1:50 PM, Gosmac <goseeped@gmail.com> wrote:

Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn’t a "typo” error of timers parameters, i have an error on global tag and can’t load the timers

I was getting this :

[Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error: No category context for line 631 of /etc/asterisk/pjsip.conf

after fix global issue

[105]
type=aor
max_contacts=1
remove_existing=yes

[105]
type=auth
auth_type=userpass
password=XXXXXXXX
username=105

[105]
type=endpoint
disallow=all
allow=ulaw
allow=alaw
context=video-test
auth=105
aors=105
direct_media=no
force_rport=yes
rewrite_contact=yes
transport=transport-udp-nat
media_encryption=no
ice_support=no
timers_min_se=90 ; Minimum session timers expiration period (default:; "90")
timers=required ; Session timers for SIP packets (default: "yes")
timers_sess_expires=3600 ; Maximum session timer expiration period

now get things working and i could see how this behave.

Thanks
Regards

Quote:

Hi guys i was trying to get working sessions-timer over PJSIP channel
i was trying to see what is supported or not about this features on
the new pjsip channel since chan_sip was kind of flexible on this ,
at the moment since wiki says pjsip support 4 modes of operation
(forced, no, required, yes) but if i try to change any of the timers
parameters (timers, timers_min_se or timers_sess_expiries) the pjsip
channel doesn?t load the endpoint or even not load the well the
channel this happens on 13.1, 13.2 and 13.3.

What is the exact configuration of the endpoint, and what is output on
the CLI? As well - you have one of the parameters incorrect above. It's
timers_sess_expires, not timers_sess_expiries. If that is incorrect in
your configuration this would be considered invalid and thus it would
not load.

Cheers,

--
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

------------------------------

Message: 3
Date: Tue, 28 Apr 2015 11:54:11 -0700
From: Chad Wallace <cwallace@lodgingcompany.com>
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] adding area code
Message-ID: <20150428115411.71697421@ws78.int.tlc>
Content-Type: text/plain; charset=US-ASCII

On Tue, 28 Apr 2015 07:21:12 -0700
Motty Cruz <motty.cruz@gmail.com> wrote:

Quote:

here is what I did and worked for me:

exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444)

exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80)

Quote:

On Tue, 28 Apr 2015 07:21:12 -0700
Motty Cruz <motty.cruz@gmail.com> wrote:

Quote:

here is what I did and worked for me:

exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444)

exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80)

jcolp at digium.com
Guest

Posted: Wed Apr 29, 2015 5:37 pm Post subject: [asterisk-users] PJSIP - sessions-timers support not working

Gosmac wrote:

Quote:

Ok , digging more into this i could see that (timers=no) and
(timers=forced) not work asterisk not pay attention to this options
when is reloaded cli not say anything and when the pjsip show
endpoint<endpoint> it show always timers=yes when (timers=no) and
(timers=forced) to that endpoint.

I wonder to force asterisk to refresh the session in some cases when
is needed .

pjsip is able to refresh the session ?

There was an issue[1] where stuff did not work/was inconsistent. This
has been resolved in git and will be in the next release.

Cheers,

[1] https://issues.asterisk.org/jira/browse/ASTERISK-24910

--
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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

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