[asterisk-users] PJSIP - sessions-timers support not working on 13.X
Gosmac
goseeped at gmail.com
Wed Apr 29 13:20:19 CDT 2015
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
> On Apr 29, 2015, at 12:30 PM, asterisk-users-request at 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 at gmail.com>
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] PJSIP - sessions-timers support not working
> on 13.X
> Message-ID: <CA1EBA47-70CF-42B0-9E7C-4D48E9C9C49B at 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 at digium.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] PJSIP - sessions-timers support not
> working on 13.X
> Message-ID: <553FC362.10306 at digium.com>
> Content-Type: text/plain; charset=UTF-8; format=flowed
>
> Gosmac wrote:
>> 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 at lodgingcompany.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] adding area code
> Message-ID: <20150428115411.71697421 at ws78.int.tlc>
> Content-Type: text/plain; charset=US-ASCII
>
> On Tue, 28 Apr 2015 07:21:12 -0700
> Motty Cruz <motty.cruz at gmail.com> wrote:
>
>> 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/${EXTEN:1},80)
>
> I find it hard to believe this is working.
>
> First, you don't have a leading underscore on your patterns. Your
> users aren't literally dialing the N's and X's are they?
>
> Second, what's with the plus in the extension? You want your users to
> dial that?
>
> Third, that's two different extensions, one with priority 1 and one
> with priority 2. The first one will set a variable and hangup, and the
> second.... there's no priority 1 for that extension... I've never tried
> that... I'm assuming it just won't work.
>
>
> --
>
> C. Chad Wallace, B.Sc.
> The Lodging Company
> http://www.lodgingcompany.com/
> OpenPGP Public Key ID: 0x262208A0
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 28 Apr 2015 12:27:10 -0700
> From: Motty Cruz <motty.cruz at gmail.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] adding area code
> Message-ID: <553FDF0E.7000404 at gmail.com>
> Content-Type: text/plain; charset="windows-1252"; Format="flowed"
>
> I apologize, I coppied the wrong code,
> here is the code I am using:
>
> ; Adding Area code and striping 9 for local numbers
> exten => _9XXXXXXX,n,Set(CALLERID(all)= <3817383444>)
> exten => _9XXXXXXX,n,Dial(SIP/intelepeer/1381${EXTEN:1},80)
>
>
> Thanks,
> motty
>
> On 04/28/2015 11:54 AM, Chad Wallace wrote:
>> On Tue, 28 Apr 2015 07:21:12 -0700
>> Motty Cruz <motty.cruz at gmail.com> wrote:
>>
>>> 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.
>>
>>
>
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> ------------------------------
>
> Message: 5
> Date: Tue, 28 Apr 2015 16:01:38 -0400
> From: Jeremy Kister <asterisk-03 at jeremykister.com>
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Asterisk 13/PJSIP + registration
> Message-ID: <553FE722.6000906 at 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 at 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 at pack-net.co.uk>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] Asterisk 1.8.32.3 chan_sip deadlock
> Message-ID:
> <CAHE6+j0ao6LD85D2cZFDYyZQURYEabLRsx_BX3zrGY-1TgQ_Fg at 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 at 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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