[asterisk-users] PJSIP - sessions-timers support not working on 13.X
Gosmac
goseeped at gmail.com
Wed Apr 29 17:31:17 CDT 2015
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
> On Apr 29, 2015, at 1:50 PM, Gosmac <goseeped at 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
>
>> On Apr 29, 2015, at 12:30 PM, asterisk-users-request at lists.digium.com wrote:
>>
>> Send asterisk-users mailing list submissions to
>> asterisk-users at lists.digium.com
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> or, via email, send a message with subject or body 'help' to
>> asterisk-users-request at lists.digium.com
>>
>> You can reach the person managing the list at
>> asterisk-users-owner at lists.digium.com
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of asterisk-users digest..."
>>
>>
>> Today's Topics:
>>
>> 1. 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.
>>>
>>>
>>
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150428/6fa782f1/attachment-0001.html>
>>
>> ------------------------------
>>
>> 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
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150429/4318194c/attachment-0001.html>
>>
>> ------------------------------
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> End of asterisk-users Digest, Vol 129, Issue 33
>> ***********************************************
>
More information about the asterisk-users
mailing list