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, [email protected] wrote: > > Send asterisk-users mailing list submissions to > [email protected] > > 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 > [email protected] > > You can reach the person managing the list at > [email protected] > > 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 <[email protected]> > To: [email protected] > Subject: [asterisk-users] PJSIP - sessions-timers support not working > on 13.X > Message-ID: <[email protected]> > 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 <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] PJSIP - sessions-timers support not > working on 13.X > Message-ID: <[email protected]> > 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 <[email protected]> > To: [email protected] > Subject: Re: [asterisk-users] adding area code > Message-ID: <[email protected]> > Content-Type: text/plain; charset=US-ASCII > > On Tue, 28 Apr 2015 07:21:12 -0700 > Motty Cruz <[email protected]> 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/${EXTEN:1},80) > > I find it hard to believe this is working. > > First, you don't have a leading underscore on your patterns. Your > users aren't literally dialing the N's and X's are they? > > Second, what's with the plus in the extension? You want your users to > dial that? > > Third, that's two different extensions, one with priority 1 and one > with priority 2. The first one will set a variable and hangup, and the > second.... there's no priority 1 for that extension... I've never tried > that... I'm assuming it just won't work. > > > -- > > C. Chad Wallace, B.Sc. > The Lodging Company > http://www.lodgingcompany.com/ > OpenPGP Public Key ID: 0x262208A0 > > > > > ------------------------------ > > Message: 4 > Date: Tue, 28 Apr 2015 12:27:10 -0700 > From: Motty Cruz <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [asterisk-users] adding area code > Message-ID: <[email protected]> > Content-Type: text/plain; charset="windows-1252"; Format="flowed" > > I apologize, I coppied the wrong code, > here is the code I am using: > > ; Adding Area code and striping 9 for local numbers > exten => _9XXXXXXX,n,Set(CALLERID(all)= <3817383444>) > exten => _9XXXXXXX,n,Dial(SIP/intelepeer/1381${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 <[email protected]> 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 <[email protected]> > To: [email protected] > Subject: [asterisk-users] Asterisk 13/PJSIP + registration > Message-ID: <[email protected]> > 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:[email protected] > 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 <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: [asterisk-users] Asterisk 1.8.32.3 chan_sip deadlock > Message-ID: > <cahe6+j0ao6ld85d2czfdyyzquryeablrsx_bx3zrgy-1tgq...@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: [email protected] > 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 > *********************************************** -- _____________________________________________________________________ -- 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
