search for: callevents

...example,dc=com Pass=123456 ----------------------------------------------------------------------- vim /etc/asterisk/extconfig.conf sipusers => ldap,"dc=example,dc=com",sip ------------------------------------------------------------------------ vim /etc/asterisk/sip.conf [general] callevents=yes rtcachefriends=yes but i got the follwoing error : PBX*CLI> module reload res_config_ldap.so ? ? -- Reloading module 'res_config_ldap.so' (LDAP realtime interface) ? == Parsing '/etc/asterisk/res_ldap.conf': ? == Found [Nov ?8 09:38:06] WARNING[8687]: res_config_ldap.c:...

...ging exten => _X.,n,Progress() exten => _X.,n,Wait(5) exten => _X.,n,Dial(SIP/123&SIP/456,30,oxX) ... exten => fax,1,NoOp(**** FAX DETECTED ****) exten => fax,n,Goto(fax-rx,receive,1) in the sip.conf i specified [general] sendrpid=rpid trustrpid=yes language=de videosupport=yes callevents=yes caninvite=yes qualify=yes nat=force_rport,comedia faxdetect=yes t38pt_udptl=yes ... [abcde] type=peer insecure=invite defaultuser=12345678912 fromuser=12345678912 fromdomain=abcde.ab secret=guess-what host=abcde.ab qualify=yes context=from-sip dtmfmode=rfc2833 callbackextension=12345678912...

...st_invite: Failed to authenticate user "585415198" <sip:585415198 at 82.208.46.240> <sip:585415198 at 82.208.46.240>;tag=as18abefe8 Can someone help me out of this? I have Asterisk 1.2 on the Ubuntu 7.04. Outcoming and internal calls functions well. Thanks sip.conf: [general] callevents=yes register => username:password at sip.mujtelefon.cz/username [100] callerid=100 secret=password type=friend context=internal srvlookup=yes type=friend qualify=yes nat=no host=dynamic canreinvite=no context=internal call-limit=1 [101] callerid=101 secret=password type=friend context=interna...

...adWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Max-Forwards: 70 Content-Length: 0 Here's the configs: subscribecontext = device-hints allowexternaldomains = yes allowguest = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyr...

...s my sip.conf with one client: [general] context=incoming realm=softpbx bindport=5060 bindaddr=0.0.0.0 srvlookup=yes useclientcode=yes defaultexpirey=3600 vmexten=voicemail disallow=all allow=alaw allow=ulaw allow=gsm ;qualify=no ;canreinvite=no musicclass=default language=de useragent=ipanlage callevents=yes nat=yes rtcachefriends=no rtupdate=no rtautoclear=no ignoreregexpire=yes amaflags=omit canreinvite=no subscribecontext=outcust limitonpeers=yes allowsubscribe=yes notifyringing=yes [xxx] type=friend context=outcust nat=yes qualify=yes secret=yyyy username=xxx callerid="bla bla" accou...

...ing from CLI: sip qualify peer device load no result. What I did not configured? My sip.conf [general] context = default allowguest = no bindport = 5060 bindaddr = 0.0.0.0 allowexternaldomains = no allowoverlap = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = no checkmwi = 10 compactheaders = no defaultexpiry = 120 domain=sop-korniychuk domain=172.30.8.13 domain=172.30.8.13:5060 dumphistory = no externrefresh = 10 g726nonstandard = no notifyringing = yes srvlookup = yes t1min = 100 t38pt_udptl = no ;tos_audio = none ;tos_sip = none ;t...

...39;200' is now UNREACHABLE! Last qualify: 53 The phone (gxp1450) is configured to use symetric rtp with disabled crypto lifetime and srtp mode forced. Any ideas what could be wrong? [general] allowguest=no alwaysauthreject=yes nat=force_rport,comedia sendrpid=rpid trustrpid=yes language=de callevents=yes qualify=yes faxdetect=yes t38pt_udptl=no disallow=all allow=ulaw allow=alaw ;-------------------------Encryption----- encryption=yes tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/var/lib/asterisk/keys/asterisk.pem tlscafile=/var/lib/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 tlsd...

> On 27 Jun 2017, at 13:25, Peter Smith <peter.smith at linaro.org> wrote: > > Hello Alessandro, > > Despite the statement in the HowToCrossCompileLLVM guide "If you’re > using Clang as the cross-compiler, there is a problem in the LLVM ARM > back-end that is producing absolute relocations on > position-independent code (R_ARM_THM_MOVW_ABS_NC), so for now, you

Oh, so it looks like I hit a bit of a wall there :-) I’ll take a look thanks. That bug talks about R_ARM_THM_CALL which I assume are thumb related. Will your implementation fix also R_ARM_CALL errors? > On 28 Jun 2017, at 17:15, Peter Smith <peter.smith at linaro.org> wrote: > > Hello Alessandro, > > The LLD ARM port doesn't currently support range extension thunks,

I've successfully used Peter's patches to get past those relocation errors. On 6/28/17, 9:36 AM, "llvm-dev on behalf of Peter Smith via llvm-dev" <llvm-dev-bounces at lists.llvm.org on behalf of llvm-dev at lists.llvm.org> wrote: Yes it should cover the following relocations: R_ARM_CALL (ARM BL/BLX) R_ARM_JUMP24 (ARM B) R_ARM_THM_CALL (Thumb BL/BLX)

The bottom of the bug has the revision numbers (e.g. D34035). That one corresponds to e.g. https://reviews.llvm.org/D34035 There's also https://reviews.llvm.org/D34634 which contains all of Peter's patches, but it's not going to rebase cleanly once the individual patches start going in. On 6/28/17, 10:56 AM, "Alessandro Pistocchi" <apukfreelance at gmail.com> wrote:

At a guess that looks like your llvm and lld checkouts are not quite in synch. It will be worth updating llvm and lld to top of trunk. I've rebased the consolidated patch https://reviews.llvm.org/D34634 this morning, it might be worth trying that if you are seeing problems. Peter On 29 June 2017 at 22:09, Alessandro Pistocchi <apukfreelance at gmail.com> wrote: > Hi, I tried

...uthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, > ;regcontext=sipregistrations > ;registertimeout=20 ; retry registration calls every 20 seconds (default) > ;registerattempts=10 ; Number of registration attempts before we give up > callevents=no ; generate manager events when sip ua performs events (e.g. hold) > externip=The_IP_of_my_router ; Address that we're going to put in outbound SIP messages > ;externhost=foo.dyndns.net ; Alternatively you can specify an > ;externrefresh=10 ; H...

....11.0(11.20.0) disallow=all allow=g723 allow=ulaw allow=gsm allow=alaw allow=g729 allow=speex allow=g722 allow=h264 allow=h263p allow=h263 allow=h261 tlsenable=yes tlsbindaddr=0.0.0.0 tlscipher=ALL tlsclientmethod=tlsv1 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt callevents=no jbenable=no videosupport=yes allowguest=no srvlookup=no defaultexpiry=120 minexpiry=60 maxexpiry=3600 registerattempts=0 registertimeout=20 g726nonstandard=no maxcallbitrate=384 canreinvite=no rtptimeout=30 rtpholdtimeout=300 rtpkeepalive=0 checkmwi=10 notifyringing=yes notifyhold=yes nat=yes [...

...attempts=10 ; Number of registration attempts before we give up ; 0 = continue forever, hammering the other server until it ; accepts the registration ; Default is 0 tries, continue forever ;callevents=no ; generate manager events when sip ua performs events (e.g. hold) ;----------------------------------------- NAT SUPPORT ------------------------ ; The externip, externhost and localnet settings are used if you use Asterisk ; behind a NAT device to communicate with services...

Commit-ID: ef5fd2060f3c7d1a4a22a079bc1c32b61964bb5f Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=ef5fd2060f3c7d1a4a22a079bc1c32b61964bb5f Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Tue, 7 Jun 2016 16:47:59 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Fri, 25 Jan 2019 02:57:21 +0000 [klibc] eval: Return status in

Commit-ID: 777b77571a451d5fc5dfc04749854ea40abe8093 Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=777b77571a451d5fc5dfc04749854ea40abe8093 Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Tue, 7 Jun 2016 16:47:59 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Sat, 28 Mar 2020 21:42:54 +0000 [klibc] dash: eval: Return

> On 26 Jun 2017, at 16:25, Rui Ueyama <ruiu at google.com> wrote: > > On Sun, Jun 25, 2017 at 6:40 AM, Alessandro Pistocchi via llvm-dev <llvm-dev at lists.llvm.org <mailto:llvm-dev at lists.llvm.org>> wrote: > Hi, > > I am trying to build a completely GNU free linux toolchain for the raspberry pi. > > I successfully managed to compile llvm and clang

...(no NAT) with precisely the same symptoms. The same symptoms persist whether the PSTN or the CPE initiate the call. My example configs are as follows: SIP - [general] limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes disallow=all allow=g722 allow=ulaw allow=gsm allowoverlap=no callevents=yes allowguest=no directmedia=no bindport=bind_here bindaddr=to_this_address srvlookup=yes maxexpiry=7200 defaultexpiry=3600 [authentication] [test-voice] type=friend host=dynamic secret=not_my_secret context=users disallow=all allow=ulaw nat=yes directmedia=no qualify=yes trunk=no IAX2 - [gener...

...wait(2) at the correct place. But i'm still unable to receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short after the Fax session has started. My sip.conf includes [general] allowguest=no alwaysauthreject=yes sendrpid=rpid trustrpid=yes language=de videosupport=yes callevents=yes qualify=yes nat=force_rport,comedia faxdetect=yes t38pt_udptl=yes,redundancy,maxdatagram=400 directrtpsetup=yes disallow=all allow=ulaw allow=alaw and the corresponding Peer [sipcall.ch] type=peer insecure=invite defaultuser=123456789 fromuser=123456789 fromdomain=voipdomain.com secret=gueswh...