search for: allowsubscribe

...ername=1001 secret=blah subscribecontext=default regexten=1001 callerid="blah" <XXXXXXXXXX> host=dynamic nat=yes canreinvite=no mailbox=1001 at default registertrying=yes [testuser] type=friend secret=blah callerid="blah" <XXXXXXXXX> host=dynamic nat=yes qualify=yes allowsubscribe=yes canreinvite=no context=default [testuser2] type=friend username=testuser2 secret= callerid="blah" <blah> host=dynamic nat=yes qualify=yes allowsubscribe=yes canreinvite=no context=default Someone is able to connect to my server and make a call since they can access the defau...

...5060) bindaddr = 192.168.1.243 ; x = Asterisk server IP address disallow=all ;allow = ulaw ; Allow all codecs ;allow = alaw context = bogon-calls ; Send SIP callers that we don't know about here canreinvite=no directrtpsetup=yes nat=no ;subscribecontext= localextensions ;default allowsubscribe=yes ; Disable support for subscriptions. (Default is yes) [App] type=friend username=App ;regexten=1234 ; When they register, create extension 1234 ;secret=password host=dynamic context=from-sip mailbox=App disallow=all allow = alaw ;canreinvite=no ;d...

...is 5060) bindaddr = 192.168.1.243 ; x = Asterisk server IP address disallow=all ;allow = ulaw ; Allow all codecs ;allow = alaw context = from-sip ; Send SIP callers that we don't know about here canreinvite=no directrtpsetup=yes nat=no ;subscribecontext= localextensions ;default allowsubscribe=yes ; Disable support for subscriptions. (Default is yes) [App] type=friend username=App ;regexten=1234 ; When they register, create extension 1234 ;secret=password host=dynamic context=from-sip mailbox=App disallow=all allow = alaw ;canreinvite=no ;d...

...SPA2102). Shouldn't the UDPTL stream go through Asterisk? Have i missed sometheng else? Asterisk SVN-trunk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201] secret=xyz caller...

...w=gsm allow=h263p canreinvite=no limitonpeer=yes notifyringing=yes notifyhold=yes externip=xx.xx.xx.xx.xx fromdomain=xx.xx.xx.xx localnet=192.168.0.0/255.255.255.0 [yy.yy.yy.yy] type=friend host=yy.yy.yy.yy insecure=port,invite [699] type=friend secret=1234 dial=SIP/699 callerid=Video <699> allowsubscribe=yes host=dynamic context=from-internal insecure=port,invite In addition here's the relevant portions of the SIP.CONF from the main server: [general] videosupport=yes disallow=all allow=ulaw allow=gsm allow=h263p canreinvite=no fromdomain=yy.yy.yy.yy externip=yy.yy.yy.yy localnet=10.200.26.0/...

...6103,hint,SIP/103 exten => 6104,hint,SIP/104 exten => 6105,hint,SIP/105 exten => _1XX,1,SIPAddHeader(Alert-Info:\;info=ring3) exten => _1XX,2,Dial(SIP/${EXTEN},20,Tt) exten => _1XX,3,VoiceMail(${EXTEN}@default,u) exten => _1XX,104,VoiceMail(${EXTEN}@default,b) sip.conf [general] allowsubscribe=yes ;subscribecontext = default notifyringing=no notifyhold=yes ;limitonpeers=yes [100] type=peer context=demo callerid=Back Office <100> username=100 secret=(Private) host=dynamic nat=no qualify=yes canreinvite=no dtmfmode=rfc2833 call-limit=5 ma...

...out any problem in the same network. After we had downgrade to 1.2.32 everything works fine again on these phones. my question is, had there been a big change in sip.conf or codec handling which cause this problem, cause i used the same sip.conf just adding notifyringing=yes, limitonpeers=yes and allowsubscribe=yes. Here is 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 usera...

...Here's what I've got: [hft0] type=friend username=hft0 secret=mysecret context=outtrunk-office host=192.168.200.99 disallow=all allow=ulaw dtmfmode=rfc2833 progressinband=no ;Polycom phones have trouble with the progressinband=never callerid="HFT Booth 0 <(619) 364-4850>" allowsubscribe=yes And some of the Polycom phone config: reg reg.1.displayName="HFT0" reg.1.address="6193644850" reg.1.label="4850" reg.1.type="private" reg.1.lcs="" reg.1.csta="" reg.1.thirdPartyName="" reg.1.a...

...before I go the bug route I'd like someone to just verify my configuration files make sure I'm not doing something stupid. SIP.CONF: [general] callerid=Unknown Caller disallow=all allow=ulaw allow=gsm [717] type=friend dial=SIP/717 callerid=Ken Williams <717> mailbox=717 at default allowsubscribe=yes host=dynamic context=from-internal [727] type=friend secret=1234 dial=SIP/727 callerid=Conference Room <727> mailbox=727 at default <mailto:mailbox=727 at default> allowsubscribe=yes host=dynamic context=from-internal EXTENSIONS.CONF: [from-internal] include => parked...

...resources, enabled few settings in sip.conf, but this still doesn't change. my sip.conf is: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 ignoreregexpire=yes limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes call-limit=1 and the corresponding realtime entry is: name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: device <21168> canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: NULL host: dynamic insecure: NUL...

...etc. We have customers that require both BLF and call waiting at the same time. We are running Asterisk 1.8.11-cert7 I've made the following additions to sip.conf [general]: callcounter=yes counteronpeer=yes (undocumented? Supposed to replace limitonpeers?) (old relevant values, unchanged) allowsubscribe=yes subscribecontext=blf notifyringing=yes notifyhold=yes limitonpeers=yes I also tried may other suggestions I've found like placing the hints in the same context as the extensions and removing subscribecontext. Is there something I'm missing? Is something not working correctly? Thank...

....21.1* 3. I'm using SIP realtime peers, *sip.conf *configuration follows: [general] bindport=5060 bindaddr=0.0.0.0 context=default language=es rtcachefriends=yes disallow=all allow=ulaw allow=alaw allow=gsm rtpholdtimeout=300 rtptimeout=300 dtmfmode=rfc2833 videosupport=yes progressinband=yes allowsubscribe=yes subscribecontext=extensiones notifyringing=yes notifyhold= yes limitonpeers= yes Daniel Arohuanca Lagos +51 1 994149553 Lima-Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081014/72edad82/attac...

...> > issue reports to implode - what's the configuration? > > > > Here's the sip.conf (only showing a single extension since they're all the > same): > [general] > directmedia=no > directrtpsetup=no > dtmfmode=rfc2833 > context=asterisk-internal > allowsubscribe=no > qualify=no > disallow=all > allow=ulaw > allow=alaw > allow=gsm > localnet=10.10.32.0/255.255.248.0 > localnet=192.168.32.0/255.255.255.0 > > [146] > secret= > host=dynamic > type=friend > > From the aforementioned sip debug capture, 146 is on the 10...

...Hi Ishfaq > Look into Busy Lamp Field/Presence > > Here's a starting point: > > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Thanks a lot, but it does not seems to work... Here my configuration: sip.conf: [general] allowsubscribe=yes subscribecontext = default notifyringing = yes notifycid = yes callcounter = yes extensions.conf: [anika_incoming] exten => _00493512222222,hint,SIP/00493511111111 exten => _00493512222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}]) exten => _00493512222222,n,Dial(local/2222222...

Andrew Martin wrote: > ----- Original Message ----- >> From: "Joshua Colp"<jcolp at digium.com> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users at lists.digium.com> >> Sent: Monday, May 11, 2015 12:32:06 PM >> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32

...t causes the call to be dropped if an ACK to the INVITE is not received within 32 seconds. How can I determine if this is the case and how can I resolve this "Retransmission timeout" problem? Here is my sip.conf: general] directmedia=no directrtpsetup=no dtmfmode=rfc2833 context=internal allowsubscribe=no qualify=no disallow=all allow=ulaw allow=alaw allow=gsm localnet=10.10.32.0/255.255.248.0 [123] secret=111111 host=dynamic type=friend Thanks! Andrew Martin

...ll times. Now I have my Sip.conf setup with externip= X.Y.Z.250 [general] port = 5060 bindaddr = 0.0.0.0 context = default allowoverlap=no srvlookup = yes : externip = externip = x.y.z.250 localnet=10.202.17.0/255.255.255.0 qualify=yes nat=yes register = xxxxxxx:SipServer/xxxxxxxx limitonpeers=yes allowsubscribe=yes notifyringing=yes notifyhold=yes useclientcode=yes canreinvite=no I have pfsense setup to forward ports 5060 and RTP ports over UDP back to the internal asterisk server. And a firewall rule to allow this traffic from only my ITSP SipServer. I can make a call from any phone on the local phones...

...an't seem to get to the bottom of it. I have multiple SIP DEBUG console logs and DEBUG/VERBOSE set to 4 logs around the time SIP stops responding. SIP.CONF: [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=gsm context=from-internal allowsubscribe=yes notifyhold=no limitonpeers=yes [701] type=friend secret=blahblah port=5060 host=dynamic dtmfmode=rfc2833 dial=SIP/701 context=from-internal canreinvite=no reinvite=no mailbox=701@default call-limit=9 allowsubscribe=yes Thanks for any help, Ken -------------- next part -------------- An HTML a...

...9.22>;tag=as4fffe111 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.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 = 3...

...insecure: NULL trustrpid: NULL progressinband: NULL promiscredir: NULL useclientcode: NULL accountcode: NULL setvar: NULL callerid: NULL amaflags: NULL callcounter: NULL busylevel: NULL allowoverlap: NULL allowsubscribe: NULL videosupport: NULL maxcallbitrate: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL t38pt_usertpsource: NULL regexten: NULL fromdomain: testers.com f...