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...end dbsecret=dundi/secret context=internal [voipprovider] type=friend host=voipprovider.web dtmfmode=rfc2833 insecure=port,invite disallow=all allow=g729 context=external [300] type=peer callerid=300 username=300 secret=secret host=dynamic context=internal mailbox=300 at default notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 [301] type=peer callerid=301 username=301 secret=secret host=dynamic context=internal mailbox=301 at default notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed....

...longer in 1.4... # sip.conf [general] bindport = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw tos=0x68 notifyringing=yes notifyhold=yes limitonpeers=yes [120] type=friend secret=secret record_out=Adhoc record_in=Adhoc qualify=yes port=5060 pickupgroup= nat=yes mailbox=120 at default host=dynamic dtmfmode=inband disallow= dial=SIP/120 context=from-internal canreinvite=...

...XX,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 mailbox=100 at default disallow=all allow=ulaw allow=alaw ;allow=g723.1 allow=g729 ;calling...

...address="station1" reg.1.label="101" sip.conf ;Polycom Phone [station1] disallow=all allow=ulaw allow=alaw type=peer context=internal host=dynamic dtmfmode=rfc2833 callerid="101" <101> nat=no qualify=yes canreinvite=no notifyringing=yes notifyhold=yes call-limit=99 ;X-Lite softphone [station2] disallow=all allow=ulaw allow=alaw type=peer context=internal host=dynamic dtmfmode=rfc2833 callerid="102" <102> nat=no qualify=yes canreinvite=no notifyringing=yes notifyhold=yes mailbox=1234 call-limit=99 exte...

...nd lot of other 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: dy...

I think I have my echo problem solved, now i need to tackle the MWI. I can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a section from my sip.conf for my test phone: [general] context=internal allowguest=no allowoverlap=no allowtransfer=yes notifyhold=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes vmexten=9998 at internal ;vmexten=*97 disallow=all allow=ulaw allow=ilbc mohinterpret=default mohsuggest=default language=en useragent=TCTC PBX ;dtmfmode = info fromdomain=10.10.60.253 ;relaxdtmf=yes [15] username=15 host=dynamic type=f...

...ting 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? Thanks in advance, Pan -------------- next part -----------...

...ion 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/attachment.htm

...ite=update directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=90 session-minse=120 session-refresher=uac register => 104:xxxxx at xxxxxx.com/104 defaultip=192.168.xx.xxx mailbox=104 disallow=all allow=ulaw,alaw artcachefriends=yes notifyhold=yes incominglimit=1 call-limit=3 Other information will be provided as asked for. Thanks in advance for any help you can provide. Danny Nicholas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachme...

...up 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 network to the outside world via...

...0.245 Http Traffic goes to 192.168.100.2 then via another internet link (ADSL 8bps connection) Router is preventing any traffic other than VoIP. for example we tried to pass HTTP requests via the internet link .. but did not go through. Asterisk Side: sip.conf sample: [GENERAL] notifyringing=yes notifyhold=yes limitonpeers=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes t38pt_udptl = yes bindport=5070 externip=SERVER_IP rtptimeout=60 session-timers=originate session-expires=600 session-minse=90 session-refresher=uas rtpholdtimeout=120 rtpkeepalive=20 allow=gsm t38pt_udptl=yes sendrpi...

...; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls subscribecontext = default ; Set a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes ; Notify subscriptions on HOLD state (default: no) limitonpeers=yes allow=ulaw [21] ;Bill Salmons type=peer username=21 callerid=Bill Salmons <21> secret=21 host=dynamic context=default mailbox=21 canreinvite=no nat=1 qualify=yes Subscribecontext=default...

....0.3-1.fc23.x86_64) Google Chrome (google-chrome-stable-47.0.2526.111-1.x86_64) SIP.js 0.7.2 I set up my SIP configuration to have two SIP accounts. Account 1000 is the Linphone and 1001 is the webrtc: [general] faxdetect=no vmexten=*97 context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes useragent=FPBX-2.11.0(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=tlsv...

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

...n extension (NOT a trunk). So, F1 = 201, F2 = 202, F3 = 203, and so on... I'm googled thousand of pages and many more confusing concepts are in my mind. My server uses extensions with numbering 2XX placed in context 'phones'. I set yet in sip.conf: limitonpeer=yes notifyringing=yes notifyhold=yes allowsubscribe=yes sip show peer 222 (222 is my test phone) give me... * Name : 222 Secret : <Set> MD5Secret : <Not set> Context : phones Subscr.Cont. : phones Language : it AMA flags : Unknown Transfer mode: open CallingPres : Presentat...

...d = yes call-limit = 6 My general section of sip.conf : [general] qualify=yes context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=srvcentral.meudominio.com.br tos_sip=cs3 tos_audio=ef tos_video=af41 language=pt_BR rtptimeout=60 rtpholdtimeout=300 notifyringing = no notifyhold = no limitonpeers = yes nat=yes rtcachefriends=yes rtsavesysname=yes rtupdate=yes Registration is working fine, the only problem I can see is qualify. Anybody can help me ? Marcelo H. Terres mhterres at gmail.com **************************************** ICQ: 6649932 MSN: mhterres at hotmail.com...

...language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100 disallow=all allow=g729 allow=g723 allow=ulaw allow=alaw allow=ilbc allow=gsm allow=h263 allow=h263p allow=h264 ***sip.conf*** [general] . . . . notifyringing=yes notifyhold=no rtupdate=yes rtcachefriends=yes ***extensions.conf*** [pbx9] exten => 8340,hint,SIP/8340 include => pruebas switch => Realtime/pbx9 at extensions In the Asterisk CLI i could see this message: [Jul 15 18:43:14] WARNING[15867]: chan_sip.c:15766 handle_request_subscribe: SUBSCRIBE fail...

...language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100 disallow=all allow=g729 allow=g723 allow=ulaw allow=alaw allow=ilbc allow=gsm allow=h263 allow=h263p allow=h264 ***sip.conf*** [general] . . . . notifyringing=yes notifyhold=no rtupdate=yes rtcachefriends=yes ***extensions.conf*** [pbx9] exten => 8340,hint,SIP/8340 include => pruebas switch => Realtime/pbx9 at extensions In the Asterisk CLI i could see this message: [Jul 15 18:43:14] WARNING[15867]: chan_sip.c:15766 handle_request_subscribe: SUBSCRIBE fail...

...; Enable DNS SRV lookups on outbound calls allowsubscribe=yes ; Disable support for subscriptions. (Default is yes) subscribecontext = default ; Set a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes ; Notify subscriptions on HOLD state (default: no) limitonpeers = yes ; Apply call limits on peers only. This will improve useclientcode=yes When more information is needed, please ask..............

Can anyone help with BLF for Snom 300s ? (Asterisk 1.4.10.1) I've setup hints for a couple of Snom 300's but Asterisk doesn't send Extension Changed messages to subscribed phones unless the second 'line' button is used (I've tried Snom's version 6 and 7 and two difference 300s). On the Asterisk Console I don't see any message when picking up a Snom 300 and dialing