search for: notifyringing

...s just not doing quite what I want. It may not be possible, but I figured it was worth asking about. The details: Asterisk 11.6.0 Polycom SoundPoint IP650 phones running 4.03 firmware. We have a queue with 4 phones in it. ringinuse is set to yes and the stategy is ringall. In sip.conf, we have notifyringing set to yes as well. Asterisk is sending messages of the type application/dialog-info+xml to the phones. This works nicely in almost every scenario. We have one person on the queue who answers the phones first, the rest of us only pick up if he is already on another call and not picking up. We...

" Hello, I have been trying to get my Grandstream GXP2000 phones to stop showing ringing state on monitored extensions. But no matter where I put notifyringing=no asterisk still sends the ringing state to the phones. Is this a bug I should report or is there another way around it. Here is how i have my subscriptions setup: extensions.conf [demo] exten => 6100,hint,SIP/100 exten => 6101,hint,SIP/101 exten => 6102,hint,SIP/102 exten => 6103,h...

...; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; 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 conte...

...hone" and so it sends "Ringing". This makes the "busy lights" less than useful, if a call makes multiple phones ring you can't tell, looking at the busy lights, which ones are busy, and so less likely to answer. In the chan_sip configuration there is an option "notifyringing": notifyringing *notifyringing* enables or disables notifications for the RINGING state when an extension is already INUSE. Only affects subscriptions using the *dialog-info* event package. Option can be configured in the general section only. It cannot be set per-pe...

Hello With 1.4.23.1, I can't really see any difference between setting this value to yes or no. Can you explain ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090304/ccf0f222/attachment.htm

Hello list, I read this in sip.conf : notifyringing = no ; Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes) What does this mean ?! Does this mean that when I mark this as "yes", a phone that already has taken a call will be send a second and third call ?! I want that if...

Hi, I set verbose to 3, but I do not see any RINGING notification in the CLI. On the contrary, when the phone goes UNREACHABLE I get: [Dec 13 21:10:06] NOTICE[9988]: chan_sip.c:25533 sip_poke_noanswer: Peer '152' is now UNREACHABLE! Last qualify: 130 == Extension Changed 152[blf] new state Unavailable for Notify User 154 [Dec 13 21:11:08] NOTICE[9988]: chan_sip.c:20196

..."Ringing". > > This makes the "busy lights" less than useful, if a call makes multiple > phones ring you can't tell, looking at the busy lights, which ones are > busy, and so less likely to answer. > > In the chan_sip configuration there is an option "notifyringing": > > notifyringing > > *notifyringing* enables or disables notifications for the RINGING state > when an extension is already INUSE. Only affects subscriptions using the > *dialog-info* event package. Option can be configured in the general > section only. It cannot be se...

Hi All Has anyone managed to get the hint priority with Swissvoice IP10S phones working? I have 2 phones: a Snom 360, setup as the reception phone on extension 11, and a Swissvoice IP10S on extension 12. When calling each other (tested both ways) I can only ever see the Snom 360 in the Active State from 'show hints'. The Swissvoice stubbornly remains in the Idle State when on a call!

...undified] type=friend 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 attachmen...

...rked, but it does not 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-intern...

...RIBE in my sip debug traces. I have problem to understand how hint priority works. I follow the instructions from http://www.voip-info.org/wiki/index.php page=Asterisk+standard+extensions but it still doesn't work. [2001] type=friend context=local host=dynamic dtmfmode=rfc2833 incominglimit=1 notifyringing=yes subscribecontext=notify disallow=all allow=alaw allow=ulaw [2002] type=friend context=local host=dynamic dtmfmode=rfc2833 incominglimit=1 notifyringing=yes subscribecontext=notify disallow=all allow=alaw allow=ulaw [local] exten => 2001,1,Dial(SIP/2001,10,tr) exten => 2002,1,Dial(SIP/20...

...ne1.cfg: reg.1.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-l...

...a snom360 and xlite could dial out without 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...

...pe=friend username = 235 callerid="R Vermeeren mobiel" <235> host = dynamic secret = 235 context = default qualify = yes login = 235 callgroup = 3 pickupgroup = 3 disallow = all allow = alaw call-limit = 6 default section of sip.conf [general] dtmfmode=rfc2833 rfc2833compensate=yes notifyringing=yes context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) ; bindport is the local UDP...

...ked UPGRADE.txt, and 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...

...h 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? Thanks in advance, Pan -------------- next p...

...p.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/attachment.htm

Hello Asterisk community, I'm trying to use BLF with Asterisk Realtime, i've been searching for some info but nothing seems to be clear, can anyone help me eith some ideas to make this work ok? I'va my dialplan with Realtime Thanks in advance -- Saludos Danny Dias SkypeID: danny.dias1

...rst disallow all codecs allow=g729 allow=gsm allow=alaw ; Allow codecs in order of preference dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes I have the definition of the phone in DB. voip-test-01*CLI> sip show peer 0625037998 voip-test-01*CLI> * Name : 0625037998 Realtime peer: No Secret : <Set> MD5Secret : <Not set> Context : sipresidential Subscr.Con...