search for: rtpholdtimeout

Hello everybody, I'm having issues with calls being dropped on Aastra phones, when the call is on hold. Tested with models 6863i and 6867i. I've figured that the call is dropped by Asterisk when it reaches the rtpholdtimeout limit. I've reported the issue to Aastra, asking them to implement some kind of "RTP keep-alive" feature on their phones. Maybe the phone could send some RTCP frame (or an empty RTP frame) just to prove it is alive. Unfortunately Aastra said the hold behaviour on the phone is corre...

...bug stuff, and all I can see is my asterisk sending the registration packets, but no answer is received. Here's the relevant parts of my sip.conf: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes tos=0x18 checkmwi=10 videosupport=yes allow=all relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 register => XXXXX:pass@sipgate.de/XXXXX ;XXXXX == sipgateid [XXXXX] type=friend insecure=very nat=yes username=XXXXX fromuser=XXXXX fromdomain=sipgate.de secret=pass host=sipgate.de qualify=yes -- Michiel van Baak michiel@vanbaak.info http://michiel.vanbaak.info GnuPG key: http://pgp.mi...

...but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good response time to be check from time to time. defaultexpiry = 900 defaultexpirey = 900 maxexpiry = 300 maxexpirey = 300 minexpiry = 60 registerattempts = 5 registertimeout = 5 rtpholdtimeout = 900 rtptimeout = 60 jbmaxsize = 60 jbresyncthreshold = 200 qualify = yes qualify = 600 qualifyfreq = 60 Thank you. P.S. If you consider that i use too much options you can tell me what to drop. I use asterisk 1.8.6.0. -------------- next part -------------- An HTML attachment was scrubbed... UR...

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [10000] type=friend secret=test host=dynamic nat=yes -------------------------- ----- extensions.conf: ----- [common] exten => 501,1,MeetMe(12,MI) exten => 501,n,Hangup() exten => i,1,Hangup() exten => h,1...

...port | varchar(5) | | | | | | qualify | varchar(4) | YES | | NULL | | | restrictcid | char(1) | YES | | NULL | | | rtptimeout | char(3) | YES | | NULL | | | rtpholdtimeout | char(3) | YES | | NULL | | | secret | varchar(30) | YES | | NULL | | | type | varchar(6) | | | | | | username | varchar(30) | | | | | | allow...

...gus-calls bindport=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) maxexpirey=3600 ; Must be larger than the re-register timeout on the router defaultexpirey=3600 notifymimetype=text/plain rtptimeout=60 rtpholdtimeout=300 disallow=all allow=ulaw ; ; This section is because i'm behind nat ; register=>6477235412:<mypassword>@sip.unlimitel.ca/6477235412 externip=<mystaticIPaddress> ;Outside address localnet=192.168.0.148/255.255.255.0 ;Inside Network ; ;******************************************...

Hi All, UA <--------> Asterisk Server <---------> UB if there is no rtp for a specified number of minutes / seconds then I want to disconnect the call. I've tried using rtptimout and rtpholdtimeout but no luck pls guide. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070917/7ee005e6/attachment.htm

I have a couple asterisk boxes, running sip between both boxes. 1.4.43 on both. both are installed from source, both have default settings, My config for one box is: [devgeis] type=friend defaultname=devgeis username=devgeis secret=yes disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 host=192.168.1.8 context=panel The other box is the same. There are times when "sip show peers" has Unspecified like: devgeis/devgeis (Unspecified) D a 0 Unmonitored So the registration is lost. But a short time later...

...e DID dialed and am hoping someone on the list can assist me. Here's the relevant info: Ingress SIP trunk: IP: 123.45.45.3456 DID's XXX-XXX-XX00-XX10 sip.conf: [general] useragent=Asterisk port=5060 context=default tos=lowdelay disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=300 rtpholdtimeout=600 // // My thought in this context is I will grab any incoming SIP call from the IP address of my SIP trunk and pass it to my sip-defaul-in context // // in extensions.conf [sip-default-in] type=friend defaultip=123.45.3456 host=123.45.3456 nat=no insecure=very context=sip-default-in canreinvit...

We have had 20 calls over the last month where the SIP channel has not identified that the person on the receiving end has hung up. Is there a way of fixing this ? TIA Julian

...realtime). Sometimes a call will disconnect 30 seconds after the SIP phone hits the mute button but it doesn't happen all the time. I've done a sip debug while watching this happen and that doesn't show anything other than a BYE message being sent out of the blue. The rtptimeout and rtpholdtimeout are both set to 0 on a global level and for the sip extension the sip table row has NULL in both columns. I've tried playing with those 2 values, both on a global and sip extension level but regardless to what they are set to, if the call gets disconnected it is always 30 seconds after the...

...== Using SIP RTP CoS mark 5 -- Called SIP/526 -- SIP/526-000000ac is ringing 526 is the extension in question. (my definition follows): [526] type=friend defaultname=526 defaultuser=526 secret=XXXXXXXXX dtmfmode=RFC2833 host=dynamic description=Polycom context=sip qualify=yes rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 callerid="Polycom " qualify=no canreinvite=yes timezone=1 nat=force_rport,comedia disallow=all allow=ulaw allow=alaw allow=gsm Thoughts on what is happening here or what to try? Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <...

...efault-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: NULL host: dynamic insecure: NULL language: NULL mailbox: 21168 at device md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: xxx type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1200593168 ipaddr: xxx.xxx.xxx.xxx regexten: cancallforward: yes setvar: Any help would be appreciated. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. atis at iq-labs.net Skype: atis...

...ueid | name | accountcode | amaflags | callgroup | callerid | canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | host | incominglimit | outgoinglimit | insecure | language | mailbox | md5secret | nat | permit | deny | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | type | username | allow | disallow | regseconds | ipaddr | auth | +----------+------+-------------+----------+-----------+----------+-------------+---------+-----------+----------+----------+------------+---------+---------------+---------------+----------+----------+---------+-----...

...11313fc173a 00102/00000 0x0 (nothing) No Tx: CANCEL 2. Asterisk version: *1.4.21.1* 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... UR...

...`permit` varchar(95) default NULL, `mask` varchar(95) default NULL, `musiconhold` varchar(100) default NULL, `pickupgroup` varchar(10) default NULL, `qualify` char(3) default NULL, `regexten` varchar(80) default NULL, `restrictcid` char(3) default NULL, `rtptimeout` char(3) default NULL, `rtpholdtimeout` char(3) default NULL, `secret` varchar(80) default NULL, `setvar` varchar(100) default NULL, `disallow` varchar(100) default 'all', `allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw', `fullcontact` varchar(80) NOT NULL default '', `ipaddr` varchar(15) NOT NULL def...

...charging and this is a wrong. This problem was not appearing when Asterisk machine was having static real IP address because I was enabling the rtptimeout paramters. But now as the asterisk box IP address is private and it is behind NATing then it is appearing even I enabled the (rtptimeout=50 and rtpholdtimeout=120). What should I do? Regards Bilal

...peers' (I have confirmed that this is dependent on the order in the conf file, not numeric order) sip.conf :- [general] port = 5060 bindaddr = 0.0.0.0 pedantic = no autocreatepeer = no context = sip registertimeout=20 localnet = 10.10.10.0/255.255.255.0 srvlookup = yes tos=0xb8 rtptimeout=300 rtpholdtimeout=1800 maxexpirey=3600 defaultexpirey=1200 [sip-101] ; Aastra 480i phones for general office type=peer insecure=very disallow=all allow=ulaw allow=alaw host=dynamic dtmfmode=auto canreinvite=no context=office-dial qualify=yes username=101 secret=xxxxxx mailbox=101 callerid="User 1" <101...

...| accountcode | amaflags | callgroup | callerid | canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | fullcontact | host | insecure | language | mailbox | md5secret | nat | deny | permit | mask | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | type | username | disallow | allow | musiconhold | regseconds | ipaddr | regexten | cancallforward | setvar | +----+-----------------------+-------------+----------+-----------+------------+-------------+---------+-----------+----------+----------+------------+---------...

...`permit` varchar(95) default NULL, `deny` varchar(95) default NULL, `mask` varchar(95) default NULL, `pickupgroup` varchar(10) default NULL, `port` varchar(5) NOT NULL default '', `qualify` char(3) default NULL, `restrictcid` char(1) default NULL, `rtptimeout` char(3) default NULL, `rtpholdtimeout` char(3) default NULL, `secret` varchar(80) default NULL, `type` varchar(6) NOT NULL default 'friend', `username` varchar(80) NOT NULL default '', `disallow` varchar(100) default 'all', `allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw', `musiconhold` varch...