Atis Lezdins
2008-Apr-22 11:56 UTC
[asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS
Hi, I experience my log flooded with warning messages like this: [Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match request NOTIFY to call '5239fdfb15d20e5153a2f7365ce2ea4d at 69.80.215.12'. Giving up I traced this down to point when we added to sip.conf status notifications: allowsubscribe=yes rtcachefriends=yes So, those notifications allow for queue to display (In Use) etc, and creates no warnings for other devices except Audiocodes gateway. I wonder is there any way how to disable this message in Asterisk, or make Audiocodes act correctly? Below is the sip debug for this (xx.xx.xx.xx is Audiocodes, yy.yy.yy.yy is Asterisk). Regards, Atis ------------------------------------------------------------------------------------- [Apr 14 01:30:24] VERBOSE[19514] logger.c: Scheduling destruction of SIP dialog '5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy' in 32000 ms (Method: NOTIFY) [Apr 14 01:30:24] VERBOSE[19514] logger.c: Reliably Transmitting (NAT) to xx.xx.xx.xx:5060: NOTIFY sip:90170 at xx.xx.xx.xx SIP/2.0 Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport From: "Unknown" <sip:Unknown at yy.yy.yy.yy>;tag=as436bf308 To: <sip:90170 at xx.xx.xx.xx> Contact: <sip:Unknown at yy.yy.yy.yy> Call-ID: 5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:asterisk at yy.yy.yy.yy Voice-Message: 0/0 (0/0) --- [Apr 14 01:30:24] VERBOSE[19514] logger.c: <--- SIP read from xx.xx.xx.xx:5060 ---> SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport From: "Unknown" <sip:Unknown at yy.yy.yy.yy>;tag=as436bf308 To: <sip:90170 at xx.xx.xx.xx>;tag=1c73477527 Call-ID: 5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy CSeq: 102 NOTIFY Contact: <sip:xx.xx.xx.xx> Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> [Apr 14 01:30:24] VERBOSE[19514] logger.c: --- (10 headers 0 lines) --- [Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match request NOTIFY to call '5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy'. Giving up. -- Atis Lezdins, VoIP Project Manager / Developer, atis at iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835
Grey Man
2008-Apr-22 12:15 UTC
[asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS
For blind transfers Asterisk will send the call back to the dial plan and into the TRANSFER (I think, could be a different name) context if it exists. Within that context you can access the channel that was answered on the original call using ${DIALEDPEERNUMBER}. Note that this mechanism cannot be use for attended transfers as they are not sent back to the dial plan for processing. Regards, Greyman.
Johansson Olle E
2008-Apr-23 14:42 UTC
[asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS
22 apr 2008 kl. 13.56 skrev Atis Lezdins:> Hi, > > I experience my log flooded with warning messages like this: > > [Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match > request NOTIFY to call > '5239fdfb15d20e5153a2f7365ce2ea4d at 69.80.215.12'. Giving up > > I traced this down to point when we added to sip.conf status > notifications: > > allowsubscribe=yes > rtcachefriends=yesNo, that is wrong. What you have below is a voicemail notification. You have to remove the mailbox= in the peer configuration in order to not have any voicemail notifications. The reason why this error message is coming up, is that the device should SUBSCRIBE for the notifications in order to get them. Out of habit, Asterisk sends these without subscriptions as default, but you can configure asterisk to handle this on a subscription basis. /O> > > So, those notifications allow for queue to display (In Use) etc, and > creates no warnings for other devices except Audiocodes gateway. > > I wonder is there any way how to disable this message in Asterisk, or > make Audiocodes act correctly? > > Below is the sip debug for this (xx.xx.xx.xx is Audiocodes, > yy.yy.yy.yy is Asterisk). > > Regards, > Atis > > ------------------------------------------------------------------------------------- > > > [Apr 14 01:30:24] VERBOSE[19514] logger.c: Scheduling destruction of > SIP dialog '5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy' in 32000 ms > (Method: NOTIFY) > [Apr 14 01:30:24] VERBOSE[19514] logger.c: Reliably Transmitting (NAT) > to xx.xx.xx.xx:5060: > NOTIFY sip:90170 at xx.xx.xx.xx SIP/2.0 > Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport > From: "Unknown" <sip:Unknown at yy.yy.yy.yy>;tag=as436bf308 > To: <sip:90170 at xx.xx.xx.xx> > Contact: <sip:Unknown at yy.yy.yy.yy> > Call-ID: 5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy > CSeq: 102 NOTIFY > User-Agent: Asterisk PBX > Max-Forwards: 70 > Event: message-summary > Content-Type: application/simple-message-summary > Content-Length: 92 > > Messages-Waiting: no > Message-Account: sip:asterisk at yy.yy.yy.yy > Voice-Message: 0/0 (0/0) > > --- > [Apr 14 01:30:24] VERBOSE[19514] logger.c: > <--- SIP read from xx.xx.xx.xx:5060 ---> > SIP/2.0 481 Call/Transaction Does Not Exist > Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport > From: "Unknown" <sip:Unknown at yy.yy.yy.yy>;tag=as436bf308 > To: <sip:90170 at xx.xx.xx.xx>;tag=1c73477527 > Call-ID: 5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy > CSeq: 102 NOTIFY > Contact: <sip:xx.xx.xx.xx> > Supported: em,timer,replaces,path > Allow: > REGISTER > ,OPTIONS > ,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE > Content-Length: 0 > > > <-------------> > [Apr 14 01:30:24] VERBOSE[19514] logger.c: --- (10 headers 0 lines) > --- > [Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match > request NOTIFY to call '5239fdfb15d20e5153a2f7365ce2ea4d at yy.yy.yy.yy'. > Giving up. > > > -- > Atis Lezdins, > VoIP Project Manager / Developer, > atis at iq-labs.net > Skype: atis.lezdins > Cell Phone: +371 28806004 > Cell Phone: +1 800 7300689 > Work phone: +1 800 7502835 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users--- * Olle E Johansson - oej at edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden