[asterisk-users] voicemail quota
Hi, I am using asterisk-1.4.11. Voicemail quotas only apply to the new messages in the INBOX. Browsing quickly through the 1.6 app_voicemail it seems that 1.6 does implement voicemail quota for both INBOX and Old messages. Is that correct? If so, is there an existing patch available that anyone would know of? Thanks, Sebastien. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no audio, firewall problem?
Hello, I am runing asterisk on a embedded linux and am having some RTP audio issues at the beginning of the call: the comfort noise packet seems to be opening the pinhole in the firewall though I don't understand why it is not already opened. Then audio is then transferred correctly between caller and callee through the asterisk bridge. The SIP INVITE is received on a WAN interface and then I dial out to another SIP channel through the same interface. CLI output with RTP debug shows that Packet2Packet is only started and RTP is only sent by asterisk after the first rtpkeepalive timeout. If I sniff at a mirroring port in the network I can see the first RTP packet going from my caller to the asterisk server yet it seems that it is never received (or it never reaches) asterisk (it is a direct route). All firewall rules on the asterisk box are setup for the range of ports defined by rtp.conf (10k-11k in mycase); that is consistent with the SDP signaling generated by asterisk for the INVITE OUT and for the 200 OK back to the caller in the media description attribute. Watching iptables live activation does not show any RTP packet blocked at the beginning of the call. netstat shows: netstat -an | grep udp | grep 10 netstat: no support for 'AF INET6 (tcp)' on this system netstat: no support for 'AF INET6 (udp)' on this system netstat: no support for 'AF INET6 (raw)' on this system udp0 0 216.54.141.148:105540.0.0.0:* udp0 0 216.54.141.148:105550.0.0.0:* udp0 0 216.54.141.148:101020.0.0.0:* udp0 0 216.54.141.148:101030.0.0.0:* as I am using bindaddr=0.0.0.0 in the sip.conf. I have multiple NICs on that box, could it be a problem or ...? Thanks for any suggestion, Sebastien. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP sent before the INVITE ACK (for voicemail app)
Thanks, in my case though it looks like the originating party (polycom softphone) is hearing a clipped voicemail prompt because of that; should I look into having that fixed into their firmware? As a workaround, I was thinking to just add a few seconds delay in app_voicemail, or wait through AGI before calling voicemail, makes sense? > Date: Wed, 1 Oct 2008 15:43:37 +0100 > From: [EMAIL PROTECTED] > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] RTP sent before the INVITE ACK (for voicemail > app) > > > > > CLI output does not show any error that I see. Is there any reason other > > than a SIP 183 that would trigger this and isn't asterisk supposed to > > ACK/answer the channel before playing any prompt? > > > > Asterisk wil start the audio as soon as it sends back the 200 Ok > response it doesn't wait for the ACK. Most SIP servers will work like > that. The matching of ACK requests to a SIP transaction is not a > particulalrly robust mechanism (for instance if a user agent puts its > IP address in the Call-ID and a SIP ALG fiddles with the SIP packet > for INVITEs but ignores ACKs then there will be a mismatch. This > happens more frequently then you would think) so sending RTP after an > OK response is the correct thing to do. > > I think Asterisk will actually cut off the call after 32s if it > doesn't get an ACK which is not such a great idea but that may have > been changed in later versions. The arrival of an RTP packet from the > remote end should be used as the definitive indication of an answered > call not the ACK. > > Regards, > > Greyman. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP sent before the INVITE ACK (for voicemail app)
Hello, With asterisk 1.4.11, I am calling AGI "exec voicemail" upon a SIP INVITE invite -> asterisk <- 100 <- 200 <- RTP ACK -> ... asterisk is sending the RTP for the greeting before the original invite is ACK-ed (confirmed with a tcpdump) as if playing the prompt as soon as it is received from the AGI. I don't see any 183 so I don't think early media should apply. CLI output does not show any error that I see. Is there any reason other than a SIP 183 that would trigger this and isn't asterisk supposed to ACK/answer the channel before playing any prompt? Thanks, Sebastien. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rtpkeepalive problem ?
Hello, I'm having a problem when registering a x-lite to my asterisk server and bridging the xlite SIP channel to a PSTN SIP channel; in such case, the audio paths are only created x seconds after rtpkeepalive expires. If I set rtpkeepalive to 0, I never get the audio paths. I wiresharked it and can see the ComfortNoise packet beeing sent 10 seconds later (my rtpkeepalive value); I tried using nat=yes, withtout any change. Any help on where I should start? Thanks. Sebastien. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
