search for: rtpkeepalive

Hi list! When I receive an incoming call from a SIP peer where I've configured disallow=all allow=alaw (and no other codec) I can see the following NOTICE on the console: Dropping incompatible voice frame SIP/peer07-0000007c of format ulaw since our native format has changed to (alaw) My question is: where can I change the native format from ulaw to alaw (or something else)? Is ulaw, as

...oks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0' In sip.conf I have rtpkeepalive=15 but that does not seem to help. Does anyone know what I can do to fix this, other than increase the timeout on Bria? Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080408/0f205c55/att...

...N Gateway supports symmetric RTP and early media using 183 Session Progress. So If I call a PSTN number which has IVR message played before the call is connected (via 183), those media RTP packets do not reach the asterisk inside till asterisk sends out media packet to the PSTN gateway. I have used rtpkeepalive option and set it to 1 sec. But it seems that I drop rtp voice packets in the initial instructions played by the IVR. How do I make sure that asterisk sends RTP packets (null rtp) to the PSTN gateway just after receiving the media details in 183 SDP to open the firewall port from inside? Re...

...f 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,Hangup() exten =&...

Hi Joshua > The "rtp_keepalive" option can be used to have the RTP stack send an > RTP packet out. Try that and see what happens. Once again 'bullseye' that fixed the problem. Thank you! Mit freundlichen Gr?ssen -Beno?t Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29

Hey guys, I'm using asterisk 1.6.2.13 and have an endpoint which uses silence suppression which I can't turn off. I've set rtpkeepalive=10 in sip.conf [general], as well as under the peer details for our sip provider but it doesn't seem to do anything. Rtp debug shows that we are receiving RTP from the SIP provider, and forwarding it to the end point, but no RTP packets are sent back to the provider (ie. No keep alives). I did...

...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 I look again an...

...s is happening only in the analoge channels. What other than the rtptimeout, the hangup in the extensions.conf, the voicemail? Is there anything I have to take care for it that might cause this stuck and keeping the channel openned? By the way, for such cases, what should I place the value of the rtpkeepalive as currently it is 0? What other things I have to take care for it? Regards Bilal

...oS 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: <http://lists.dig...

Greetings, (Apologies if this is an FAQ, but I've Googled for hours and haven't come up with anything yet.) I have an Asterisk system deployed at a customer's site. It is connected to the outside world by a local SIP provider. When someone calls in through the trunk to leave a voicemail, Asterisk is not sending any RTP packets back through the trunk after the beep is played. This

...terns. *sip.conf:* registerattempts=0 registertimeout=20 *peer confifuration:* [XXXX-friend] disallow=all host=192.168.1.1 defaultuser=<phone number> fromuser=<phone number> callerid=<phone number> secret=<ISP secret> type=friend qualify=yes allow=ulaw allow=alaw nat=no rtpkeepalive=10 dtmfmode=rfc2833 insecure=port,invite context=from-trunk-ISP1 fromdomain=<ISP domain> *registration string:* register=<phone number>:<ISP secret>@<ISP domain>/<phone number> *where:* <phone number> is our ISP-provided phone number <ISP secret> is ou...

...fromuser: NULL qualify: NULL defaultip: NULL outboundproxy: PU.BL.IC.IP contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/keys/ca.crt sippasswd: md5ofmypwd rpid: NULL domain: testers....

...ing) if silence suppression is disabled. Just as I would expect any end point to send 'silence' if it was muted when silence suppression was disabled. It seems that RTP keepalives would serve this purpose, however this doesn't seem to be available either... Should I file a bug report re rtpkeepalive? Sent from my iPhone On 29/01/2011, at 12:55 AM, "Kevin P. Fleming" <kpfleming at digium.com> wrote: > On 01/27/2011 10:52 PM, Ryan Tucker wrote: >> So, I've done some more testing and got some more info. >> >> I have one endpoint that does silence suppre...

...nf 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 sendrpid=yes trustrpid=no directrtpsetup=yes [USERNAME] deny=0.0.0.0/0.0.0.0 type=friend secret=PASSWORD qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes host=dynamic dtmfmode=rfc2833 disallow=all allow=gsm context=from-callcenter canreinvite=no we have a call re...

...m fromuser: 660 qualify: NULL defaultip: NULL outboundproxy: 1.1.1.1 contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/keys/ca.crt sippasswd: a84a4ddcda13d13c9573d5294047b6a2 rpid: NULL...

...auth: NULL fullname: NULL trunkname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL parkinglot: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL autoframing: NULL rtpkeepalive: NULL call-limit: NULL g726nonstandard: NULL ignoresdpversion: NULL allowtransfer: NULL dynamic: NULL path: NULL supportpath: NULL sippasswd: my-md5-pwd rpid: NULL domain: testers.com sippasswd2: NULL I...

...255.0 localnet=192.168.1.0/255.255.252.0 localnet=10.0.0.0/255.255.255.0 One phone config: (both are the same) [YYYYY] type=friend defaultname=YYYYY defaultuser=YYYYY secret=notshown dtmfmode=RFC2833 host=dynamic description=testing. context=some-context-that-works rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 callerid=YYYYYY qualify=yes insecure= canreinvite=yes timezone=0 nat=force_rport,comedia disallow=all allow=ulaw allow=alaw allow=gsm Which accounts for all locations. Why might I not be getting audio ? Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <...

...appreciated. And by the way, my Asterisk box is talking to a Level 3 SIP gateway with the following configuration: [bandwidth] type=peer host=x.x.x.x dtmfmode=rfc2833 canreinvite=no disallow=all allow=ulaw context=incoming reinvite=no canreinvite=no nat=no directrtpsetup=yes rfc2833compensate=yes rtpkeepalive=60 Thanks in advance! - Justin Tunney

Hi, I've got a problem with rtp keepalives. I'm using basically the same config on 2 hosts, but one of them sends rtp comfort noise when it's on hold, the other doesn't. The only difference I can think of now is that one of the machines is multihomed, but that might be unrelated. rtpkeepalive is set to 2 and I can confirm is by doing `sip show settings`. I've tried all combinations of nat and qualify for the peer that has problems - rtp comfort noise is simply not sent. After trying to make it work for a day or so, I reported it as a bug (https://issues.asterisk.org/view.php?id=1546...

I have had a case where after a hangup on the Alsa channel asterisk still thinks the line or call is active. I have: rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 in my sip.conf file to help with this but it had no effect. How can I ensure a session HANGS up and is not stale???? Is there a way for the next incoming call to VERIFY that console/ALSA channel is still valid. I dont want to hangup a real connection - I want to give a busy tone for sure....