Re: [asterisk-users] RTP port ranges
Maybe this could help you: http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf Am 13.09.2013 11:49, schrieb Jonas Kellens: Hello, and when I define 11500 - 11954 it should use a random port in this range. Where is it stated that you MUST use 1-2 ??? Someone else please ? Jonas. On 09/13/2013 11:46 AM, Andrew Colin wrote: Because normally it will use a random port between them On 9/13/2013 11:43 AM, Jonas Kellens wrote: On 09/13/2013 11:41 AM, Andrew Colin wrote: Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP not being switched between both SIP endpoints
We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call coming from opensips. Directmedia is not specifically defined so its using the asterisk default value. * Name : vmpubopensips3 Description : Secret : Not set MD5Secret: Not set Remote Secret: Not set Context : from-pubopensips Record On feature : automon Record Off feature : automon Subscr.Cont. : Not set Language : Tonezone : Not set AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : no Force rport : Auto (No) Symmetric RTP: No ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 88.x.x.x Addr-IP : 88.x.x.x:5060 Defaddr-IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : (gsm|ulaw|alaw) Codec Order : (alaw:20,ulaw:20,gsm:20) Auto-Framing : No Status : Unmonitored Useragent: Reg. Contact : Qualify Freq : 6 ms Keepalive: 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No When the call comes in the SDP contains :- v=0. o=root 973184584 973184584 IN IP4 81.x.x.x s=session. c=IN IP4 81.x.x.x t=0 0. m=audio 11370 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and we reply back with :- v=0. o=root 822402971 822402971 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10428 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. When we send the outbound SIP information we advertise the following SDP :- v=0. o=root 431105643 431105643 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10144 RTP/AVP 8 3 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and the other end replies with :- v=0. o=hksbc1a 609621538 609621538 IN IP4 203.x.x.x s=sip call. c=IN IP4 203.x.x.x t=0 0. m=audio 34146 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. In the Dial() command the only option we are using is M() which is used to run a macro when the call is answered. This is used to update cdr records and perform other features if they are enabled. In this case we are just updating the cdr records so I would expect the audio to be switched as soon as the macro finishes. Any ideas what could be wrong? We are running Asterisk PBX 11.2-cert2 Thanks Gareth -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8 sends SIP/2.0 481 Call/Transaction Does Not Exist to INVITE
To: sip:8009499...@x.yyy.32.10:5060 ;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 In your call sample To has a tag. if this is the first Invite it can't have a tag at the end, otherwise Asterisk will look for an existing dialog in its database and will show an error, if can't find any. It looks like the other end is never closing the previous dialog?.. is Asterisk sending a proper 200 OK after receiving a BYE? NAT problem? Thanks, I think you are correct on that... there are no NAT problems... the dialog ends with Asterisk sending a 481 because the dialog does not exist. Im going to try to have the customer remove that tag from the To header. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Goto(${PREV_CONTEXT},s,1) functionality?
I have a moderately complicated ivr menu, implemented in an AEL dialplan. A
feature I'm struggling to add now is going back from current context to a
previous one, so that user could browse almost the whole ivr menu manually.
The difficulty is that when current context has several entry points i.e.
there are several contexts that can lead to current, it is impossible to
implement a go back feature using simple static Goto(${foo},s,1).
At some point, I thought that Gosub() was a solution, but 1) it is not
recommended for using in *.ael (with warnings everywhere) and 2) it can be
only used to return to some defined point of prev context, not its start!
Maybe I'm missing something and there actually *is* some ${PREV_CONTEXT}
channel variable, helping alot with navigation?
If not, what is the best way to implement a go back feature there is?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-1.8.23.1 mysql cdr
Hi list, Reply to my own question http://lists.digium.com/pipermail/asterisk-users/2013-September/280541.html I come up with a patch that enable timezone support. add new configuration cdrzone option in cdr_mysql.conf. in cdr_mysql.conf add cdrzone= any valid timezone(consult /usr/share/zoneifo) disable all existing options usegmtime etc. added new cli option cdr mysql cdrzone. it will show you selected timezone. patch can be download from http://www.world-call-trade.com/asterisk/cdr_mysql_cdrzone.patch please report back here. BEST REGARDS Asghar Mohammad -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
