search for: rtpstart

...fe to do? I've got the "Allow Anonymous Inbound SIP Calls" box unchecked in freePBX. Is there anything else I need to do? Isn't there an issue with the extension/secret being passed in clear text? It looks like I need to open port 5060, and whatever ports are inbetween the rtpstart/rtpend values in /etc/asterisk/rtp.conf. Is that right? Right now thats 9999 ports, I've read that you can chop that down to 20 ports for just a few calls. We want to have 5-6 simultaneous calls, so if I set rtpstart to 10001 and rtpend to 10100, then open up those ports, is that adequat...

Hello, using Asterisk 1.4.28, I encountered a problem with SIP RTP port allocation. I found some entries in mailinglist and bugtracker regarding this issue, but only old ones. My rtp.conf has [general] rtpstart=30000 rtpend=30100 so 100 ports available. I know that up to 4 ports per channel can be used and so up to 25 channels are possible. But even earlier I often get the error about "No RTP ports remaining". I had a look at netstat -nuap and it shows that a lot of ports are still assigne...

...vider and after internal manage it makes a second call to other destination--DID--. For AGI compatibility issues I could not use Version 1.4.22 (issues whith DeadAGI for billing purpuses). This is my rtp.conf [general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 ; rtpstart=10000 rtpend=20000 This is my sip.conf for the TRUNK [TRUNK] type=peer nat=never host=destination.public.ip.address fromdomain=my.public.ip.address dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 Please help. -- Juan E. Rodr?guez -------------- next pa...

...rks fine.</font></div> <div><font face=3D"Arial" size=3D"2">What is the problem? Thank you in=20 advance</font></div></blockquote><div><br>a pointer to check, in rtp.conf j= ust make sure that rtp start port is set explicitly rtpstart=3D10000, cause= default rtpstart is 5000 so opening port 10000-20000 in router without set= ting this may not help.<br> &nbsp; <br></div></div>-- <br>Thanks &amp; Regards,<br>Godson Gera<br><a hr= ef=3D"http://godson.in/voip-asterisk-consul...

Hello may situation is the next: Asterisk <--> NAT1 (router)<---> internet <--> NAT2 (router) <--> x-lite ^ | ip phone (cisco) Asterisk and de cisco phone are in the same LAN. I want to make a call between the x-lite and the ip phone. I can do the call but there is only audio from de ip-phone

...;t have the RTP ports open, either. I had already been wondering about this because I thought there would have to be ports open for 'canreinvite=NO' to work. > Any source to UDP ports X to Y (10000 to 20000 by default) allow. Are you saying that the ports specified in rtp.conf ('rtpstart' and 'rtpend') specify with ports asterisk uses regardless whether RTP or SRTP is being used? Is that why you speak of "media" (ports)? (That would have been and would answer my original question: Where to specify the SRTP ports?) > What you can't do is limit the...

...=<username> secret=<password> port=5060 canreinvite=YES dtmfmode=rfc2833 I've tried opening all ports to test this but, still doesn't work. Now, I need to know which especific ports to open in order to allow sip flow correctly. Also enabled/opened ports 5060 - 5070 and the rtp: rtpstart=10000 rtpend=20000 Don't know what else to try. Please help. Thanks in advanced for your help.

...bound calls allow = all [1000] nat=yes type=friend secret=Polycom context=internal host=dynamic canreinvite=no mailbox=1000@default callerid=TESTUSER1 <1000> * ----------------- [extensions.conf] exten => 1000,1,Macro(stdexten,1000@default,SIP/1000) ---------------- [rtp.conf] [general] rtpstart=12000 rtpend=12005 dtmftimeout=3000 What are we missing? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/732aebfd/attachment.htm

...tmfmode=rfc2833 nat=yes mailbox=200 at home disallow=all allow=ulaw I can make calls in my LAN but i can?t ear comunications with another client trought Internet. My Asterisk is in my LAN and i not have a DMZ. I search in the list and find something about "rtp" ==> rtp.conf. I found rtpstart and rtpend and forward those Ports on my firewall, but this don?t work for me. What?s wrong??? If you need some info please tell me. Thanks in advance! Emiliano Vazquez.

...rt: 5004 Destination port: 5004 17 0.055282 192.168.0.5 192.168.1.137 ICMP Destination unreachable (Port unreachable) 18 0.077638 192.168.1.137 2427 192.168.0.5 2727 MGCP 250 9 Connection was deleted 19 0.100808 192.168.1.137 2427 192.168.0.5 2727 MGCP 200 10 OK rtp.conf: [general] rtpstart=5004 rtpend=5005 mgcp.conf: [general] port = 2727 bindaddr = 0.0.0.0 [192.168.1.137] host = 192.168.1.137 context = home callerid = "ShoreTel" <4368> dtmfmode = inband accountcode = 1000 amaflags = billing callwaiting = no callretur...

Hi D'arcy Have you checked your RTP port ranges (I'm sure you have), and also that the server IP for RTP as specified in the initial SIP is correct? Not sure how this will relate to your setup, but we had something similar here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP service provider in the middle. We had slightly different parameters, e. g. that we would

Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording.

...ulaw ;allow=alaw context=default connection=asterisk jabber.conf [general] debug=yes autoprune=yes autoregister=yes [asterisk] type=client serverhost=talk.google.com username=tharanga12345 at gmail.com secret=xxxxxx port=5222 usetls=yes usesasl=yes timeout=1000 rtp.conf iam using lower ports... rtpstart=1650 rtpend=4560 sip.conf [general] context=default bindport=5060 bindaddr=0.0.0.0 ;srvlookup=yes dtmfmode=rfc2833 relaxdtmf=no disallow=all allow=ulaw ;allow=alaw ;allow=gsm maxexpirey=30 defaultexpirey=180 canreinvite=yes ;nat=no UserAgent=Asterisk [312] type=friend context=default regexten=...

...allow=alaw ; only alaw works with sip1... nat=no canreinvite=no qualify=yes insecure=port,invite username=imagi-justvoip fromuser=00491785450880 secret=xxxxxxxxxxxx registerattempts=0 ; keep trying to register (normally times out after 10 attempts) context=from-external from rtp.conf rtpstart=19000 rtpend=20000 -- Simon Tennant _____________________________________________ fixed: .uk +44 20 7043 6756 .de +49 89 420 955 854 mob: .uk +44 78 5335 6047 .de +49 17 8545 0880 xmpp: simon at buddycloud.com

I see in sip debug it says Audio is at port 10342 Express Talk suggests Audio at UDP 8000-8020 I've tried changing Express Talk to 10000 and forwarded 10000-10400. Is there a possibility Express Talk won't work in the 10000 range? Is it possible to limit Asterisk to 8000-8020? Thank you, Gary

To Jonas: I have an asterisk box at home and I have this line in my rtp.conf file: rtpstart=10000 rtpend=10100 And My FW is setup to forward all incoming ports of range 10000-10100 to the asterisk PC. I've never had a problem since one year, but I have never received more than two simultaneous calls with SIP clients. Message: 5 Date: Fri, 13 Sep 2013 11:49:59 +0200 From: Jonas...

...[IPV6](!,my-codecs) dtmfmode=rfc2833 context=sip-out type=friend host=dynamic transport=tls encryption=yes nat=no qualify=yes the phone it's self contains [200](IPV6) context=abc callerid=123 defaultuser=123 fromuser=123 secret=secret mailbox=123 at default The rtp ports are defined via rtpstart=15000 rtpend=20000 and the Firewall is open at TCP 5061 and udp 15000:20000 what did i miss in my configuration? Best Regards Jakob -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 473 bytes Desc: OpenPGP...

hi all ! we've got IAX2 protocol working between several Asterisk servers. Now we are concerned with doing bandwidth management to maintain an acceptable voice quality. We thought of prioritizing the udp traffic. ( Giving a high priority to those IAX2 udp ports.) I know that IAX2 uses udp/4569. Is there any other traffic/ports that we need to consider for bandwidth shaping w.r.t IAX2.

...en => 6011,1,Dial(SIP/6011) exten => 6011,n,Hangup exten => 6012,1,Dial(SIP/6012) exten => 6012,n,Hangup include => outgoing [outgoing] ;Route calls starting with 0 to VOSP exten => _0.,1,Dial(SIP/vosp/${EXTEN}) exten => _0.,n,Hangup ;====================== rtp.conf [general] rtpstart=10000 ;1 even port for (symetric) RTP + 1 odd port for RTCP ;for a total of 10 concurrent conversations rtpend=10020

...=> s,n,Dial(SIP/1000,20,r) exten => s,n,Hangup() ;Outgoing exten => 100,1,JabberStatus(google,YYYY at gmail.com,STATUS) exten => 100,n,NoOp(Jabber Status=${STATUS}) exten => 100,n,Dial(Gtalk/google/invoxgtalk at gmail.com/Talk) exten => 100,n,Hangup() # /etc/asterisk/rtp.conf rtpstart=1650 rtpend=4560 ports opened on the router tcp 443 -incoming, outgoing tcp 5222-incoming,outgoing udp- all open incoming, outgoing -> i am able to call from my external gtalk client to the server configured user # this case is working fine Executing [s at google-in:1] NoOp("Gta...