search for: externaddr

...nf and netstat shows that Asterisk is listing also on IPv6. My Asterisk server is behind a IPv4 NAT and was working absolutely perfect. But after my bindaddr change I got a problem with external calls. I spend some time to investigate this issue and found out the outbound calls are working. The externaddr is used in the SIP INVITE. If I received a inbound call the externaddr isn't used any more in SDP part of the answer from the Asterisk. The result is one way audio. In addition I saw the following message in the Asterisk log: [May 25 19:18:18] WARNING[3674] chan_sip.c: Address remapping...

...time, it is random. The audio is flowing normally and the call is dropped. Has anyone ever experienced this? My settings changed below: allowoverlap = no udpbindaddr = 0.0.0.0 tcpenable = no tcpbindaddr = 0.0.0.0 transport = udp, ws, wss srvlookup = yes directmedia = no rtcachefriends = yes externaddr = my ip address externhost = my domain address ;   foo.dyndns.net; refreshed periodically externrefresh = 180       localnet = 172.31.40.21 / 255.255.240.0; AWS NETWORK       localnet = 192.168.0.0 / 255.255.0.0; RFC 1918 addresses       localnet = 10.0.0.0 / 255.0.0.0; Also RFC1918       loc...

...veals that I'm trying to do this using the private IP of my Asterisk box (Nat, again, is on the firewall at data center), and I get TCP retransmissions. so the fact it isn't working makes sense, because my local box doesn't know how to get to a private IP address. I've tried using externaddr in sip.conf to no avail. Is there some command I'm missing? Obviously if I put an interface with a public IP on the outside I'd bet that would resolve this problem, but sort of like having that guy behind a hardware firewall :) I'm to the point of telling them to fire up a VPN on be...

...en=${EXTEN:12:3} exten => _j.,n,NoOp("Callerid is " ${CALLERID(all)} ) exten => _j.,n,GoTo(from-outside,${3digitexten},1) [from-outside] exten => 123,1,NoOp() exten => 123,n,Answer() exten => 123,n,Dial(SIP/jnctn/1212xxxyyyy) exten => 123,n,HangUp() sip.conf: [general] externaddr=xx.yyy.zz.aa nat=yes directmedia=no ; tried nonat sip show peer jnctn: Insecure : invite Force rport : Yes ......... DirectMedia : No sip show peer teliax: Insecure : port,invite Force rport : Yes ........ DirectMedia : No And the cli doesn't show any p...

I have a multi-homed machine (quite a few IP addresses on one of the interfaces) For SIP I found that using externaddr in sip.conf would make it much more reliable with ICE and RTP using the correct IP Is there an equivalent setting for XMPP / motif.conf? I saw bindaddr in gtalk.conf but it doesn't appear to be mentioned in the source code for chan_motif

...; My settings changed below: > > allowoverlap = no > udpbindaddr = 0.0.0.0 > tcpenable = no > tcpbindaddr = 0.0.0.0 > > transport = udp, ws, wss > > srvlookup = yes > > directmedia = no > > rtcachefriends = yes > > externaddr = my ip address > > externhost = my domain address ; foo.dyndns.net > <http://foo.dyndns.net>; refreshed periodically > externrefresh = 180 > >        localnet = 172.31.40.21 / 255.255.240.0; AWS NETWORK >        localnet = 192.168.0.0 / 255.255.0.0; R...

...PBX 11.4.0 Date: Fri, 02 Aug 2013 00:27:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 I really cannot understand what is wrong, I have checked my sip.conf configuration and it is the same as in past versions. externaddr and localnet are set to the proper values. Any ideas? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001

Dear list, I'm trying to get Asterisk to work dual-stack on Linux and I'm left with a question. Imagine that a user (on the road) connects to Asterisk from various places. Many of them probably don't have IPv6 support yet. However, his house and office do have IPv6 connectivity. I would like to make sure that whenever IPv6 is available, the connection will be made over IPv6, but

...one configuration on the remote server for the site is [general] session-timers=refuse disallow=all allow=g729:20 allow=ulaw allow=alaw fromuser=xxx useragent=xxx callcounter=yes alwaysauthreject=yes allowguest=no jbnable=yes jbforce=no jbimpl=adaptive jblog=no jbmaxsize=200 jbresyncthreshold=1000 externaddr=xx.xx.xx.xx localnet=xx.xx.xx.xx/255.255.255.0 [xxxx] type=peer user=xxxx secret=xxxx host=dynamic disallow=all allow=g729 allow=ulaw allow=alaw dtmfmode=rfc2833 context=xxxx call-limit=1 limitonpeers=yes callgroup=1 pickupgroup=1 deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.xxx/255.255.255.255 nat=for...

...s anyone ever experienced this? > > My settings changed below: > > allowoverlap = no > udpbindaddr = 0.0.0.0 > tcpenable = no > tcpbindaddr = 0.0.0.0 > > transport = udp, ws, wss > > srvlookup = yes > > directmedia = no > > rtcachefriends = yes > > externaddr = my ip address > > externhost = my domain address ; foo.dyndns.net; refreshed periodically > externrefresh = 180 > > localnet = 172.31.40.21 / 255.255.240.0; AWS NETWORK > localnet = 192.168.0.0 / 255.255.0.0; RFC 1918 addresses > localnet = 10.0.0.0 / 2...

...ase of an address change. This a lot less trivial, maybe unnecessary, and probably covered by the previous item. I would seriously consider this patch for addition to 1.4 and 1.2. The code is very little intrusive, and it would solve in a correct way the nat traversal problems for which externip/externaddr/stunaddr are only a partial and expensive workaround. __________________________________________________________________

...asterisk not using stun returned public ip address for any of the sip requests going out of my network. i have done settings as below res_stun_monitor.conf settings: [general] stunaddr = stun.ideasip.com stunrefresh = 30 stun show status Hostname Port Period Retries Status ExternAddr externport stun.ideasip.com 3478 30 3 OK 61.12.17.171 39710 sip.conf localnet=192.168.0.0/255.255.255.0 register=>jai9999:123456:jai9999 at sip2sip.info/jai9999 when above command runs , it is sending register method with my private ip address. REGISTER sip:sip2s...

Hi I have installed Asterisk 11.13.1 on Fedora running in VirtualBox. The VB network interface is configured to use NAT. The host machine is Windows 7 and is connected to a SIP server using a VPN connection. I have configured ?externaddr?, ?localnet? and ?nat=force_rport,comedia?. Asterisk registration is successful, I see in Wireshark the packets send between Asterisk and SIP server. However, when I try to call using a softphone, installed directly on the Windows host, I see no incoming packets from the SIP server to Asterisk, so...

...d below: >> >> allowoverlap = no >> udpbindaddr = 0.0.0.0 >> tcpenable = no >> tcpbindaddr = 0.0.0.0 >> >> transport = udp, ws, wss >> >> srvlookup = yes >> >> directmedia = no >> >> rtcachefriends = yes >> >> externaddr = my ip address >> >> externhost = my domain address ; foo.dyndns.net; refreshed periodically >> externrefresh = 180 >> >> localnet = 172.31.40.21 / 255.255.240.0; AWS NETWORK >> localnet = 192.168.0.0 / 255.255.0.0; RFC 1918 addresses >>...

...8 at 83.186.238.111> (Checking From) --From tag 7a82b127 --To-tag [Feb 12 15:13:59] DEBUG[25824] acl.c: For destination '83.186.238.111', our source address is '172.26.19.13'. [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Target address 83.186.238.111:5060 is not local, substituting externaddr [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 212.105.99.108:5060 [Feb 12 15:13:59] DEBUG[25824] chan_sip.c: Allocating new SIP dialog for 857d4ed8 at 83.186.238.111<mailto:857d4ed8 at 83.186.238.111> - MESSAGE (No RTP) [Feb 12 15:13:59] VERBOSE[25824] chan...

CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten

...eo: 6 802.1p CoS RTP text: 5 Jitterbuffer enabled: Yes Jitterbuffer forced: No Jitterbuffer max size: 200 Jitterbuffer resync: 1200 Jitterbuffer impl: fixed Jitterbuffer log: No Network Settings: --------------------------- SIP address remapping: Enabled using externhost Externhost: pbx.domain.com externaddr: 11.22.33.44:0 Externrefresh: 10 Localnet: 192.168.101.0/255.255.255.0 Global Signalling Settings: --------------------------- Codecs: 0x60e (gsm|ulaw|alaw|speex|ilbc) Codec Order: ulaw:20,alaw:20,gsm:20,speex:20,ilbc:30 Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: No Compact SIP headers...