search for: externip

#!/bin/bash # checksetexternip.sh # Author: John Cahill email at johncahill.net # Licence: GPL v3 # Description: script that queries checkip.dyndns.com to find the server's external IP address. Updates asterisk's externip value and does a sip reload if necessary. # Last modified 06/02/2012 is_ip(){ input=$1 oct...

Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=1...

Does anyone know if the 'externip=' in sip.conf is resolved just once at startup or on an on going basis? I would like to use a DNS name through one of the dynamic DNS providers, but if the DNS updates, and asterisk continues using the old resolved value, this could get tricky. Thanks, Dennis

Hi, We have some Asterisk servers that we are moving behind a NAT to preserve public addresses and make room for growth. This is Asterisk 1.4 NAT works very good with the externip/localnet-setting when we are connected directly to our teleco. But when I try to use NAT and put them behind our Kamailio something interesting happens: The media-address in the SDP is the internal ip and not the external. This is the setup: Teleco <-> Kamailio <-> Asterisk SIP...

I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[password]@voiptalk.org/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP Debugging Enabled Jan 26 18:20:04 NOTICE[9226]: chan_sip.c:...

...standard BT home DSL, which means I cannot have a static IP address, therefore i'm forced to use NAT, I subscribe to a DDNS service and have written a VB app which polls the router every 10 seconds and updates the DDNS if appropriate. This is fine but I need to be able to modify my sip.conf (externip = w.x.y.z) and reload sip, does anyone know of a script/app which does an nslookup and modifies the conf file, then reloads sip? Regards, Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050824/7ffe...

I would like to access my shell environment variable MYIP from within sip.conf to put in externip. I've tried some variations of syntax after reading "The Future of Telephony" but it's not working yet. Should it be externip=${ENV{$MYIP}} or some other syntax?? Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux

Will this do the intended thing? This is in sip_nat.conf which is included in sip.conf: externip=192.168.0.200 localnet=192.168.0.200/255.255.255.0 externip=64.168.237.110 localnet=192.168.1.2/255.255.255.0 I have Asterisk running on a box with two Ethernet interfaces and bound to both. One interface, 192.168.1.2 services clients outside the firewall who are led to believe that Asterisk is r...

Hi all, I've tried search this problem on the list... no luck... The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the private 192 address to the outside...) when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending the packets t...

Can i use a domain name instead of an IP address for externip (sip.conf) Because im using dynamic dns. Not sure what i'm trying to achieve as yet but, i want to know if it is possible?

Hi, What is the equal option for externip in asterisk 13 with pjsip. I have tried external_media_address=XX.XX.XX.XX external_signaling_address=XX.XX.XX.XX but asterisk 13 writes local ip to the from header. any suggestions? Best Regards, Madushan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http:/...

ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: <sip:test at ekiga.net>;tag=as64618445 To: <sip:test at e...

We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. ?To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b66661d72f1328c918@%EXTERNIP% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b66661d72f1328c918@%INTERNIP%. ?I can not figure...

...sible to specify more than 1 localnet? I know this is an odd question. I have a customer that has multiple sites linked by VPN. Main range is 192.168.33.0/24 and a remote site is 10.1.1.0/24 We want to allow some access to the public IP address at the main site. For this to work I need to use the externip and localnet directive. If I do this it rewrites the SDP with the external IP address of the main site on dialog with the VPN'd sites. This means that I can either have the VPN endpoints working or have people accessing from outside.. Any work arounds? Thanks Brian -------------- next part -...

Hi, Is there some parameter that I should pay attention to when using externip parameter on sip.conf? I ask this because after using sipsak I've noticed maybe the reason that i don't get voice in one direction could be because at the SDP/SIP messages are references to the 192.168.1.50 (server internal ip) and not the ip from the provider. Is there a way to alter SDP...

I have an Asterisk 1.4.19.1 server that is behind a Fortinet firewall. Localnet is 192.168.2.0/255.255.255.0 and all external sip devices look as if they are on the same local network because the Fortinet rewrites the incoming IP as its own address. The problem I have is that when I set "externip=148.XXX.XXX.XXX" it is being ignored and I can see SDP packets that have the internal network number of the server (192.168.2.24). Since the SPA3102 has symmetric RTP it simply ignores the internal IP address and sends the rtp back to the same address it received it from they work fine. The...

...---->ISP internet Connection <-----Firewall3 | remote office | ----Remote User Phone 192.168.97.74/24 There is a Lan2Lan VPN tunnel between the Firewall2 and the Remote Office Firewall3 I can Ping the remote office phone from the asterisk PBX at all times. Now I have my Sip.conf setup with externip= X.Y.Z.250 [general] port = 5060 bindaddr = 0.0.0.0 context = default allowoverlap=no srvlookup = yes : externip = externip = x.y.z.250 localnet=10.202.17.0/255.255.255.0 qualify=yes nat=yes register = xxxxxxx:SipServer/xxxxxxxx limitonpeers=yes allowsubscribe=yes notifyringing=yes notifyhold=yes u...

On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad <faheem2084 at gmail.com> wrote: > > > On Wednesday, 14 September 2016, Madushan Geethanga < > mgliyanage.rc at gmail.com> wrote: > >> Hi, >> >> What is the equal option for externip in asterisk 13 with pjsip. I have >> tried >> >> external_media_address=XX.XX.XX.XX >> external_signaling_address=XX.XX.XX.XX >> >> but asterisk 13 writes local ip to the from header. any suggestions? >> > Setting 'from_domain' on the endpoint wi...

...s long as the phone is ringing, everything is fine. But after the pickup, asterisk sends a SIP/SDP package with its private address (192.168.100.10). After the softphone received this package, it tries to send RTP data to this address! Obviously those packages never reach asterisk... Does 'externip' just works for SIP and not for RTP? Where does the the internal IP-address come from and how can I set the right one? My configuration: [general] externip = 85.XXX.XXX.XXX nat = yes localnet = 192.168.100.0/24 [42] deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=XXX qualify=yes port=...

Hi everybody, I finally want to get rid of 1-way audio problem. Please help me here. I have 3 scenarios. 1. Audio is always one way. Caller who dialed can't listen the called party but called party can listen him. In this scenatio Asterisk is on dynamic IP with dyndns FQDN. sip.conf has externip = abc.dyndns.org and localnet = xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. Where is the voice getting lost from the called party? NAT is there but Asterisk is in DMZ. 2. Conversation is going fine when all of a sudden you realize that other parth has started saying 'hello, hello'...