search for: localmask

Asterisk stopped answering the calls. I'm just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0<:1000@10.0.0.101> my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid="PSTN GW" <3000> deny=0.0.0.0 permit=10.0.0.154 ;SPA-3000 IP address dtmfmode=rfc2833 canreinvite=no cantransfer=yes My extension.conf [globals] PSTN_GW=10.0.0....

Lance, Have you configured your sip.conf to use these aprameters under General? ;externip=66.213.227.66 ;localnet=192.168.1.0 ;localmask=255.255.255.0 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lance Grover Sent: Thursday, June 02, 2005 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] asterisk on...

...ajor changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600...

...reply to critical packet" How is Asterisk supposed to be configured? Currently this: externip = 94.18.x.x ; Address that we're going to put in outbound SIP messages ; if we're behind a NAT localnet = 192.168.5.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask ; The externip, localnet and localmask is used ; when registering and communication with other proxies ; that we're registered with tcpbindaddr=0.0.0.0...

...tensions ;srvlookup = yes ;pedantic = yes ;tos=lowdelay ;maxexpirey=3600 ;defaultexpirey=120 ;notifymimetype=text/plain ;videosupport=yes externip = xxx.xxx.xxx.xxx localnet = 192.168.1.0 localmask = 255.255.255.0 [5001] qualify=yes type=friend context=sip-extensions username=5001 callerid=5001 <5001> host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 mailbox=5001 [5002] qualify=yes context=sip-extensions type=friend username=5002 callerid=5002 <5002> host=dynamic nat=ye...

...ve the external IP on the asterisk server, which I have done I have fowared 5060UDP, 8000UDP, and 35000 to 37000 UDP to the internal IP (192.168.1.115) I have put 35000 and 37000 into the rtp.conf as the start/end ports extracts of sip.conf: externip = 60.234.129.154 localnet = 192.168.1.115 localmask = 255.255.255.0 [88] type=friend secret=********** dtmfmode=rfc2833 nat=yes host=dynamic canreinvite=no Trying with xlite at the other end Registered ok, can dial both ways, just no audio at all. In the log of xlite (cant see it at the moment as im not vnc'd in at the moment) it showed t...

...at=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP address? If Asterisk is behind the NAT, sip.conf must have in [globals] externip = <External IP address> localnet = < Internal NETWORK address> localmask = <mask of localnet> Is that right? If someone have clear documents explaining configuration of Asterisk and many softphone with differents cases of deployement with NAT, these would be placed on the voip-info.org wiki....

...ex.php/Asterisk_Setup . This has not worked. Right now, "sip show registry" shows this: Host Username Refresh State sip.broadvoice.com:5060 6033751414@s 120 Auth. Sent I added this to my sip.conf: externip=xxx.xxx.xxx.xxx localnet=10.1.0.0 localmask=255.255.0.0 nat=yes My /etc/asterisk/sip_additional.conf contains this relevant portion: register=> nnnnnnnnnn@sip.broadvoice.com:pppppppppp:nnnnnnnnnn@sip.broadvoice.com/2 00 [from-broadvoice] username=nnnnnnnnnn user=nnnnnnnnnn type=user secret=pppppppppp nat=yes insecure=very host=sip.broad...

...ut either I get 1/2 second of audio or no audio. No matter how long I wait there is just no audio or just a short snippet of audio at the beginning. Here is parts of my sip.conf; [general] port = 5060 ; Port to bind to localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask externip = 206.40.161.235 context = intern ; Default for incoming calls maxexpirey=3600 defaultexpirey=300 disallow=all ; Disallow all codecsa allow=gsm allow=alaw allow=ulaw tos=reliability register => xxx:xxx@sipgate.de/...

...; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video externip = xxxxxxxxxxxxxxxxx ; Address that we're going to put in ; if we're behind a NAT localnet = 192.168.0.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask disallow=all allow=ulaw allow=alaw allow=gsm [4001] type=friend secret=4001 host=dynamic defaultip=192.168.0.201 mailbox=4001@default context=default [4002] type=friend userid=4002 secret=4002 host=dynamic defaultip=192.168.0.202 mailbox=4002@default cont...

...Newbie) ;-------------Testing------------------ [general] port = 5060 bindaddr = 0.0.0.0 allow=ulaw ; dtmfmode=info ; nat=yes ; This section is because i'm behind nat externip = x.x.x.x ;Outside address localnet = 10.73.73.133 ;Inside address localmask = 255.255.255.0 ;Inside subnet context = sip ; Default context for incoming calls register => ##########:secret@sip.stanaphone.com/1000 register => ##########:secret@sip.provider.net/4078 register => ##########:secret@sip.provider.net/4077 [stanaph...

My Asterisk box also does NAT for internal network, and establishes site-to-site VPN tunnel(s). As a result I have several internal interfaces with private addresses on them, and only one public interface. By trial-and-error I've found out that FWD (SIP) won't work unless I disable my VPN tunnels - it would send the internal IP address to FWD's SIP server instead of public one. I

Hi, just to repeat my previous post (and trying to find a solution): Setup is * behind NAT. I can use FWD (time service, echo server) without problems when I add this to sip.conf: externip=a.b.c.d ; a.b.c.d is the IP of the router (Linux/Nat) outside_addr=a.b.c.d My ICH however now responds with: -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from

...if anybody can help: ----------------------------------------- [general] port = 5060?????????????????????; Port to bind to bindaddr = 190.100.200.1????????????????; Address to bind to ;bindaddr = 0.0.0.0 externip = <extern ip> nat=1 localnet = 190.100.200.0 ? ? ? ? ; Internal NETWORK address localmask = 255.255.255.0 ? ? ?; Internal netmask context = default???????????????; Default for incoming calls srvlookup = yes?????????; Enable SRV lookups on outbound calls disallow=all???? allow=gsm???????????????; Disallow all codecs allow=ulaw allow=alaw??????????????????????; Allow codecs in order of pr...

...derful solution for Asterisk (at least for my use). Below is a sanitized snippet from my working sip.conf; your mileage may vary: [general] dtmfmode=inband port=5060 bindaddr=<enter yours> context=incoming disallow=all allow=ulaw externip = <enter yours> localnet = <enter yours> localmask = 255.255.255.0 nat=yes register => .. ... register => 16125551212:password@sphone.vopr.vonage.net:5061/99612 [sip99612] secret=password username=16125551212 host=sphone.vopr.vonage.net port=5061 type=peer nat=yes canreinvite=no dtmfmode=rfc2833 fromuser=16125551212 context=incoming fromdom...

...sisge echosquelch=1 isdnmode=ptp devices=2 msn=492 incomingmsn=* controller=1 softdtmf=0 context=sisge echosquelch=1 isdnmode=ptp devices=2 SIP.CONF [general] port = 5060 bindaddr = 0.0.0.0 context = sisge tos = lowdelay disallow = all allow = ulaw allow = alaw allow = gsm localnet = 192.168.1.0 localmask = 255.255.255.0 language = it canreinvite= no [492] context=sisge username=492 type=friend secret=492 host=dynamic qualify=yes callerid=492 dtmfmode=rfc2833 EXTENSIONS.CONF [general] static=yes writeprotect=no TRUNK=CAPI [globals] [sisge] exten => 492,1,Dial(SIP/492,60,tr) exten => 492,2...

...t; 10005. > > > As for your [general] section, I would maybe try something like this: > > > [general] > > port = 5060 ; Port to bind to > > context=sip-extensions > > externip = xxx.xxx.xxx.xxx > > localnet = 192.168.1.0/24 > > localmask is no longer used. Not sure if there is backwords > compatibility (and I'm doing this from memory, I don't have Asterisk > behind NAT anymore) > > I don't know, maybe it will work, maybe it won't :) > > HTH, > Leif Madsen > http://www.asteriskdocs.org >...

...general] section, I would maybe try something like this: > > > > > [general] > > > port = 5060 ; Port to bind to > > > context=sip-extensions > > > externip = xxx.xxx.xxx.xxx > > > localnet = 192.168.1.0/24 > > > > localmask is no longer used. Not sure if there is backwords > > compatibility (and I'm doing this from memory, I don't have Asterisk > > behind NAT anymore) > > > > I don't know, maybe it will work, maybe it won't :) > > > > HTH, > > Leif Madsen &g...

...ecause of the breakage between FWD and Vonage that I saw mentioned on this list. But going through FWD seems like a hack. I'd like to contact them directly using SIP. Obviously this is difficult because of the NAT firewall. I'm running asterisk 1.0.2. In my sip.conf I've got localnet, localmask, and externip defined. If I turn on sip debug, it looks like the packets are getting rewritten correctly. My entry for vonage looks like this: [vonage] type=peer host=sip.vonage.net context=default canreinvite=no dtmfmode=rfc2833 insecure=very I tried telling my firewall to port forward all 5060...

I found this configuration file on Wiki for FWD behind firewall ; SIP Configuration for Asterisk ; [general] disallow=all allow=ulaw port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind SIP channel to externip=xxx.xxx.xxx.xxx localnet=172.16.1.0 localmask=255.255.255.0 context=inbound-sip ; Default context for incoming calls maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register => FWD#:secret@fwd.pulver.com/FWD# bindaddr=0.0.0.0 - is this the address of my asterisk server externip=xxx.xxx.xxx.xxx - this is my external IP b...