search for: sip_nat

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 fir...

...ntact: <sip:99084611234 at Z.Z.247.106>^M Content-Type: application/sdp^M Content-Length: 315^M ^M v=0^M o=root 1021147583 1021147583 IN IP4 Z.Z.247.106^M s=Asterisk PBX 1.6.0.17^M c=IN IP4 Z.Z.247.106^M t=0 0^M m=audio 18702 RTP/AVP 0 8 3 101^M I have the following in the sip_nat.conf localnet=Y.Y.47.149/255.255.0.0 externhost=Z.Z.247.106 externrefresh=10 fromdomain=att.com nat=yes qualify=yes canreinvite=no I think the SDP should have give the Y.Y.47.149 IP on the local net side to the phone. But I am unable to figure how make it do that. The Asterisk log...

how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring. Edit sip_nat.conf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to /etc/asterisk/sip_custom.conf allow=gsm all...

...addr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ; allow=h263p ; H.263p is the enhanced video codec context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf And I left only H.263 basic in codec's configuration in Video Phone. No chance to get the communication in H.263 protocol. I saw that to use H.263+ protocol I need Asterisk CVS. I am not using asterisk CVS I am using asterisk 1.0.9 (l...

Hello all, For some reason I am not showing registration in SIP. Can anyone give me an idea what can cause this? asterisk1*CLI> sip show registry Host Username Refresh State -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/bd3a7577/attachment.htm

...solve translation problems. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_additional.conf register => xxxxxxxxxx@sip.broadvoice.com:pppppppppp:xxxxxxxxxx@sip.broadvoice.com/2197 [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=xxxxxxxxxx secret=pppppppppp username=xxxxxxxxxx insecure=very cont...

...ng module 'res_phoneprov' (HTTP Phone Provisioning) == Parsing '/etc/asterisk/sip.conf': == Found == Parsing '/etc/asterisk/sip_general_additional.conf': == Found == Parsing '/etc/asterisk/sip_general_custom.conf': == Found == Parsing '/etc/asterisk/sip_nat.conf': == Found == Parsing '/etc/asterisk/sip_registrations_custom.conf': == Found == Parsing '/etc/asterisk/sip_registrations.conf': == Found == Parsing '/etc/asterisk/sip_custom.conf': == Found == Parsing '/etc/asterisk/sip_additional.conf': =...

I have a little confusion about the general settings (other than the register values) in the SIP General area. I understand that for examle in a SIP context like [FWD] or [BROADVOICE] the entries in those areas are ths settings that take effect in any communication woth FWD and/or BROADVOICE. However, I'm confused as to the purpose of the "general" settings -- to what or which

...out of the via headers. Here's my scenario: UA -> Snom NATf -> Snom 4S Proxy -> Asterisk Echo Test Function NATf, the proxy, and Asterisk are all on public IPs. So my question is: In chan_sip.c, copy_via_headers function, I see an if statement testing for "(ast_test_flag(p, SIP_NAT) == SIP_NAT_ALWAYS)" What in sip.conf do I do to toggle/change SIP_NAT to try to match this if statement? Following is my sip.conf for the proxy. Note I've tried nat=yes, nat=no, nat=always but the darn thing always takes the "else" instead of matching the if. #sip.conf ;...

...al] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) externip = 62.219.XXX.XXX disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown nat = yes #include sip_nat.conf #include sip_additional.conf [root@crystalclear root]# cat /etc/asterisk/sip_additional.conf register=TollIPdemo1:somesecret@sipdevice.FQDN.net [sip-devices] username=TollIPdemo1 type=friend secret=somesecret host=sipdevice.FQDN.net fromuser=TollIPdemo1 context=from-pstn canreinvite=no call...

...port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 202.69.190.244 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_additional.conf If there are some parameters that i hould define please let me know... if this work all the configuration will be posted as reference for others... THank you all. -------------- next part -------------- An HTML attachment was scrubbed......

...or whatever your localnet mask) nat=yes port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_additional.conf register={userid}:{password}@{domain}.net [200] username=200 type=friend secret=??? qualify=no port=5060 pickupgroup= nat=never mailbox= host=dynamic dtmfmode=rfc2833 disallow= context=from-internal canreinvite=no callgroup= callerid="D...

...blems. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=alaw allow=g729 allow=g723 context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown language=es #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf ---------------------------------------------------------------------------- --- SIP_ADDITIONAL.CONF ---------------------------------------------------------------------------- --- [as5300] type=peer qualify=yes host=xxx.xxx.xxx.xxx (AS53...

hi, making may way through all this...internal sip registration works,(cant call yet but anyhow)... the asterisk box is obvisoulsy behind a router. im not 100% sure if i should go with port forwarding or NAT and if a or b, what additional setup is actually correct? sip_nat.conf # this is when i got the NAT -route, right? #gets all the dyndns-stuff #externip = home.mydomain.com (Enter your DynamicDNS domain name. Obviously it's just easier to get a static IP address and avoid using DynamicDNS altogether.) externhost = home.mydomain.com externrefresh = 5 (which mea...

Has anyone seen something like this before. Randomly, on longish calls, the local side of the call audio goes dead. Meaning remote caller can hear us but we cannot hear the remote person? Linux voip 2.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:10:25 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. WANPIPE Release: 3.4.1 Wanpipe Config:

...en dialing *97 for voicemail wont' give me any audio. Picture posted here shows my DD-WRT NAT setting: *http://tinypic.com/r/21cuqlu/5* Any input will be much appreciated. This is running latest PBXinaFLASH (which has FreePBX) and I tried using externip=x.x.x.x/255.255.248.0 in /etc/asterisk/sip_nat.conf but it was of no use. Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100323/fdbdeeb7/attachment.htm

...; solve translation problems. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf [1000] username=1000 secret=abc123 context=mytest host=dynamic ----------------- /etc/asterisk/extensions.conf : [general] static=yes writeprotect=yes ;Suport phones ;SUPPORTPHONES=SIP/2205&SIP/2206&SIP/2207&SIP/2208&SIP/...

Has anyone out there successfully set up their * box to terminate their VONAGE calls? I (and I am sure lots of others) would love to hear how you did it. I'd like to be able to get rid of the extra hardware I have hanging around here and use the ASTERISK machine to handle the SIP termination instead of needing to have a Linksys modem (w/phone) and an additional X100P card. Thanks.

I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting "Explicit MWI Subscription" to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Thanks! -- Warm Regards, Lee

Hi to everyone! I have succerfully instaled my new Asterisk 1.4 on my debian etch. I have my users in sip.conf like this: [200] type=peer host=dynamic context=home secret=200 callerid= 200 dtmfmode=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