similar to: SIP behind NAT - use of "externip" option

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

Trying now to set up the final part of my * switch. I must admit I've had great fun over the last week or so playing with it, and would like to thank you guys for all the assistance (past and present, since I've been trawling a lot of old posts!!!). Scenario - using voiptalk.org to supply the incoming gateway, tied to an 0845 number for convenience in testing. Internal 7960 -> 7960

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 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=192.168.200.0/255.255.255.0 Do

#!/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=$1 octet1=$(echo $input | cut -d "." -f1) octet2=$(echo $input

I have several of incoming numbers on IAX from voiptalk and magrathea but have a problem with IAX.conf. If I follow the example from voiptalk [VoIPTalk Incoming Number] type=friend username=VoIPTalk Incoming Number context=[XXXXXXXX] and make incoming entries in IAX.conf for the numbers like below with a different entry for each number pointing to a different context, incoming numbers always

Why is it that newcomers always feel like inserting 'Answer' is a necessary step in their extension.conf entries? >[voiptalk.org] >;forwards any calls starting with an "8" thru voiptalk.org >exten => _8.,1,Answer >exten => _8.,3,SetCIDNum(55555555) >exten => _8.,4,SetCIDName(My Name And Surname) >exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)

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

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

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, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm

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:

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

who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as:

Hi, We're deploying a small VoIP solution for a group of teleworkers. Naturally, this exposes us to all sorts of fun, most of which we seem to have working properly. However, some NAT issues are still bugging us and we have noticed that often these situations didn't exist when users were connected directly to our VoIP provider, voiptalk.org. They have something which they call a

Hi, I use VoIPTalk as my provider and unsure of a minor issue. When people call me they get a US ring tone instead of UK. Is this a Asterisk configuration issue or one for VoIPTalk ? I am running 1.6.2.0 and IAX2 trunks. Thanks - Phil

I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have