I have an Asterisk with multiple IP's, on the same subnet. When a call comes in, I need to send it back out via SIP, but need that only one IP is used as originating IP for all calls. For example machines has 192.168.50.3 192.168.50.4 192.168.50.5 .... but when I originate the second leg of a call, the IP address that is supposed to be read as source IP must be 192.168.50.5, regardless of how the call arrived. How do I do that? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100530/2cd1be14/attachment.htm
ayodele abejide
2010-May-30 15:18 UTC
[asterisk-users] Implementing Asterisk on a WAN or an Intranet
Hi everyone, I want to Implement asterisk on my company's intranet comprising of 100 local area networks I don't know how to configure my dialplan and my sip.conf files Thanks Date: Sun, 30 May 2010 10:06:12 -0400 From: venefax at gmail.com To: asterisk-users at lists.digium.com Subject: [asterisk-users] How to use one single IP as origination I have an Asterisk with multiple IP's, on the same subnet. When a call comes in, I need to send it back out via SIP, but need that only one IP is used as originating IP for all calls. For example machines has 192.168.50.3 192.168.50.4 192.168.50.5 .... but when I originate the second leg of a call, the IP address that is supposed to be read as source IP must be 192.168.50.5, regardless of how the call arrived. How do I do that? _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100530/467f77dd/attachment.htm
Peter den Hartog
2010-May-31 08:18 UTC
[asterisk-users] How to use one single IP as origination
Try it with the from= in sip.conf You can give a from IP there. On Sun, May 30, 2010 at 4:06 PM, CDR <venefax at gmail.com> wrote:> I have an Asterisk with multiple IP's, on the same subnet. When a call > comes in, I need to send it back out via SIP, but need that only one IP is > used as originating IP for all calls. > For example > machines has > 192.168.50.3 > 192.168.50.4 > 192.168.50.5 > .... > but when I originate the second leg of a call, the IP address that is > supposed to be read as source IP must be 192.168.50.5, regardless of how the > call arrived. > > How do I do that? > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Groet // Kind regards, Peter den Hartog -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100531/f35e4844/attachment.htm
See bindaddr here: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf That should do exactly what you want. Regards, Mike From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of CDR Sent: Sunday, May 30, 2010 10:06 To: asterisk-users at lists.digium.com Subject: [asterisk-users] How to use one single IP as origination I have an Asterisk with multiple IP's, on the same subnet. When a call comes in, I need to send it back out via SIP, but need that only one IP is used as originating IP for all calls. For example machines has 192.168.50.3 192.168.50.4 192.168.50.5 .... but when I originate the second leg of a call, the IP address that is supposed to be read as source IP must be 192.168.50.5, regardless of how the call arrived. How do I do that? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100531/d4d1fd57/attachment.htm
Zeeshan Zakaria
2010-Jul-27 00:58 UTC
[asterisk-users] Configuring X-lite for a remote user
Configuring x-lite is a smaller problem here, do you have on your router
your public IP ported to private IP at all and have you tested it before?
As for x-lite check it on my website at
http://visionvoip.com/help/x-lite.php
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-26 8:51 PM, "ayodele abejide" <ayodeleabejide at
hotmail.com> wrote:
I have asterisk running at home, a friend would be traveling out of the
country and I want him to be able to put a call through from his remote
location, I am wondering how I would configure the X-lite client on his pc
so he would be able to call through assuming my public address is A.B.C.D
and the static address the asterisk machine is on is 192.168.0.3.
Thanks in anticipation
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Adolphe Cher-aime
2010-Jul-27 01:14 UTC
[asterisk-users] Configuring X-lite for a remote user
To have your asterisk box reachable from internet you must configure static nat on your router to get sip traffic to the public Ip redirected to your internal ip. Make sure that sip and rtp traffic are not bloked by firewall. And configure xlite to connect to your public ip address. Adolphe Cher-aime From my Iphone On Jul 26, 2010, at 7:48 PM, ayodele abejide <ayodeleabejide at hotmail.com> wrote:> I have asterisk running at home, a friend would be traveling out of > the country and I want him to be able to put a call through from his > remote location, I am wondering how I would configure the X-lite > client on his pc so he would be able to call through assuming my > public address is A.B.C.D and the static address the asterisk > machine is on is 192.168.0.3. > > Thanks in anticipation > > Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Si > gn up now. > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100726/28d08ff4/attachment.htm
Kyle Kienapfel
2010-Jul-27 02:14 UTC
[asterisk-users] Configuring X-lite for a remote user
On Mon, Jul 26, 2010 at 6:14 PM, Adolphe Cher-aime <acheraime at gmail.com> wrote:> To have your asterisk box reachable from internet you must configure static nat on your router to get sip traffic to the public Ip redirected to your internal ip. Make sure that sip and rtp traffic are not bloked by firewall. > And configure xlite to connect to your public ip address. > > > Adolphe Cher-aime > From my Iphone > On Jul 26, 2010, at 7:48 PM, ayodele abejide <ayodeleabejide at hotmail.com> wrote: > > I have asterisk running at home, a friend? would be traveling out of the country and I want him to be able to put a call through from his remote location, I am wondering how I would configure the X-lite client on his pc so he would be able to call through assuming my public address is A.B.C.D and the static address the asterisk machine is on is 192.168.0.3. > > Thanks in anticipation > > ________________________________ > Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up now. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > ??????????????http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ??http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > ? ? ? ? ? ? ? http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >Dynamic hostnames are pretty useful too