[Asterisk-Users] Connecting 2 * Together-Pulling hair out
Tim Pushor
timp at crossthread.com
Thu May 5 16:51:04 MST 2005
You don't need to have multiple connections defined. (I think you can
IAX trunk multiple conversations over ONE IAX connection (one data
stream) but you can run multiple streams without problem (if you had
concurrent multiple connections you may be able to gain some efficiency
using IAX trunking).
I am batching it this weekend and probably will be hacking away at my
setup, so if you want, give me a call at fwd #561293, ext 100 if you are
pulling your hair out ;-)
Tim
mr. barker wrote:
>Thank you to both Chris and Tim
>
>I could not get my head around this .. after seeing the examples it now
>makes sense what needs to be done. I will give both a whirl tonight.
>
>I do like the RSA key idea.
>
>One question is this, will I need multiple accounts on the Static IP
>machines so the Dynamic machine has the ability to make more then one
>concurrent SIP call through the Static IP machine ?
>
>If I could get the Static IP box to go through the my SMC router it would be
>great. I tried opening the ports. 5060udp/tcp, 10000-20000udp/tcp.
>Tried even setting the machine in the DMZ zone. I think the VOIP provider
>just has problems translating through the NAT or something.
>The linux box is running *@home no firewall setting that I know of.
>To much of a Newbie at linux .. lol and I have been at it for almost 1 year
>now and still have soooo much to learn.
>
>
>-----Original Message-----
>From: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chris
>Sent: Thursday, May 05, 2005 4:46 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
>
> I haven't gotten to keys yet.
>The documentation out there doesn't seem to be very good.
>
>Chris
>
>
>----- Original Message -----
>From: "Tim Pushor" <timp at crossthread.com>
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
><asterisk-users at lists.digium.com>
>Sent: Thursday, May 05, 2005 4:06 PM
>Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
>
>
>
>
>>Personally, if I owned both boxes and had full control of the dialplan
>>on both, I'd stay away from passwords. (but be careful what I say, I'm a
>>hack)
>>
>>I have a bunch of boxes connected together via IAX and authenticating
>>via RSA. The entries in iax.conf are simple, and dialing across the
>>connection is simple (no passwords in the dialplan) (thanks again Rich
>>for taking the time).
>>
>>Tim
>>
>>Here is a sample of iax.conf entries on machine a:
>>
>>[machineb]
>>type=user
>>host=machineb.internal.net
>>auth=rsa
>>inkeys=machineb
>>username=machineb
>>context=inbound
>>
>>[machineb]
>>type=peer
>>host=machineb.internal.net
>>auth=rsa
>>outkey=machinea
>>username=machinea
>>
>>And an example dialplan entry to dial an extention on machineb (in the
>>inbound context):
>>
>>exten => 333,1,Dial(IAX2/machineb/333)
>>
>>And on machinea, the opposite of machineb:
>>
>>[machinea]
>>type=user
>>host=machinea.internal.net
>>auth=rsa
>>inkeys=machinea
>>username=machinea
>>context=inbound
>>
>>[machinea]
>>type=peer
>>host=machinea.internal.net
>>auth=rsa
>>outkey=machineb
>>username=machineb
>>
>>To generate the keys:
>>
>>on machinea:
>>
>>astgenkey -n machinea
>>mv machinea.* /var/lib/asterisk/keys
>>
>>copy machinea.pub to machineb's /var/lib/asterisk/keys
>>
>>on machineb:
>>
>>astgenkey -n machineb
>>mv machineb.* /var/lib/asterisk/keys
>>
>>copy machineb.pub to machinea's /var/lib/asterisk/keys
>>
>>
>>Chris wrote:
>>
>>
>>
>>> I have something similar. Both of my servers are behind a firewall
>>>
>>>
>and NAT. You will need to allow UDP 4569 through the firewall for IAX2. If
>you have NAT you will need to redirect 4569 to the internal server.
>
>
>>> I would suggest using AMP and then looking at IAX_ADDITIONAL.CONF to
>>>
>>>
>see how it's done. You can modify the IAX.CONf because I don't believe AMP
>rewrites that file.
>
>
>>> I think the user and passwords are required. I would suggest using
>>>
>>>
>a strong password or someone may decide to make a few phone calls. After
>this you will need the routing in Extensions.conf to allow calls to be made
>on this trunk.
>
>
>>> Asterisk will handle the SIP > IAX. All my clients are SIP and
>>>
>>>
>they have no trouble going over a IAX trunk to other SIP devices on the
>other server.
>
>
>>>This is what my IAX_ADDITIONAL.CONF looks like
>>>
>>>SiteA - Dynamic IP
>>>--------------
>>>[boxb-peer]
>>>username=boxa-user
>>>type=peer
>>>trunk=yes
>>>secret=mypassword
>>>host=thehost.dyndns.org
>>>
>>>[boxb-user]
>>>type=user
>>>secret=mypassword2
>>>host=thehost.dyndns.org
>>>context=from-internal
>>>
>>>---------------
>>>Site b - Static IP
>>>----------------
>>>
>>>[boxa-peer]
>>>username=boxb-user
>>>type=peer
>>>trunk=yes
>>>secret=mypassword2
>>>host=xxx.xxx.xxx.xxx
>>>
>>>[boxa-user]
>>>type=user
>>>secret=mypassword
>>>host=xxx.xxx.xxx.xxx
>>>context=from-internal
>>>
>>>
>>>Regards,
>>>
>>>Chris
>>>
>>>
>>>----- Original Message -----
>>>From: "mr. barker" <cabalitomb at shaw.ca>
>>>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>>>
>>>
><asterisk-users at lists.digium.com>
>
>
>>>Sent: Thursday, May 05, 2005 1:58 PM
>>>Subject: RE: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
>>>
>>>
>>>
>>>
>>>
>>>
>>>>Yes trying to connect to boxes together.
>>>>
>>>>One sits outside the internal firewall and is on the inside.
>>>>
>>>>I am using AMP. However I can just put whatever I need in the
>>>>
>>>>
>custom.conf
>
>
>>>>sections.
>>>>The users agents are SIP .. can SIP call go over a IAX trunk ? if so
>>>>
>>>>
>great.
>
>
>>>>To create the trunk do I need to use a users name and password ? or ?
>>>>
>>>>I need to have the *box that is behind the firewall to be able to place
>>>>
>>>>
>a
>
>
>>>>call out through the *box that has a public ip.
>>>>
>>>>Thank you
>>>>
>>>>-----Original Message-----
>>>>From: asterisk-users-bounces at lists.digium.com
>>>>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Chris
>>>>Sent: Thursday, May 05, 2005 8:20 AM
>>>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>>Subject: Re: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
>>>>
>>>> I am not sure what you are trying to do. I have created an IAX2
>>>>
>>>>
>trunk
>
>
>>>>between the servers over an internet connection.
>>>>Then all you have to do is put in call routing on the trunks to forward
>>>>
>>>>
>the
>
>
>>>>call to the right place. Are you using AMP or trying to do it manually.
>>>>I found everything a little confusing as well, but it is simple now that
>>>>
>>>>
>I
>
>
>>>>understand it.
>>>>
>>>>
>>>>Chris
>>>>
>>>>----- Original Message -----
>>>>From: "mr. barker" <cabalitomb at shaw.ca>
>>>>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>>>><asterisk-users at lists.digium.com>
>>>>Sent: Thursday, May 05, 2005 4:43 AM
>>>>Subject: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>> _____
>>>>>
>>>>>Subject: [Asterisk-Users] Connecting 2 * Together-Pulling hair out
>>>>>
>>>>>
>>>>>
>>>>>I have read the docs on connecting 2* together but am unsure of a few
>>>>>
>>>>>
>>>>>
>>>>>
>>>>things
>>>>
>>>>
>>>>
>>>>
>>>>>Do I need a different account for each number that will be called from
>>>>>
>>>>>
>one
>
>
>>>>>box to the other ? ie. Do I set up a user account on one and then have
>>>>>
>>>>>
>the
>
>
>>>>>other box log into that account when it whats to make a call ?
>>>>>
>>>>>
>>>>>
>>>>>I have 2 asterisk boxes and only one of them has the ability to access
>>>>>
>>>>>
>a
>
>
>>>>>VoipAccount and PSTN connections.(*box 1). The other holds the SIP
>>>>>extensions for the internal SIP users/exten(*box2)
>>>>>
>>>>>I would like to be able to have the box with the Sip UA(*box2) on it to
>>>>>
>>>>>
>be
>
>
>>>>>able to place a call using the box that has the VoipAccount and PSTN
>>>>>connection. I am able to make multiple UA calls on the VoipAccount and
>>>>>
>>>>>
>3
>
>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>on
>>>>
>>>>
>>>>
>>>>
>>>>>the PSTN lines (only have 3 lines coming in). I can get it to work if
>>>>>
>>>>>
>I
>
>
>>>>>create a user exten on *box1 and map a trunk(which is really only an
>>>>>
>>>>>
>>>>>
>>>>>
>>>>exten)
>>>>
>>>>
>>>>
>>>>
>>>>>using the user/password login to that exten from *box2. However when I
>>>>>
>>>>>
>>>>>
>>>>>
>>>>try
>>>>
>>>>
>>>>
>>>>
>>>>>to place a second call when the VOIP line is in use it gives me error (
>>>>>basically saying can't use the trunk because it is in use) I would
>>>>>
>>>>>
>like
>
>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>to
>>>>
>>>>
>>>>
>>>>
>>>>>be able to have this exten/trunk to be able to use multiple connections
>>>>>
>>>>>
>on
>
>
>>>>>it.
>>>>>
>>>>>
>>>>>
>>>>>There must be an easier way to do this I am just not sure how. I
>>>>>
>>>>>
>looked
>
>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>at
>>>>
>>>>
>>>>
>>>>
>>>>>creating IAX trunks but still come up with the Trunk is really an Exten
>>>>>name/password .
>>>>>
>>>>>
>>>>>
>>>>>Any help would be appreciated. (my brain is boiling eggs)
>>>>>
>>>>>
>>>>>
>>>>>Thank you.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>--------------------------------------------------------------------------
>>>
>>>
>--
>
>
>>>>----
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>>_______________________________________________
>>>>>Asterisk-Users mailing list
>>>>>Asterisk-Users at lists.digium.com
>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>To UNSUBSCRIBE or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>>
>>>>>
>>>>>
>>>>_______________________________________________
>>>>Asterisk-Users mailing list
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>>>>
>>>>------------------------------------------------------------------------
>>>>
>>>>_______________________________________________
>>>>Asterisk-Users mailing list
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>>>>
>>>>
>>>>
>>_______________________________________________
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>
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