[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
Sonny Rajagopalan
sonny.rajagopalan at gmail.com
Thu Jan 8 13:32:43 CST 2015
Thank you for your note, Scott.
I set rewrite_contact=yes for both contacts, and I also had to do
remove_existing=yes because I had to remove the existing contact
information (max_contacts = 1 was preventing new contact information)
using pjsip
qualify demo-alice etc., after which the right IP addresses showed in pjsip
show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is
now
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.1.0/24
;Templates for the necessary config sections
[endpoint_internal](!)
type=endpoint
context=from-internal
disallow=all
allow=ulaw
[auth_userpass](!)
type=auth
auth_type=userpass
[aor_dynamic](!)
type=aor
max_contacts=1
remove_existing=yes
;Definitions for our phones, using the templates above
[demo-alice](endpoint_internal)
auth=demo-alice
aors=demo-alice
mailboxes=box_a
rewrite_contact=yes
[demo-alice](auth_userpass)
password=demo-alice ; put a strong, unique password here instead
username=demo-alice
[demo-alice](aor_dynamic)
[demo-bob](endpoint_internal)
auth=demo-bob
aors=demo-bob
mailboxes=box_b
rewrite_contact=yes
[demo-bob](auth_userpass)
password=demo-bob ; put a strong, unique password here instead
username=demo-bob
[demo-bob](aor_dynamic)
Thank you for your help!
On Thu, Jan 8, 2015 at 11:48 AM, Scott Griepentrog <sgriepentrog at digium.com>
wrote:
> It would appear that you have the Asterisk server on a public IP address,
> your two endpoints are behind a NAT, and you have rewrite_contact enabled
> in pjsip.conf.
>
> In which case, what you are seeing is correct. In order to be able to
> send a call to an extension where it is behind NAT, Asterisk must update
> the contact to have the current IP and port that the phone registered via
> (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state
> for).
>
> On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan <
> sonny.rajagopalan at gmail.com> wrote:
>
>> I am following the instructions in
>> https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I
>> am trying to make a call from extension Alice (6001) to extension for Bob
>> (6002). When I make the call, I can hear the ringing on Alice's phone
>> (caller), but Bob's phone (callee) doesn't ring, or show a call coming in
>> from Alice. My setup and environment is as follows: Alice, Bob and Asterisk
>> all in the same 192.168.1.0/24 network, and they are able to register to
>> the Asterisk server running 13.1.0/PJSIP. The rest of the configuration is
>> the same as the aforementioned wiki page, but is shown here for clarity:
>>
>> root at asterisk13FFP:/var/log/asterisk# more /etc/asterisk/extensions.conf
>> [from-internal]
>> exten=>6001,1,Dial(PJSIP/demo-alice)
>> exten=>6002,1,Dial(PJSIP/demo-bob)
>> exten=>6003,1,Answer()
>> same =>6003,n,Playback(hello-world)
>> same =>6003,n,Hangup()
>>
>>
>> What I do observe is that I when I request the output of pjsip show
>> endpoints, I get Contact information for the two SIP peers that have
>> registered different from their actual IP addresses. I suspect this has
>> something to do with their calls being routed elsewhere. If my assumption
>> is correct--how do I fix this? Alice should be at 192.168.1.50 and Bob
>> should be at 192.168.1.149, instead, they (both) show IP address
>> 146.115.163.234. Any help is deeply appreciated. Thanks.
>>
>> asterisk13FFP*CLI> pjsip show endpoints
>>
>> Endpoint: <Endpoint/CID.....................................>
>> <State.....> <Channels.>
>> I/OAuth:
>> <AuthId/UserName...........................................................>
>> Aor: <Aor............................................>
>> <MaxContact>
>> Contact: <Aor/ContactUri...............................>
>> <Status....> <RTT(ms)..>
>> Transport: <TransportId........> <Type> <cos> <tos>
>> <BindAddress..................>
>> Identify:
>> <Identify/Endpoint.........................................................>
>> Match: <ip/cidr.........................>
>> Channel: <ChannelId......................................>
>> <State.....> <Time(sec)>
>> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
>>
>> =========================================================================================
>>
>> Endpoint: demo-alice
>> Unavailable 0 of inf
>> InAuth: demo-alice/demo-alice
>> Aor: demo-alice 1
>> Contact: demo-alice/sip:demo-alice@*146.115.163.234*:38519
>> Unknown nan
>>
>> Endpoint: demo-bob Not in
>> use 0 of inf
>> InAuth: demo-bob/demo-bob
>> Aor: demo-bob 1
>> Contact: demo-bob/sip:demo-bob@*146.115.163.234*:38321;tra
>> Unknown nan
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
>
>
>
> --
> [image: Digium logo]
> Scott Griepentrog
> Digium, Inc · Software Developer
> 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
> direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
> Check us out at: http://digium.com · http://asterisk.org
>
> --
> _____________________________________________________________________
> -- 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/20150108/ced0d1c0/attachment.html>
More information about the asterisk-users
mailing list