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Sonny Rajagopalan

2015-Jan-08 16:15 UTC

[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue

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
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Scott Griepentrog

2015-Jan-08 16:48 UTC

head link

[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue

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
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Sonny Rajagopalan

2015-Jan-08 19:32 UTC

head link

[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue

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
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