[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
Sonny Rajagopalan
sonny.rajagopalan at gmail.com
Thu Jan 8 19:03:38 CST 2015
Well, I thought it worked, but it actually doesn't--I am able to get the
caller pick up the phone, but for some reason, I cannot hear anything on
either side no matter who does the calling. Again, my two SIP phones are on
the local 192.168.1.0/24 network (do not go over the Internet) and the
Asterisk server is located in the same network (not accessed over the
Internet). Any help is appreciated.
Does the fact that Asterisk is running on a VirtualBox VM on the same
machine as one of the SIP phones matter? I am able to access the ARI REST
interface of the Asterisk server quite fine on the host machine.
I suspect it has to do with RTP not being set up, but all the codec support
is there. Here's a log for the SIP request from 192.168.1.50:
<--- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009 --->
INVITE sip:6002 at 192.168.1.139;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 146.115.163.234:64009
;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-
Max-Forwards: 70
Contact: <sip:demo-alice at 146.115.163.234:64009;transport=UDP>
To: <sip:6002 at 192.168.1.139;transport=UDP>
From: <sip:demo-alice at 192.168.1.139;transport=UDP>;tag=b661670b
Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.21933 r21903
Authorization: Digest
username="demo-alice",realm="asterisk",nonce="[removed]",uri="
sip:6002 at 192.168.1.139
;transport=UDP",response="[removed]",cnonce="[removed]",nc=00000001,qop=auth,algorithm=md5,opaque="[removed]"
Allow-Events: presence, kpml
Content-Length: 245
v=0
o=Z 0 0 IN IP4 146.115.163.234
s=Z
c=IN IP4 146.115.163.234
t=0 0
m=audio 8000 RTP/AVP 0 3 110 8 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<--- Transmitting SIP response (319 bytes) to UDP:192.168.1.50:64009 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 146.115.163.234:64009
;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-
Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.
From: <sip:demo-alice at 192.168.1.139>;tag=b661670b
To: <sip:6002 at 192.168.1.139>
CSeq: 2 INVITE
Content-Length: 0
Any help is appreciated. A topology is shown below in ASCII.
< ( Big bad Internet ) >
GW/NAPT/Router
|
----------------------------------------------------------
/ \
| |
Host A Host B
-----------------
-----------------
| Alice | | Bob
|
| 192.168.1.50 | |
192.168.1.149 |
|---------------|
|---------------|
| Asterisk sr |
| (VM) |
| 192.168.1.239 |
|---------------|
On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> 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
>>>
>>>
>>> --
>>> _____________________________________________________________________
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>>>
>>
>>
>>
>> --
>> [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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