Asterisk 13.1.0/PJSIP Peer IP Address Issue

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I am following the instructions in 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 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@asterisk13FFP:/var/log/asterisk# more /etc/asterisk/extensions.conf
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 and Bob should be at, instead, they (both) show IP address Any help is deeply appreciated. Thanks.

asterisk13FFP*CLI> pjsip show endpoints








Exten: CLCID:
Endpoint: demo-alice Unavailable 0 of inf
InAuth: demo-alice/demo-alice
Aor: demo-alice 1
Contact: demo-alice/sip:demo-alice@**:38519 Unknown

Endpoint: demo-bob Not in use
0 of inf
InAuth: demo-bob/demo-bob
Aor: demo-bob 1
Contact: demo-bob/sip:demo-bob@**:38321;tra Unknown

5 thoughts on - 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).

  • 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

    type=transport protocol=udp bind=
    ;Templates for the necessary config sections

    type=endpoint context=from-internal disallow=all allow=ulaw

    type=auth auth_type=userpass

    type=aor max_contacts=1
    ;Definitions for our phones, using the templates above

    auth=demo-alice aors=demo-alice mailboxes=box_a rewrite_contact=yes
    password=demo-alice ; put a strong, unique password here instead username=demo-alice


    auth=demo-bob aors=demo-bob mailboxes=box_b rewrite_contact=yes
    password=demo-bob ; put a strong, unique password here instead username=demo-bob


    Thank you for your help!

  • 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 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

    < --- Received SIP request (1229 bytes) from UDP: --->
    INVITE sip:6002@;transport=UDP SIP/2.0
    Via: SIP/2.0/UDP
    Max-Forwards: 70
    From: ;tag=b661670b Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. CSeq: 2 INVITE
    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=”

    Allow-Events: presence, kpml Content-Length: 245

    o=Z 0 0 IN IP4
    c=IN IP4
    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

    < --- Transmitting SIP response (319 bytes) to UDP: --->
    SIP/2.0 100 Trying Via: SIP/2.0/UDP
    Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. From: ;tag=b661670b To:
    CSeq: 2 INVITE
    Content-Length: 0

    Any help is appreciated. A topology is shown below in ASCII.

    < ( Big bad Internet ) >


  • Solved!

    The issue was that RTP flows were being established to the wrong IP address.

    I figured out this issue–I had to disable STUN in both SIP phones for this to work correctly.

    Still, I wish a working configuration for Asterisk, and two SIP phones in the same network would have helped me tremendously.

  • ​To sort out RTP problems, I would recommend:

    1) on all endpoints use codec of allow=!all,ulaw — this is or should be supported by all endpoints and eliminates any issues of mismatch, translation, etc., and can be adjusted later once everything is working

    2) add an Echo() application to your dialplan so you can call it and check
    ​RTP to and from Asterisk

    3) start with direct_media=no to run all the RTP through Asterisk first

    4) packet capture at/on the asterisk server, as well as at endpoints if need be, to identfy if and where RTP streams are being sent and received.

    The goal being to get two way audio calls up through Asterisk, and then change one thing at a time towards your desired configuration and retest.