My suspicion would be that the line o=Z 0 0 IN IP4 146.115.163.234
is causing the problem. Your SIP client is reporting it's external IP address for the audio stream rather than it's internal one. I would look at the settings in your sip client to see if it has any automatic NAT stuff (like using a STUN server) and disable it. Regards, Patrick. ________________________________ From: asterisk-users-boun...@lists.digium.com <asterisk-users-boun...@lists.digium.com> on behalf of Sonny Rajagopalan <sonny.rajagopa...@gmail.com> Sent: 09 January 2015 01:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue [Spam score:10%] 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<http://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<http://192.168.1.50/>: <--- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009<http://192.168.1.50:64009/> ---> INVITE sip:6002@192.168.1.139<mailto:sip%3A6002@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@146.115.163.234:64009;transport=UDP> To: <sip:6002@192.168.1.139<mailto:sip%3A6002@192.168.1.139>;transport=UDP> From: <sip:demo-alice@192.168.1.139<mailto:sip%3Ademo-alice@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@192.168.1.139<mailto:sip%3A6002@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<http://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@192.168.1.139<mailto:sip%3Ademo-alice@192.168.1.139>>;tag=b661670b To: <sip:6002@192.168.1.139<mailto:sip%3A6002@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.rajagopa...@gmail.com<mailto:sonny.rajagopa...@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<http://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 <sgriepent...@digium.com<mailto:sgriepent...@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.rajagopa...@gmail.com<mailto:sonny.rajagopa...@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<http://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@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<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 -- [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://digium.com/> · http://asterisk.org<http://asterisk.org/> -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com<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
-- _____________________________________________________________________ -- 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
