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 192.168.1.0/24 network would have helped me tremendously. On Thu, Jan 8, 2015 at 8:03 PM, Sonny Rajagopalan < sonny.rajagopa...@gmail.com> wrote: > 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@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;transport=UDP> > From: <sip:demo-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 > ;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@192.168.1.139>;tag=b661670b > To: <sip:6002@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> 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 < >> 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> 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@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 >>> >> >> >
-- _____________________________________________________________________ -- 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