[asterisk-users] Problem with SIP trunk I've set up between two * boxes.

Ken D'Ambrosio ken at jots.org
Mon Dec 10 15:53:24 CST 2012 

On 2012-12-10 16:16, Danny Nicholas wrote:
> Does each box show up in the others "SIP SHOW PEERS"?

Yup -- each shows in the other's. Sorry I didn't mention that.

-Ken

>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ken 
> D'Ambrosio
> Sent: Monday, December 10, 2012 2:59 PM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Problem with SIP trunk I've set up between 
> two *
> boxes.
>
> Hi!  I'm trying to set up a SIP trunk so that I can test calls, etc.,
> between a new Asterisk box, and an old 1.4 box.
>
> 
> ---------------------------------------------------------------------------
>
> New box:
> root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
>
> siptrunk.conf:
> [box1] ; All box1 extensions; see extensions.conf type=peer
> context=adhearsion
> host=172.17.0.17  ; IP for old system
> disallow=all
> allow=g729
> canreinvite=yes
> qualify=no
>
>
> Old box:
> root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
>
> siptrunk.conf:
> [box2] ; All box2 extensions; see extensions.conf type=peer
> context=local_SIP
> host=172.17.145.145 ; IP for new system
> disallow=all
> allow=g729
> canreinvite=yes
> qualify=no
>
> extensions.conf snippet:
> [local_SIP]
> include => aggregate
> include => passthrough
> exten => _7XXX,1,Dial(SIP/box2/${EXTEN}) exten => _7XXX,2,Hangup()
>
> 
> -----------------------------------------------------------------------
> When I dial, all I get is (I'll attach the full dialog up to that 
> point from
> SIP debug, below.)
>      -- Executing [7444 at local_SIP:1] Dial("SIP/6110-08291cb0",
> "SIP/box2/7444") in new stack
>      -- Couldn't call box2/7444
> Scheduling destruction of SIP dialog
> '1f18dd4b4ee8f7583041de280f307c18 at 172.17.0.17' in 32000 ms (Method:
> INVITE)
>    == Everyone is busy/congested at this time (0:0/0/0)
> 
> -----------------------------------------------------------------------
>
> Where am I goofing up?  Any pointers?
>
> Thanks!
>
> -Ken
>
>
>
>
> 
> -----------------------------------------------------------------------
> INVITE sip:7444 at 172.17.0.17 SIP/2.0
> Via: SIP/2.0/UDP
> 
> 172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
> Max-Forwards: 70
>  From: <sip:6110 at 172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
> To: <sip:7444 at 172.17.0.17>
> Contact: <sip:6110 at 172.17.9.1:55388;ob>
> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
> CSeq: 24152 INVITE
> Route: <sip:172.17.0.17;transport=udp;lr>
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, 
> REFER,
> MESSAGE, OPTIONS
> Supported: replaces, 100rel, timer, norefersub
> Session-Expires: 1800
> Min-SE: 90
> User-Agent: CSipSimple_d2vzw-16/r1916
> Content-Type: application/sdp
> Content-Length:   354
>
> v=0
> o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 
> 172.17.9.1
> t=0 0
> m=audio 4006 RTP/AVP 96 3 0 8 101
> c=IN IP4 172.17.9.1
> a=rtcp:4007 IN IP4 172.17.9.1
> a=sendrecv
> a=rtpmap:96 SILK/8000
> a=fmtp:96 useinbandfec=0
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> <------------->
> --- (16 headers 16 lines) ---
> Sending to 172.17.9.1 : 55388 (NAT)
> Using INVITE request as basis request - 
> nUiGauUpyxjNOJfcZog476ws.Art7jZS
>
> <--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
> 
> 172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1
> 72.17.9.1;rport=55388
>  From: <sip:6110 at 172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
> To: <sip:7444 at 172.17.0.17>;tag=as595faea1
> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
> CSeq: 24152 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", 
> nonce="16883b72"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog 
> 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
> in 32000 ms (Method: INVITE)
> Found user '6110'
>
> <--- SIP read from 172.17.9.1:55388 ---> ACK sip:7444 at 172.17.0.17 
> SIP/2.0
> Via: SIP/2.0/UDP
> 
> 172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
> Max-Forwards: 70
>  From: <sip:6110 at 172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
> To: <sip:7444 at 172.17.0.17>;tag=as595faea1
> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
> CSeq: 24152 ACK
> Route: <sip:172.17.0.17;transport=udp;lr>
> Content-Length:  0
>
>
> <------------->
> --- (9 headers 0 lines) ---
>
> <--- SIP read from 172.17.9.1:55388 --->
> INVITE sip:7444 at 172.17.0.17 SIP/2.0
> Via: SIP/2.0/UDP
> 
> 172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
> Max-Forwards: 70
>  From: <sip:6110 at 172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
> To: <sip:7444 at 172.17.0.17>
> Contact: <sip:6110 at 172.17.9.1:55388;ob>
> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
> CSeq: 24153 INVITE
> Route: <sip:172.17.0.17;transport=udp;lr>
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
> REFER, MESSAGE, OPTIONS
> Supported: replaces, 100rel, timer, norefersub
> Session-Expires: 1800
> Min-SE: 90
> User-Agent: CSipSimple_d2vzw-16/r1916
> Proxy-Authorization: Digest username="6110", realm="asterisk",
> nonce="16883b72", uri="sip:7444 at 172.17.0.17",
> response="b75389c5938b4f185b3d31bd4463abf3", algorithm=MD5
> Content-Type: application/sdp
> Content-Length:   354
>
> v=0
> o=- 3564161970 3564161970 IN IP4 172.17.9.1
> s=pjmedia
> c=IN IP4 172.17.9.1
> t=0 0
> m=audio 4006 RTP/AVP 96 3 0 8 101
> c=IN IP4 172.17.9.1
> a=rtcp:4007 IN IP4 172.17.9.1
> a=sendrecv
> a=rtpmap:96 SILK/8000
> a=fmtp:96 useinbandfec=0
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> <------------->
> --- (17 headers 16 lines) ---
> Sending to 172.17.9.1 : 55388 (NAT)
> Using INVITE request as basis request -
> nUiGauUpyxjNOJfcZog476ws.Art7jZS
> Found user '6110'
> Found RTP audio format 96
> Found RTP audio format 3
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 101
> Peer audio RTP is at port 172.17.9.1:4006
> Found unknown media description format SILK for ID 96
> Found audio description format GSM for ID 3
> Found audio description format PCMU for ID 0
> Found audio description format PCMA for ID 8
> Found audio description format telephone-event for ID 101
> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xe
> (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
> Peer audio RTP is at port 172.17.9.1:4006
> Looking for 7444 in local_SIP (domain 172.17.0.17)
> list_route: hop: <sip:6110 at 172.17.9.1:55388;ob>
>
> <--- Transmitting (no NAT) to 172.17.9.1:55388 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 
> 172.17.9.1:55388;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1;received=1
> 72.17.9.1;rport=55388
>  From: <sip:6110 at 172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
> To: <sip:7444 at 172.17.0.17>
> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
> CSeq: 24153 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:7444 at 172.17.0.17>
> Content-Length: 0
>
>
> <------------>
>      -- Executing [7444 at local_SIP:1] Dial("SIP/6110-08293240",
> "SIP/box2/7444") in new stack
>      -- Couldn't call box2/7444
> Scheduling destruction of SIP dialog
> '2e08d34c5211d82d7e9afa67550458cb at 172.17.0.17' in 32000 ms (Method:
> INVITE)
>    == Everyone is busy/congested at this time (0:0/0/0)
>
>
>
>
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> --
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> _____________________________________________________________________
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>                http://www.asterisk.org/hello
>
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