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

Mon Dec 10 16:35:06 CST 2012

Looks like a connectivity issue, doesn't it?

IP of box2, 172.17.145.145, doesn't show up even once in the SIP dialogues.

What happens on box2 (asterisk -vvvr and tcpdump port 5060) in the 
moment that you place a call through box1 to box2?

Also what's strange is that you are trying to call from box2 to box2? 
Because local_SIP is the context on box2, and on box1 it's "adhearsion". 
The console message you pasted shows @local_SIP however, so it looks 
like you are calling from box2 to box2?

Am 10.12.2012 22:53, schrieb Ken D'Ambrosio:
> 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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>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
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>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>