Hi, Ken
I have almost the same setup as yours: new
asterisk-----SIP-----Trixbox(Asterisk 1.4)---PRI----pots
Here are my configs:
new box sip.conf:
[126]
directmedia=no
type=friend
host=<trixbox_IP_addr>
secret=my_secret
username=126 ;this is for outgoing calls from new asterisk via trixbox
fromuser=126 ;this is for outgoing calls from new asterisk via trixbox
context=default
disallow=all
allow=alaw
allow=ulaw
qualify=yes
qualifyfreq=60
nat=yes
pickupgroup=1
callgroup=1
trixbox
[126]
type=friend
secret=mysecret
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=
host=dynamic
dtmfmode=rfc2833
dial=SIP/126
context=from-internal
canreinvite=no
callgroup=
callerid=device <126>
accountcode=
call-limit=50
New box's account (126) registers to the Trixbox so as to make incoming calls
from trixbox to new box possible.
The config in the new box implies that the trixbox require authorization in
calls from the new box (username and fromuser options are necessary for this).
Actually looking through the sip.conf in 1.8 asterisk I found that there are
"auth " option as well as "remotesecret and remoteuser" - but I can not
understand how they work in case if I need to authorise my outgoing calls
(probably sip.conf will be more logical in the future 12th version).
Hope this helps.
Dmitry Pavlenko
________________________________
From: Ken D'Ambrosio <[email protected]>
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Sent: Tuesday, December 11, 2012 3:53 AM
Subject: Re: [asterisk-users] Problem with SIP trunk I've set up between two *
boxes.
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: [email protected]
> [mailto:[email protected]] On Behalf Of Ken
> D'Ambrosio
> Sent: Monday, December 10, 2012 2:59 PM
> To: [email protected]
> 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@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@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@local_SIP:1] Dial("SIP/6110-08291cb0",
> "SIP/box2/7444") in new stack
> -- Couldn't call box2/7444
> Scheduling destruction of SIP dialog
> '[email protected]' 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:[email protected] SIP/2.0
> Via: SIP/2.0/UDP
>
> 172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
> Max-Forwards: 70
> From: <sip:[email protected]>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
> To: <sip:[email protected]>
> Contact: <sip:[email protected]: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:[email protected]>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
> To: <sip:[email protected]>;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:[email protected]
> SIP/2.0
> Via: SIP/2.0/UDP
>
> 172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
> Max-Forwards: 70
> From: <sip:[email protected]>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
> To: <sip:[email protected]>;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:[email protected] SIP/2.0
> Via: SIP/2.0/UDP
>
> 172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
> Max-Forwards: 70
> From: <sip:[email protected]>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
> To: <sip:[email protected]>
> Contact: <sip:[email protected]: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:[email protected]",
> 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:[email protected]: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:[email protected]>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
> To: <sip:[email protected]>
> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
> CSeq: 24153 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:[email protected]>
> Content-Length: 0
>
>
> <------------>
> -- Executing [7444@local_SIP:1] Dial("SIP/6110-08293240",
> "SIP/box2/7444") in new stack
> -- Couldn't call box2/7444
> Scheduling destruction of SIP dialog
> '[email protected]' in 32000 ms (Method:
> INVITE)
> == Everyone is busy/congested at this time (0:0/0/0)
>
>
>
>
> --
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