<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div>Hi, Ken</div><div><br></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal; ">I have almost the same setup as yours: new asterisk-----SIP-----Trixbox(Asterisk 1.4)---PRI----pots</div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal; ">Here are my configs:</div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal; "><br></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal; "><font
class="Apple-style-span" face="Arial" size="3"><span class="Apple-style-span" style="font-size: 13px;">new box sip.conf:</span></font></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: Arial; background-color: transparent; font-style: normal; "><font class="Apple-style-span" face="Arial" size="3"><span class="Apple-style-span" style="font-size: 13px;"><div style="background-color: transparent; ">[126]</div><div style="background-color: transparent; ">directmedia=no</div><div style="background-color: transparent; ">type=friend</div><div style="background-color: transparent; ">host=<trixbox_IP_addr></div><div style="background-color: transparent; ">secret=my_secret</div><div style="background-color: transparent; ">username=126 ;this is for outgoing calls from new asterisk via trixbox</div><div style="background-color: transparent; ">fromuser=126 ;this is for outgoing calls from new asterisk via
trixbox</div><div style="background-color: transparent; ">context=default</div><div style="background-color: transparent; ">disallow=all</div><div style="background-color: transparent; ">allow=alaw</div><div style="background-color: transparent; ">allow=ulaw</div><div style="background-color: transparent; ">qualify=yes</div><div style="background-color: transparent; ">qualifyfreq=60</div><div style="background-color: transparent; ">nat=yes</div><div style="background-color: transparent; ">pickupgroup=1</div><div style="background-color: transparent; ">callgroup=1</div><div style="background-color: transparent; "><br></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: Arial; background-color: transparent; font-style: normal; ">trixbox</div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: Arial; background-color: transparent; font-style: normal; "><div style="background-color: transparent; ">[126]</div><div
style="background-color: transparent; ">type=friend</div><div style="background-color: transparent; ">secret=mysecret</div><div style="background-color: transparent; ">record_out=Adhoc</div><div style="background-color: transparent; ">record_in=Adhoc</div><div style="background-color: transparent; ">qualify=yes</div><div style="background-color: transparent; ">port=5060</div><div style="background-color: transparent; ">pickupgroup=</div><div style="background-color: transparent; ">nat=yes</div><div style="background-color: transparent; ">mailbox=</div><div style="background-color: transparent; ">host=dynamic</div><div style="background-color: transparent; ">dtmfmode=rfc2833</div><div style="background-color: transparent; ">dial=SIP/126</div><div style="background-color: transparent; ">context=from-internal</div><div style="background-color: transparent; ">canreinvite=no</div><div style="background-color: transparent; ">callgroup=</div><div
style="background-color: transparent; ">callerid=device <126></div><div style="background-color: transparent; ">accountcode=</div><div style="background-color: transparent; ">call-limit=50</div></div></span></font></div><div><br></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal; ">New box's account (126) registers to the Trixbox so as to make incoming calls from trixbox to new box possible.</div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal; ">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).</div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color:
transparent; font-style: normal; ">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).</div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal; "><br></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal; "><br></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal; ">Hope this helps.</div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif;
background-color: transparent; font-style: normal; "><br></div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal; ">Dmitry Pavlenko</div><div style="color: rgb(0, 0, 0); font-size: 16px; font-family: 'times new roman', 'new york', times, serif; background-color: transparent; font-style: normal; "><br></div> <div style="font-size: 12pt; font-family: 'times new roman', 'new york', times, serif; "> <div style="font-size: 12pt; font-family: 'times new roman', 'new york', times, serif; "> <div dir="ltr"> <font size="2" face="Arial"> <hr size="1"> <b><span style="font-weight:bold;">From:</span></b> Ken D'Ambrosio <ken@jots.org><br> <b><span style="font-weight: bold;">To:</span></b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> <br> <b><span style="font-weight: bold;">Sent:</span></b> Tuesday,
December 11, 2012 3:53 AM<br> <b><span style="font-weight: bold;">Subject:</span></b> Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.<br> </font> </div> <br>
On 2012-12-10 16:16, Danny Nicholas wrote:<br>> Does each box show up in the others "SIP SHOW PEERS"?<br><br>Yup -- each shows in the other's. Sorry I didn't mention that.<br><br>-Ken<br><br>><br>> -----Original Message-----<br>> From: <a ymailto="mailto:asterisk-users-bounces@lists.digium.com" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>> [mailto:<a ymailto="mailto:asterisk-users-bounces@lists.digium.com" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Ken <br>> D'Ambrosio<br>> Sent: Monday, December 10, 2012 2:59 PM<br>> To: <a ymailto="mailto:asterisk-users@lists.digium.com" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>> Subject: [asterisk-users] Problem with SIP trunk I've set up between <br>> two *<br>> boxes.<br>><br>> Hi! I'm trying to set up a
SIP trunk so that I can test calls, etc.,<br>> between a new Asterisk box, and an old 1.4 box.<br>><br>> <br>> ---------------------------------------------------------------------------<br>><br>> New box:<br>> root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf<br>><br>> siptrunk.conf:<br>> [box1] ; All box1 extensions; see extensions.conf type=peer<br>> context=adhearsion<br>> host=172.17.0.17 ; IP for old system<br>> disallow=all<br>> allow=g729<br>> canreinvite=yes<br>> qualify=no<br>><br>><br>> Old box:<br>> root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf<br>><br>> siptrunk.conf:<br>> [box2] ; All box2 extensions; see extensions.conf type=peer<br>> context=local_SIP<br>> host=172.17.145.145 ; IP for new system<br>> disallow=all<br>> allow=g729<br>> canreinvite=yes<br>> qualify=no<br>><br>> extensions.conf
snippet:<br>> [local_SIP]<br>> include => aggregate<br>> include => passthrough<br>> exten => _7XXX,1,Dial(SIP/box2/${EXTEN}) exten => _7XXX,2,Hangup()<br>><br>> <br>> -----------------------------------------------------------------------<br>> When I dial, all I get is (I'll attach the full dialog up to that <br>> point from<br>> SIP debug, below.)<br>> -- Executing [7444@local_SIP:1] Dial("SIP/6110-08291cb0",<br>> "SIP/box2/7444") in new stack<br>> -- Couldn't call box2/7444<br>> Scheduling destruction of SIP dialog<br>> '1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:<br>> INVITE)<br>> == Everyone is busy/congested at this time (0:0/0/0)<br>> <br>> -----------------------------------------------------------------------<br>><br>> Where am I goofing up? Any pointers?<br>><br>>
Thanks!<br>><br>> -Ken<br>><br>><br>><br>><br>> <br>> -----------------------------------------------------------------------<br>> INVITE sip:7444@172.17.0.17 SIP/2.0<br>> Via: SIP/2.0/UDP<br>> <br>> 172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0<br>> Max-Forwards: 70<br>> From: <sip:6110@172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN<br>> To: <sip:7444@172.17.0.17><br>> Contact: <sip:6110@172.17.9.1:55388;ob><br>> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS<br>> CSeq: 24152 INVITE<br>> Route: <sip:172.17.0.17;transport=udp;lr><br>> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, <br>> REFER,<br>> MESSAGE, OPTIONS<br>> Supported: replaces, 100rel, timer, norefersub<br>> Session-Expires: 1800<br>> Min-SE: 90<br>> User-Agent: CSipSimple_d2vzw-16/r1916<br>> Content-Type: application/sdp<br>>
Content-Length: 354<br>><br>> v=0<br>> o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 <br>> 172.17.9.1<br>> t=0 0<br>> m=audio 4006 RTP/AVP 96 3 0 8 101<br>> c=IN IP4 172.17.9.1<br>> a=rtcp:4007 IN IP4 172.17.9.1<br>> a=sendrecv<br>> a=rtpmap:96 SILK/8000<br>> a=fmtp:96 useinbandfec=0<br>> a=rtpmap:3 GSM/8000<br>> a=rtpmap:0 PCMU/8000<br>> a=rtpmap:8 PCMA/8000<br>> a=rtpmap:101 telephone-event/8000<br>> a=fmtp:101 0-15<br>><br>> <-------------><br>> --- (16 headers 16 lines) ---<br>> Sending to 172.17.9.1 : 55388 (NAT)<br>> Using INVITE request as basis request - <br>> nUiGauUpyxjNOJfcZog476ws.Art7jZS<br>><br>> <--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---><br>> SIP/2.0 407 Proxy Authentication Required<br>> Via: SIP/2.0/UDP<br>> <br>> 172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1<br>>
72.17.9.1;rport=55388<br>> From: <sip:6110@172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN<br>> To: <sip:7444@172.17.0.17>;tag=as595faea1<br>> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS<br>> CSeq: 24152 INVITE<br>> User-Agent: Asterisk PBX<br>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>> Supported: replaces<br>> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", <br>> nonce="16883b72"<br>> Content-Length: 0<br>><br>><br>> <------------><br>> Scheduling destruction of SIP dialog <br>> 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'<br>> in 32000 ms (Method: INVITE)<br>> Found user '6110'<br>><br>> <--- SIP read from 172.17.9.1:55388 ---> ACK sip:7444@172.17.0.17 <br>> SIP/2.0<br>> Via: SIP/2.0/UDP<br>> <br>> 172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0<br>> Max-Forwards: 70<br>> From:
<sip:6110@172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN<br>> To: <sip:7444@172.17.0.17>;tag=as595faea1<br>> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS<br>> CSeq: 24152 ACK<br>> Route: <sip:172.17.0.17;transport=udp;lr><br>> Content-Length: 0<br>><br>><br>> <-------------><br>> --- (9 headers 0 lines) ---<br>><br>> <--- SIP read from 172.17.9.1:55388 ---><br>> INVITE sip:7444@172.17.0.17 SIP/2.0<br>> Via: SIP/2.0/UDP<br>> <br>> 172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1<br>> Max-Forwards: 70<br>> From: <sip:6110@172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN<br>> To: <sip:7444@172.17.0.17><br>> Contact: <sip:6110@172.17.9.1:55388;ob><br>> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS<br>> CSeq: 24153 INVITE<br>> Route: <sip:172.17.0.17;transport=udp;lr><br>> Allow: PRACK, INVITE, ACK, BYE,
CANCEL, UPDATE, SUBSCRIBE, NOTIFY,<br>> REFER, MESSAGE, OPTIONS<br>> Supported: replaces, 100rel, timer, norefersub<br>> Session-Expires: 1800<br>> Min-SE: 90<br>> User-Agent: CSipSimple_d2vzw-16/r1916<br>> Proxy-Authorization: Digest username="6110", realm="asterisk",<br>> nonce="16883b72", uri="sip:7444@172.17.0.17",<br>> response="b75389c5938b4f185b3d31bd4463abf3", algorithm=MD5<br>> Content-Type: application/sdp<br>> Content-Length: 354<br>><br>> v=0<br>> o=- 3564161970 3564161970 IN IP4 172.17.9.1<br>> s=pjmedia<br>> c=IN IP4 172.17.9.1<br>> t=0 0<br>> m=audio 4006 RTP/AVP 96 3 0 8 101<br>> c=IN IP4 172.17.9.1<br>> a=rtcp:4007 IN IP4 172.17.9.1<br>> a=sendrecv<br>> a=rtpmap:96 SILK/8000<br>> a=fmtp:96 useinbandfec=0<br>> a=rtpmap:3 GSM/8000<br>> a=rtpmap:0 PCMU/8000<br>> a=rtpmap:8 PCMA/8000<br>> a=rtpmap:101 telephone-event/8000<br>> a=fmtp:101
0-15<br>><br>> <-------------><br>> --- (17 headers 16 lines) ---<br>> Sending to 172.17.9.1 : 55388 (NAT)<br>> Using INVITE request as basis request -<br>> nUiGauUpyxjNOJfcZog476ws.Art7jZS<br>> Found user '6110'<br>> Found RTP audio format 96<br>> Found RTP audio format 3<br>> Found RTP audio format 0<br>> Found RTP audio format 8<br>> Found RTP audio format 101<br>> Peer audio RTP is at port 172.17.9.1:4006<br>> Found unknown media description format SILK for ID 96<br>> Found audio description format GSM for ID 3<br>> Found audio description format PCMU for ID 0<br>> Found audio description format PCMA for ID 8<br>> Found audio description format telephone-event for ID 101<br>> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xe<br>> (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)<br>> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1<br>>
(telephone-event), combined - 0x1 (telephone-event)<br>> Peer audio RTP is at port 172.17.9.1:4006<br>> Looking for 7444 in local_SIP (domain 172.17.0.17)<br>> list_route: hop: <sip:6110@172.17.9.1:55388;ob><br>><br>> <--- Transmitting (no NAT) to 172.17.9.1:55388 ---><br>> SIP/2.0 100 Trying<br>> Via: SIP/2.0/UDP<br>> <br>> 172.17.9.1:55388;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1;received=1<br>> 72.17.9.1;rport=55388<br>> From: <sip:6110@172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN<br>> To: <sip:7444@172.17.0.17><br>> Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS<br>> CSeq: 24153 INVITE<br>> User-Agent: Asterisk PBX<br>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>> Supported: replaces<br>> Contact: <sip:7444@172.17.0.17><br>> Content-Length: 0<br>><br>><br>> <------------><br>> --
Executing [7444@local_SIP:1] Dial("SIP/6110-08293240",<br>> "SIP/box2/7444") in new stack<br>> -- Couldn't call box2/7444<br>> Scheduling destruction of SIP dialog<br>> '2e08d34c5211d82d7e9afa67550458cb@172.17.0.17' in 32000 ms (Method:<br>> INVITE)<br>> == Everyone is busy/congested at this time (0:0/0/0)<br>><br>><br>><br>><br>> --<br>> This mail was scanned by BitDefender<br>> For more information please visit<br>> <a href="http://www.bitdefender.com/links/en/frams.html" target="_blank">http://www.bitdefender.com/links/en/frams.html</a><br>><br>><br>><br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>>
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