Does the mytel gateway show up fine in sip show peers? PaulH
On Thu, 2007-09-13 at 15:06 +1000, Joshua Small wrote: > You can ignore this. I mistyped the password, and once it was fixed, > and registered correctly, both links failed to work again. > > I have some extended information from sip debug. Again, this shows up > as soon as I try to register two connections. > > > > <--- SIP read from 203.166.103.242:5060 ---> > > SIP/2.0 403 Forbidden > > Via: SIP/2.0/UDP > 192.168.107.4:5060;branch=z9hG4bK454ad99d;received=59.167.248.154;rport=53487 > > From: "Joshua Small" <sip:[EMAIL PROTECTED]>;tag=as3d465ba3 > > To: <sip:[EMAIL PROTECTED]>;tag=as5937f41d > > Call-ID: [EMAIL PROTECTED] > > CSeq: 103 INVITE > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > Content-Length: 0 > > > > Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 > 959 | www.visinet.com.au > > This e-mail is intended for use by the named recipients only and > contains confidential information. Opinions and other information in > this message that pertain to the sender's employer and its products > and services represent the opinion of the sender and not > necessarily those of the employer. > > > > > From: Joshua Small > Sent: Thursday, 13 September 2007 1:38 PM > To: '[email protected]' > Subject: FW: [asterisk-users] Problems with two trunks > > > > > Update on this: > > > > I found that by changing insecure = very to insecure = invite, adding > the second trunk no longer stopped calls working. > > I’ve read the documentation on this switch and still don’t see how it > applies/is meant to get used. > > > > Anyway, with this change in place, the following may help: > > > > asterisk*CLI> sip show registry > > Host Username Refresh State > Reg.Time > > gw02.mytel.net.au:5060 11111 120 Request > Sent > > gw02.mytel.net.au:5060 22222 105 Registered > Thu, 13 Sep 2007 23:33:47 > > > > I have set a dial plan so that some handsets use the 2222 (not the > real number) extension (which work) and now I only need to determine > why 11111 never seems to register. > > > > If I remove all traces of the 2222 connection from my config, 11111 > registers fine. > > Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 > 959 | www.visinet.com.au > > This e-mail is intended for use by the named recipients only and > contains confidential information. Opinions and other information in > this message that pertain to the sender's employer and its products > and services represent the opinion of the sender and not > necessarily those of the employer. > > > > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Joshua > Small > Sent: Thursday, 13 September 2007 10:44 AM > To: [email protected] > Subject: [asterisk-users] Problems with two trunks > > > > > Hi, > > > > I am attempting to setup an asterisk server, current specs: > > CentOS release 5 (Final) > > Asterisk 1.4.11 > > Asterisk-gui checked out from SVN last week > > > > I started with a fairly basic setup involving one VOIP provider who > provided one dial in number, and a couple of handsets. Config files > are below. It was pretty much totally built by Asterisk-gui, except > for the fact I had to add “insecure=very” into users.conf in order to > stop the dialin from our provider presenting an authentication error. > Advice on any more correct approach would be appreciated, but is not > the focus of this post: > > > > Users.conf > > ;several handsets setup like this... > > [6001] > > callwaiting = yes > > context = numberplan-custom-1 > > email = [EMAIL PROTECTED] > > fullname = Joshua Small > > hasagent = yes > > hasdirectory = yes > > hasiax = no > > hasmanager = no > > hassip = yes > > hasvoicemail = no > > host = dynamic > > mailbox = 6001 > > secret = XXXXX > > threewaycalling = yes > > registeriax = no > > registersip = yes > > canreinvite = no > > nat = no > > dtmfmode = rfc2833 > > vmsecret = 1234 > > > > ;some PSTNS > > [trunk_2] > > callerid = asreceived > > context = DID_trunk_2 > > group = 2 > > hasexten = no > > hasiax = no > > hassip = no > > trunkname = Ports 1,2,3,4 > > trunkstyle = analog > > zapchan = 1,2,3,4 > > > > ;my IP trunk > > [trunk_3] > > allow = all > > context = DID_trunk_3 > > dialformat = ${EXTEN:1} > > hasexten = no > > hasiax = no > > hassip = yes > > host = gw02.mytel.net.au > > port = 5060 > > registeriax = no > > registersip = yes > > secret = XXXXXXXX > > trunkname = Custom - MyTel2 > > trunkstyle = customvoip > > username = XXXXXXXX > > type = friend > > nat = yes > > > > ;extensions.conf > > [numberplan-custom-1] > > plancomment = DialPlan1 > > include = default > > include = parkedcalls > > exten = _0XXXXX!,1,Macro(trunkdial,${trunk_3}/${EXTEN:1}) > > comment = _0XXXXX!,1,First,standard > > ;a failover to PSTN, not yet enabled > > ;exten = _0XXXXX!,2,Macro(trunkdial,${trunk_2}/0${EXTEN:1}) > > ;comment = _0XXXXX!,1,First,standard > > > > At this point, everything appears to work fine. We can make calls from > our several handsets using our voip link no problems. > > We have two different accounts with our provider, the goal being > certain handsets will connect to this account and therefore be billed > separately. I haven’t gotten as far as to add the extra handsets and > set an appropriate dialplan, all I did was add this to users.conf: > > > > [trunk_extra] > > allow = all > > context = DID_trunk_3 > > dialformat = ${EXTEN:1} > > hasexten = no > > hasiax = no > > hassip = yes > > host = gw02.mytel.net.au > > port = 5060 > > registeriax = no > > registersip = yes > > secret = XXXXXXXX > > trunkname = Custom - MyTel Two > > trunkstyle = customvoip > > username = XXXXXXXXXX > > type = friend > > nat = yes > > > > From this point on, my existing handsets don’t appear to be able to > get a line out. My console looks like this (from the first call out): > > Connected to Asterisk 1.4.11 currently running on asterisk (pid = > 31999) > > -- Remote UNIX connection > > Verbosity is at least 8 > > -- Executing [EMAIL PROTECTED]:1] > Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new > stack > > -- Executing [EMAIL PROTECTED]:1] Dial("SIP/8001-b7d0bb20", > "SIP/trunk_3/0425298582") in new stack > > -- Called trunk_3/0425298582 > > [Sep 13 20:42:16] WARNING[32011]: chan_sip.c:12016 > handle_response_invite: Received response: "Forbidden" from '"Joshua > Small" <sip:[EMAIL PROTECTED]>;tag=as29bb274d' > > -- SIP/trunk_3-097ac708 is circuit-busy > > == Everyone is busy/congested at this time (1:0/1/0) > > -- Executing [EMAIL PROTECTED]:2] Goto("SIP/8001-b7d0bb20", > "s-CONGESTION|1") in new stack > > -- Goto (macro-trunkdial,s-CONGESTION,1) > > -- Executing [EMAIL PROTECTED]:1] > NoOp("SIP/8001-b7d0bb20", "") in new stack > > == Auto fallthrough, channel 'SIP/8001-b7d0bb20' status is > 'CONGESTION' > > > > > > Any advice on why our trunk_3 becomes congested, just because > trunk_extra is set to exist, is appreciated. > > Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 > 959 | www.visinet.com.au > > This e-mail is intended for use by the named recipients only and > contains confidential information. Opinions and other information in > this message that pertain to the sender's employer and its products > and services represent the opinion of the sender and not > necessarily those of the employer. > > > > > _______________________________________________ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
