The other thing to check is your codecs that you're using. From the console, run: asterisk -rx 'core show translation recalc 1'
It should show you all the codecs that your system supports. Make sure that you've not accidentally restricted it to something like g729 when you don't have it on your box.
Do you have a local DDI or have you got yourself one of the free 028 numbers?
ChillingSilence: What other inbound routes do you have?
The other thing to check is your codecs that you're using. From the console, run: asterisk -rx 'core show translation recalc 1'
It should show you all the codecs that your system supports. Make sure that you've not accidentally restricted it to something like g729 when you don't have it on your box.
Do you have a local DDI or have you got yourself one of the free 028 numbers?
I did have a local DDI number for a bit, but that was only for a month. Right now I'm trying to use an 028 number, because basically I want it so I can just change the details and move over to slingshot, so I'm not without a homeline for days on end till I get it sorted.
I checked the codecs, and they seem to be fine, they all match up and should work...
It shouldn't, but it'll be costing you every time you try and ring it ;)
Post your trunk config here, minus the passwords.
So, just to clarify, you have just the *one* catch-all inbound route? Can you make a call and post the asterisk output again please, now you've got that?
ChillingSilence: It shouldn't, but it'll be costing you every time you try and ring it ;)
Post your trunk config here, minus the passwords.
So, just to clarify, you have just the *one* catch-all inbound route? Can you make a call and post the asterisk output again please, now you've got that?
Thats alright, I'm borrowing a mates 028 account as well, so it doesnt cost :P
It's also worth noting that if I use the register string from 2talks website phonenumber:password@2talk.co.nz/phonenumber it goes to the 2talk voicemail system for my number.
If I use phonenumber:password@2talk.co.nz it goes the goodbye messages.
So I will do a test with both, and give you the output:
with phonenumber in the register string:
Connected to Asterisk 1.6.0.26-FONCORE-r78 currently running on trixbox1 (pid = 2555) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 trixbox1*CLI>
Without the number in the string:
Connected to Asterisk 1.6.0.26-FONCORE-r78 currently running on trixbox1 (pid = 2555) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 -- Executing [s@default:1] Playback("SIP/2talk-0000000e", "vm-goodbye") in new stack -- <SIP/2talk-0000000e> Playing 'vm-goodbye.gsm' (language 'en') -- Executing [s@default:2] Macro("SIP/2talk-0000000e", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/2talk-0000000e", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/2talk-0000000e", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/2talk-0000000e", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/2talk-0000000e", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/2talk-0000000e' in macro 'hangupcall' == Spawn extension (default, s, 2) exited non-zero on 'SIP/2talk-0000000e' trixbox1*CLI>
It's a bit of a concern it plays "vm-goodbye". I'm not sure if that's voicemail-specific or if it's also used as the generic "goodbye". Can you also run: asterisk -rx 'sip show peers'
It should show your Desktop Ext as "OK (5ms)" for example.
One thing you can try - Add a Misc Destionation Call it "Echo Test" Dial: *43 Submit that
Adjust your Inbound route so that it goes through to the Echo Test instead of your Ext, then try calling, and post the asterisk log output.
ChillingSilence: Take out the ilbc&gsm from the trunk
It's a bit of a concern it plays "vm-goodbye". I'm not sure if that's voicemail-specific or if it's also used as the generic "goodbye". Can you also run: asterisk -rx 'sip show peers'
It should show your Desktop Ext as "OK (5ms)" for example.
One thing you can try - Add a Misc Destionation Call it "Echo Test" Dial: *43 Submit that
Adjust your Inbound route so that it goes through to the Echo Test instead of your Ext, then try calling, and post the asterisk log output.
My account is showing up fine:
[trixbox1.localdomain ~]# [trixbox1.localdomain ~]# asterisk -rx 'sip show peers' Name/username Host Dyn Nat ACL Port Status 2talk/02825502572 202.180.76.163 N 5060 Unmonitored 103 (Unspecified) D N A 5060 UNKNOWN 101/101 192.168.0.5 D N A 48640 OK (103 ms) 3 sip peers [Monitored: 1 online, 1 offline Unmonitored: 1 online, 0 offline]
I couldnt find that Misc Destination option you were talking about..
There was custom destination and what not, but that didnt work.. So I set the inbound route to go to the phonebook directory, and tried other options like playing hold music and that didnt work
ChillingSilence: So it shows the exact same thing when you dial through and go to the Phonebook?
You can go into Tools --> Module Admin --> Check in there and make sure Misc Destination is installed and enabled :)
Checked there too, still nothing toat says "Misc Destinations" but there was a panel for Custom Destinations, but that again, doesnt work.
Oh yes, sorry forgot to post that, but yes, it shows the same messages
Connected to Asterisk 1.6.0.26-FONCORE-r78 currently running on trixbox1 (pid = 2555) Verbosity is at least 3 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 -- Executing [s@default:1] Playback("SIP/2talk-0000000f", "vm-goodbye") in new stack -- <SIP/2talk-0000000f> Playing 'vm-goodbye.gsm' (language 'en') -- Executing [s@default:2] Macro("SIP/2talk-0000000f", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/2talk-0000000f", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf("SIP/2talk-0000000f", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("SIP/2talk-0000000f", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup("SIP/2talk-0000000f", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/2talk-0000000f' in macro 'hangupcall' == Spawn extension (default, s, 2) exited non-zero on 'SIP/2talk-0000000f' == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 -- ast_get_srv: SRV lookup for '_sip._UDP.2talk.co.nz' mapped to host fep8.2talk.co.nz, port 5060
Sorry also forgot to mention. The moment I take off those extra codecs, it starts going to the 2talk voicemail box. then when I put them back, it starts saying goodbye :|
ChillingSilence: Sorry, just realized you're using "default" as your context. Change that to "from-trunk" and it'll work :)
Mate, you are brilliant! That was it. Such a tiny detail. Just made a call without any issues.
One thing I am curious about those is the codec thing.. but thats okay, that can be ironed out at a later date... Will have to read up about it a bit more.. Thanks for that though. awesome!
Make sure that you've got them all enabled in the 2talk control panel. You can set your preferences in there. I personally like the iLBC codec the most myself. It's lower bandwidth, better than gsm, but also free. 2talk support it but WxC do not. You've just gotta make sure that your box also supports it.
I seem to be having issues getting my iPhone to work over 3G now :( it works over WiFi no problems, but i cant get any audio over 3G. It rings and registers fine, but just can't hear a damn thing. :( any ideas?
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