Help connecting Trixbox to callwithus
Submitted by mpatc on Wed, 01/30/2008 - 6:44am.
I have set up a Trixbox 2.4 box on an Dell GX270 with 1 Polycom 500, two Nortel i2004, a X-lite softphone and a Zoiper softphone. All phones appear to connect to trixbox/asterisk ok and receive time, date, and missed call messages. I can call internally from any phone to any other phone.
I have tried to set up trunks with callwithus (both SIP and IAX) and with FWD. To date I can get the softphones to connect directly to callwithus and can make outgoing calls. Based on this, I know my callwithus account is good.
When I try to place an outgoing call from any phone through trixbox to a PSTN number the call does not go through. On the zioper softphone, I see 'early media', hear nothing and then see 'service or option unavailable'. When I pick up a polycom or nortel phone, I hear a dialtone until I dial, after that I hear nothing, they will show that they are dialing, and then show 'congestion'.
When I dial internally the asterisk CLI looks like this:
"
-- Executing [100@from-internal:1] Macro("SIP/198-09e02f10", "exten-vm|novm|100") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/198-09e02f10", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/198-09e02f10", "user-callerid: device 198") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/198-09e02f10", "AMPUSER=198") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/198-09e02f10", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] GotoIf("SIP/198-09e02f10", "0?start") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/198-09e02f10", "REALCALLERIDNUM=198") in new stack
-- Executing [s@macro-user-callerid:6] NoOp("SIP/198-09e02f10", "REALCALLERIDNUM is 198") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/198-09e02f10", "AMPUSER=198") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/198-09e02f10", "AMPUSERCIDNAME=Matt - Zoiper") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/198-09e02f10", "0?report") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/198-09e02f10", "AMPUSERCID=198") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/198-09e02f10", "CALLERID(all)="Matt - Zoiper" <198>") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/198-09e02f10", "REALCALLERIDNUM=198") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/198-09e02f10", "TTL: ARG1: novm") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/198-09e02f10", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("SIP/198-09e02f10", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/198-09e02f10", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/198-09e02f10", "Using CallerID "Matt - Zoiper" <198>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/198-09e02f10", "FROMCONTEXT=exten-vm") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/198-09e02f10", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/198-09e02f10", "EXTTOCALL=100") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/198-09e02f10", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/198-09e02f10", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/198-09e02f10", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/198-09e02f10", "record-enable|100|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/198-09e02f10", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/198-09e02f10", "recordingcheck|20080130-080607|1201698367.114") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080130-080607|1201698367.114: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("SIP/198-09e02f10", "No recording needed") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/198-09e02f10", "dial||tr|100") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/198-09e02f10", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/198-09e02f10", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'Matt - Zoiper' number is '198'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 100 to extension map
-- dialparties.agi: Extension 100 cf is disabled
-- dialparties.agi: Extension 100 do not disturb is disabled
> dialparties.agi: extnum 100 has: cw: 1; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: ExtensionState: 0
-- dialparties.agi: dbset CALLTRACE/100 to 198
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:10] Dial("SIP/198-09e02f10", "USTM/100@NOC||tr") in new stack
-- unistim_request(100@NOC)
-- Called 100@NOC
-- USTM/100@NOC-0 is ringing
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/198-09e02f10' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/198-09e02f10' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/198-09e02f10'
-- Executing [h@macro-dial:1] Macro("SIP/198-09e02f10", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/198-09e02f10", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/198-09e02f10", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/198-09e02f10", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/198-09e02f10", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/198-09e02f10", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/198-09e02f10", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/198-09e02f10' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/198-09e02f10'
trixbox1*CLI>
"
When I try to call a PSTN outside number it looks like this
"
-- Executing [1614XXXXXXX@from-internal:1] ResetCDR("SIP/198-09e02f10", "") in new stack
-- Executing [1614XXXXXXX@from-internal:2] NoCDR("SIP/198-09e02f10", "") in new stack
-- Executing [1614XXXXXXX@from-internal:3] Wait("SIP/198-09e02f10", "1") in new stack
-- Executing [1614XXXXXXX@from-internal:4] Playback("SIP/198-09e02f10", "silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer") in new stack
--
--
--
-- Executing [1614XXXXXXX@from-internal:5] Wait("SIP/198-09e02f10", "1") in new stack
-- Executing [1614XXXXXXX@from-internal:6] Congestion("SIP/198-09e02f10", "20") in new stack
== Spawn extension (from-internal, 1614XXXXXXX, 6) exited non-zero on 'SIP/198-09e02f10'
-- Executing [h@from-internal:1] Macro("SIP/198-09e02f10", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/198-09e02f10", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/198-09e02f10", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/198-09e02f10", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/198-09e02f10", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/198-09e02f10", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/198-09e02f10", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/198-09e02f10' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/198-09e02f10'
trixbox1*CLI>
"
My outbound trunk for callwithus appears to connect over IAX (and SIP too when I was using that instead of IAX)
64.85.162.136:4569 N 519XXXXXX 64.XXX.XXX.XXX:4569 60 Registered
The dial plan of my outbound route that uses the IAX trunk is
1+NXXNXXXXXX
1614+NXXXXXX
1800NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX
Over the past 2 weeks I have read countless pages and tried many things, most of the time not knowing exactly what I was doing. If you have any suggestions, I would greatly appreciate it. Let me know what .conf files you would like to see and I will post them.
Thank you very much,
Matt
My setup
Wed, 01/30/2008 - 12:17pm
My outbound route has these dial patterns in them:
1NXXNXXXXXX
NXXNXXXXXX
NXXXXXX
My CallWithUs trunk settings has these dial patterns in them:
1320+NXXXXXX
1+NXXNXXXXXX
My outgoing settings for CallWithUs look like:
allow=g729&ilbc&ulaw&gsm
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=callwithus.com
insecure=invite
secret=secret
type=friend
username=username
Hope that helps.
Tom, Thanks for the
Thu, 01/31/2008 - 6:03am
Tom, Thanks for the suggestions. I tried the few lines you had that I did not but I still have the get the same result.
It seems like somewhere between dialing the number, selecting the route dial plan and the trunk things get lost. From the CLI it appears that asterisk does not even try to start a call, it just plays the 'cannot connect message'
Until further notice, be advised that I don't know what I am doing.
Maybe it has nothing to do,
Thu, 01/31/2008 - 3:08pm
Maybe it has nothing to do, but have just added a sip trunk to my box and I was getting a "all conections are busy" and similar messages and it was a SIP authorization problem, I was able to troubleshoot it by using "sip debug" and forgetting about the asterisk log...
In my case I saw that the responses from the sip provider where being sent to the wrong ip, and i just realized that I forgot to set a externip in my sip.conf
Hope it helps
Similar Issue
Thu, 04/10/2008 - 10:16am
I can likewise receive calls but can't dial out. I use CallWithUs for IAX as well, and I've had it working with other Asterisk installations, but can't get it working with FreePBX.
Here are my settings. Am I missing something obvious?
Trunk: Outgoing Dial Rules Dial Rules: 1952+NXXXXXX 1+NXXXXXXXXX 1NXXXXXXXXX Outbound Dial Prefix: 9 Outgoing Settings Trunk Name: CallWithUs PEER Details: allow=g729&ilbc&ulaw&gsm context=from-trunk disallow=all dtmfmode=rfc2833 host=callwithus.com insecure=invite secret=ssssss type=friend username=uuuuuuuuu Incoming Settings USER Context: uuuuuuuuu USER Details: type=user username=uuuuuuuuu context=from-trunk Registration Register String: uuuuuuuuu:ssssss@west.callwithus.com Outbound Route: Route Name: Everything Route Password: (blank) Emergency Dialing: [ ] Intra Company Route: [ ] Music on Hold? [default] Dial Patterns: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk Sequence: 0 IAX2/CallWithUs Inbound Route: Route: InCallWithUs DID Number: [ ] Caller ID Number: [ ] Fax Extension: [disabled] Set Destination: ( ) Terminate Call (X) [<774> Andrew Rothman] ( ) Voicemail
Okay, so it turns out the
Fri, 04/11/2008 - 4:06pm
Okay, so it turns out the whole time that the issue is I don't have to dial "9" for an outside trunk.
But suppose I want to? How do I set it up so that people DO have to dial 9 for an outside line? I tried putting this in the trunk:
9|1NXXNXXXXXX
...and so on, but no dice. I just managed to dial India. :( There's five and a half cents I'll never get back...
I have these rules in my
Sun, 04/13/2008 - 10:53am
I have these rules in my Outbound Route:
9|1NXXNXXXXXX
9|NXXNXXXXXX
9|NXXXXXX
1NXXNXXXXXX
NXXNXXXXXX
NXXXXXX
Then in my Trunk I have these rules:
1320+NXXXXXX
1+NXXNXXXXXX
Basically, the outbound route selects which route to take (with or without a 9). The Trunk rules "fix" the number if they need to be fixed (adding a 1 or the 1320 prefix).
Member Since:
2008-01-15