Update: I am no longer worrying about the Xlite connection. I’ve got an analog phone connected via a Sipura 2100 to the Asterisk server. So I’m having trouble dialing with the Sipura, Xlite, and any other way I could imagine.
I know there’s a bunch of junk in here from the GUI. I tried to set it up via the GUI and it wouldn’t connect, so I abandoned it for the manual way again. There are also other inbound numbers and one other extension, but I don’t believe those are relevant for this particular issue the way I understand it…
Ok. I thought by setting the “context” of [601] in users.conf to “outgoing-1”, it should try to dial out from the extension. However, it always tells me that it can’t find the number I dial in the correct context. “Looking for 9390[last 4 digits] in outgoing-1 (domain 192.168.0.76)”
I tried removing spaces in outgoing context but no matter what else, I try, I can’t get the outgoing calls to go through. Now what???
; -----------------------------------------------------------
[[[users.conf]]]
; -----------------------------------------------------------
[trunk_1]
secret = XXXXXXXXXXXX
provider =
trunkstyle = customvoip
username = 478287xxxx@sip.broadvoice.com
authname = 478287xxxx
trunkname = Custom - Broadvoice 478287xxxx (ATL) DO NOT EDIT VIA WEB
callerid =
hasexten = no
hassip = yes
hasiax = no
registeriax = no
registersip = no
host = l1.sip.broadvoice.com
dialformat = ${EXTEN:1}
context = DID_trunk_1 ; default
group =
insecure = very
fromuser = 478287xxxx
fromdomain = sip.broadvoice.com
qualify = yes
pedantic = no
[601]
fullname = My Name (Office Sipura)
secret = xxxxx
email =
cid_number = 478287xxxx
zapchan =
context = outgoing-1
hasvoicemail = yes
hasdirectory = no
hassip = yes
hasiax = no
hasmanager = no
callwaiting = yes
threewaycalling = yes
mailbox = 601
hasagent = no
group =
host = dynamic
nat = no
; -----------------------------------------------------------
[[[extensions.conf]]]
; -----------------------------------------------------------
[default]
exten => 8500,1,VoicemailMain
exten => 8500,n,Hangup
include => voicemenu-custom-1
exten => 600,1,Goto(voicemenu-custom-1|s|1)
[voicemenu-custom-1]
comment = mainmenu
exten = s,1,Answer
exten = s,2,Background(thank-you-for-calling)
exten = s,3,Background(if-u-know-ext-dial)
exten = s,4,Background(otherwise)
exten = s,5,Background(pls-hold-while-try)
exten = s,6,Background(to-reach-operator)
include = default
alias_exten = 600
[numberplan-custom-1]
plancomment = DialPlan1
include = default
exten = _91700XXXXXXX.,1,Macro(trunkdial,${}/${EXTEN:1})
comment = _91700XXXXXXX.,1,IAXTEL,standard
exten = _9256XXXXXXX.,1,Macro(trunkdial,${}/${EXTEN:4})
comment = _9256XXXXXXX.,1,Local,standard
exten = _9011XXXXXXX.,1,Macro(trunkdial,${}/${EXTEN:1})
comment = _9011XXXXXXX.,1,International,standard
exten = _911.,1,Macro(trunkdial,${trunk_1}/${EXTEN:0})
comment = _911.,1,911,standard
exten = _9XXXXXXX.,1,Macro(trunkdial,${trunk_1}/${EXTEN:1})
exten = _91XXXXXXXXXX.,1,Macro(trunkdial,${trunk_1}/${EXTEN:1})
comment = _91XXXXXXXXXX.,1,Longdistance,standard
[outgoing-1]
exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@trunk_1,30,r)
exten => _9NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@trunk_1,30,r)
exten => _9NXXXXXX,3,Dial(SIP/${EXTEN:1}@trunk_1,30,r)
exten => _1NXXNXXXXXX,4,Dial(SIP/${EXTEN}@trunk_1,30,r)
exten => _NXXNXXXXXX,5,Dial(SIP/${EXTEN}@trunk_1,30,r)
exten => _NXXXXXX,6,Dial(SIP/${EXTEN}@trunk_1,30,r)
[asterisk_guitools]
exten = executecommand,1,System(${command})
exten = executecommand,n,Hangup()
exten = record_vmenu,1,Answer
exten = record_vmenu,n,Playback(vm-intro)
exten = record_vmenu,n,Record(${var1})
exten = record_vmenu,n,Playback(vm-saved)
exten = record_vmenu,n,Playback(vm-goodbye)
exten = record_vmenu,n,Hangup
exten = play_file,1,Answer
exten = play_file,n,Playback(${var1})
exten = play_file,n,Hangup
hasbeensetup = N
[DID_trunk_1]
include = default
;exten = _X.,1,Goto(default|600|1)
exten => 478287xxxx,1,Goto(default|600|1)
[DID_trunk_2]
include = default
;exten = _X.,1,Goto(default|600|1)
exten => 478287yyyy,3,Goto(default|600|1)
[DID_trunk_3]
include = default
;exten = _X.,1,Goto(default|600|1)
exten => 404795zzzz,1,Goto(default|600|1)
; -----------------------------------------------------------
[[[DEBUG OUTPUT]]]
; -----------------------------------------------------------
— (16 headers 20 lines) —
Sending to 192.168.0.74 : 5060 (no NAT)
Using INVITE request as basis request - e2ff6526-3415ed48@192.168.0.74
Found peer ‘601’
[REMOVED AUDIO DEBUG]
Found description format PCMU for ID 0
Found description format G726-32 for ID 2
Found description format G723 for ID 4
Found description format PCMA for ID 8
Found description format G729a for ID 18
Found description format G726-40 for ID 96
Found description format G726-24 for ID 97
Found description format G726-16 for ID 98
Found description format NSE for ID 100
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.74:16420
Looking for 9390[last 4 digits] in outgoing-1 (domain 192.168.0.76)
<— Reliably Transmitting (no NAT) to 192.168.0.74:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-abd7d49c;received=192.168.0.74
From: My Name (Sipura 1L2) sip:601@192.168.0.76;tag=cb7b338effd23200o1
To: <sip:9390[last 4 digits]@192.168.0.76>;tag=as42b5e7f8
Call-ID: e2ff6526-3415ed48@192.168.0.74
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0