Can't do any calls ...

inforemp

Joined
Sep 8, 2009
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#1
You can answer either spanish or english.

I've been trying with every post and every manual, but i can't manage to get it to work,
I can't call for example, if I want to dial 100000 (my voip internal numbers), from my extension, i always get "all lines busy"
Same with incoming calls, if I dial my most likely "miss-configurated" extension, wich is 100123
from the outside, it never gets in

I'll paste everything i have until now, hoping someone will help me...




Open ATA Linksys PAP2 ports: 5060 = Line 1 5061 Line


sip.conf

externip=cuenxxxx.ath.xx
localnet=192.168.1.0/255.255.255.0

sip_nat.conf

nat=yes
externip=cuentxxx.ath.cx
localnet=192.168.1.0/255.255.255.0
externrefresh=5




trunk:

username= ? (extension user, or voip?)
type=peer
secret=xxxxx
insecure=very
host= ? (router ip, elastix, or my voip?)
dtmfmode=rfc2833
disallow=all
allow=alaw&ulaw&gsm
canredirect=no
canreinvite=no

user details

canreinvite=no
context=from-trunk
fromuser= ? (extension or voip?)
qualify=no
secret=xxxx
type=user
username=xxxx


Register String


voipuser:voippassword@200.69.159.33 (ip voip) no?


outbound routes:

dial pattern: 8|x.

trunk sequence: trunk


my sip debug:
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/3000-09f58aa8", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/3000-09f58aa8", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/3000-09f58aa8", "SIP/citarella/100000|300|") in new stack
Audio is at 192.168.1.100 port 13526
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.50:5060:
INVITE sip:100000@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed;rport
From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
To: <sip:100000@192.168.1.50>
Contact: <sip:1003000@192.168.1.100>
Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
CSeq: 102 INVITE
User-Agent: Elastix
Max-Forwards: 70
Date: Wed, 16 Sep 2009 12:24:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 3125 3125 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 13526 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv

---
-- Called citarella/100000
cuentaip*CLI>
<--- SIP read from 192.168.1.50:5060 --->
SIP/2.0 404 Not Found
To: <sip:100000@192.168.1.50>;tag=b81f97b570e4f770i0
From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed
Server: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.50:5060:
ACK sip:100000@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed;rport
From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
To: <sip:100000@192.168.1.50>;tag=b81f97b570e4f770i0
Contact: <sip:1003000@192.168.1.100>
Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
CSeq: 102 ACK
User-Agent: Elastix
Max-Forwards: 70
Content-Length: 0


---
-- SIP/citarella-09f5fa88 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/3000-09f58aa8", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/3000-09f58aa8", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/3000-09f58aa8", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [8100000@from-internal:5] Macro("SIP/3000-09f58aa8", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/3000-09f58aa8", "all-circuits-busy-now|noanswer") in new stack
Audio is at 192.168.1.100 port 11100
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
cuentaip*CLI>
<--- Transmitting (NAT) to 192.168.1.50:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
To: <sip:8100000@192.168.1.100>;tag=as6734574c
Call-ID: c4612e0e-bb4b4c81@192.168.1.50
CSeq: 102 INVITE
User-Agent: Elastix
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:8100000@192.168.1.100>
Content-Type: application/sdp
Content-Length: 355

v=0
o=root 3125 3125 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
t=0 0
m=audio 11100 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv

<------------>
-- <SIP/3000-09f58aa8> Playing 'all-circuits-busy-now' (language 'es')
Really destroying SIP dialog '21483b1906c22465184755a9611a97c7@192.168.1.100' Method: INVITE
-- Executing [s@macro-outisbusy:2] Playback("SIP/3000-09f58aa8", "pls-try-call-later|noanswer") in new stack
-- <SIP/3000-09f58aa8> Playing 'pls-try-call-later' (language 'es')
cuentaip*CLI>
<--- SIP read from 192.168.1.50:5060 --->
CANCEL sip:8100000@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61
From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
To: <sip:8100000@192.168.1.100>
Call-ID: c4612e0e-bb4b4c81@192.168.1.50
CSeq: 102 CANCEL
Max-Forwards: 70
Proxy-Authorization: Digest username="3000",realm="asterisk",nonce="65ea3d5a",uri="sip:8100000@192.168.1.100",algorithm=MD5,response="4afdab4c474409369b72ce7ae707f132"
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.50 : 5060 (NAT)

<--- Reliably Transmitting (NAT) to 192.168.1.50:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
To: <sip:8100000@192.168.1.100>;tag=as6734574c
Call-ID: c4612e0e-bb4b4c81@192.168.1.50
CSeq: 102 INVITE
User-Agent: Elastix
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 192.168.1.50:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
To: <sip:8100000@192.168.1.100>;tag=as6734574c
Call-ID: c4612e0e-bb4b4c81@192.168.1.50
CSeq: 102 CANCEL
User-Agent: Elastix
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/3000-09f58aa8' in macro 'outisbusy'
== Spawn extension (from-internal, 8100000, 5) exited non-zero on 'SIP/3000-09f58aa8'
-- Executing [h@from-internal:1] Macro("SIP/3000-09f58aa8", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/3000-09f58aa8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/3000-09f58aa8", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/3000-09f58aa8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/3000-09f58aa8", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3000-09f58aa8' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3000-09f58aa8'
cuentaip*CLI>
<--- SIP read from 192.168.1.50:5060 --->
ACK sip:8100000@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61
From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
To: <sip:8100000@192.168.1.100>;tag=as6734574c
Call-ID: c4612e0e-bb4b4c81@192.168.1.50
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="3000",realm="asterisk",nonce="65ea3d5a",uri="sip:8100000@192.168.1.100",algorithm=MD5,response="fc9351980db59df69276c8325bb1ebff"
Contact: Linea 1 <sip:3000@192.168.1.50:5060>
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog 'c4612e0e-bb4b4c81@192.168.1.50' Method: ACK
 

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