Asterisk problems with outgoing calls

Asterisk Asterisk Support
#1

Hi, I am newbie in Asterisk and I have a problem with outgoing call’s from my asterisk to my voip provider (freeconet)…

Softwear:
CentOS 6.4
Asterisk 1.8.24
FreePBX 2.9
X-Lite (soft phone)

Topology:

My provider (freeconet) - Asterisk (on CentOS) - X-lite

I can call on my voip number with succes, unfortunately when I tried to call from X-lite my call is automatically terminated.

Statement on ma soft phone:

Statement in terminal from Asterisk :

[quote]==Using SIP RTP TOS bits 184
==Using SIP RTP CoS mark 5[/quote]

Logfiles from my freePBX when i call to voip phone and from voip phone:

[quote][Nov 29 08:18:55] VERBOSE[3647] asterisk.c: Asterisk Ready.
[Nov 29 08:19:03] VERBOSE[3649] asterisk.c: – Remote UNIX connection
[Nov 29 08:19:08] VERBOSE[3672] netsock2.c: == Using SIP RTP TOS bits 184
[Nov 29 08:19:08] VERBOSE[3672] netsock2.c: == Using SIP RTP CoS mark 5
[Nov 29 08:19:08] VERBOSE[3691] pbx.c: – Executing [przychodzace1@freeconet:1] Set(“SIP/freeconet-in1-00000000”, “TOHDR=”) in new stack
[Nov 29 08:19:08] VERBOSE[3691] pbx.c: – Executing [przychodzace1@freeconet:2] GotoIf(“SIP/freeconet-in1-00000000”, “1?wew,133,1”) in new stack
[Nov 29 08:19:08] VERBOSE[3691] pbx.c: – Goto (wew,133,1)
[Nov 29 08:19:08] VERBOSE[3691] pbx.c: – Executing [133@wew:1] Dial(“SIP/freeconet-in1-00000000”, “SIP/3784433,45,Tt”) in new stack
[Nov 29 08:19:08] VERBOSE[3691] netsock2.c: == Using SIP RTP TOS bits 184
[Nov 29 08:19:08] VERBOSE[3691] netsock2.c: == Using SIP RTP CoS mark 5
[Nov 29 08:19:08] VERBOSE[3691] app_dial.c: – Called SIP/3784433
[Nov 29 08:19:09] VERBOSE[3691] app_dial.c: – SIP/3784433-00000001 is ringing
[Nov 29 08:19:12] VERBOSE[3691] pbx.c: == Spawn extension (wew, 133, 1) exited non-zero on ‘SIP/freeconet-in1-00000000’
[Nov 29 08:21:39] VERBOSE[3672] netsock2.c: == Using SIP RTP TOS bits 184
[Nov 29 08:21:39] VERBOSE[3672] netsock2.c: == Using SIP RTP CoS mark 5[/quote]

My sip configuration:

[quote] context=default
bindport=5060

srvlookup=yes
defaultexpiry=60
allowguest=no
dtmfmode=rfc2833
nat=yes

defaultexpiry=60
localnet=192.168.2.1/255.255.255.0
externip=77.252.253.170

register => login:pass@sip.freeconet.pl/przychodzace1

[freeconet-out]
type=peer

username=login
secret=pass
fromdomain=sip.freeconet.pl

context=freeconet
host=sip.freeconet.pl
port=5060
outboundproxy=sip.freeconet.pl
outboundproxyport=5060
insecure=no

[freeconet-in1]
type=peer
fromdomain=sip.freeconet.pl
port=5060
context=freeconet [/quote]

My extensions file:

[quote] [defaul]
exten => _.,1,Hangup

[freeconet]

exten =>przychodzace1,1,Set(TOHDR=${SIP_HEADER(To)})

exten =>przychodzace1,2,GotoIf($["${REGEX("223784433" ${TOHDR})}" = "1"]?wew,133,1)

[freeconet1]

include => wew

exten => t,1,Hangup

exten => h,1,Hangup

exten => _XXX.,1,SIPAddHeader(X-Fid: ${SIPCALLID})
exten => _XXX.,2,Set(CALLERID(num)=48223784433)
exten => _XXX.,3,Dial(SIP/${EXTEN}@freeconet-out)

[wew]

exten => 133,1,Dial(SIP/3784433,45,Tt)[/quote]

sip show peers:

[quote]Name/username Host Dyn Forcerport ACL Port Status
3784433/3784433 192.168.2.104 D N 5060 Unmonitored
freeconet-in1 213.218.116.65 N 5060 Unmonitored
freeconet-in2 213.218.116.66 N 5060 Unmonitored
freeconet-out/lukaszbiele 213.218.116.66 N 5060 Unmonitored
4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 4 online, 0 offline] [/quote]

sip show registry:

[quote]Host dnsmgr Username Refresh State Reg.Time
sip.freeconet.pl:5060 N lukaszbielec 45 Registered Thu, 28 Nov 2013 12:33:59
1 SIP registrations.[/quote]

sip set debug peer 3784433 and call from voip phone:

Support from my voip provider don’t want help…
In connection with this I ask you for help…
Do you have solution or any suggestions?

#2

SIP/2.0 484 Address Incomplete

They’ve sent a 3 digit number (900). Your dialplan expects a number of 4 or more digits (. means 1 or more digits).

Note that the RTP COS messages are not errors.

#3

Unfortunately, still I can’t make outgoing calls…

When i call to other number (1001213):

Log’s form terminal:

[quote]== Using SIP RTP CoS mark 5
– Executing [1001213@freeconet1:1] SIPAddHeader(“SIP/3784433-00000004”, “X-Fid: Zjc0ODFkZjQ1MzkyYTEyMGRlZDBhMThjYTUxZGQ1NjU”) in new stack
– Executing [1001213@freeconet1:2] Set(“SIP/3784433-00000004”, “CALLERID(num)=48223784433”) in new stack
– Executing [1001213@freeconet1:3] Dial(“SIP/3784433-00000004”, “SIP/1001213@freeconet-out”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/1001213@freeconet-out
– SIP/freeconet-out-00000005 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/3784433-00000004’ status is ‘CONGESTION’
– Executing [h@freeconet1:1] Hangup(“SIP/3784433-00000004”, “”) in new stack
== Spawn extension (freeconet1, h, 1) exited non-zero on ‘SIP/3784433-00000004’[/quote]

sip set debug peer 3784433:

[quote]<— SIP read from UDP:192.168.2.103:5060 —>
INVITE sip:1001213@192.168.2.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-6d988a4330254654-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:3784433@192.168.2.103:5060
To: sip:1001213@192.168.2.101
From: sip:3784433@192.168.2.101;tag=a76b7847
Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5.5 stamp 71236
Content-Length: 306

v=0
o=- 13030640618233466 1 IN IP4 192.168.2.103
s=X-Lite 4 release 4.5.5 stamp 71236
c=IN IP4 192.168.2.103
t=0 0
m=audio 50394 RTP/AVP 125 9 8 0 100 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 12 lines) —
Sending to 192.168.2.103:5060 (NAT)
Using INVITE request as basis request - MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA
Found peer ‘3784433’ for ‘3784433’ from 192.168.2.103:5060

<— Reliably Transmitting (NAT) to 192.168.2.103:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-6d988a4330254654-1—d8754z-;received=192.168.2.103;rport=5060
From: sip:3784433@192.168.2.101;tag=a76b7847
To: sip:1001213@192.168.2.101;tag=as2ee36d25
Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA
CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.24.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="03e7d85d"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.2.103:5060 —>
ACK sip:1001213@192.168.2.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-6d988a4330254654-1—d8754z-;rport
Max-Forwards: 70
To: sip:1001213@192.168.2.101;tag=as2ee36d25
From: sip:3784433@192.168.2.101;tag=a76b7847
Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.2.103:5060 —>
INVITE sip:1001213@192.168.2.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-23a3746f5069276f-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:3784433@192.168.2.103:5060
To: sip:1001213@192.168.2.101
From: sip:3784433@192.168.2.101;tag=a76b7847
Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5.5 stamp 71236
Authorization: Digest username=“3784433”,realm=“asterisk”,nonce=“03e7d85d”,uri="sip:1001213@192.168.2.101",response=“aa2d7c2da92f0a718ebc1153926b3d35”,algorithm=MD5
Content-Length: 306

v=0
o=- 13030640618233466 1 IN IP4 192.168.2.103
s=X-Lite 4 release 4.5.5 stamp 71236
c=IN IP4 192.168.2.103
t=0 0
m=audio 50394 RTP/AVP 125 9 8 0 100 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 12 lines) —
Sending to 192.168.2.103:5060 (NAT)
Using INVITE request as basis request - MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA
Found peer ‘3784433’ for ‘3784433’ from 192.168.2.103:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 125
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Found unknown media description format opus for ID 125
Found audio description format speex for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x20000100c (ulaw|alaw|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x20000100c (ulaw|alaw|speex16|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.103:50394
Looking for 1001213 in freeconet1 (domain 192.168.2.101)
list_route: hop: sip:3784433@192.168.2.103:5060

<— Transmitting (NAT) to 192.168.2.103:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-23a3746f5069276f-1—d8754z-;received=192.168.2.103;rport=5060
From: sip:3784433@192.168.2.101;tag=a76b7847
To: sip:1001213@192.168.2.101
Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA
CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.24.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1001213@192.168.2.101:5060
Content-Length: 0

<------------>
– Executing [1001213@freeconet1:1] SIPAddHeader(“SIP/3784433-00000006”, “X-Fid: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA”) in new stack
– Executing [1001213@freeconet1:2] Set(“SIP/3784433-00000006”, “CALLERID(num)=48223784433”) in new stack
– Executing [1001213@freeconet1:3] Dial(“SIP/3784433-00000006”, “SIP/1001213@freeconet-out”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/1001213@freeconet-out
– SIP/freeconet-out-00000007 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/3784433-00000006’ status is ‘CONGESTION’

<— Reliably Transmitting (NAT) to 192.168.2.103:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-23a3746f5069276f-1—d8754z-;received=192.168.2.103;rport=5060
From: sip:3784433@192.168.2.101;tag=a76b7847
To: sip:1001213@192.168.2.101;tag=as1ee85282
Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA
CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.24.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0

<------------>
– Executing [h@freeconet1:1] Hangup(“SIP/3784433-00000006”, “”) in new stack
== Spawn extension (freeconet1, h, 1) exited non-zero on ‘SIP/3784433-00000006’

<— SIP read from UDP:192.168.2.103:5060 —>
ACK sip:1001213@192.168.2.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.103:5060;branch=z9hG4bK-d8754z-23a3746f5069276f-1—d8754z-;rport
Max-Forwards: 70
To: sip:1001213@192.168.2.101;tag=as1ee85282
From: sip:3784433@192.168.2.101;tag=a76b7847
Call-ID: MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA
CSeq: 2 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘MzMwNTE1MzM0ODhlYTMyNjI4MzZhNjczMTBiZWU1NDA’ Method: ACK
[/quote]

#4

The trace isn’t consistent with the dialplan you provided. It looks like it came from a dialplan with invalid substring syntax, but there is no substringing in the dialplan you provided.

My guess is you have ${undefinedvariable}:1:3 rather than ${EXTEN:1:3}

#5

This you mean - exten => _X.,3,Dial(SIP/${undefinedvariable}:1:3@freeconet-out) ?

if so, then i show this in terminal:

[quote] == Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [1001213@freeconet1:1] SIPAddHeader(“SIP/3784433-00000000”, “X-Fid: MDIwZDZmNGRiM2Q1NmU5OThiM2Y4OTRiYWQ1ZWYyZjI”) in new stack
– Executing [1001213@freeconet1:2] Set(“SIP/3784433-00000000”, “CALLERID(num)=48223784433”) in new stack
– Executing [1001213@freeconet1:3] Dial(“SIP/3784433-00000000”, “SIP/:1:3@freeconet-out”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/:1:3@freeconet-out
– SIP/freeconet-out-00000001 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/3784433-00000000’ status is ‘CONGESTION’
– Executing [h@freeconet1:1] Hangup(“SIP/3784433-00000000”, “”) in new stack
== Spawn extension (freeconet1, h, 1) exited non-zero on ‘SIP/3784433-00000000’
[/quote]

ps. I’m just starting their adventure with asterisk :smile: