Asterisk Doesn't see the OKs?

Hi all!

I have defined a SIP trunk with a local provider. It’s work fine but only for the incoming calls… For the outgoing nothing happens… after a while (few minutes) it say: “all line are busy… etc etc” ¿?

The pict shows that Asterisk doesn’t see the OKs… How can I fix that?

Something is missing in my configuration trunk?

host=190.xxx.xxx.165
type=peer
insecure=very
allow=all
nat=no
dtmfmode=rfc2833
qualify=no
canreinvite=no
trunk=yes

Thanks!

Any Idea?

Thanks in advance.

Hi , what is the server IP address ? is it correct that there is no nat ?

also what does the dial string look like ? and also the sip debug

Ian

Please provide:

  1. the exact version of Asterisk.
  2. sip set debug output, as text (not an image).

We’ll need to see your bit of extensions.conf that calls the sip provider please.

To see detailed traffic either turn on sip debugging, or use ngrep. Does your provider require any authentication parameters such as the username, fromuser, or secret? Or does it accept all calls from your IP?

Is allow=all valid? You should probably be specifying the codecs you can handle there.

disallow=all
allow=ulaw,gsm,etc…

insecure=very is a 1.2 directive and has been changed to insecure=port,invite but that shouldn’t affect outbound.

THANKS all for helping me!

The server IP address is fine, there is no NAT.

Extensions.conf
(There is NO trouble for here… I already defined another trunks and I did use the same configuration)


[outrt-003-IPLAN]
include => outrt-003-IPLAN-custom
exten => _0114xxxxxxx,1,Macro(user-callerid,SKIPTTL,)
exten => _0114xxxxxxx,n,Set(_NODEST=)
exten => _0114xxxxxxx,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _0114xxxxxxx,n,Macro(dialout-trunk,3,${EXTEN},)
exten => _0114xxxxxxx,n,Macro(outisbusy,)


Asterisk Version:
Asterisk 1.4.22, Copyright © 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com

Sip set debug:

<--- SIP read from 200.xxx.xxx.85:62216 --->
INVITE sip:01143282800@200.xxx.xxx.87 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.85:62216;branch=z9hG4bK-d8754z-ea4af70f73383d66-1---d8754z-
Max-Forwards: 70
Contact: <sip:6000@200.xxx.xxx.85:62216>
To: "01143282800"<sip:01143282800@200.xxx.xxx.87>
From: <sip:6000@200.xxx.xxx.87>;tag=7d71501f
Call-ID: ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 371

v=0
o=- 4 2 IN IP4 200.xxx.xxx.85
s=CounterPath X-Lite 3.0
c=IN IP4 200.xxx.xxx.85
t=0 0
m=audio 6642 RTP/AVP 107 119 100 106 0 105 98 8 3 101
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 15 lines) ---
Sending to 200.xxx.xxx.85 : 62216 (no NAT)
Using INVITE request as basis request - ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.

<--- Reliably Transmitting (no NAT) to 200.xxx.xxx.85:62216 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 200.xxx.xxx.85:62216;branch=z9hG4bK-d8754z-ea4af70f73383d66-1---d8754z-;received=200.xxx.xxx.85
From: <sip:6000@200.xxx.xxx.87>;tag=7d71501f
To: "01143282800"<sip:01143282800@200.xxx.xxx.87>;tag=as20292f29
Call-ID: ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a5590c3"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.' in 32000 ms (Method: INVITE)
Found user '6000'

<--- SIP read from 200.xxx.xxx.85:62216 --->
ACK sip:01143282800@200.xxx.xxx.87 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.85:62216;branch=z9hG4bK-d8754z-ea4af70f73383d66-1---d8754z-
To: "01143282800"<sip:01143282800@200.xxx.xxx.87>;tag=as20292f29
From: <sip:6000@200.xxx.xxx.87>;tag=7d71501f
Call-ID: ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
asterisco*CLI> 
<--- SIP read from 200.xxx.xxx.85:62216 --->
INVITE sip:01143282800@200.xxx.xxx.87 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.85:62216;branch=z9hG4bK-d8754z-3207d64bff7cd420-1---d8754z-
Max-Forwards: 70
Contact: <sip:6000@200.xxx.xxx.85:62216>
To: "01143282800"<sip:01143282800@200.xxx.xxx.87>
From: <sip:6000@200.xxx.xxx.87>;tag=7d71501f
Call-ID: ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="6000",realm="asterisk",nonce="0a5590c3",uri="sip:01143282800@200.xxx.xxx.87",response="8f0cf54f63e1c9e1ab14410f46739d89",algorithm=MD5
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 371

v=0
o=- 4 2 IN IP4 200.xxx.xxx.85
s=CounterPath X-Lite 3.0
c=IN IP4 200.xxx.xxx.85
t=0 0
m=audio 6642 RTP/AVP 107 119 100 106 0 105 98 8 3 101
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (13 headers 15 lines) ---
Sending to 200.xxx.xxx.85 : 62216 (no NAT)
Using INVITE request as basis request - ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.
Found user '6000'
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 200.xxx.xxx.85:6642
Found unknown media description format BV32 for ID 107
Found unknown media description format BV32-FEC for ID 119
Found audio description format SPEEX for ID 100
Found unknown media description format SPEEX-FEC for ID 106
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|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 200.xxx.xxx.85:6642
Looking for 01143282800 in from-internal (domain 200.xxx.xxx.87)
list_route: hop: <sip:6000@200.xxx.xxx.85:62216>

<--- Transmitting (no NAT) to 200.xxx.xxx.85:62216 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 200.xxx.xxx.85:62216;branch=z9hG4bK-d8754z-3207d64bff7cd420-1---d8754z-;received=200.xxx.xxx.85
From: <sip:6000@200.xxx.xxx.87>;tag=7d71501f
To: "01143282800"<sip:01143282800@200.xxx.xxx.87>
Call-ID: ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:01143282800@200.xxx.xxx.87>
Content-Length: 0


<------------>
Audio is at 200.xxx.xxx.87 port 3910
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 (no NAT) to 190.xxx.xxx.165:5060:
INVITE sip:01143282800@190.xxx.xxx.165 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.87:5060;branch=z9hG4bK328f296c;rport
From: "X-Lite" <sip:6000@200.xxx.xxx.87>;tag=as7ea15fa5
To: <sip:01143282800@190.xxx.xxx.165>
Contact: <sip:6000@200.xxx.xxx.87>
Call-ID: 779e001f2cae659a05314061255843e4@200.xxx.xxx.87
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Nov 2008 15:21:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 10449 10449 IN IP4 200.xxx.xxx.87
s=session
c=IN IP4 200.xxx.xxx.87
t=0 0
m=audio 3910 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:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #1 (no NAT) to 190.xxx.xxx.165:5060:
INVITE sip:01143282800@190.xxx.xxx.165 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.87:5060;branch=z9hG4bK328f296c;rport
From: "X-Lite" <sip:6000@200.xxx.xxx.87>;tag=as7ea15fa5
To: <sip:01143282800@190.xxx.xxx.165>
Contact: <sip:6000@200.xxx.xxx.87>
Call-ID: 779e001f2cae659a05314061255843e4@200.xxx.xxx.87
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Nov 2008 15:21:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 10449 10449 IN IP4 200.xxx.xxx.87
s=session
c=IN IP4 200.xxx.xxx.87
t=0 0
m=audio 3910 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:off - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #2 (no NAT) to 190.xxx.xxx.165:5060:
INVITE sip:01143282800@190.xxx.xxx.165 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.87:5060;branch=z9hG4bK328f296c;rport
From: "X-Lite" <sip:6000@200.xxx.xxx.87>;tag=as7ea15fa5
To: <sip:01143282800@190.xxx.xxx.165>
Contact: <sip:6000@200.xxx.xxx.87>
Call-ID: 779e001f2cae659a05314061255843e4@200.xxx.xxx.87
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Nov 2008 15:21:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 10449 10449 IN IP4 200.xxx.xxx.87
s=session
c=IN IP4 200.xxx.xxx.87
t=0 0
m=audio 3910 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:off - - - -
a=ptime:20
a=sendrecv

---
asterisco*CLI> 
<--- SIP read from 200.xxx.xxx.85:62216 --->
CANCEL sip:01143282800@200.xxx.xxx.87 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.85:62216;branch=z9hG4bK-d8754z-3207d64bff7cd420-1---d8754z-
To: "01143282800"<sip:01143282800@200.xxx.xxx.87>
From: <sip:6000@200.xxx.xxx.87>;tag=7d71501f
Call-ID: ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.
CSeq: 2 CANCEL
Proxy-Authorization: Digest username="6000",realm="asterisk",nonce="0a5590c3",uri="sip:01143282800@200.xxx.xxx.87",response="f2025acef909cd5e8f2bebc02bfa6111",algorithm=MD5
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Sending to 200.xxx.xxx.85 : 62216 (no NAT)
asterisco*CLI> 
<--- Reliably Transmitting (no NAT) to 200.xxx.xxx.85:62216 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 200.xxx.xxx.85:62216;branch=z9hG4bK-d8754z-3207d64bff7cd420-1---d8754z-;received=200.xxx.xxx.85
From: <sip:6000@200.xxx.xxx.87>;tag=7d71501f
To: "01143282800"<sip:01143282800@200.xxx.xxx.87>;tag=as2c1aed1c
Call-ID: ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
asterisco*CLI> 
<--- Transmitting (no NAT) to 200.xxx.xxx.85:62216 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 200.xxx.xxx.85:62216;branch=z9hG4bK-d8754z-3207d64bff7cd420-1---d8754z-;received=200.xxx.xxx.85
From: <sip:6000@200.xxx.xxx.87>;tag=7d71501f
To: "01143282800"<sip:01143282800@200.xxx.xxx.87>;tag=as2c1aed1c
Call-ID: ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:01143282800@200.xxx.xxx.87>
Content-Length: 0


<------------>
Reliably Transmitting (no NAT) to 190.xxx.xxx.165:5060:
CANCEL sip:01143282800@190.xxx.xxx.165 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.87:5060;branch=z9hG4bK328f296c;rport
From: "X-Lite" <sip:6000@200.xxx.xxx.87>;tag=as7ea15fa5
To: <sip:01143282800@190.xxx.xxx.165>
Call-ID: 779e001f2cae659a05314061255843e4@200.xxx.xxx.87
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '779e001f2cae659a05314061255843e4@200.xxx.xxx.87' in 32000 ms (Method: INVITE)
asterisco*CLI> 
<--- SIP read from 200.xxx.xxx.85:62216 --->
ACK sip:01143282800@200.xxx.xxx.87 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.85:62216;branch=z9hG4bK-d8754z-3207d64bff7cd420-1---d8754z-
To: "01143282800"<sip:01143282800@200.xxx.xxx.87>;tag=as2c1aed1c
From: <sip:6000@200.xxx.xxx.87>;tag=7d71501f
Call-ID: ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'ZTgwZjBmNTQzY2U4MzY1MjliNzU3NGJmMTYzMjYzYTY.' Method: ACK
Retransmitting #1 (no NAT) to 190.xxx.xxx.165:5060:
CANCEL sip:01143282800@190.xxx.xxx.165 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.87:5060;branch=z9hG4bK328f296c;rport
From: "X-Lite" <sip:6000@200.xxx.xxx.87>;tag=as7ea15fa5
To: <sip:01143282800@190.xxx.xxx.165>
Call-ID: 779e001f2cae659a05314061255843e4@200.xxx.xxx.87
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Retransmitting #2 (no NAT) to 190.xxx.xxx.165:5060:
CANCEL sip:01143282800@190.xxx.xxx.165 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.87:5060;branch=z9hG4bK328f296c;rport
From: "X-Lite" <sip:6000@200.xxx.xxx.87>;tag=as7ea15fa5
To: <sip:01143282800@190.xxx.xxx.165>
Call-ID: 779e001f2cae659a05314061255843e4@200.xxx.xxx.87
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Done. I changed that.

[quote]
Or does it accept all calls from your IP?
[/quote]Yes, that’s correct.

Please, remember that incoming calls work fine…

thanks!

The Acks aren’t reaching Asterisk. Look for a firewall problem. Note that the working case is using non-standard port numbers.

I imagine the person who asked for extensions.conf wanted the relevant parts of the macros, as well.

And that macro does…

EDIT : David beat me to it :smile:

EDIT : EDIT : Maybe changing to this in sip.conf as well:
disallow = all
allow = ulaw
allow = alaw

From the trace, I see a phone calling through asterisk to an outside line, but everything going out to the provider is not being responded to.
Traffic between
200.xxx.xxx.85 and 200.xxx.xxx.87 are working as intended.

Traffic when being sent out to 190.xxx.xxx.165 are not being responded to.

This is the outbound invite:

[code]INVITE sip:01143282800@190.xxx.xxx.165 SIP/2.0
Via: SIP/2.0/UDP 200.xxx.xxx.87:5060;branch=z9hG4bK328f296c;rport
From: “X-Lite” sip:6000@200.xxx.xxx.87;tag=as7ea15fa5
To: sip:01143282800@190.xxx.xxx.165
Contact: sip:6000@200.xxx.xxx.87
Call-ID: 779e001f2cae659a05314061255843e4@200.xxx.xxx.87
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 17 Nov 2008 15:21:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 10449 10449 IN IP4 200.xxx.xxx.87
s=session
c=IN IP4 200.xxx.xxx.87
t=0 0
m=audio 3910 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:off - - - -
a=ptime:20
a=sendrecv
[/code]
Which is followed up by a retry since it didn’t get a response:

[code]

Retransmitting #1 (no NAT) to 190.xxx.xxx.165:5060: [/code]

Are there any firewalls blocking the inbound SIP response from the provider? If not, I’d contact your provider to help troubleshoot further.

Edit: also try adding canreinvite=no to the sip peer.

There is no firewall, I did opened all ports.

[quote]Maybe changing to this in sip.conf as well:
disallow = all
allow = ulaw
allow = alaw[/quote]
Done. There is no change…

Yes! so, It could be a Extensions.conf issues? I think not…

Extensions.conf

[macro-dialout-dundi]
include => macro-dialout-dundi-custom
exten => s,1,Set(DIAL_TRUNK=${ARG1})
exten => s,n,ExecIf($[$["${ARG3}" != ""] & $["${DB(AMPUSER/${AMPUSER}/pinless)}" != "NOPASSWD"]],Authenticate,${ARG3})
exten => s,n,GotoIf($["x${OUTDISABLE_${DIAL_TRUNK}}" = "xon"]?disabletrunk,1)
exten => s,n,Set(DIAL_NUMBER=${ARG2})
exten => s,n,Set(DIAL_TRUNK_OPTIONS=${DIAL_OPTIONS})
exten => s,n,Set(GROUP()=OUT_${DIAL_TRUNK})
exten => s,n,GotoIf($["${OUTMAXCHANS_${DIAL_TRUNK}}foo" = "foo"]?nomax)
exten => s,n,GotoIf($[ ${GROUP_COUNT(OUT_${DIAL_TRUNK})} > ${OUTMAXCHANS_${DIAL_TRUNK}} ]?chanfull)
exten => s,n(nomax),GotoIf($["${INTRACOMPANYROUTE}" = "YES"]?skipoutcid)
exten => s,n,Set(DIAL_TRUNK_OPTIONS=${TRUNK_OPTIONS})
exten => s,n,Macro(outbound-callerid,${DIAL_TRUNK})
exten => s,n(skipoutcid),AGI(fixlocalprefix)
exten => s,n,Set(OUTNUM=${OUTPREFIX_${DIAL_TRUNK}}${DIAL_NUMBER})
exten => s,n,GotoIf($[$["${MOHCLASS}" = "default"] | $["foo${MOHCLASS}" = "foo"]]?gocall)
exten => s,n,Set(DIAL_TRUNK_OPTIONS=M(setmusic^${MOHCLASS})${DIAL_TRUNK_OPTIONS})
exten => s,n(gocall),Macro(dialout-dundi-predial-hook,)
exten => s,n,GotoIf($["${PREDIAL_HOOK_RET}" = "BYPASS"]?bypass,1)
exten => s,n,GotoIf($["${custom}" = "AMP"]?customtrunk)
exten => s,n,Macro(dundi-${DIAL_TRUNK},${OUTNUM})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s,n(chanfull),Noop(max channels used up)
exten => s-BUSY,1,Noop(Dial failed due to trunk reporting BUSY - giving up)
exten => s-BUSY,n,Playtones(busy)
exten => s-BUSY,n,Busy(20)
exten => s-NOANSWER,1,Noop(Dial failed due to trunk reporting NOANSWER - giving up)
exten => s-NOANSWER,n,Playtones(congestion)
exten => s-NOANSWER,n,Congestion(20)
exten => s-CANCEL,1,Noop(Dial failed due to trunk reporting CANCEL - giving up)
exten => s-CANCEL,n,Playtones(congestion)
exten => s-CANCEL,n,Congestion(20)
exten => _s-.,1,GotoIf($["x${OUTFAIL_${ARG1}}" = "x"]?noreport)
exten => _s-.,n,AGI(${OUTFAIL_${ARG1}})
exten => _s-.,n(noreport),Noop(TRUNK Dial failed due to ${DIALSTATUS} - failing through to other trunks)
exten => s,n(skipoutcid),AGI(fixlocalprefix)
exten => s,n,Set(OUTNUM=${OUTPREFIX_${DIAL_TRUNK}}${DIAL_NUMBER})
exten => s,n,GotoIf($[$["${MOHCLASS}" = "default"] | $["foo${MOHCLASS}" = "foo"]]?gocall)
exten => s,n,Set(DIAL_TRUNK_OPTIONS=M(setmusic^${MOHCLASS})${DIAL_TRUNK_OPTIONS})
exten => s,n(gocall),Macro(dialout-dundi-predial-hook,)
exten => s,n,GotoIf($["${PREDIAL_HOOK_RET}" = "BYPASS"]?bypass,1)
exten => s,n,GotoIf($["${custom}" = "AMP"]?customtrunk)
exten => s,n,Macro(dundi-${DIAL_TRUNK},${OUTNUM})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s,n(chanfull),Noop(max channels used up)
exten => s-BUSY,1,Noop(Dial failed due to trunk reporting BUSY - giving up)
exten => s-BUSY,n,Playtones(busy)
exten => s-BUSY,n,Busy(20)
exten => s-NOANSWER,1,Noop(Dial failed due to trunk reporting NOANSWER - giving up)
exten => s-NOANSWER,n,Playtones(congestion)
exten => s-NOANSWER,n,Congestion(20)
exten => s-CANCEL,1,Noop(Dial failed due to trunk reporting CANCEL - giving up)
exten => s-CANCEL,n,Playtones(congestion)
exten => s-CANCEL,n,Congestion(20)
exten => _s-.,1,GotoIf($["x${OUTFAIL_${ARG1}}" = "x"]?noreport)
exten => _s-.,n,AGI(${OUTFAIL_${ARG1}})
exten => _s-.,n(noreport),Noop(TRUNK Dial failed due to ${DIALSTATUS} - failing through to other trunks)
exten => disabletrunk,1,Noop(TRUNK: ${OUT_${DIAL_TRUNK}} DISABLED - falling through to next trunk)
exten => bypass,1,Noop(TRUNK: ${OUT_${DIAL_TRUNK}} BYPASSING because dialout-dundi-predial-hook)
exten => h,1,Macro(hangupcall,)

The complete file:
turboupload.com/files/get/tj … ional.conf

Thanks ALL!!!

Nevermind! That macro-to-a-macro-to-a… dialplan is giving me headache. I think Dave with his ITSP problem sounds more like it.

Good luck :smile:.

mmm you think?

I believe it’s exactly otherwise… if you look the ethereal capture, is the Asterisk who doesn’t respond the OK from provider.

Ah! right now, the calls are reaching the destiny, but the asterisk still showing “all line are busy…”

¿¿¿???

mmm you think?

I believe it’s exactly otherwise… if you look the ethereal capture, is the Asterisk who doesn’t respond the OK from provider.

Ah! right now, the calls are reaching the destiny, but the asterisk still showing “all line are busy…”

¿¿¿???[/quote]
Ethereal may be seeing it, but asterisk is not. At least not in the sip debugging you pasted. Check your local firewall.

Ok but, if this is a firewall trouble, when I make the inbound calls, the same thing should to happen… or not?

Disable your firewall temporarily to rule it out as the issue.

The multiple 200 OK’s from your provider as seen in the ethereal trace mean that the provider is not seeing an ACK to their 200 OK. Asterisk is not sending an ACK back since it doesn’t see the 200 OK in the first place.

Ok but, if this is a firewall trouble, when I make the inbound calls, the same thing should to happen… or not?[/quote]

Why are we seeing internal address space now? 192.168.10.119

[quote]Why are we seeing internal address space now? 192.168.10.119[/quote]Ignore that, my mistake…

[quote]Disable your firewall temporarily to rule it out as the issue. [/quote]Done. I already made that… but same thing…

The provider is ok. They have the same configuration for all clients… so…

I’m running out of ideas :smile: Does the server have multiple IP’s? Are you binding to a particular IP for SIP? bindaddr in sip.conf.

As I already said, the one that is working is not using standard SIP port numbers, but the one that isn’t is using standard port numbers, at least for the remote side.