Hangup on second incoming call

Hi,

when i’m receiving call from fxo port (dahdi/1) of my tdm410 board towards dahdi/4-1 and then i receive a call from a sip account directed to dahdi/4-2 there is an hangup.

I can’t understand what is happening.

Some suggestion?

Here is the cli output wit sip verbose enabled:

– Executing [1@Internal:1] Macro(“SIP/2-0000005d”, “normal,1,DAHDI/4”) in new stack
– Executing [s@macro-normal:1] Verbose(“SIP/2-0000005d”, “1,Incoming call from “2” <2>”) in new stack
Incoming call from “2” <2>
– Executing [s@macro-normal:2] Dial(“SIP/2-0000005d”, “DAHDI/4,40,KkTt”) in new stack
– Called 4I>
– DAHDI/4-2 is ringing
ServerCentos*CLI>
<— Transmitting (NAT) to 192.168.10.101:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.101:5060;branch=z9hG4bKRSTaskNAbi8IqVpT;received=192.168.10.101;rport=5060
From: “2” sip:2@192.168.10.99;tag=W9lVPFqaBfnhWa3H
To: “1” sip:1@192.168.10.99;tag=as41f0e1fa
Call-ID: BbtYYzScxSta0kPt@192.168.10.101
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:1@192.168.10.99
Content-Length: 0

<------------>
– CPE supports Call Waiting CallerID. Sending '2/2’
ServerCentos
CLI>
<— SIP read from UDP://192.168.10.98:5060 —>
REGISTER sip:192.168.10.99 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.98:5060;rport;branch=z9hG4bKbe63bb433f
From: “gsm box” sip:4003@192.168.10.99;tag=77f95f7b
To: “gsm box” sip:4003@192.168.10.99
Call-ID: 36b8687004d0ad21201653ff19e49ace@192.168.10.98
Contact: sip:4003@192.168.10.98:5060
CSeq: 6536 REGISTER
Max-Forwards: 70
Expires: 60
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Authorization: Digest
username=“4003”,realm=“sip.messagenet.it”,nonce=“60ca20e2”,response=“56a89edfaeccc44349ff1c2cb5f437e4”,uri=“sip:192.168.10.99”,algorithm=MD5
User-Agent: Mv-37x (904290)
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.10.98 : 5060 (NAT)

<— Transmitting (NAT) to 192.168.10.98:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.98:5060;branch=z9hG4bKbe63bb433f;received=192.168.10.98;rport=5060
From: “gsm box” sip:4003@192.168.10.99;tag=77f95f7b
To: “gsm box” sip:4003@192.168.10.99;tag=as43a86e79
Call-ID: 36b8687004d0ad21201653ff19e49ace@192.168.10.98
CSeq: 6536 REGISTER
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“sip.messagenet.it”, nonce="60a15673"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘36b8687004d0ad21201653ff19e49ace@192.168.10.98’ in 32000 ms (Method: REGISTER)
ServerCentos*CLI>
<— SIP read from UDP://192.168.10.98:5060 —>
REGISTER sip:192.168.10.99 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.98:5060;rport;branch=z9hG4bK1093759681
From: “gsm box” sip:4003@192.168.10.99;tag=77f95f7b
To: “gsm box” sip:4003@192.168.10.99
Call-ID: 36b8687004d0ad21201653ff19e49ace@192.168.10.98
Contact: sip:4003@192.168.10.98:5060
CSeq: 6537 REGISTER
Max-Forwards: 70
Expires: 60
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Authorization: Digest
username=“4003”,realm=“sip.messagenet.it”,nonce=“60a15673”,response=“accc0b1a97c269fc1ec29f29061648d4”,uri=“sip:192.168.10.99”,algorithm=MD5
User-Agent: Mv-37x (904290)
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.10.98 : 5060 (NAT)
ServerCentos*CLI>
<— Transmitting (NAT) to 192.168.10.98:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.98:5060;branch=z9hG4bK1093759681;received=192.168.10.98;rport=5060
From: “gsm box” sip:4003@192.168.10.99;tag=77f95f7b
To: “gsm box” sip:4003@192.168.10.99;tag=as43a86e79
Call-ID: 36b8687004d0ad21201653ff19e49ace@192.168.10.98
CSeq: 6537 REGISTER
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: sip:4003@192.168.10.98:5060;expires=60
Date: Mon, 08 Feb 2010 10:14:25 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘36b8687004d0ad21201653ff19e49ace@192.168.10.98’ in 32000 ms (Method: REGISTER)
ServerCentos*CLI>
<— SIP read from UDP://212.97.59.76:5061 —>

<------------->
– Channel 4 still has (callwait) call, ringing phone
– Hungup ‘DAHDI/4-1’
== Spawn extension (macro-normal, s, 2) exited non-zero on ‘DAHDI/1-1’ in macro ‘normal’
== Spawn extension (Incoming1, s, 2) exited non-zero on ‘DAHDI/1-1’
– Hungup ‘DAHDI/1-1’