Fix Spawn extension (from_trunk_GMSC, 206, 1) exited non-zero on 'SIP/trunk_GMSC-00000002'

quang151080 · November 15, 2017, 3:28pm

Dear,
I have connect SIP with telco and i have proplem:
Show sip pees is OK but when dail shortcode 206 then have proplem
– Executing [206@from_trunk_GMSC:1] Answer(“SIP/trunk_GMSC-00000002”, “”) in new stack
== Spawn extension (from_trunk_GMSC, 206, 1) exited non-zero on ‘SIP/trunk_GMSC-00000002’

This is detail debug sip:

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘4b7680a96b11af0214ccceaf4c45ed06@10.228.37.149:5060’ Method: OPTIONS
Reliably Transmitting (no NAT) to 10.228.72.4:5060:
OPTIONS sip:10.228.72.4 SIP/2.0
Via: SIP/2.0/UDP 10.228.37.149:5060;branch=z9hG4bK05d818e7
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.228.37.149;tag=as102a4570
To: sip:10.228.72.4
Contact: sip:asterisk@10.228.37.149:5060
Call-ID: 0c5ece5c4faf599738cca0445c9c02d4@10.228.37.149:5060
CSeq: 102 OPTIONS
User-Agent: NATCOMVOIP
Date: Tue, 14 Nov 2017 14:07:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.228.72.4:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.228.37.149:5060;branch=z9hG4bK05d818e7
Call-ID: 0c5ece5c4faf599738cca0445c9c02d4@10.228.37.149:5060
From: "asterisk"sip:asterisk@10.228.37.149;tag=as102a4570
To: sip:10.228.72.4;tag=vfq7qons
CSeq: 102 OPTIONS
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘0c5ece5c4faf599738cca0445c9c02d4@10.228.37.149:5060’ Method: OPTIONS

<— SIP read from UDP:10.228.8.4:5065 —>
INVITE sip:206@10.228.37.149;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.228.8.4:5065;branch=z9hG4bKtslikosil0jbwsrt0hisrkmlh;X-DptMsg=139
Route: sip:10.228.37.149:5060;transport=udp;lr
Call-ID: jqrvsmpqivvwstto0hbbqslkkhmqhpit@10.18.5.64
From: "33123456"sip:33123456@10.228.8.4;transport=udp;user=phone;tag=pvwsvssb-CC-37
To: "206"sip:206@10.228.37.149;transport=udp;user=phone
CSeq: 1 INVITE
P-Access-Network-Info: GEN-ACCESS;"area-number=+50933001303"
Max-Forwards: 70
Contact: sip:10.228.8.4:5060
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
P-Asserted-Identity: tel:33123456
Supported: 100rel,timer,histinfo,precondition
Min-SE: 90
Session-Expires: 1800;refresher=uac
P-Early-Media: supported
Content-Length:332
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1108417851 1108417852 IN IP4 10.228.9.11
s=SipCall
c=IN IP4 10.228.9.11
t=0 0
m=audio 62872 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC
<------------->
— (18 headers 14 lines) —
Sending to 10.228.8.4:5065 (no NAT)
Using INVITE request as basis request - jqrvsmpqivvwstto0hbbqslkkhmqhpit@10.18.5.64
Found peer ‘trunk_GMSC’ for ‘33123456’ from 10.228.8.4:5065
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 116
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 116
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.228.9.11:62872
Looking for 206 in from_trunk_GMSC (domain 10.228.37.149)
list_route: hop: sip:10.228.8.4:5060

<— Transmitting (no NAT) to 10.228.8.4:5065 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.228.8.4:5065;branch=z9hG4bKtslikosil0jbwsrt0hisrkmlh;X-DptMsg=139;received=10.228.8.4
From: "33123456"sip:33123456@10.228.8.4;transport=udp;user=phone;tag=pvwsvssb-CC-37
To: "206"sip:206@10.228.37.149;transport=udp;user=phone
Call-ID: jqrvsmpqivvwstto0hbbqslkkhmqhpit@10.18.5.64
CSeq: 1 INVITE
Server: NATCOMVOIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: sip:206@10.228.37.149:5060
Content-Length: 0

<------------>
– Executing [206@from_trunk_GMSC:1] Answer(“SIP/trunk_GMSC-00000002”, “”) in new stack
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 10.228.8.4:5065 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.228.8.4:5065;branch=z9hG4bKtslikosil0jbwsrt0hisrkmlh;X-DptMsg=139;received=10.228.8.4
From: "33123456"sip:33123456@10.228.8.4;transport=udp;user=phone;tag=pvwsvssb-CC-37
To: "206"sip:206@10.228.37.149;transport=udp;user=phone;tag=as1cbc6f89
Call-ID: jqrvsmpqivvwstto0hbbqslkkhmqhpit@10.18.5.64
CSeq: 1 INVITE
Server: NATCOMVOIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: sip:206@10.228.37.149:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 72099609 72099609 IN IP4 10.228.37.149
s=Asterisk PBX 1.8.5.0
c=IN IP4 10.228.37.149
t=0 0
m=audio 11010 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

<— SIP read from UDP:10.228.8.4:5065 —>
ACK sip:206@10.228.37.149:5060 SIP/2.0
Via: SIP/2.0/UDP 10.228.8.4:5065;branch=z9hG4bKislshotvtkpwjvtqj0lrw0vok;X-DptMsg=139
Call-ID: jqrvsmpqivvwstto0hbbqslkkhmqhpit@10.18.5.64
From: "33123456"sip:33123456@10.228.8.4;transport=udp;user=phone;tag=pvwsvssb-CC-37
To: "206"sip:206@10.228.37.149;transport=udp;user=phone;tag=as1cbc6f89
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

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

<— SIP read from UDP:10.228.8.4:5065 —>
BYE sip:206@10.228.37.149:5060 SIP/2.0
Via: SIP/2.0/UDP 10.228.8.4:5065;branch=z9hG4bKjlspbsqv00khj00jthtrko0sq;X-DptMsg=139
Call-ID: jqrvsmpqivvwstto0hbbqslkkhmqhpit@10.18.5.64
From: "33123456"sip:33123456@10.228.8.4;transport=udp;user=phone;tag=pvwsvssb-CC-37
To: "206"sip:206@10.228.37.149;transport=udp;user=phone;tag=as1cbc6f89
CSeq: 2 BYE
Reason: SIP;text="SS120001F159L790 Session Timer Check Message Failed for INVITE 2xx"
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 10.228.8.4:5065 (no NAT)
Scheduling destruction of SIP dialog ‘jqrvsmpqivvwstto0hbbqslkkhmqhpit@10.18.5.64’ in 6400 ms (Method: BYE)

<— Transmitting (no NAT) to 10.228.8.4:5065 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.228.8.4:5065;branch=z9hG4bKjlspbsqv00khj00jthtrko0sq;X-DptMsg=139;received=10.228.8.4
From: "33123456"sip:33123456@10.228.8.4;transport=udp;user=phone;tag=pvwsvssb-CC-37
To: "206"sip:206@10.228.37.149;transport=udp;user=phone;tag=as1cbc6f89
Call-ID: jqrvsmpqivvwstto0hbbqslkkhmqhpit@10.18.5.64
CSeq: 2 BYE
Server: NATCOMVOIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (from_trunk_GMSC, 206, 1) exited non-zero on ‘SIP/trunk_GMSC-00000002’

david551 · November 15, 2017, 3:36pm

exited non-zero generally just means that someone hungup, and someone was either the called party, or it was the calling party and the context suggested the end of the call, e.g. Dial after a successful call. It is not an error and is normal in almost every call.

In your case, it has happened because the caller has ended the call.

They have provided some diagnostic information: SS120001F159L790 Session Timer Check Message Failed for INVITE 2xx Providing that information is optional, and it appears to be free text.

I cannot see why they would produced such a diagnostic. You will need to ask the people that maintain it. As I guess it relates to session timers, you could see if it is happier if you refuse them.

quang151080 · November 15, 2017, 3:56pm

Thank you very much David551,
This means that because my provider has stopped the call (Reason: SIP;text=“SS120001F159L790 Session Timer Check Message Failed for INVITE 2xx”).

I have more one question.
I have received SIP on IP 10.228.8.4 (log: SIP read from UDP:10.228.8.4:5065) but media in other IP(c=IN IP4 10.228.9.11).
and i do not know two line log :
a=curr:qos local none
a=curr:qos remote none

Information log at below:

<— SIP read from UDP:10.228.8.4:5065 —>
INVITE sip:206@10.228.37.149;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.228.8.4:5065;branch=z9hG4bKtslikosil0jbwsrt0hisrkmlh;X-DptMsg=139
Route: sip:10.228.37.149:5060;transport=udp;lr
Call-ID: jqrvsmpqivvwstto0hbbqslkkhmqhpit@10.18.5.64
From: "33123456"sip:33123456@10.228.8.4;transport=udp;user=phone;tag=pvwsvssb-CC-37
To: "206"sip:206@10.228.37.149;transport=udp;user=phone
CSeq: 1 INVITE
P-Access-Network-Info: GEN-ACCESS;"area-number=+50933001303"
Max-Forwards: 70
Contact: sip:10.228.8.4:5060
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
P-Asserted-Identity: tel:33123456
Supported: 100rel,timer,histinfo,precondition
Min-SE: 90
Session-Expires: 1800;refresher=uac
P-Early-Media: supported
Content-Length:332
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1108417851 1108417852 IN IP4 10.228.9.11
s=SipCall
c=IN IP4 10.228.9.11
t=0 0
m=audio 62872 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC

david551 · November 15, 2017, 9:41pm

Having media on a separate address is quite common.

See RFC 3312 for the curr:qos and des:qos attributes, although I have never seen these before and think Asterisk ignores it.

There is something about your other obscure attribute in http://www.etsi.org/deliver/etsi_ts/129200_129299/129235/08.03.00_60/ts_129235v080300p.pdf but very few SIP implementations appear to understand that one.

quang151080 · November 16, 2017, 2:54am

Thanks for your reply,
Reason: SIP;text=“SS120001F159L790 Session Timer Check Message Failed for INVITE 2xx”

You are right, My provider has drop call by itself. They said to me that in my asterisk don’t config Section timer parameter. I have never see this parameter.
When they remove this parameter out their system (with my acount) then SIP has worked fine.
Can you tell to me how to config Section timer in asterisk (i am using asteterisk 1.8)

david551 · November 16, 2017, 9:31am

Do you mean session timer?

Asterisk is doing a normal session timer negotiation, and has clear documentation on how to configure them in the sample sip.conf file.

I’ve never heard of section timers, and I’m sure that Asterisk hasn’t heard of them.

quang151080 · February 27, 2018, 5:02pm

Dear David,
With this issue i have fix by
in Sip.conf

externip = 10.228.37.149
localnet = 10.228.37.149/255.255.255.0
…
session-timers=refuse
session-expires=1800
session-minse=90
session-refresher=uac

But with the same asterisk on another server with NAT
externip = 10.228.30.26
localnet = 10.228.37.149/255.255.255.0
…
session-timers=refuse
session-expires=1800
session-minse=90
session-refresher=uac

i have received the same err
Reason: SIP;text=“SS120001F159L790 Session Timer Check Message Failed for INVITE 2xx”

Pls help me,

quang151080 · February 27, 2018, 6:40pm

Dear David,
Sory because no NAT,
In my server have two IP (two card)

IP 10.228.30.26 connect to SIP server
IP 10.228.37.149 connect to Database
I have difine a SIP acount

[trunk_GMSC]
type=peer
host=10.228.8.4
context=from_trunk_GMSC
qualify=yes
;nat=no
;keepalive=45
dtmfmode=rfc2833
;disallow=all
;allow=gsm
;allow=alaw
;allow=ulaw
Canreinvite = no
insecure=port,invite

session-timers=refuse
session-expires=1800
session-minse=90
session-refresher=uac