Call often hangup after few seconds

Asterisk Asterisk SIP
1

Hi all,
I have a cisco ip phone 303 and freepbx.
Sometimes, when i call out of my lan, calls hangup after few seconds.

[general]
faxdetect=yes
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
disallow=all ;First disallow all codecs
allow=g729
allow=alaw
allow=gsm
allow=ulaw ; Allow codecs in order of preference
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
videosupport=no ; Turn on support for SIP video. You need to turn this on
callevents=yes ; generate manager events when sip ua
alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
rtpkeepalive=5 ; Send keepalives in the RTP stream to keep NAT open
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes ; Apply call limits on peers only. This will improve
t38pt_udptl = yes ; Default false
registertimeout=15 ; retry registration calls every 20 seconds (default)
externip = 93.57.87.34 ; Address that we’re going to put in outbound SIP
localnet=192.168.1.0/255.255.255.0 ; DF AGGIUNTO
localnet=192.168.0.0/255.255.0.0 ; All RFC 1918 addresses are local networks
nat=yes ; Global NAT settings (Affects all peers and users)
canreinvite=no ; Asterisk by default tries to redirect the
jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 30 ; Max length of the jitterbuffer in milliseconds.
Wjbresyncthreshold = 1000 ; Jump in the frame timestamps over which the
jitterbuffer is jbimpl = adaptive ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to “no”.
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers
dtmfmode=inband

[account]
[JOBs]
disallow=all
allow=alaw
secret=xxxxxx
callerid=“yyyyyy”
mailbox=zzzz
context=cccccc
type=friend
host=dynamic
nat=yes ; DF - nat=no
port=5060
dtmfmode=rfc2833

when not in call and looking at sip debug i see

<— Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-140da15b;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=fbd306646654a9bo0
To: sip:192.168.1.246;tag=as3c1dd163
Call-ID: 5c2d9c40-98ba4e6f@192.168.1.217
CSeq: 84 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

please help

2

You need to provide a sip set debug on log of this actually occurring. The small piece you’ve provided is normal. We were sent a NOTIFY and we rejected it. It is not involving an active call.

3

hope this will be usefull

<— SIP read from 192.168.1.217:5060 —>
NOTIFY sip:192.168.1.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-b80acbda
From: “JOBs” sip:JOBs@192.168.1.246;tag=fbd306646654a9bo0
To: sip:192.168.1.246
Call-ID: 5c2d9c40-98ba4e6f@192.168.1.217
CSeq: 196 NOTIFY
Max-Forwards: 70
Contact: “JOBs” sip:JOBs@192.168.1.217:5060
Event: keep-alive
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.217 : 5060 (NAT)

<— Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-b80acbda;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=fbd306646654a9bo0
To: sip:192.168.1.246;tag=as1ed6203b
Call-ID: 5c2d9c40-98ba4e6f@192.168.1.217
CSeq: 196 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

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

<— SIP read from 192.168.1.217:5060 —>
NOTIFY sip:192.168.1.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-dca4d939
From: “JOBs” sip:JOBs@192.168.1.246;tag=fbd306646654a9bo0
To: sip:192.168.1.246
Call-ID: 5c2d9c40-98ba4e6f@192.168.1.217
CSeq: 197 NOTIFY
Max-Forwards: 70
Contact: “JOBs” sip:JOBs@192.168.1.217:5060
Event: keep-alive
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.217 : 5060 (NAT)

<— Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-dca4d939;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=fbd306646654a9bo0
To: sip:192.168.1.246;tag=as6ffda2ac
Call-ID: 5c2d9c40-98ba4e6f@192.168.1.217
CSeq: 197 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
– Nobody picked up in 60000 ms
– Hungup ‘IAX2/PBXPA-765’
– Executing [63207141492@JOBSINFO:2] Hangup(“SIP/JOBs-0000005d”, “”) in new stack
== Spawn extension (JOBSINFO, 63207141492, 2) exited non-zero on 'SIP/JOBs-0000005d’
Scheduling destruction of SIP dialog ‘40b3b9df-c5baa55a@192.168.1.217’ in 32000 ms (Method: INVITE)

4

There is nothing for the SIP call in that trace either, are you limiting it at all or starting it after the call?

5

I don’t understand, what i have to set?

6

Type “sip set debug on” before you do a call. Also do “core set verbose 9”. Do a call. Provide the output. That should give a full picture of what is going on.

7

I have setted sip set debug ip and asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
(i have set now core set verbose 9 and system show me “verbosity was 60”)

in the web interface of the cisco phone i found in the ext1 configuration
NAT mapping enable=no;
NAT keep alive enable = no;

8

here a hangup call from the start. the first time call hangup

<— SIP read from 192.168.1.217:5060 —>
INVITE sip:63208895442@192.168.1.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-bf12b03b
From: “JOBs” sip:JOBs@192.168.1.246;tag=50114e42553447a3o0
To: sip:63208895442@192.168.1.246
Call-ID: c0c55dee-1419be7@192.168.1.217
CSeq: 101 INVITE
Max-Forwards: 70
Contact: “JOBs” sip:JOBs@192.168.1.217:5060
Expires: 240
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 397
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 188030 188030 IN IP4 192.168.1.217
s=-
c=IN IP4 192.168.1.217
t=0 0
m=audio 10338 RTP/AVP 8 0 2 9 18 96 97 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (14 headers 18 lines) —
Sending to 192.168.1.217 : 5060 (NAT)
Using INVITE request as basis request - c0c55dee-1419be7@192.168.1.217

<— Reliably Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-bf12b03b;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=50114e42553447a3o0
To: sip:63208895442@192.168.1.246;tag=as333d5996
Call-ID: c0c55dee-1419be7@192.168.1.217
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="12a8ff22"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘c0c55dee-1419be7@192.168.1.217’ in 32000 ms (Method: INVITE)
Found user ‘JOBs’

<— SIP read from 192.168.1.217:5060 —>
ACK sip:63208895442@192.168.1.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-bf12b03b
From: “JOBs” sip:JOBs@192.168.1.246;tag=50114e42553447a3o0
To: sip:63208895442@192.168.1.246;tag=as333d5996
Call-ID: c0c55dee-1419be7@192.168.1.217
CSeq: 101 ACK
Max-Forwards: 70
Contact: “JOBs” sip:JOBs@192.168.1.217:5060
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 0

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

<— SIP read from 192.168.1.217:5060 —>
INVITE sip:63208895442@192.168.1.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-b83d8b0b
From: “JOBs” sip:JOBs@192.168.1.246;tag=50114e42553447a3o0
To: sip:63208895442@192.168.1.246
Call-ID: c0c55dee-1419be7@192.168.1.217
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username=“JOBs”,realm=“asterisk”,nonce=“12a8ff22”,uri="sip:63208895442@192.168.1.246",algorithm=MD5,response="de3db489712fa909b51211c6f53853da"
Contact: “JOBs” sip:JOBs@192.168.1.217:5060
Expires: 240
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 397
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 188030 188030 IN IP4 192.168.1.217
s=-
c=IN IP4 192.168.1.217
t=0 0
m=audio 10338 RTP/AVP 8 0 2 9 18 96 97 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (15 headers 18 lines) —
Sending to 192.168.1.217 : 5060 (NAT)
Using INVITE request as basis request - c0c55dee-1419be7@192.168.1.217
Found user 'JOBs’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G722 for ID 9
Found audio description format G729a for ID 18
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=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 192.168.1.217:10338
Looking for 63208895442 in JOBSINFO (domain 192.168.1.246)
list_route: hop: sip:JOBs@192.168.1.217:5060

<— Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-b83d8b0b;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=50114e42553447a3o0
To: sip:63208895442@192.168.1.246
Call-ID: c0c55dee-1419be7@192.168.1.217
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:63208895442@192.168.1.246
Content-Length: 0

<------------>
– Executing [63208895442@JOBSINFO:1] Dial(“SIP/JOBs-0000008b”, “IAX2/PBXPA:PBXPA@192.168.1.4:4569/63208895442|60|Tto”) in new stack
– Call accepted by 192.168.1.4 (format alaw)
– Format for call is alaw
– Called PBXPA:PBXPA@192.168.1.4:4569/63208895442
– IAX2/PBXPA-5343 is ringing

<— Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-b83d8b0b;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=50114e42553447a3o0
To: sip:63208895442@192.168.1.246;tag=as2be451b4
Call-ID: c0c55dee-1419be7@192.168.1.217
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:63208895442@192.168.1.246
Content-Length: 0

<------------>
– IAX2/PBXPA-5343 is making progress passing it to SIP/JOBs-0000008b
Audio is at 192.168.1.246 port 18030
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-b83d8b0b;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=50114e42553447a3o0
To: sip:63208895442@192.168.1.246;tag=as2be451b4
Call-ID: c0c55dee-1419be7@192.168.1.217
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:63208895442@192.168.1.246
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 3111 3111 IN IP4 192.168.1.246
s=session
c=IN IP4 192.168.1.246
t=0 0
m=audio 18030 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
– IAX2/PBXPA-5343 stopped sounds
Reliably Transmitting (NAT) to 192.168.1.217:5060:
OPTIONS sip:JOBs@192.168.1.217:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.246:5060;branch=z9hG4bK20b10e34;rport
From: “asterisk” sip:asterisk@192.168.1.246;tag=as49989a70
To: sip:JOBs@192.168.1.217:5060
Contact: sip:asterisk@192.168.1.246
Call-ID: 4f3f01a4074a3ae74cb11b123bc13899@192.168.1.246
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 09 Feb 2017 14:40:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<— SIP read from 192.168.1.217:5060 —>
SIP/2.0 200 OK
To: sip:JOBs@192.168.1.217:5060;tag=2ff013349f772566i0
From: “asterisk” sip:asterisk@192.168.1.246;tag=as49989a70
Call-ID: 4f3f01a4074a3ae74cb11b123bc13899@192.168.1.246
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.246:5060;branch=z9hG4bK20b10e34
Server: Cisco/SPA303-7.4.9a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘4f3f01a4074a3ae74cb11b123bc13899@192.168.1.246’ Method: OPTIONS
– Nobody picked up in 60000 ms
– Hungup ‘IAX2/PBXPA-5343’
– Executing [63208895442@JOBSINFO:2] Hangup(“SIP/JOBs-0000008b”, “”) in new stack
== Spawn extension (JOBSINFO, 63208895442, 2) exited non-zero on 'SIP/JOBs-0000008b’
Scheduling destruction of SIP dialog ‘c0c55dee-1419be7@192.168.1.217’ in 32000 ms (Method: INVITE)

<— Reliably Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-b83d8b0b;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=50114e42553447a3o0
To: sip:63208895442@192.168.1.246;tag=as2be451b4
Call-ID: c0c55dee-1419be7@192.168.1.217
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

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

<— SIP read from 192.168.1.217:5060 —>
ACK sip:63208895442@192.168.1.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-b83d8b0b
From: “JOBs” sip:JOBs@192.168.1.246;tag=50114e42553447a3o0
To: sip:63208895442@192.168.1.246;tag=as2be451b4
Call-ID: c0c55dee-1419be7@192.168.1.217
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username=“JOBs”,realm=“asterisk”,nonce=“12a8ff22”,uri="sip:63208895442@192.168.1.246",algorithm=MD5,response="de3db489712fa909b51211c6f53853da"
ontact: “JOBs” sip:JOBs@192.168.1.217:5060
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 0

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

<— SIP read from 192.168.1.217:5060 —>
INVITE sip:63208895442@192.168.1.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-e91df87b
From: “JOBs” sip:JOBs@192.168.1.246;tag=7670d1dcce5c24d3o0
To: sip:63208895442@192.168.1.246
Call-ID: d1046950-5d4653ef@192.168.1.217
CSeq: 101 INVITE
Max-Forwards: 70
Contact: “JOBs” sip:JOBs@192.168.1.217:5060
Expires: 240
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 397
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 195200 195200 IN IP4 192.168.1.217
s=-
c=IN IP4 192.168.1.217
t=0 0
m=audio 10340 RTP/AVP 8 0 2 9 18 96 97 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (14 headers 18 lines) —
Sending to 192.168.1.217 : 5060 (NAT)
Using INVITE request as basis request - d1046950-5d4653ef@192.168.1.217

<— Reliably Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-e91df87b;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=7670d1dcce5c24d3o0
To: sip:63208895442@192.168.1.246;tag=as45c8b626
Call-ID: d1046950-5d4653ef@192.168.1.217
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="35caff4d"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘d1046950-5d4653ef@192.168.1.217’ in 32000 ms (Method: INVITE)
Found user ‘JOBs’

<— SIP read from 192.168.1.217:5060 —>
ACK sip:63208895442@192.168.1.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-e91df87b
From: “JOBs” sip:JOBs@192.168.1.246;tag=7670d1dcce5c24d3o0
To: sip:63208895442@192.168.1.246;tag=as45c8b626
Call-ID: d1046950-5d4653ef@192.168.1.217
CSeq: 101 ACK
Max-Forwards: 70
Contact: “JOBs” sip:JOBs@192.168.1.217:5060
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 0

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

<— SIP read from 192.168.1.217:5060 —>
INVITE sip:63208895442@192.168.1.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-e675b527
From: “JOBs” sip:JOBs@192.168.1.246;tag=7670d1dcce5c24d3o0
To: sip:63208895442@192.168.1.246
Call-ID: d1046950-5d4653ef@192.168.1.217
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username=“JOBs”,realm=“asterisk”,nonce=“35caff4d”,uri="sip:63208895442@192.168.1.246",algorithm=MD5,response="f32abe38e0d5376a3eb234392d629a4f"
Contact: “JOBs” sip:JOBs@192.168.1.217:5060
Expires: 240
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 397
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 195200 195200 IN IP4 192.168.1.217
s=-
c=IN IP4 192.168.1.217
t=0 0
m=audio 10340 RTP/AVP 8 0 2 9 18 96 97 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (15 headers 18 lines) —
Sending to 192.168.1.217 : 5060 (NAT)
Using INVITE request as basis request - d1046950-5d4653ef@192.168.1.217
Found user 'JOBs’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G722 for ID 9
Found audio description format G729a for ID 18
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=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 192.168.1.217:10340
Looking for 63208895442 in JOBSINFO (domain 192.168.1.246)
list_route: hop: sip:JOBs@192.168.1.217:5060

<— Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-e675b527;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=7670d1dcce5c24d3o0
To: sip:63208895442@192.168.1.246
Call-ID: d1046950-5d4653ef@192.168.1.217
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:63208895442@192.168.1.246
Content-Length: 0

<------------>
– Executing [63208895442@JOBSINFO:1] Dial(“SIP/JOBs-0000008c”, “IAX2/PBXPA:PBXPA@192.168.1.4:4569/63208895442|60|Tto”) in new stack
– Called PBXPA:PBXPA@192.168.1.4:4569/63208895442
– Call accepted by 192.168.1.4 (format alaw)
– Format for call is alaw
Reliably Transmitting (NAT) to 192.168.1.217:5060:
OPTIONS sip:JOBs@192.168.1.217:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.246:5060;branch=z9hG4bK06349c92;rport
From: “asterisk” sip:asterisk@192.168.1.246;tag=as299d20bf
To: sip:JOBs@192.168.1.217:5060
Contact: sip:asterisk@192.168.1.246
Call-ID: 62a1418018c878ac1670a713094506f6@192.168.1.246
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 09 Feb 2017 14:41:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<— SIP read from 192.168.1.217:5060 —>
SIP/2.0 200 OK
To: sip:JOBs@192.168.1.217:5060;tag=2ff013349f772566i0
From: “asterisk” sip:asterisk@192.168.1.246;tag=as299d20bf
Call-ID: 62a1418018c878ac1670a713094506f6@192.168.1.246
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.246:5060;branch=z9hG4bK06349c92
Server: Cisco/SPA303-7.4.9a
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘62a1418018c878ac1670a713094506f6@192.168.1.246’ Method: OPTIONS
– IAX2/PBXPA-1880 is ringing

<— Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-e675b527;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=7670d1dcce5c24d3o0
To: sip:63208895442@192.168.1.246;tag=as5e96853e
Call-ID: d1046950-5d4653ef@192.168.1.217
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:63208895442@192.168.1.246
Content-Length: 0

<------------>
– IAX2/PBXPA-1880 is making progress passing it to SIP/JOBs-0000008c
Audio is at 192.168.1.246 port 14712
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-e675b527;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=7670d1dcce5c24d3o0
To: sip:63208895442@192.168.1.246;tag=as5e96853e
Call-ID: d1046950-5d4653ef@192.168.1.217
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:63208895442@192.168.1.246
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 3111 3111 IN IP4 192.168.1.246
s=session
c=IN IP4 192.168.1.246
t=0 0
m=audio 14712 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
– IAX2/PBXPA-1880 stopped sounds
– IAX2/PBXPA-1880 answered SIP/JOBs-0000008c
Audio is at 192.168.1.246 port 14712
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-e675b527;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=7670d1dcce5c24d3o0
To: sip:63208895442@192.168.1.246;tag=as5e96853e
Call-ID: d1046950-5d4653ef@192.168.1.217
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:63208895442@192.168.1.246
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 3111 3112 IN IP4 192.168.1.246
s=session
c=IN IP4 192.168.1.246
t=0 0
m=audio 14712 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<— SIP read from 192.168.1.217:5060 —>
ACK sip:63208895442@192.168.1.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-392a254c
From: “JOBs” sip:JOBs@192.168.1.246;tag=7670d1dcce5c24d3o0
To: sip:63208895442@192.168.1.246;tag=as5e96853e
Call-ID: d1046950-5d4653ef@192.168.1.217
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username=“JOBs”,realm=“asterisk”,nonce=“35caff4d”,uri="sip:63208895442@192.168.1.246",algorithm=MD5,response="f32abe38e0d5376a3eb234392d629a4f"
Contact: “JOBs” sip:JOBs@192.168.1.217:5060
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘c0c55dee-1419be7@192.168.1.217’ Method: ACK

<— SIP read from 192.168.1.217:5060 —>
BYE sip:63208895442@192.168.1.246 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-5f1c11d1
From: “JOBs” sip:JOBs@192.168.1.246;tag=7670d1dcce5c24d3o0
To: sip:63208895442@192.168.1.246;tag=as5e96853e
Call-ID: d1046950-5d4653ef@192.168.1.217
CSeq: 103 BYE
Max-Forwards: 70
Proxy-Authorization: Digest username=“JOBs”,realm=“asterisk”,nonce=“35caff4d”,uri="sip:63208895442@192.168.1.246",algorithm=MD5,response="81ac3b2e6c408171144d3d4909d69098"
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.1.217 : 5060 (NAT)

<— Transmitting (NAT) to 192.168.1.217:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.217:5060;branch=z9hG4bK-5f1c11d1;received=192.168.1.217
From: “JOBs” sip:JOBs@192.168.1.246;tag=7670d1dcce5c24d3o0
To: sip:63208895442@192.168.1.246;tag=as5e96853e
Call-ID: d1046950-5d4653ef@192.168.1.217
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
– Hungup ‘IAX2/PBXPA-1880’
== Spawn extension (JOBSINFO, 63208895442, 1) exited non-zero on 'SIP/JOBs-0000008c’
Really destroying SIP dialog ‘d1046950-5d4653ef@192.168.1.217’ Method: BYE

9

“Nobody picked up in 60000 ms”. Your outgoing call leg using IAX2 did not indicate that the other party had answered, so it hung the call up. I’d investigate what is on the other side of that IAX2 connection to determine if it was answered.

10

yes, call was answered, after few seconds or minutes of call, the call hangup

11

Right, it appears answered to you but the signaling does not show it as answered. That’s why it was hung up. If the thing on the other side of the IAX2 connection doesn’t tell us it is answered we don’t know.

12

can I bypass this?
What can I try?

13

Noone here can answer that question. We don’t know how your stuff is set up, what the IAX2 connection is used for, what’s on the other side.

14

so the 192.168.1.246 call the 192.168.1.4 that call the customer.
Can be useful a sip debug from the 1.4?

15

Yes, if you are using SIP to connect to what is dialed on the 1.4 side.

16

meanwhile the iax.conf of the 1.4 is

[general]
bindport=4569 ; bindport and bindaddr may be specified
bindaddr=0.0.0.0 ; more than once to bind to multiple
adsi=no
language=it
mohinterpret=passthrough
mohsuggest=default
disallow=all
allow=alaw
allow=ulaw
allow=g729
jitterbuffer=no
forcejitterbuffer=no
maxjitterbuffer=200
maxjitterinterps=10
resyncthreshold=1000
trunkfreq=20 ; How frequently to send trunk msgs (in ms)
trunktimestamps=yes
minregexpire = 360
maxregexpire = 3600
iaxthreadcount = 50
iaxmaxthreadcount = 250
autokill=yes

[PBXPA]
disallow=all
allow=alaw
type=friend
username=PBXPA
secret=PBXPA
auth=md5
qualify=yes
transfer=yes
trunk=yes
context=PBXTOPBX
host=192.168.1.246
port=4569

17

still waiting for a call :smiley: