Why Asterisk not working properly with android sip client

zain · January 9, 2013, 7:14pm

I am developing SIP client in Android using native android SIP Stack (developer.android.com/reference/ … mmary.html). Problem is that my android application is working perfectly with elastix server but not working properly with asterisk server. I am facing two problems
1 .Low audio quality with distortion
2. Most of times only one way audio

I have installed asterisk server 1.6.2.11 . I have tested my app on android 2.3 and android 4.0.3.

navaismo · January 9, 2013, 8:18pm

Elastix use FreePBX, FreePBX is a GUI for Asterisk, so You are using Asterisk in both cases. Then Youe need to look at your configs when you are using plain asterisk, also provides traces of the cli and sip debug for both cases and finally check at your NAT settings for avoid the one way audio.

david55 · January 9, 2013, 9:21pm

Also using a version of Asterisk with many known bugs and which is a sub-version of a version on which bug fixing stopped over a year ago is not going to help.

zain · January 11, 2013, 6:23am

First of all thank you for your wonderful support…I have now updated my asterisk server from 1.6.2.11 to 1.8.11.0 but still only one way audio ??
On my centos i have installed following applications

Asterisk 1.8.11.0
Apache
Apache Tomcat
Eclipse
MySql
phpMyAdmin

I dont know if any of these installed applications creating problem or its Asterisk problem ???

rgpasterisk · January 11, 2013, 2:43pm

Do this:

Generate sip extension from freepbx in asterisknow, example: 100.
Lookfor “[100]” in the file /etc/asterisk/sip_additional.conf
Use the configuration used in your asterisk only installation.
If it does not work you can see /etc/asterisk/sip_general_additional.conf
core restart now from cli

zain · January 12, 2013, 4:20pm

Thanks for taking interest…I have tried all as mentioned by “rgpasterisk” but still no luckk
Now i am facing another problem. I establish outgoing call from my android client , the called client (android,xlite or zoiper) rings, but if i ends call, the called client still rings…why asterisk dont read end call function from android.
On the other hand if i make call from (zoiper/xlite) to (zoiper/xlite) it ends call perfectly …i dont know what is that problem ??

navaismo · January 12, 2013, 5:17pm

Are you configuring the Nat settings for your asterisk? Asterisk needs some config in order to work properly with remote extensions. This is not plug&play also you need to configure your modem/router.

zain · January 12, 2013, 5:42pm

I am using asterisk on local network (LAN)…

My dial plan in extensions.conf is:
[incoming-calls-wildcard]
exten => _2XX,hint,(SIP/${EXTEN},120)
exten => _2XX,1,Dial(SIP/${EXTEN},120)
exten => _2XX,n,Hangup
My sip account is:
[215]
deny=0.0.0.0/0.0.0.0
secret=very123
dtmfmode=rfc2833
canreinvite=no
context=incoming-calls-wildcard
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/215
mailbox=215@device
permit=0.0.0.0/0.0.0.0
callerid=device <215>
callcounter=yes
faxdetect=no

david55 · January 12, 2013, 8:07pm

Why do you have nat=yes, if everything is on the same LAN? This is overused, even if there is some NAT involved.

zain · January 12, 2013, 8:10pm

When i call from my pc using zoiper/xlite to android (native android sip client) now i can hear audio from both sides but when i make call from android to pc (zoiper/xlite) i cannot hear anything on android.
pc(zoiper) ip 192.168.15.27
android ip 192.168.15.71
asterisk server ip 192.168.15.118
Sip debug when calling from android to zoiper .

[code]
<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK5996b0d9;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as05233e7d
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 14151c2d29c039983aa449b56ce419e0@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘14151c2d29c039983aa449b56ce419e0@192.168.15.118:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.15.71:45616 —>
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: 5f7229b44e43a9e0eeaccabafd9a1e4f@192.168.15.71
CSeq: 7757 OPTIONS
From: “211” sip:211@192.168.15.118;tag=1758376458
To: “211” sip:211@192.168.15.118
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK6c57f04e92e9e34153367ca0e6e29675353134;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Looking for s in default (domain 192.168.15.118)

<— Transmitting (NAT) to 192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK6c57f04e92e9e34153367ca0e6e29675353134;received=192.168.15.71;rport=45616
From: “211” sip:211@192.168.15.118;tag=1758376458
To: “211” sip:211@192.168.15.118;tag=as6a8e1b47
Call-ID: 5f7229b44e43a9e0eeaccabafd9a1e4f@192.168.15.71
CSeq: 7757 OPTIONS
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:192.168.15.118:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5f7229b44e43a9e0eeaccabafd9a1e4f@192.168.15.71’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK55bfb5a3;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as4d70d20a
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 2a013bfc627ee6d66ee6cd1147523dca@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK55bfb5a3;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as4d70d20a
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 2a013bfc627ee6d66ee6cd1147523dca@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘2a013bfc627ee6d66ee6cd1147523dca@192.168.15.118:5060’ Method: OPTIONS
Really destroying SIP dialog ‘ebeb1ae28a83c525ddb70282998f4bdb@192.168.15.71’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK50992e76;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as58f34282
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 1acdcd6b772b0ec563983f4c404f085c@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK50992e76;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as58f34282
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 1acdcd6b772b0ec563983f4c404f085c@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘1acdcd6b772b0ec563983f4c404f085c@192.168.15.118:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.15.27:5060 —>

<------------->
Reliably Transmitting (NAT) to 192.168.15.27:5060:
OPTIONS sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1dc068d1;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as18e028af
To: sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 60d2d7de6893aaaf4281f02c7f7dfd14@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.27:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1dc068d1;rport=5060
Contact: sip:192.168.15.27:5060
To: sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP;tag=5512d67c
From: "asterisk"sip:asterisk@192.168.15.118;tag=as18e028af
Call-ID: 60d2d7de6893aaaf4281f02c7f7dfd14@192.168.15.118:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.39 r16838
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘60d2d7de6893aaaf4281f02c7f7dfd14@192.168.15.118:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK72d2ee91;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as398ea7d4
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 776e44155f168cc902981c5159d5a57c@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK72d2ee91;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as398ea7d4
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 776e44155f168cc902981c5159d5a57c@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘776e44155f168cc902981c5159d5a57c@192.168.15.118:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.15.71:45616 —>
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: e24339a14c0eb923956a010bc85add06@192.168.15.71
CSeq: 7471 OPTIONS
From: “211” sip:211@192.168.15.118;tag=3555156406
To: “211” sip:211@192.168.15.118
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKdaef11c4331a1f58ad2dd04c58ccdc1e353134;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Looking for s in default (domain 192.168.15.118)

<— Transmitting (NAT) to 192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKdaef11c4331a1f58ad2dd04c58ccdc1e353134;received=192.168.15.71;rport=45616
From: “211” sip:211@192.168.15.118;tag=3555156406
To: “211” sip:211@192.168.15.118;tag=as1a539872
Call-ID: e24339a14c0eb923956a010bc85add06@192.168.15.71
CSeq: 7471 OPTIONS
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:192.168.15.118:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘e24339a14c0eb923956a010bc85add06@192.168.15.71’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1ecea84c;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as167765df
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 5f4d210800aa07065fd35fba215e9783@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1ecea84c;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as167765df
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 5f4d210800aa07065fd35fba215e9783@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘5f4d210800aa07065fd35fba215e9783@192.168.15.118:5060’ Method: OPTIONS
Really destroying SIP dialog ‘5e5f98ad4818911a86d4b438d054e39f@192.168.15.71’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK15f7593b;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as53340ecf
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK15f7593b;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as53340ecf
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.15.71:45616 —>
BYE sip:215@192.168.15.118:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKad00f5add2a1f2e940a91e9a3f827b26353134
CSeq: 5511 BYE
From: “211” sip:211@192.168.15.118;tag=2465683119
To: sip:215@192.168.15.118;tag=as573c52b3
Call-ID: d188757cd6c044783fd413d11f8a982f@192.168.15.71
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.15.71:45616 (NAT)
Scheduling destruction of SIP dialog ‘d188757cd6c044783fd413d11f8a982f@192.168.15.71’ in 6400 ms (Method: BYE)

<— Transmitting (NAT) to 192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKad00f5add2a1f2e940a91e9a3f827b26353134;received=192.168.15.71;rport=45616
From: “211” sip:211@192.168.15.118;tag=2465683119
To: sip:215@192.168.15.118;tag=as573c52b3
Call-ID: d188757cd6c044783fd413d11f8a982f@192.168.15.71
CSeq: 5511 BYE
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0d06690561757f98637241394cc8dbba@192.168.15.118:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP for address/port to send to
set_destination: set destination to 115.167.21.82:5060
Reliably Transmitting (NAT) to 192.168.15.27:5060:
BYE sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport
Max-Forwards: 70
From: “device” sip:211@192.168.15.118;tag=as404f0eb0
To: sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP;tag=96055240
Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.11.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

== Spawn extension (incoming-calls-wildcard, 215, 1) exited non-zero on 'SIP/211-00000008’
Retransmitting #1 (NAT) to 192.168.15.27:5060:
BYE sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport
Max-Forwards: 70
From: “device” sip:211@192.168.15.118;tag=as404f0eb0
To: sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP;tag=96055240
Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.11.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from UDP:192.168.15.27:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport=5060
Contact: sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP
To: sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP;tag=96055240
From: "device"sip:211@192.168.15.118;tag=as404f0eb0
Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
CSeq: 103 BYE
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0

<------------->
– (9 headers 0 lines) —
Really destroying SIP dialog ‘0d06690561757f98637241394cc8dbba@192.168.15.118:5060’ Method: INVITE

<— SIP read from UDP:192.168.15.27:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport=5060
Contact: sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP
To: sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP;tag=96055240
From: "device"sip:211@192.168.15.118;tag=as404f0eb0
Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
CSeq: 103 BYE
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0

<------------->
– (9 headers 0 lines) —
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK30d9d70f;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as4f0724aa
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 05f27bf10f0a0759637b90252dc4366a@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK30d9d70f;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as4f0724aa
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 05f27bf10f0a0759637b90252dc4366a@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘05f27bf10f0a0759637b90252dc4366a@192.168.15.118:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.15.71:45616 —>
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: a5a311df861221d42844a8c485d4fee8@192.168.15.71
CSeq: 5815 OPTIONS
From: “211” sip:211@192.168.15.118;tag=3109248316
To: “211” sip:211@192.168.15.118
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK1e09e6f6c30b4b74ddc93c590abd3383353134;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Looking for s in default (domain 192.168.15.118)

<— Transmitting (NAT) to 192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK1e09e6f6c30b4b74ddc93c590abd3383353134;received=192.168.15.71;rport=45616
From: “211” sip:211@192.168.15.118;tag=3109248316
To: “211” sip:211@192.168.15.118;tag=as51223faf
Call-ID: a5a311df861221d42844a8c485d4fee8@192.168.15.71
CSeq: 5815 OPTIONS
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:192.168.15.118:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘a5a311df861221d42844a8c485d4fee8@192.168.15.71’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1f8e44dc;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as7a9a1ea3
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 7ebcafc7159379fd047075a85c424588@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1f8e44dc;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as7a9a1ea3
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 7ebcafc7159379fd047075a85c424588@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘7ebcafc7159379fd047075a85c424588@192.168.15.118:5060’ Method: OPTIONS
Really destroying SIP dialog ‘d188757cd6c044783fd413d11f8a982f@192.168.15.71’ Method: BYE
Really destroying SIP dialog ‘a81e6a5f591141abd73f9dad478a6b56@192.168.15.71’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK017f2b0a;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as5367b37c
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:44:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK017f2b0a;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as5367b37c
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118:5060’ Method: OPTIONS[/code]

When calling from pc (zoiper) to android

[code]<— SIP read from UDP:192.168.15.71:45616 —>
BYE sip:215@192.168.15.118:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKe2f448b90c482342407298e309be5aed353134
CSeq: 1 BYE
From: sip:211@192.168.15.71:45616;transport=udp;tag=4162167884
To: “device” sip:215@192.168.15.118;tag=as5805dc66
Call-ID: 2732e4564ce8534c5765a456045a9960@192.168.15.118:5060
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Sending to 192.168.15.71:45616 (NAT)
Scheduling destruction of SIP dialog ‘2732e4564ce8534c5765a456045a9960@192.168.15.118:5060’ in 8576 ms (Method: BYE)

<— Transmitting (NAT) to 192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKe2f448b90c482342407298e309be5aed353134;received=192.168.15.71;rport=45616
From: sip:211@192.168.15.71:45616;transport=udp;tag=4162167884
To: “device” sip:215@192.168.15.118;tag=as5805dc66
Call-ID: 2732e4564ce8534c5765a456045a9960@192.168.15.118:5060
CSeq: 1 BYE
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (incoming-calls-wildcard, 211, 1) exited non-zero on 'SIP/215-0000000a’
Scheduling destruction of SIP dialog ‘MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:215@115.167.21.82:5060;transport=UDP for address/port to send to
set_destination: set destination to 115.167.21.82:5060
Reliably Transmitting (NAT) to 192.168.15.27:5060:
BYE sip:215@115.167.21.82:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport
Max-Forwards: 70
From: sip:211@192.168.15.118;transport=UDP;tag=as10377813
To: sip:215@192.168.15.118;transport=UDP;tag=50312112
Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.11.0
Proxy-Authorization: Digest username=“215”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.15.118”, nonce="", response="c897390cc8e4f674d7e9cd1efa7319a6"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

Retransmitting #1 (NAT) to 192.168.15.27:5060:
BYE sip:215@115.167.21.82:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport
Max-Forwards: 70
From: sip:211@192.168.15.118;transport=UDP;tag=as10377813
To: sip:215@192.168.15.118;transport=UDP;tag=50312112
Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.11.0
Proxy-Authorization: Digest username=“215”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.15.118”, nonce="", response="c897390cc8e4f674d7e9cd1efa7319a6"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from UDP:192.168.15.27:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport=5060
Contact: sip:215@115.167.21.82:5060;transport=UDP
To: sip:215@192.168.15.118;transport=UDP;tag=50312112
From: sip:211@192.168.15.118;transport=UDP;tag=as10377813
Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
CSeq: 102 BYE
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0

<------------->
— (9 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.’ Method: ACK

<— SIP read from UDP:192.168.15.27:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport=5060
Contact: sip:215@115.167.21.82:5060;transport=UDP
To: sip:215@192.168.15.118;transport=UDP;tag=50312112
From: sip:211@192.168.15.118;transport=UDP;tag=as10377813
Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
CSeq: 102 BYE
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK01e7e47d;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as73902c1e
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK01e7e47d;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as73902c1e
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.15.71:45616 —>
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: c51761a244f0dc9639f87a5cf9af1b3f@192.168.15.71
CSeq: 9273 OPTIONS
From: “211” sip:211@192.168.15.118;tag=740019322
To: “211” sip:211@192.168.15.118
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK5ac51b5cca5ad84320fc342692fdbb2d353134;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Looking for s in default (domain 192.168.15.118)

<— Transmitting (NAT) to 192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK5ac51b5cca5ad84320fc342692fdbb2d353134;received=192.168.15.71;rport=45616
From: “211” sip:211@192.168.15.118;tag=740019322
To: “211” sip:211@192.168.15.118;tag=as1bed6ef2
Call-ID: c51761a244f0dc9639f87a5cf9af1b3f@192.168.15.71
CSeq: 9273 OPTIONS
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:192.168.15.118:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘c51761a244f0dc9639f87a5cf9af1b3f@192.168.15.71’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK6cd92a69;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as6b3be4fc
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 7985a87e4599439b4d6449d53d5b940a@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Really destroying SIP dialog ‘ed5a7d84b136ece8e4bb07860b907b49@192.168.15.71’ Method: OPTIONS

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK6cd92a69;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as6b3be4fc
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 7985a87e4599439b4d6449d53d5b940a@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘7985a87e4599439b4d6449d53d5b940a@192.168.15.118:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK5a693bf9;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as258e14cc
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 7ef504a71c3bd1bd77b7a18d5741ac3d@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK5a693bf9;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as258e14cc
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 7ef504a71c3bd1bd77b7a18d5741ac3d@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘7ef504a71c3bd1bd77b7a18d5741ac3d@192.168.15.118:5060’ Method: OPTIONS
Really destroying SIP dialog ‘2732e4564ce8534c5765a456045a9960@192.168.15.118:5060’ Method: BYE
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK3c24b1cb;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as54c6581a
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 7742af6c24e3e9a33efa0ba73051a29e@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK3c24b1cb;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as54c6581a
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 7742af6c24e3e9a33efa0ba73051a29e@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘7742af6c24e3e9a33efa0ba73051a29e@192.168.15.118:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.15.71:45616 —>
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: 18347a0db6841423591b08250847a1e0@192.168.15.71
CSeq: 3824 OPTIONS
From: “211” sip:211@192.168.15.118;tag=841349553
To: “211” sip:211@192.168.15.118
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK79bb575a5d0f832291b861987849d9a2353134;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Looking for s in default (domain 192.168.15.118)

<— Transmitting (NAT) to 192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK79bb575a5d0f832291b861987849d9a2353134;received=192.168.15.71;rport=45616
From: “211” sip:211@192.168.15.118;tag=841349553
To: “211” sip:211@192.168.15.118;tag=as3f82607b
Call-ID: 18347a0db6841423591b08250847a1e0@192.168.15.71
CSeq: 3824 OPTIONS
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:192.168.15.118:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘18347a0db6841423591b08250847a1e0@192.168.15.71’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK5c3f2dd7;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as6574e057
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 274a65d40a8be3a6640563fe01c2c576@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK5c3f2dd7;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as6574e057
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 274a65d40a8be3a6640563fe01c2c576@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘274a65d40a8be3a6640563fe01c2c576@192.168.15.118:5060’ Method: OPTIONS
Really destroying SIP dialog ‘162194e2b5ce568ffb6d14160e5f2184@192.168.15.71’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK5a8a8009;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as7679352e
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 457d48dc05fac68e66fbe8ac306c1578@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK5a8a8009;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as7679352e
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 457d48dc05fac68e66fbe8ac306c1578@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘457d48dc05fac68e66fbe8ac306c1578@192.168.15.118:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.15.27:5060 —>

<------------->
Reliably Transmitting (NAT) to 192.168.15.27:5060:
OPTIONS sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK5e85260b;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as278c95d1
To: sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 3e4b5e291c1220bb152edff61a6e97d8@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.27:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK5e85260b;rport=5060
Contact: sip:192.168.15.27:5060
To: sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP;tag=17419722
From: "asterisk"sip:asterisk@192.168.15.118;tag=as278c95d1
Call-ID: 3e4b5e291c1220bb152edff61a6e97d8@192.168.15.118:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.39 r16838
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘3e4b5e291c1220bb152edff61a6e97d8@192.168.15.118:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1edd900b;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as3d0d122c
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 351988395e3e01fd6180a9372bbd19d7@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1edd900b;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as3d0d122c
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 351988395e3e01fd6180a9372bbd19d7@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘351988395e3e01fd6180a9372bbd19d7@192.168.15.118:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.15.71:45616 —>
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: 09809b7ed9a68b7f270f33f1d8de13dc@192.168.15.71
CSeq: 4619 OPTIONS
From: “211” sip:211@192.168.15.118;tag=4017391219
To: “211” sip:211@192.168.15.118
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK40d81d1686d23c9133e52f4180c2accb353134;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Looking for s in default (domain 192.168.15.118)

<— Transmitting (NAT) to 192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK40d81d1686d23c9133e52f4180c2accb353134;received=192.168.15.71;rport=45616
From: “211” sip:211@192.168.15.118;tag=4017391219
To: “211” sip:211@192.168.15.118;tag=as52fe1845
Call-ID: 09809b7ed9a68b7f270f33f1d8de13dc@192.168.15.71
CSeq: 4619 OPTIONS
Server: Asterisk PBX 1.8.11.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:192.168.15.118:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘09809b7ed9a68b7f270f33f1d8de13dc@192.168.15.71’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK13b8cca8;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as6e6638f8
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 76f6f373769041c3462ebb676641f1b7@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.15.71:45616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK13b8cca8;rport=5060;received=192.168.15.118
From: “asterisk” sip:asterisk@192.168.15.118;tag=as6e6638f8
To: sip:211@192.168.15.71:45616;transport=udp
Call-ID: 76f6f373769041c3462ebb676641f1b7@192.168.15.118:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘76f6f373769041c3462ebb676641f1b7@192.168.15.118:5060’ Method: OPTIONS
Really destroying SIP dialog ‘9eeee094f46eec920ac462e291314bde@192.168.15.71’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK47a8a134;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.15.118;tag=as76426de6
To: sip:211@192.168.15.71:45616;transport=udp
Contact: sip:asterisk@192.168.15.118:5060
Call-ID: 3a98a25b41dc3b3e699ee4383669e984@192.168.15.118:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.0
Date: Sat, 12 Jan 2013 19:54:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0[/code]

zain · January 13, 2013, 5:06pm

I also tried but didn’t work…

david55 · January 13, 2013, 7:17pm

Please provide a trace which includes the whole call, in particular the invites. It would help if you could remove the OPTIONS and their responses from the trace,

There seem to be retransmissions on the BYEs, which suggests that the network is in distress.

zain · January 14, 2013, 8:59am

[quote=“david55”]Please provide a trace which includes the whole call, in particular the invites. It would help if you could remove the OPTIONS and their responses from the trace,

There seem to be retransmissions on the BYEs, which suggests that the network is in distress.[/quote]

I have tested the scenario on three different networks with the same results…
David can please elaborate which trace you are talking…is it wireshark trace???

david55 · January 14, 2013, 10:43am

The same sort of trace as you provided above, except, this time, with the relevant packets included.

zain · January 16, 2013, 5:14am

Thanks for every one in this forum for anticipating specially for David…I have manged to solve the problem…The two devices “xlite/zoiper” and “android native sip” client uses different default audio codecs.

default codec for xlite is BroadVoice-32
default codec for zoiper is GSM
default codec for android G.711 uLaw

As these devices should use same codec while communicating with each other…In my scenario these devices were using different codecs which results in one way audio (when calling from android to xlite/zoiper).
While creating the SIP accounts in sip.conf while we can enforce two communicating clients to use same audio codec in following manner.

[code][211]
deny=0.0.0.0/0.0.0.0
secret=123456
dtmfmode=rfc2833
canreinvite=no
context=incoming-calls-wildcard
host=dynamic
nat=yes
type=friend
port=5060
qualify=yes
callgroup=
pickupgroup=
permit=0.0.0.0/0.0.0.0
callcounter=yes
faxdetect=no
disallow=all (disable default audio codec)
allow=ulaw (allow uLaw audio codec)

[215]
deny=0.0.0.0/0.0.0.0
secret=123456
dtmfmode=rfc2833
canreinvite=no
context=incoming-calls-wildcard
host=dynamic
nat=no
type=friend
port=5060
qualify=yes
callgroup=
pickupgroup=
permit=0.0.0.0/0.0.0.0
callcounter=yes
faxdetect=no
disallow=all (disable default audio codec)
allow=ulaw (allow uLaw audio codec)
[/code]

We can also configure audio codec settings on client side by selecting the same audio codec on both sides…

Best Regards:
Zain Ali Mughal

mady · February 23, 2015, 10:20am

Hello,

I want to build an android voip client by using this voip sip android example. Can it be used as a mobile extension in an Asterisk PBX?

Thanks in advance