Why asterisk not properly working with android sip client?

3
votes

Asterisk= 1.8.11.0

Android= 2.3/4.0.3

Android Sip client=Native Android sip client/sipdemo

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. On the other hand i have tested this scenario on elastix (which also uses asterisk 1.8.11.0) with no problem in audio. 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 .

<--- 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:[email protected]>;tag=as05233e7d
To: <sip:[email protected]:45616;transport=udp>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.15.71:45616 --->
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: [email protected]
CSeq: 7757 OPTIONS
From: "211" <sip:[email protected]>;tag=1758376458
To: "211" <sip:[email protected]>
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:[email protected]>;tag=1758376458
To: "211" <sip:[email protected]>;tag=as6a8e1b47
Call-ID: [email protected]
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


<--- 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:[email protected]>;tag=as167765df
To: <sip:[email protected]:45616;transport=udp>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Really destroying SIP dialog '[email protected]' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:[email protected]: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:[email protected]>;tag=as53340ecf
To: <sip:[email protected]:45616;transport=udp>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>;tag=as53340ecf
To: <sip:[email protected]:45616;transport=udp>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.15.71:45616 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKad00f5add2a1f2e940a91e9a3f827b26353134
CSeq: 5511 BYE
From: "211" <sip:[email protected]>;tag=2465683119
To: <sip:[email protected]>;tag=as573c52b3
Call-ID: [email protected]
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 '[email protected]' 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:[email protected]>;tag=2465683119
To: <sip:[email protected]>;tag=as573c52b3
Call-ID: [email protected]
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 '[email protected]:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:[email protected]: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:[email protected]: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:[email protected]>;tag=as404f0eb0
To: <sip:[email protected]:5060;rinstance=49cb21467969bef8;transport=UDP>;tag=96055240
Call-ID: [email protected]: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:[email protected]: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:[email protected]>;tag=as404f0eb0
To: <sip:[email protected]:5060;rinstance=49cb21467969bef8;transport=UDP>;tag=96055240
Call-ID: [email protected]: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:[email protected]:5060;rinstance=49cb21467969bef8;transport=UDP>
To: <sip:[email protected]:5060;rinstance=49cb21467969bef8;transport=UDP>;tag=96055240
From: "device"<sip:[email protected]>;tag=as404f0eb0
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: Zoiper for Windows 2.39 r16838
Content-Length: 0
<------------->
-- (9 headers 0 lines) ---
Really destroying SIP dialog '[email protected]: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:[email protected]:5060;rinstance=49cb21467969bef8;transport=UDP>
To: <sip:[email protected]:5060;rinstance=49cb21467969bef8;transport=UDP>;tag=96055240
From: "device"<sip:[email protected]>;tag=as404f0eb0
Call-ID: [email protected]: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:[email protected]: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:[email protected]>;tag=as4f0724aa
To: <sip:[email protected]:45616;transport=udp>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>;tag=as4f0724aa
To: <sip:[email protected]:45616;transport=udp>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.15.71:45616 --->
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: [email protected]
CSeq: 5815 OPTIONS
From: "211" <sip:[email protected]>;tag=3109248316
To: "211" <sip:[email protected]>
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:[email protected]>;tag=3109248316
To: "211" <sip:[email protected]>;tag=as51223faf
Call-ID: [email protected]
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 '[email protected]' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:[email protected]: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:[email protected]>;tag=as7a9a1ea3
To: <sip:[email protected]:45616;transport=udp>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>;tag=as7a9a1ea3
To: <sip:[email protected]:45616;transport=udp>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Really destroying SIP dialog '[email protected]' Method: BYE
Really destroying SIP dialog '[email protected]' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:[email protected]: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:[email protected]>;tag=as5367b37c
To: <sip:[email protected]:45616;transport=udp>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>;tag=as5367b37c
To: <sip:[email protected]:45616;transport=udp>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

When calling from pc (zoiper) to android

<--- SIP read from UDP:192.168.15.71:45616 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKe2f448b90c482342407298e309be5aed353134
CSeq: 1 BYE
From: <sip:[email protected]:45616;transport=udp>;tag=4162167884
To: "device" <sip:[email protected]>;tag=as5805dc66
Call-ID: [email protected]:5060
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Sending to 192.168.15.71:45616 (NAT)
Scheduling destruction of SIP dialog '[email protected]: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:[email protected]:45616;transport=udp>;tag=4162167884
To: "device" <sip:[email protected]>;tag=as5805dc66
Call-ID: [email protected]: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:[email protected]: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:[email protected]:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport
Max-Forwards: 70
From: <sip:[email protected];transport=UDP>;tag=as10377813
To: <sip:[email protected];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:[email protected]:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport
Max-Forwards: 70
From: <sip:[email protected];transport=UDP>;tag=as10377813
To: <sip:[email protected];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:[email protected]:5060;transport=UDP>
To: <sip:[email protected];transport=UDP>;tag=50312112
From: <sip:[email protected];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:[email protected]:5060;transport=UDP>
To: <sip:[email protected];transport=UDP>;tag=50312112
From: <sip:[email protected];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:[email protected]: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:[email protected]>;tag=as73902c1e
To: <sip:[email protected]:45616;transport=udp>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>;tag=as73902c1e
To: <sip:[email protected]:45616;transport=udp>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.15.71:45616 --->
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: [email protected]
CSeq: 9273 OPTIONS
From: "211" <sip:[email protected]>;tag=740019322
To: "211" <sip:[email protected]>
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:[email protected]>;tag=740019322
To: "211" <sip:[email protected]>;tag=as1bed6ef2
Call-ID: [email protected]
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

<--- 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:[email protected]>;tag=as54c6581a
To: <sip:[email protected]:45616;transport=udp>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.15.71:45616 --->
OPTIONS sip:192.168.15.118 SIP/2.0
Call-ID: [email protected]
CSeq: 3824 OPTIONS
From: "211" <sip:[email protected]>;tag=841349553
To: "211" <sip:[email protected]>
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)


<------------->
--- (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:[email protected]>;tag=4017391219
To: "211" <sip:[email protected]>;tag=as52fe1845
Call-ID: [email protected]
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 '[email protected]' in 32000 ms (Method: OPTIONS)
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:[email protected]: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:[email protected]>;tag=as6e6638f8
To: <sip:[email protected]:45616;transport=udp>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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:[email protected]>;tag=as6e6638f8
To: <sip:[email protected]:45616;transport=udp>
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
Really destroying SIP dialog '[email protected]' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.15.71:45616:
OPTIONS sip:[email protected]: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:[email protected]>;tag=as76426de6
To: <sip:[email protected]:45616;transport=udp>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]: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

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
3
androidvoipasterisk

3 Answers

3
votes

Thanks for every one in this forum for anticipating...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 is 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.

[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)

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

0
votes

Well,

Configuration wise, it appear that your configuration is ok. However, what seems odd to me is the following IP address inside your trace - 115.167.21.82

Can you please elaborate what this IP address is? My assumption is that you are located behind a NAT firewall and this is your external IP address. In that case, look in /etc/asterisk/sip.conf for the "localnet" definition, and define your local network address range.

When working with Elastix, it is able to setup that one automatically - but when doing a manual install, you have to set that one up.

Nir S