Why asterisk not properly working with android sip client?
14,066
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..
Comments
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Zain almost 2 years
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: OPTIONSWhen 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: 0I 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,HangupMy 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