[asterisk-users] Asterisk with PJSIP
エムディーシー太郎
mdc.taro at gmail.com
Wed Sep 10 05:00:32 CDT 2014
Thank you for your reply.
After setting "pjsip set logger on",
the following message is displayed.
It seems that the 9002(SIP client) refuse INVITE message.
Are SIP methods too many?
Thanks,
MMEEGGAA
--------------------
<--- Transmitting SIP request (449 bytes) to UDP:192.168.177.180:16060 --->
OPTIONS sip:9001 at 192.168.177.180:16060 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPj3c090235-9385-4cea-9c56-83f2755606d1
From: <sip:1e00dac6-cbe1-4bfa-96f4-b24085b848df at 192.168.177.190
>;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b
To: <sip:9001 at 192.168.177.180>
Contact: <sip:1e00dac6-cbe1-4bfa-96f4-b24085b848df at 192.168.177.190:5060>
Call-ID: 6294a204-c645-41a8-ac27-e5aaa5f6c08c
CSeq: 44261 OPTIONS
Content-Length: 0
<--- Received SIP response (333 bytes) from UDP:192.168.177.180:16060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPj3c090235-9385-4cea-9c56-83f2755606d1
From: <sip:1e00dac6-cbe1-4bfa-96f4-b24085b848df at 192.168.177.190
>;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b
To: <sip:9001 at 192.168.177.180>;tag=EF1my
Call-ID: 6294a204-c645-41a8-ac27-e5aaa5f6c08c
CSeq: 44261 OPTIONS
<--- Transmitting SIP request (443 bytes) to UDP:192.168.177.191:5060 --->
OPTIONS sip:9002 at 192.168.177.191 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPj98982748-5a3d-4a4c-9d90-ad4988e8d933
From: <sip:b4bf1474-a5c6-475b-985b-5ff48c550a7f at 192.168.177.190
>;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49
To: <sip:9002 at 192.168.177.191>
Contact: <sip:b4bf1474-a5c6-475b-985b-5ff48c550a7f at 192.168.177.190:5060>
Call-ID: eb7a8bee-b06f-4a30-a5d9-eb437e104f76
CSeq: 16803 OPTIONS
Content-Length: 0
<--- Received SIP response (333 bytes) from UDP:192.168.177.191:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPj98982748-5a3d-4a4c-9d90-ad4988e8d933
From: <sip:b4bf1474-a5c6-475b-985b-5ff48c550a7f at 192.168.177.190
>;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49
To: <sip:9002 at 192.168.177.191>;tag=hSl7b
Call-ID: eb7a8bee-b06f-4a30-a5d9-eb437e104f76
CSeq: 16803 OPTIONS
<--- Received SIP request (1170 bytes) from UDP:192.168.177.180:16060 --->
INVITE sip:9002 at 192.168.177.190 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.G4FYrVaiY;rport
From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
To: sip:9002 at 192.168.177.190
CSeq: 20 INVITE
Call-ID: 2c1KLd1INo
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 633
User-Agent: Linphone/3.6.99 (belle-sip/1.2.4)
Contact: <sip:9001 at 192.168.177.180:16060
>;+sip.instance="<urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2>"
v=0
o=9001 2189 3894 IN IP4 192.168.177.180
s=Talk
c=IN IP4 192.168.177.180
t=0 0
a=ice-pwd:000030810000373f00004003
a=ice-ufrag:00004c02
m=audio 17590 RTP/AVP 0 110 3 8 101
c=IN IP4 61.117.138.218
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:21548
a=candidate:1 1 UDP 2130706431 192.168.177.180 7078 typ host
a=candidate:1 2 UDP 2130706430 192.168.177.180 7079 typ host
a=candidate:2 1 UDP 1694498815 XXX.XXX.XXX.XXX 17590 typ srflx raddr
192.168.177.180 rport 7078
a=candidate:2 2 UDP 1694498814 XXX.XXX.XXX.XXX 21548 typ srflx raddr
192.168.177.180 rport 7079
<--- Transmitting SIP response (431 bytes) to UDP:192.168.177.180:16060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.177.180:16060
;rport;received=192.168.177.180;branch=z9hG4bK.G4FYrVaiY
Call-ID: 2c1KLd1INo
From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
To: <sip:9002 at 192.168.177.190>;tag=z9hG4bK.G4FYrVaiY
CSeq: 20 INVITE
WWW-Authenticate: Digest
realm="asterisk",nonce="1410336707/cd97e01134333d7d5769e49872f750a4",opaque="58e109d10f49a371",algorithm=md5,qop="auth"
Content-Length: 0
<--- Received SIP request (373 bytes) from UDP:192.168.177.180:16060 --->
ACK sip:9002 at 192.168.177.190 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.G4FYrVaiY;rport
Call-ID: 2c1KLd1INo
From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
To: <sip:9002 at 192.168.177.190>;tag=z9hG4bK.G4FYrVaiY
Contact: <sip:9001 at 192.168.177.180:16060
>;+sip.instance="<urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2>"
Max-Forwards: 70
CSeq: 20 ACK
<--- Received SIP request (1428 bytes) from UDP:192.168.177.180:16060 --->
INVITE sip:9002 at 192.168.177.190 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.00879CXMH;rport
From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
To: sip:9002 at 192.168.177.190
CSeq: 21 INVITE
Call-ID: 2c1KLd1INo
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 633
User-Agent: Linphone/3.6.99 (belle-sip/1.2.4)
Contact: <sip:9001 at 192.168.177.180:16060
>;+sip.instance="<urn:uuid:aa35e441-baa3-4e07-af2f-cd1ff2dc99a2>"
Authorization: Digest realm="asterisk",
nonce="1410336707/cd97e01134333d7d5769e49872f750a4",
opaque="58e109d10f49a371", username="9001", uri="sip:9002 at 192.168.177.190",
response="a822c66fe1c1d30492beeb08e6daaae5", cnonce="faaa92d5",
nc=00000001, qop=auth
v=0
o=9001 2189 3894 IN IP4 192.168.177.180
s=Talk
c=IN IP4 192.168.177.180
t=0 0
a=ice-pwd:000030810000373f00004003
a=ice-ufrag:00004c02
m=audio 17590 RTP/AVP 0 110 3 8 101
c=IN IP4 61.117.138.218
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:21548
a=candidate:1 1 UDP 2130706431 192.168.177.180 7078 typ host
a=candidate:1 2 UDP 2130706430 192.168.177.180 7079 typ host
a=candidate:2 1 UDP 1694498815 XXX.XXX.XXX.XXX 17590 typ srflx raddr
192.168.177.180 rport 7078
a=candidate:2 2 UDP 1694498814 XXX.XXX.XXX.XXX 21548 typ srflx raddr
192.168.177.180 rport 7079
<--- Transmitting SIP response (256 bytes) to UDP:192.168.177.180:16060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.177.180:16060
;rport;received=192.168.177.180;branch=z9hG4bK.00879CXMH
Call-ID: 2c1KLd1INo
From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
To: <sip:9002 at 192.168.177.190>
CSeq: 21 INVITE
Content-Length: 0
-- Executing [9002 at internal:1] Dial("PJSIP/9001-00000006",
"PJSIP/9002,20") in new stack
-- Called PJSIP/9002
debug
== debug1 (0|0:0/0/0)
== debug2 (2|1:0/0/0)
<--- Transmitting SIP request (910 bytes) to UDP:192.168.177.191:5060 --->
INVITE sip:9002 at 192.168.177.191 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124
From: <sip:9001 at 192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002 at 192.168.177.191>
Contact: <sip:b9b2034d-e72e-4a18-bcd5-4e84d967afbc at 192.168.177.190:5060>
Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9
CSeq: 18942 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 273
v=0
o=- 278980317 278980317 IN IP4 localhost.localdomain
s=Asterisk
c=IN IP4 192.168.177.190
t=0 0
m=audio 10338 RTP/AVP 0 101
c=IN IP4 192.168.177.190
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP response (309 bytes) from UDP:192.168.177.191:5060 --->
SIP/2.0 400 Bad request
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124
From: <sip:9001 at 192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002 at 192.168.177.191>;tag=bajXh
Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9
CSeq: 18942 INVITE
<--- Transmitting SIP request (339 bytes) to UDP:192.168.177.191:5060 --->
ACK sip:9002 at 192.168.177.191 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124
From: <sip:9001 at 192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002 at 192.168.177.191>;tag=bajXh
Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9
CSeq: 18942 ACK
Content-Length: 0
-- PJSIP/9002-00000007 answered PJSIP/9001-00000006
-- PJSIP/9002-00000007 answered PJSIP/9001-00000006
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'PJSIP/9001-00000006' status is
'CHANUNAVAIL'
<--- Transmitting SIP response (334 bytes) to UDP:192.168.177.180:16060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.177.180:16060
;rport;received=192.168.177.180;branch=z9hG4bK.00879CXMH
Call-ID: 2c1KLd1INo
From: <sip:9001 at 192.168.177.190>;tag=~yIpJRFo9
To: <sip:9002 at 192.168.177.190>;tag=aead10f9-2194-48dd-bf38-0cc78bff561f
CSeq: 21 INVITE
Reason: Q.850;cause=34
Content-Length: 0
<--- Received SIP response (306 bytes) from UDP:192.168.177.191:5060 --->
SIP/2.0 400 Bad request
Via: SIP/2.0/UDP 192.168.177.190:5060
;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124
From: <sip:9001 at 192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002 at 192.168.177.191>;tag=bajXh
Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9
CSeq: 18942 ACK
--------------------
2014-09-05 19:24 GMT+09:00 Joshua Colp <jcolp at digium.com>:
> エムディーシー太郎 wrote:
>
>> Hi All,
>>
>> I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code
>> on CentOS7.
>> --https://wiki.asterisk.org/wiki/display/AST/Building+and+
>> Installing+pjproject
>>
>
> <snip>
>
>
>> ----------
>> 2. dial from 9001 to 9002
>>
>> *CLI> -- Executing [9002 at internal:1] Dial("PJSIP/9001-00000000",
>> "PJSIP/9002,20") in new stack
>> -- Called PJSIP/9002
>> == Everyone is busy/congested at this time (1:0/0/1)
>> -- Auto fallthrough, channel 'PJSIP/9001-00000000' status is
>> 'CHANUNAVAIL'
>>
>
> What is shown if you do "pjsip set logger on" and then try to place the
> call?
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
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