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[asterisk-users] Asterisk with PJSIP


 
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mdc.taro at gmail.com
Guest
PostPosted: Fri Sep 05, 2014 4:55 am    Post subject: [asterisk-users] Asterisk with PJSIP Reply with quote
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


The installation is OK.
But the connected SIP cilents (both Linphone on Windows7) cannot communicate.


I hope your comment such as the testing for resolving the problem.


My status is the following(1 and 2).
Why 'Everyone is busy/congested at this time (1:0/0/1)'?
(1:0/0/1<---num.nochan is 1.)


----------
1. endpoint
*CLI> pjsip show endpoints
 Endpoint:  <Endpoint/CID.....................................>  <State.....>  <Channels.>
    I/OAuth:  <AuthId/UserName...........................................................>
        Aor:  <Aor............................................>  <MaxContact>
      Contact:  <Aor/ContactUri...............................>  <Status....>  <RTT(ms)..>
  Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress..................>
   Identify:  <MatchList.................................................................>
    Channel:  <ChannelId......................................>  <State.....>  <Time(sec)>
        Exten: <DialedExten...........>  CLCID: <ConnectedLineCID.......>
 =========================================================================================
 Endpoint:  9001                                                 Not in use    0 of inf
     InAuth:  auth9001/9001
        Aor:  9001                                              10
      Contact:  9001/sip:9001@192.168.177.180:16060              Avail              25.048
  Transport:  transport-udp             udp      0      0  0.0.0.0:5060
 Endpoint:  9002                                                 Not in use    0 of inf
     InAuth:  auth9002/9002
        Aor:  9002                                              10
      Contact:  9002/sip:9002@192.168.177.189 ([email]sip%3A9002@192.168.177.189[/email])                    Avail              24.210
  Transport:  transport-udp             udp      0      0  0.0.0.0:5060


----------
2. dial from 9001 to 9002



*CLI>     -- Executing [9002@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'
----------



Thanks,
MMEEGGAA
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rainer.piper at soho-p...
Guest
PostPosted: Fri Sep 05, 2014 5:19 am    Post subject: [asterisk-users] Asterisk with PJSIP Reply with quote
Hi,

can you check the Linphone Extension 9002!!

The port is missing!
Contact:  9002/sip:9002@192.168.177.189 ([email]sip%3A9002@192.168.177.189[/email])Confused???                    Avail              24.210

Regards
Rainer

Am 05.09.2014 um 11:55 schrieb エムディーシー太郎:

Quote:
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


The installation is OK.
But the connected SIP cilents (both Linphone on Windows7) cannot communicate.


I hope your comment such as the testing for resolving the problem.


My status is the following(1 and 2).
Why 'Everyone is busy/congested at this time (1:0/0/1)'?
(1:0/0/1<---num.nochan is 1.)


----------
1. endpoint
*CLI> pjsip show endpoints
 Endpoint:  <Endpoint/CID.....................................>  <State.....>  <Channels.>
    I/OAuth:  <AuthId/UserName...........................................................>
        Aor:  <Aor............................................>  <MaxContact>
      Contact:  <Aor/ContactUri...............................>  <Status....>  <RTT(ms)..>
  Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress..................>
   Identify:  <MatchList.................................................................>
    Channel:  <ChannelId......................................>  <State.....>  <Time(sec)>
        Exten: <DialedExten...........>  CLCID: <ConnectedLineCID.......>
 =========================================================================================
 Endpoint:  9001                                                 Not in use    0 of inf
     InAuth:  auth9001/9001
        Aor:  9001                                              10
      Contact:  9001/sip:9001@192.168.177.180:16060              Avail              25.048
  Transport:  transport-udp             udp      0      0  0.0.0.0:5060
 Endpoint:  9002                                                 Not in use    0 of inf
     InAuth:  auth9002/9002
        Aor:  9002                                              10
      Contact:  9002/sip:9002@192.168.177.189 ([email]sip%3A9002@192.168.177.189[/email])                    Avail              24.210
  Transport:  transport-udp             udp      0      0  0.0.0.0:5060


----------
2. dial from 9001 to 9002



*CLI>     -- Executing [9002@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'
----------



Thanks,
MMEEGGAA








--
Rainer Piper
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
Phone: [url=callto:004922897167161]+49 228 97167161[/url]
P2P: [url=sip:7000@sip.soho-piper.de:5072]sip:7000@sip.soho-piper.de:5072[/url] (pjsip-test)
Back to top
jcolp at digium.com
Guest
PostPosted: Fri Sep 05, 2014 5:24 am    Post subject: [asterisk-users] Asterisk with PJSIP Reply with quote
エムディーシー太郎 wrote:
Quote:
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>

Quote:

----------
2. dial from 9001 to 9002

*CLI> -- Executing [9002@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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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mdc.taro at gmail.com
Guest
PostPosted: Wed Sep 10, 2014 5:01 am    Post subject: [asterisk-users] Asterisk with PJSIP Reply with quote
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@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@192.168.177.190 ([email]sip%3A1e00dac6-cbe1-4bfa-96f4-b24085b848df@192.168.177.190[/email])>;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b
To: <sip:9001@192.168.177.180 ([email]sip%3A9001@192.168.177.180[/email])>
Contact: <sip:1e00dac6-cbe1-4bfa-96f4-b24085b848df@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@192.168.177.190 ([email]sip%3A1e00dac6-cbe1-4bfa-96f4-b24085b848df@192.168.177.190[/email])>;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b
To: <sip:9001@192.168.177.180 ([email]sip%3A9001@192.168.177.180[/email])>;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@192.168.177.191 ([email]sip%3A9002@192.168.177.191[/email]) 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@192.168.177.190 ([email]sip%3Ab4bf1474-a5c6-475b-985b-5ff48c550a7f@192.168.177.190[/email])>;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49
To: <sip:9002@192.168.177.191 ([email]sip%3A9002@192.168.177.191[/email])>
Contact: <sip:b4bf1474-a5c6-475b-985b-5ff48c550a7f@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@192.168.177.190 ([email]sip%3Ab4bf1474-a5c6-475b-985b-5ff48c550a7f@192.168.177.190[/email])>;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49
To: <sip:9002@192.168.177.191 ([email]sip%3A9002@192.168.177.191[/email])>;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@192.168.177.190 ([email]sip%3A9002@192.168.177.190[/email]) SIP/2.0
Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.G4FYrVaiY;rport
From: <sip:9001@192.168.177.190 ([email]sip%3A9001@192.168.177.190[/email])>;tag=~yIpJRFo9
To: sip:9002@192.168.177.190 ([email]sip%3A9002@192.168.177.190[/email])
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@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@192.168.177.190 ([email]sip%3A9001@192.168.177.190[/email])>;tag=~yIpJRFo9
To: <sip:9002@192.168.177.190 ([email]sip%3A9002@192.168.177.190[/email])>;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@192.168.177.190 ([email]sip%3A9002@192.168.177.190[/email]) SIP/2.0
Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.G4FYrVaiY;rport
Call-ID: 2c1KLd1INo
From: <sip:9001@192.168.177.190 ([email]sip%3A9001@192.168.177.190[/email])>;tag=~yIpJRFo9
To: <sip:9002@192.168.177.190 ([email]sip%3A9002@192.168.177.190[/email])>;tag=z9hG4bK.G4FYrVaiY
Contact: <sip:9001@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@192.168.177.190 ([email]sip%3A9002@192.168.177.190[/email]) SIP/2.0
Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.00879CXMH;rport
From: <sip:9001@192.168.177.190 ([email]sip%3A9001@192.168.177.190[/email])>;tag=~yIpJRFo9
To: sip:9002@192.168.177.190 ([email]sip%3A9002@192.168.177.190[/email])
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@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@192.168.177.190 ([email]sip%3A9002@192.168.177.190[/email])", 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@192.168.177.190 ([email]sip%3A9001@192.168.177.190[/email])>;tag=~yIpJRFo9
To: <sip:9002@192.168.177.190 ([email]sip%3A9002@192.168.177.190[/email])>
CSeq: 21 INVITE
Content-Length:  0




    -- Executing [9002@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@192.168.177.191 ([email]sip%3A9002@192.168.177.191[/email]) SIP/2.0
Via: SIP/2.0/UDP 192.168.177.190:5060;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124
From: <sip:9001@192.168.177.190 ([email]sip%3A9001@192.168.177.190[/email])>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002@192.168.177.191 ([email]sip%3A9002@192.168.177.191[/email])>
Contact: <sip:b9b2034d-e72e-4a18-bcd5-4e84d967afbc@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@192.168.177.190 ([email]sip%3A9001@192.168.177.190[/email])>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002@192.168.177.191 ([email]sip%3A9002@192.168.177.191[/email])>;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@192.168.177.191 ([email]sip%3A9002@192.168.177.191[/email]) SIP/2.0
Via: SIP/2.0/UDP 192.168.177.190:5060;rport;branch=z9hG4bKPjf90aa758-0538-4145-b2c9-7d945da6f124
From: <sip:9001@192.168.177.190 ([email]sip%3A9001@192.168.177.190[/email])>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002@192.168.177.191 ([email]sip%3A9002@192.168.177.191[/email])>;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@192.168.177.190 ([email]sip%3A9001@192.168.177.190[/email])>;tag=~yIpJRFo9
To: <sip:9002@192.168.177.190 ([email]sip%3A9002@192.168.177.190[/email])>;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@192.168.177.190 ([email]sip%3A9001@192.168.177.190[/email])>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002@192.168.177.191 ([email]sip%3A9002@192.168.177.191[/email])>;tag=bajXh
Call-ID: 8a59fbf2-9cae-48b7-88de-1a2fed4ec7c9
CSeq: 18942 ACK

--------------------



2014-09-05 19:24 GMT+09:00 Joshua Colp <jcolp@digium.com (jcolp@digium.com)>:
Quote:
エムディーシー太郎 wrote:
Quote:
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>

Quote:

----------
2. dial from 9001 to 9002

*CLI>     -- Executing [9002@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

--
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marie at vtl.ee
Guest
PostPosted: Tue Sep 23, 2014 11:32 am    Post subject: [asterisk-users] Asterisk with PJSIP Reply with quote
Hi,

yes, the 9002 Linphone answers "400 Bad Request" to Asterisk's INVITE.
Does it work in the other direction (9002 calling 9001)?
Have you checked your codecs (Linphone is offering PCMA, PCMU and GSM, Asterisk just PCMU)?

Apparently, for debug logging Linphone, you should
• open a windows shell prompt
• go to c:\Program Files\Linphone
• start Linphone like this: bin/linphone --logfile "c:\Temp\logs.txt"

So maybe this way you can see some more information.

--

marie

On 10.09.2014, at 13:00, エムディーシー太郎 <mdc.taro@gmail.com> wrote:

Quote:
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@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@192.168.177.190>;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b
To: <sip:9001@192.168.177.180>
Contact: <sip:1e00dac6-cbe1-4bfa-96f4-b24085b848df@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@192.168.177.190>;tag=ef5b7ffd-6d54-4c46-8100-635d862f699b
To: <sip:9001@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@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@192.168.177.190>;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49
To: <sip:9002@192.168.177.191>
Contact: <sip:b4bf1474-a5c6-475b-985b-5ff48c550a7f@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@192.168.177.190>;tag=73ae5dfe-d5b7-4c9b-9529-5198492f5c49
To: <sip:9002@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@192.168.177.190 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.G4FYrVaiY;rport
From: <sip:9001@192.168.177.190>;tag=~yIpJRFo9
To: sip:9002@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@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@192.168.177.190>;tag=~yIpJRFo9
To: <sip:9002@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@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@192.168.177.190>;tag=~yIpJRFo9
To: <sip:9002@192.168.177.190>;tag=z9hG4bK.G4FYrVaiY
Contact: <sip:9001@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@192.168.177.190 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.180:16060;branch=z9hG4bK.00879CXMH;rport
From: <sip:9001@192.168.177.190>;tag=~yIpJRFo9
To: sip:9002@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@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@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@192.168.177.190>;tag=~yIpJRFo9
To: <sip:9002@192.168.177.190>
CSeq: 21 INVITE
Content-Length: 0


-- Executing [9002@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@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@192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002@192.168.177.191>
Contact: <sip:b9b2034d-e72e-4a18-bcd5-4e84d967afbc@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@192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002@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@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@192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002@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@192.168.177.190>;tag=~yIpJRFo9
To: <sip:9002@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@192.168.177.190>;tag=215003c4-08ad-472f-b366-bfaa8bfe303b
To: <sip:9002@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@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@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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
_____________________________________________________________________
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


--
_____________________________________________________________________
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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