[asterisk-users] trying to connect to asterisk with softphone (logs, etc)

thufir hawat.thufir at gmail.com
Mon Mar 23 18:39:43 CDT 2015




In the Asterisk log I see:

    ---
    [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29]
    <--- SIP read from UDP:198.38.7.34:5065 --->
    SIP/2.0 200 OK
    To: <sip:16046289850 at sip.babytel.ca>;tag=sd3D4swKRc
    From: <sip:16046289850 at sip.babytel.ca>;tag=as07c833c5
    Via: SIP/2.0/UDP 96.48.217.39:5060;branch=z9hG4bK13c68eb7;rport
    Call-ID: 5e070a0021f200c72308ddad6fe2521c at 192.168.0.99
    CSeq: 221 REGISTER
    Contact: <sip:16046289850 at 96.48.217.39:5060>;expires=55
    Content-Length: 0
    <------------->
    [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29] --- (8
    headers 0 lines) ---
    [Mar 23 19:25:29] NOTICE[4067] chan_sip.c: Outbound Registration:
    Expiry for nat5.babytel.ca is 55 sec (Scheduling reregistration in 40 s)
    [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29] Really
    destroying SIP dialog
    '5e070a0021f200c72308ddad6fe2521c at 192.168.0.99' Method: REGISTER
    [Mar 23 19:25:44] VERBOSE[4003] asterisk.c: [Mar 23 19:25:44]     --
    Remote UNIX connection
    [Mar 23 19:26:01] VERBOSE[15640] manager.c: [Mar 23 19:26:01]   ==
    Manager 'sendcron' logged on from 127.0.0.1
    [Mar 23 19:26:01] VERBOSE[15640] manager.c: [Mar 23 19:26:01]   ==
    Manager 'sendcron' logged off from 127.0.0.1
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04]
    <--- SIP read from UDP:192.168.0.28:5060 --->
    REGISTER sip:192.168.0.99 SIP/2.0
    Call-ID: d7ab3099e71b65e0ae104cc441aecc25 at 0:0:0:0:0:0:0:0
    CSeq: 4 REGISTER
    From: "201" <sip:201 at 192.168.0.99>;tag=5fbdd638
    To: "201" <sip:201 at 192.168.0.99>
    Max-Forwards: 70
    User-Agent: Jitsi2.6.5390Mac OS X
    Expires: 600
    Contact: "201"
    <sip:201 at 192.168.0.28:5060;transport=udp;registering_acc=192_168_0_99>;expires=600
    Via: SIP/2.0/UDP
    192.168.0.28:5060;branch=z9hG4bK-333334-5e76c348412aa7cadf05777dd72d8a4d
    Authorization: Digest
    username="201",realm="asterisk",nonce="2577db3d",uri="sip:192.168.0.99",response="c3c4a08638f1ac928b1329b312038e75",algorithm=MD5
    Content-Length: 0
    <------------->
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] ---
    (12 headers 0 lines) ---
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04]
    Sending to 192.168.0.28:5060 (NAT)
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04]
    <--- Transmitting (NAT) to 192.168.0.28:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP
    192.168.0.28:5060;branch=z9hG4bK-333334-5e76c348412aa7cadf05777dd72d8a4d;received=192.168.0.28;rport=5060
    From: "201" <sip:201 at 192.168.0.99>;tag=5fbdd638
    To: "201" <sip:201 at 192.168.0.99>;tag=as78b94599
    Call-ID: d7ab3099e71b65e0ae104cc441aecc25 at 0:0:0:0:0:0:0:0
    CSeq: 4 REGISTER
    Server: Asterisk PBX 1.8.29.0-vici
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
    INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
    nonce="43b1ba24"
    Content-Length: 0
    <------------>
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04]
    Scheduling destruction of SIP dialog
    'd7ab3099e71b65e0ae104cc441aecc25 at 0:0:0:0:0:0:0:0' in 32000 ms
    (Method: REGISTER)
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04]
    <--- SIP read from UDP:192.168.0.28:5060 --->
    REGISTER sip:192.168.0.99 SIP/2.0
    Call-ID: d7ab3099e71b65e0ae104cc441aecc25 at 0:0:0:0:0:0:0:0
    CSeq: 5 REGISTER
    From: "201" <sip:201 at 192.168.0.99>;tag=5fbdd638
    To: "201" <sip:201 at 192.168.0.99>
    Max-Forwards: 70
    User-Agent: Jitsi2.6.5390Mac OS X
    Expires: 600
    Contact: "201"
    <sip:201 at 192.168.0.28:5060;transport=udp;registering_acc=192_168_0_99>;expires=600
    Via: SIP/2.0/UDP
    192.168.0.28:5060;branch=z9hG4bK-333334-364fb1c68f6d21e3f71292e300535c15
    Authorization: Digest
    username="201",realm="asterisk",nonce="43b1ba24",uri="sip:192.168.0.99",response="ed23dc12d2effb6d02d5c7aa33a260d5",algorithm=MD5
    Content-Length: 0
    <------------->
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] ---
    (12 headers 0 lines) ---
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04]
    Sending to 192.168.0.28:5060 (NAT)
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04]
    <--- Transmitting (NAT) to 192.168.0.28:5060 --->
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP
    192.168.0.28:5060;branch=z9hG4bK-333334-364fb1c68f6d21e3f71292e300535c15;received=192.168.0.28;rport=5060
    From: "201" <sip:201 at 192.168.0.99>;tag=5fbdd638
    To: "201" <sip:201 at 192.168.0.99>;tag=as78b94599
    Call-ID: d7ab3099e71b65e0ae104cc441aecc25 at 0:0:0:0:0:0:0:0
    CSeq: 5 REGISTER
    Server: Asterisk PBX 1.8.29.0-vici
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
    INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Length: 0
    <------------>
    [Mar 23 19:26:04] NOTICE[4067] chan_sip.c: Registration from '"201"
    <sip:201 at 192.168.0.99>' failed for '192.168.0.28:5060' - Wrong password
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04]
    Scheduling destruction of SIP dialog
    'd7ab3099e71b65e0ae104cc441aecc25 at 0:0:0:0:0:0:0:0' in 32000 ms
    (Method: REGISTER)
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04]
    <--- SIP read from UDP:192.168.0.28:5060 --->
    REGISTER sip:192.168.0.99 SIP/2.0
    Call-ID: d7ab3099e71b65e0ae104cc441aecc25 at 0:0:0:0:0:0:0:0
    CSeq: 6 REGISTER
    From: "201" <sip:201 at 192.168.0.99>;tag=5fbdd638
    To: "201" <sip:201 at 192.168.0.99>
    Max-Forwards: 70
    User-Agent: Jitsi2.6.5390Mac OS X
    Expires: 600
    Contact: "201"
    <sip:201 at 192.168.0.28:5060;transport=udp;registering_acc=192_168_0_99>;expires=600
    Via: SIP/2.0/UDP
    192.168.0.28:5060;branch=z9hG4bK-333334-9eca7bab7d7d48366170c097cbf3280a
    Content-Length: 0
    <------------->
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04] ---
    (11 headers 0 lines) ---
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04]
    Sending to 192.168.0.28:5060 (NAT)
    [Mar 23 19:26:04] VERBOSE[4067] chan_sip.c: [Mar 23 19:26:04]
    <--- Transmitting (NAT) to 192.168.0.28:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP
    192.168.0.28:5060;branch=z9hG4bK-333334-9eca7bab7d7d48366170c097cbf3280a;received=192.168.0.28;rport=5060
    From: "201" <sip:201 at 192.168.0.99>;tag=5fbdd638
    To: "201" <sip:201 at 192.168.0.99>;tag=as78b94599
    Call-ID: d7ab3099e71b65e0ae104cc441aecc25 at 0:0:0:0:0:0:0:0
    CSeq: 6 REGISTER
    Server: Asterisk PBX 1.8.29.0-vici
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
    INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
    nonce="610257fc"
    Content-Length: 0
    <------------>



however, when I look in the console, with "sip set debug on," there's no 
output.  Here is the peer:

    linux-k7qk*CLI>
    linux-k7qk*CLI> sip show peers
    Name/username             Host                        Dyn Forcerport
    ACL Port     Status
    201/201                   (Unspecified)                        D   N
                 0        UNKNOWN
    202/202                   (Unspecified)                        D   N
                 0        UNKNOWN
    babytel/16046289850       198.38.7.34                            N  
               5065 Unmonitored
    gs102/gs102               (Unspecified)                        D   N
                 0        UNKNOWN
    4 sip peers [Monitored: 0 online, 3 offline Unmonitored: 1 online, 0
    offline]
    linux-k7qk*CLI>
    linux-k7qk*CLI> sip show peer 201
       * Name       : 201
       Secret       : <Set>
       MD5Secret    : <Not set>
       Remote Secret: <Not set>
       Context      : default
       Subscr.Cont. : <Not set>
       Language     : en
       Accountcode  : 201
       AMA flags    : Unknown
       Netborder CPD: No
       Transfer mode: open
       CallingPres  : Presentation Allowed, Not Screened
       Callgroup    :
       Pickupgroup  :
       MOH Suggest  : default
       Mailbox      : 201
       VM Extension : asterisk
       LastMsgsSent : 32767/65535
       Call limit   : 0
       Max forwards : 0
       Dynamic      : Yes
       Callerid     : "jitsi201" <201>
       MaxCallBR    : 384 kbps
       Expire       : -1
       Insecure     : no
       Force rport  : Yes
       ACL          : No
       DirectMedACL : No
       T.38 support : No
       T.38 EC mode : Unknown
       T.38 MaxDtgrm: 4294967295
       DirectMedia  : No
       PromiscRedir : No
       User=Phone   : No
       Video Support: No
       Text Support : No
       Ign SDP ver  : No
       Trust RPID   : No
       Send RPID    : Yes
       TrustIDOutbnd: Legacy
       Subscriptions: Yes
       Overlap dial : No
       DTMFmode     : rfc2833
       Timer T1     : 500
       Timer B      : 32000
       ToHost       :
       Addr->IP     : (null)
       Defaddr->IP  : (null)
       Prim.Transp. : UDP
       Allowed.Trsp : UDP
       Def. Username: 201
       SIP Options  : (none)
       Codecs       : 0x6 (gsm|ulaw)
       Codec Order  : (ulaw:20,gsm:20)
       Auto-Framing : No
       Status       : UNKNOWN
       Useragent    :
       Reg. Contact :
       Qualify Freq : 60000 ms
       Sess-Timers  : Accept
       Sess-Refresh : uas
       Sess-Expires : 1800 secs
       Min-Sess     : 90 secs
       RTP Engine   : asterisk
       Parkinglot   :
       Use Reason   : No
       Encryption   : No
    linux-k7qk*CLI>
    linux-k7qk*CLI>



The port seems to be open and listened to:


linux-k7qk:~ #
linux-k7qk:~ # lsof -i UDP:5060
COMMAND   PID USER   FD   TYPE DEVICE SIZE/OFF NODE NAME
asterisk 3520 root   12u  IPv4  16438      0t0  UDP *:sip
linux-k7qk:~ #


and the bind address is correct:

linux-k7qk:~ #
linux-k7qk:~ # cat /etc/asterisk/sip.conf | grep bind
bindport=5060                   ; UDP Port to bind to (SIP standard port 
is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds 
to all)
linux-k7qk:~ #




so, why, when Jitsi tries to connect to Asterisk, does it return with:

15:13:29.040 SEVERE: [29] 
service.protocol.AccountManager.doLoadStoredAccounts().213 Failed to 
load account {SERVER_PORT=6697, 
ACCOUNT_ICON_PATH=resources/images/protocol/irc/irc32x32.png, 
AUTO_CHANGE_USER_NAME=true, CHAT_ROOM_PRESENCE_TASK=true, 
NO_PASSWORD_REQUIRED=false, ACCOUNT_UID=IRC:201 at 192.168.0.99:6697, 
SERVER_ADDRESS=192.168.0.99, USER_ID=201, DEFAULT_ENCRYPTION=true, 
PROTOCOL_NAME=IRC, ENCRYPTED_PASSWORD=/hcTkghmfRJWFXrWaKDMmA==, 
CONTACT_PRESENCE_TASK=true}
java.lang.IllegalArgumentException: nick name contains invalid 
characters: only letters, digits and -, \, [, ], `, ^, {, }, |, _ are 
allowed
     at 
net.java.sip.communicator.impl.protocol.irc.IdentityManager.checkNick(IdentityManager.java:194)
     at 
net.java.sip.communicator.impl.protocol.irc.IrcStack$ServerParameters.<init>(IrcStack.java:354)
     at 
net.java.sip.communicator.impl.protocol.irc.IrcStack$ServerParameters.<init>(IrcStack.java:311)
     at 
net.java.sip.communicator.impl.protocol.irc.IrcStack.<init>(IrcStack.java:89)
     at 
net.java.sip.communicator.impl.protocol.irc.ProtocolProviderServiceIrcImpl.initialize(ProtocolProviderServiceIrcImpl.java:149)
     at 
net.java.sip.communicator.impl.protocol.irc.ProtocolProviderFactoryIrcImpl.createService(ProtocolProviderFactoryIrcImpl.java:136)
     at 
net.java.sip.communicator.service.protocol.ProtocolProviderFactory.loadAccount(ProtocolProviderFactory.java:983)
     at 
net.java.sip.communicator.service.protocol.AccountManager.doLoadStoredAccounts(AccountManager.java:204)
     at 
net.java.sip.communicator.service.protocol.AccountManager.loadStoredAccounts(AccountManager.java:446)
     at 
net.java.sip.communicator.service.protocol.AccountManager.runInLoadStoredAccountsThread(AccountManager.java:562)
     at 
net.java.sip.communicator.service.protocol.AccountManager.access$100(AccountManager.java:26)
     at 
net.java.sip.communicator.service.protocol.AccountManager$2.run(AccountManager.java:487)


The "nick name" is 201, no special characters...

I'm on an older, so want to use Jitsi because it's cross platform.



thanks,

Thufir
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