ASTERISK-09277: Unable to complete registeration / accept calls

[Home]

Summary: ASTERISK-09277: Unable to complete registeration / accept calls

Reporter: Leif Sawyer (akhepcat) Labels:

Date Opened: 2007-04-18 13:13:28 Date Closed: 2011-06-07 14:03:11

Priority: Major Regression? No

Status: Closed/Complete Components: Channels/chan_sip/Registration

Versions: Frequency of
Occurrence

Related
Issues:

Environment: Attachments:

Description: AsteriskNow :
asterisk*CLI> core show version
Asterisk 1.4.2 built by admin @ aomori on a i686 running Linux on 2007-03-23 20:40:52 UTC

Unable to complete registration with Tekelec T-7000 SIP gateway, retransmitted multiple times until EOL.

Partial registration seems to occur though, as incoming calls show up in the SIP debug, but are not passed through to the dialplan, and the call ends as "not in service"

outgoing calls are not passed through to the trunk.

****** ADDITIONAL INFORMATION ******

asterisk*CLI> core set verbose 10
Verbosity was 3 and is now 10
asterisk*CLI>
<--- SIP read from 209.165.174.124:5060 --->
INVITE sip:98680116@209.165.137.23 SIP/2.0
Via: SIP/2.0/UDP 209.165.174.124:5060;branch=z9hG4bK-8e5262a5
From: "Leif Sawyer" <sip:6116@209.165.137.23>;tag=d3e45416508757d4o0
To: <sip:98680116@209.165.137.23>
Remote-Party-ID: "Leif Sawyer" <sip:6116@209.165.137.23>;screen=yes;party=calling
Call-ID: 9e5212be-8460694c@209.165.174.124
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Leif Sawyer" <sip:6116@209.165.174.124:5060>
Expires: 240
User-Agent: Linksys/SPA941-4.1.15
Content-Length: 405
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp

v=0
o=- 16472691 16472691 IN IP4 209.165.174.124
s=-
c=IN IP4 209.165.174.124
t=0 0
m=audio 16446 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 209.165.174.124 : 5060 (no NAT)
Using INVITE request as basis request - 9e5212be-8460694c@209.165.174.124
Found peer '6116'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 209.165.174.124:16446
Found description format PCMU for ID 0
Found description format G726-32 for ID 2
Found description format G723 for ID 4
Found description format PCMA for ID 8
Found description format G729a for ID 18
Found description format G726-40 for ID 96
Found description format G726-24 for ID 97
Found description format G726-16 for ID 98
Found description format telephone-event for ID 101
Capabilities: us - 0x809ae (gsm|ulaw|alaw|g726|adpcm|lpc10|g729|h263), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x90c (ulaw|alaw|g726|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.165.174.124:16446
Looking for 98680116 in numberplan-custom-1 (domain 209.165.137.23)

<--- Reliably Transmitting (no NAT) to 209.165.174.124:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 209.165.174.124:5060;branch=z9hG4bK-8e5262a5;received=209.165.174.124
From: "Leif Sawyer" <sip:6116@209.165.137.23>;tag=d3e45416508757d4o0
To: <sip:98680116@209.165.137.23>;tag=as04346313
Call-ID: 9e5212be-8460694c@209.165.174.124
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '9e5212be-8460694c@209.165.174.124' in 32000 ms (Method: INVITE)
asterisk*CLI>
<--- SIP read from 209.165.174.124:5060 --->
ACK sip:98680116@209.165.137.23 SIP/2.0
Via: SIP/2.0/UDP 209.165.174.124:5060;branch=z9hG4bK-8e5262a5
From: "Leif Sawyer" <sip:6116@209.165.137.23>;tag=d3e45416508757d4o0
To: <sip:98680116@209.165.137.23>;tag=as04346313
Call-ID: 9e5212be-8460694c@209.165.174.124
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Leif Sawyer" <sip:6116@209.165.174.124:5060>
User-Agent: Linksys/SPA941-4.1.15
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '9e5212be-8460694c@209.165.174.124' Method: ACK
asterisk*CLI>
asterisk*CLI>
asterisk*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
trunk_1/9078650105 24.237.163.85 0 Unmonitored
6111 (Unspecified) D 0 Unmonitored
6116/6116 209.165.174.124 D 5060 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 2 offline]
asterisk*CLI> sip reload
Reloading SIP
== Parsing '/etc/asterisk/sip.conf': Found
[Apr 18 10:01:29] WARNING[16878]: chan_sip.c:16477 reload_config: To disallow external domains, you need to configure local SIP domains.
== Parsing '/etc/asterisk/users.conf': Found
[Apr 18 10:01:29] WARNING[16878]: frame.c:1289 ast_parse_allow_disallow: Cannot disallow unknown format ''
[Apr 18 10:01:29] WARNING[16878]: frame.c:1289 ast_parse_allow_disallow: Cannot allow unknown format ''
== Parsing '/etc/asterisk/sip_notify.conf': Found
[Apr 18 10:01:29] NOTICE[16878]: chan_sip.c:7134 sip_reregister: -- Re-registration for 9078650105@24.237.163.84
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 24.237.163.84:5060:
REGISTER sip:24.237.163.84 SIP/2.0
Via: SIP/2.0/UDP 209.165.137.23:5060;branch=z9hG4bK311bf006;rport
From: <sip:9078650105@24.237.163.84>;tag=as60f8ae2b
To: <sip:9078650105@24.237.163.84>
Call-ID: 3998781b60f5b2315f2f77726d122e71@209.165.137.23
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 3600
Contact: <sip:6116@209.165.137.23>
Event: registration
Content-Length: 0

---
asterisk*CLI>
<--- SIP read from 24.237.163.84:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.165.137.23:5060;branch=z9hG4bK311bf006;rport
From: <sip:9078650105@24.237.163.84>;tag=as60f8ae2b
To: <sip:9078650105@24.237.163.84>
Call-ID: 3998781b60f5b2315f2f77726d122e71@209.165.137.23
CSeq: 102 REGISTER
WWW-Authenticate: Digest algorithm=MD5,nonce="0149de91",realm="Tekelec7000"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Responding to challenge, registration to domain/host name 24.237.163.84
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 24.237.163.84:5060:
REGISTER sip:24.237.163.84 SIP/2.0
Via: SIP/2.0/UDP 209.165.137.23:5060;branch=z9hG4bK11e65e41;rport
From: <sip:9078650105@24.237.163.84>;tag=as16e8b1b4
To: <sip:9078650105@24.237.163.84>
Call-ID: 3998781b60f5b2315f2f77726d122e71@209.165.137.23
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="9078650105", realm="Tekelec7000", algorithm=MD5, uri="sip:24.237.163.84", nonce="0149de91", response="45c2e7430f6e4137235c8d15d288a309", opaque=""
Expires: 3600
Contact: <sip:6116@209.165.137.23>
Event: registration
Content-Length: 0

---
Retransmitting #1 (no NAT) to 24.237.163.84:5060:
REGISTER sip:24.237.163.84 SIP/2.0
Via: SIP/2.0/UDP 209.165.137.23:5060;branch=z9hG4bK11e65e41;rport
From: <sip:9078650105@24.237.163.84>;tag=as16e8b1b4
To: <sip:9078650105@24.237.163.84>
Call-ID: 3998781b60f5b2315f2f77726d122e71@209.165.137.23
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="9078650105", realm="Tekelec7000", algorithm=MD5, uri="sip:24.237.163.84", nonce="0149de91", response="45c2e7430f6e4137235c8d15d288a309", opaque=""
Expires: 3600
Contact: <sip:6116@209.165.137.23>
Event: registration
Content-Length: 0

---
Really destroying SIP dialog '30a4c48d44f243ef18a289a1087afa15@209.165.137.23' Method: NOTIFY
Retransmitting #2 (no NAT) to 24.237.163.84:5060:
REGISTER sip:24.237.163.84 SIP/2.0
Via: SIP/2.0/UDP 209.165.137.23:5060;branch=z9hG4bK11e65e41;rport
From: <sip:9078650105@24.237.163.84>;tag=as16e8b1b4
To: <sip:9078650105@24.237.163.84>
Call-ID: 3998781b60f5b2315f2f77726d122e71@209.165.137.23
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="9078650105", realm="Tekelec7000", algorithm=MD5, uri="sip:24.237.163.84", nonce="0149de91", response="45c2e7430f6e4137235c8d15d288a309", opaque=""
Expires: 3600
Contact: <sip:6116@209.165.137.23>
Event: registration
Content-Length: 0

---
Scheduling destruction of SIP dialog '4b59fcf67b12065f79700f472660afdc@209.165.137.23' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 209.165.174.124:5060:
NOTIFY sip:6116@209.165.174.124:5060 SIP/2.0
Via: SIP/2.0/UDP 209.165.137.23:5060;branch=z9hG4bK524eb305;rport
From: "asterisk" <sip:asterisk@209.165.137.23>;tag=as61d270ff
To: <sip:6116@209.165.174.124:5060>
Contact: <sip:asterisk@209.165.137.23>
Call-ID: 4b59fcf67b12065f79700f472660afdc@209.165.137.23
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 94

Messages-Waiting: no
Message-Account: sip:asterisk@209.165.137.23
Voice-Message: 0/0 (0/0)

---
asterisk*CLI>
<--- SIP read from 209.165.174.124:5060 --->
SIP/2.0 200 OK
To: <sip:6116@209.165.174.124:5060>;tag=72ae796dbb90c14ci0
From: "asterisk" <sip:asterisk@209.165.137.23>;tag=as61d270ff
Call-ID: 4b59fcf67b12065f79700f472660afdc@209.165.137.23
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 209.165.137.23:5060;branch=z9hG4bK524eb305
Server: Linksys/SPA941-4.1.15
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4b59fcf67b12065f79700f472660afdc@209.165.137.23' Method: NOTIFY
Retransmitting #3 (no NAT) to 24.237.163.84:5060:
REGISTER sip:24.237.163.84 SIP/2.0
Via: SIP/2.0/UDP 209.165.137.23:5060;branch=z9hG4bK11e65e41;rport
From: <sip:9078650105@24.237.163.84>;tag=as16e8b1b4
To: <sip:9078650105@24.237.163.84>
Call-ID: 3998781b60f5b2315f2f77726d122e71@209.165.137.23
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="9078650105", realm="Tekelec7000", algorithm=MD5, uri="sip:24.237.163.84", nonce="0149de91", response="45c2e7430f6e4137235c8d15d288a309", opaque=""
Expires: 3600
Contact: <sip:6116@209.165.137.23>
Event: registration
Content-Length: 0

---
Retransmitting #4 (no NAT) to 24.237.163.84:5060:
REGISTER sip:24.237.163.84 SIP/2.0
Via: SIP/2.0/UDP 209.165.137.23:5060;branch=z9hG4bK11e65e41;rport
From: <sip:9078650105@24.237.163.84>;tag=as16e8b1b4
To: <sip:9078650105@24.237.163.84>
Call-ID: 3998781b60f5b2315f2f77726d122e71@209.165.137.23
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="9078650105", realm="Tekelec7000", algorithm=MD5, uri="sip:24.237.163.84", nonce="0149de91", response="45c2e7430f6e4137235c8d15d288a309", opaque=""
Expires: 3600
Contact: <sip:6116@209.165.137.23>
Event: registration
Content-Length: 0

---
Retransmitting ASTERISK-1 (no NAT) to 24.237.163.84:5060:
REGISTER sip:24.237.163.84 SIP/2.0
Via: SIP/2.0/UDP 209.165.137.23:5060;branch=z9hG4bK11e65e41;rport
From: <sip:9078650105@24.237.163.84>;tag=as16e8b1b4
To: <sip:9078650105@24.237.163.84>
Call-ID: 3998781b60f5b2315f2f77726d122e71@209.165.137.23
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="9078650105", realm="Tekelec7000", algorithm=MD5, uri="sip:24.237.163.84", nonce="0149de91", response="45c2e7430f6e4137235c8d15d288a309", opaque=""
Expires: 3600
Contact: <sip:6116@209.165.137.23>
Event: registration
Content-Length: 0

---
Retransmitting ASTERISK-2 (no NAT) to 24.237.163.84:5060:
REGISTER sip:24.237.163.84 SIP/2.0
Via: SIP/2.0/UDP 209.165.137.23:5060;branch=z9hG4bK11e65e41;rport
From: <sip:9078650105@24.237.163.84>;tag=as16e8b1b4
To: <sip:9078650105@24.237.163.84>
Call-ID: 3998781b60f5b2315f2f77726d122e71@209.165.137.23
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="9078650105", realm="Tekelec7000", algorithm=MD5, uri="sip:24.237.163.84", nonce="0149de91", response="45c2e7430f6e4137235c8d15d288a309", opaque=""
Expires: 3600
Contact: <sip:6116@209.165.137.23>
Event: registration
Content-Length: 0

---
[Apr 18 10:01:49] WARNING[16878]: chan_sip.c:1899 retrans_pkt: Maximum retries exceeded on transmission 3998781b60f5b2315f2f77726d122e71@209.165.137.23 for seqno 103 (Critical Request)
Really destroying SIP dialog '3998781b60f5b2315f2f77726d122e71@209.165.137.23' Method: REGISTER
asterisk*CLI>
<--- SIP read from 24.237.163.84:5060 --->
INVITE sip:9078650105@209.165.137.23:5060 SIP/2.0
Via: SIP/2.0/UDP 24.237.163.84:5060;branch=z9hG4bK-00101-lwk35cQn1Hi4rca-0
Max-Forwards: 6
From: "GCI_INTERNET " <sip:9078680116@24.237.163.84:5060>;tag=lwk35cQn1Hi4rca-IPTrunk-780-17-20at24.237.163.84
To: sip:9078650105@209.165.137.23:5060
Call-ID: lwk35cQn1Hi4rca@24.237.163.84
CSeq: 101 INVITE
Remote-Party-Id: "GCI_INTERNET " <sip:9078680116@24.237.163.84:5060>;party=calling;privacy=off;id-type=subscriber
Expires: 180
Allow: REGISTER,INVITE,CANCEL,BYE,ACK
Contact: sip:24.237.163.84:5060
User-Agent: Tekelec-7000/r4.0
Content-Type: application/sdp
Content-Length: 177

v=0
o=- 1 1 IN IP4 24.237.163.84
s=
c=IN IP4 24.237.163.85
t=0 0
m=audio 49190 RTP/AVP 0 18
a=ptime:20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
<------------->
--- (14 headers 10 lines) ---
Sending to 24.237.163.84 : 5060 (no NAT)
Using INVITE request as basis request - lwk35cQn1Hi4rca@24.237.163.84
Found no matching peer or user for '24.237.163.84:5060'
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 24.237.163.85:49190
Found description format G729 for ID 18
Found description format PCMU for ID 0
Capabilities: us - 0x809ae (gsm|ulaw|alaw|g726|adpcm|lpc10|g729|h263), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 24.237.163.85:49190
Looking for 9078650105 in default (domain 209.165.137.23)

<--- Reliably Transmitting (no NAT) to 24.237.163.84:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 24.237.163.84:5060;branch=z9hG4bK-00101-lwk35cQn1Hi4rca-0;received=24.237.163.84
From: "GCI_INTERNET " <sip:9078680116@24.237.163.84:5060>;tag=lwk35cQn1Hi4rca-IPTrunk-780-17-20at24.237.163.84
To: sip:9078650105@209.165.137.23:5060;tag=as712a9076
Call-ID: lwk35cQn1Hi4rca@24.237.163.84
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'lwk35cQn1Hi4rca@24.237.163.84' in 32000 ms (Method: INVITE)
asterisk*CLI>
<--- SIP read from 24.237.163.84:5060 --->
ACK sip:9078650105@209.165.137.23:5060 SIP/2.0
Via: SIP/2.0/UDP 24.237.163.84:5060;branch=z9hG4bK-00101-lwk35cQn1Hi4rca-0
Max-Forwards: 6
From: "GCI_INTERNET " <sip:9078680116@24.237.163.84:5060>;tag=lwk35cQn1Hi4rca-IPTrunk-780-17-20at24.237.163.84
To: sip:9078650105@209.165.137.23:5060;tag=as712a9076
Call-ID: lwk35cQn1Hi4rca@24.237.163.84
CSeq: 101 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'lwk35cQn1Hi4rca@24.237.163.84' Method: ACK

Comments: By: Leif Sawyer (akhepcat) 2007-04-18 13:33:45

This is the users.conf relevant section
[trunk_1]
disallow =
allow =
callerid =
contact =
context = DID_trunk_1
dialformat = ${EXTEN:1}
fromdomain =
fromuser =
group =
hasexten = no
hasiax = no
hassip = yes
host = 24.237.163.85
insecure = port,invite
port =
provider =
registeriax = no
registersip = no
secret = xxxxxx
trunkname = Custom - GCI-T7000
trunkstyle = customvoip
username = 9078650105
By: Leif Sawyer (akhepcat) 2007-04-18 13:34:48

sip.conf:
[general]
context = default ; Default context for incoming calls
allowoverlap = no ; Disable overlap dialing support. (Default is yes)
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)
srvlookup = yes ; Enable DNS SRV lookups on outbound calls
allow = g726,g729,adpcm,ulaw,alaw,gsm,lpc10
allowexternaldomains = no
allowexternalinvites = no
allowguest = yes
allowsubscribe = no
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
compactheaders = no
disallow = ilbc,speex
dumphistory = no
g726nonstandard = no
insecure = port,invite
ignoreregexpire = no
jbenable = no
jbforce = no
jblog = no
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 3600
defaultexpiry = 3600
notifyringing = no
pedantic = yes
promiscredir = no
recordhistory = no
register = 9078650105:xxxxxx@24.237.163.84/6116
relaxdtmf = no
rtcachefriends = no
rtsavesysname = no
rtupdate = no
sendrpid = no
sipdebug = no
t1min = 100
t38pt_udptl = no
trustrpid = no
usereqphone = no
videosupport = no
registertimeout = 3600
registerattempts = 0
progressinband = no
maxexpiry = 3600 ; Maximum allowed time of incoming registrations
minexpiry = 3600 ; Minimum length of registrations/subscriptions (default 60)
defaultexpiry = 3600 ; Default length of incoming/outgoing registration
canreinvite = yes
nat = no
By: Leif Sawyer (akhepcat) 2007-04-18 13:50:40

And finally....

------------->
--- (8 headers 0 lines) ---
[Apr 18 10:48:49] NOTICE[16878]: chan_sip.c:11997 handle_response_register: Failed to authenticate on REGISTER to '9078650105@24.237.163.85' (Tries 3)
Really destroying SIP dialog '4dba33ca6965e84920aeda1b297733e7@209.165.137.23' Method: REGISTER
asterisk*CLI>
asterisk*CLI>
asterisk*CLI>
asterisk*CLI>
asterisk*CLI>
Really destroying SIP dialog '1063fb8d66ad14f726a0aebe7828beb0@209.165.137.23' Method: NOTIFY
asterisk*CLI>
By: Olle Johansson (oej) 2007-04-18 15:19:51

please add all debug output as attached files! Thank you.
By: Olle Johansson (oej) 2007-04-18 15:21:28

Please, this is not a support forum. You haven't located a bug, you just have a bad configuration.
By: Olle Johansson (oej) 2007-04-18 15:22:30

Please use the asterisk-users mailing list for support. Feel free to open another issue report when you have located a bug. Thanks :-)