State of No Authentication in Asterisk SIP Registr...

I have been trying lots of combinations and permutations of registration entries in Asterisk's SIP.conf file.   I would be grateful if someone would let me know exactly what userid and password I need to include
The bulk of instructions I have been reading over the past half a day indicate that this form is correct.
register => [edited for privacy]
where bmdemo3 is the userid and REDACTED is the password for the Skype user I created under Skype Manager to represent Asterisk to Skype's SIP trunk.
In the output to the sip show registry command for Asterisk, there is one entry that suggests a registration for sip.skype.com:5060, however the column to the far right (entitled) State says "No Authentication".   That does not seem good.
The aforementioned userid and password log me in to that skype user's account (which I activated) and I can make and receive direct Skype calls using it.   Is there another userid and password I should be using instead?
I would greatly appreciate any guidance as to what I might be doing wrong.
Cheers,
Michael

Here is the output from the sip show registry command
ubuntu*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sip.skype.com:5060 N bmdemo3 120 No Authentication
1 SIP registrations.
Here is the SIP REGISTER message being rejected:
<--- SIP read from UDP:63.209.144.201:5060 --->
SIP/2.0 403 Forbidden
From: <sip:[email protected]>;tag=as333ae3e0
To: <sip:[email protected]>;tag=c990d13f-f1dd03ad-0-9c8aac96-0
Call-ID: [email protected]
CSeq: 102 REGISTER
Via: SIP/2.0/UDP 172.16.164.33:5060;branch=z9hG4bK0ae22ba3;rport=55060;received=96.241.233.63
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------->
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 0 [ 21]: SIP/2.0 403 Forbidden
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 1 [ 48]: From: <sip:[email protected]>;tag=as333ae3e0
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 2 [ 66]: To: <sip:[email protected]>;tag=c990d13f-f1dd03ad-0-9c8aac96-0
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 3 [ 51]: Call-ID: [email protected]
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 4 [ 18]: CSeq: 102 REGISTER
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 5 [ 93]: Via: SIP/2.0/UDP 172.16.164.33:5060;branch=z9hG4bK0ae22ba3;rport=55060;received=96.241.233.63
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 6 [ 12]: Expires: 120
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 7 [ 35]: Contact: <sip:[email protected]:5060>
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Header 8 [ 17]: Content-Length: 0
[Jul 1 16:21:41] VERBOSE[1081] chan_sip.c: --- (9 headers 0 lines) ---
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: = Looking for Call ID: [email protected] (Checking To) --From tag as333ae3e0 --To-tag c990d13f-f1dd03ad-0-9c8aac96-0
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #756
[Jul 1 16:21:41] DEBUG[1081] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found
[Jul 1 16:21:41] WARNING[1081] chan_sip.c: Forbidden - wrong password on authentication for REGISTER for 'bmdemo3' to 'sip.skype.com'

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SIP Registration Request not reaching provider

I have a home lab setup and recently decided to add PSTN access via a SIP trunk and termination. I got the applicable information from the provider, set up the IOS configuration, but I'm not registering. I checked the CCSIP messages logs, no response. I talked to the provider, they are not getting my messages at all. Other traffic to and from the internet passes just fine (including SSL VPN).
I've talked to my ISP many times, and they continually insist they do not block any traffic except port 25.
Here is the applicable configuration:
sip
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And here's the debug output from ccsip all:
Apr 10 21:17:33.091: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone CDT to SIP default timezone = GMT
Apr 10 21:17:33.091: //594/000000000000/SIP/Info/sipSPISendRegister: Associated container=0x4F7E3670 to Register
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipSPISendRegister: Sending REGISTER to the transport layer
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipSPITransportSendMessage: msg=0x4F3BB008, addr=XX.XX.XX.XX, port=5060, sentBy_port=0, local_addr=10.XX.XX.XX, is_req=1, transport=1, switch=0, callBack=0x0
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
Apr 10 21:17:33.091: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:Xx.XX.XX.XX, rport:5060 with laddr:Xx.XX.XX.Xx
Apr 10 21:17:33.091: //594/000000000000/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x4F3BB008
Apr 10 21:17:33.091: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4F3BB008, addr=XX.XX.XX.XX, port=5060, local_addr=10.Xx.XX.XX, connId=3 for UDP
I took out the message because I don't think it's relevant, and has too many IPs to XX out :)
Apr 10 21:17:37.091: //594/000000000000/SIP/Error/act_sent_register_wait_100: act_sent_register_wait_100: Out of retries
Apr 10 21:17:37.091: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDecrementOverloadCount: Count:Local 0 Global 1
Apr 10 21:17:37.091: //594/000000000000/SIP/Error/ccsip_api_register_result_ind: Message Code Class 4xx Method Code 100 received for REGISTER
Apr 10 21:17:37.091: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_register_reset_dns_cache: CCSIP_REGISTER:: registrar 0 DNS resolved addr reset
Apr 10 21:17:37.091: //594/000000000000/SIP/Error/ccsip_api_register_result_ind: SIP Registration Retries Exhausted
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPIRegPthruProcessResponse: Processing response w/ resp code == 408
Apr 10 21:17:37.091: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetRPCBFromRCB: Retreiving RCB [0x4FA37D28] from RPCB [0x0]
Apr 10 21:17:37.091: //594/000000000000/SIP/Error/sipSPIRegPthruProcessResponse: Error NO RPCB
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Apr 10 21:17:37.091: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[576] removed.
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPIUdeleteCcbFromUACTable: ****Deleting from UAC table.
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x4B6CC660 key=AD7E2293-C01A11E3-80AEE452-E7A949D0
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/ccsip_offer_ans_delete:
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/ccsip_iwf_delete:
Apr 10 21:17:37.091: //594/000000000000/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 4B6CC660

Nevermind; I downloaded a SIP phone app for my smart phone, got it to connect in the same network. After that, I put the gateway in a DMZ and it started working. Not entirely sure why it was required, but hey, it works!
Cheers

HTTP Authentication Digest for SIP messages in a trunk SIP CUCME

Hello,
we would like to implement HTTP Authentication Digest for SIP messages in a trunk SIP between a Cisco 2851 and an Asterisk server.
We are using CUCM Express with 15.1(4)M (CME 8.6) as voice gateway to connect to PSTN.
According to Cisco documentation:
"To configure a gateway to use HTTP Authentication Digest, give the following command in each dial peer or SIP-UA configuration mode:
authentication username username password password [realm realm]."
The problem is that when call is from CISCO to ASTERISK, Asterisk sends a challenge to Cisco to do Authentication:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.70.11:5060;branch=z9hG4bK3E205D
Remote-Party-ID: "DN1001" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "DN1001" <sip:[email protected]>;tag=5317D4-2271
To: <sip:[email protected]>
Date: Thu, 20 Feb 2014 10:55:56 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1679566433-2572423651-2156454406-1292596908
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1392893756
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 208
<--- Reliably Transmitting (no NAT) to 10.0.70.11:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.70.11:5060;branch=z9hG4bK3E205D;received=10.0.70.11
From: "DN1001" <sip:[email protected]>;tag=5317D4-2271
To: <sip:[email protected]>;tag=as665c9410
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 11.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="559bd1d2"
Content-Length: 0
However, when call is for ASTERISK to Cisco, there is no challenge sent.
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
Max-Forwards: 70
From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0
Date: Thu, 20 Feb 2014 09:58:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282
<--- SIP read from UDP:10.0.70.11:60829 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
To: <sip:[email protected]>
Date: Thu, 20 Feb 2014 10:58:27 GMT
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
To: <sip:[email protected]>;tag=556830-757
Date: Thu, 20 Feb 2014 10:58:27 GMT
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "DN1001" <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
My configuration in Cisco device is:
dial-peer voice 1 voip
description **Calls to ASTERISK **
destination-pattern 9T
session protocol sipv2
session target sip-server
codec g711ulaw
sip-ua
keepalive target ipv4:10.1.32.70
authentication username CCME password 7 070E234F4A realm asterisk
sip-server ipv4:10.1.32.70:5060
To avoid that the ASTERISK is blocked by Cisco TOLLFRAUD_APP I have added:
voice service voip
ip address trusted list
ipv4 10.1.32.70 255.255.255.255
allow-connections sip to sip
sip
registrar server
The issue is that I would like that Cisco also send a challenge to asterisk server to authenticate SIP messages.
Any ideas?.
Regards.

Hello,
yes, but credentials command configure credentials that are used when Cisco UA must register in a server.
I do not need register Cisco into Asterisk server. What I want is that Cisco authenticate SIP messages that receive. I know
that can be enough with TOLLFRAUD_AP where remote IP is checked, but I want to do something like others routing
protocols (as OSPF, BGP) where every message must be authenticated.
Thanks.
Regards.

Connecting Asterisk SIP PBX to Skype

Hi,
I'm a new Skype Business customer, I have a few user registered and I wish to connect my company Asterisk PBX to Skype. In details, I bought a Skype-In number, I would like that calls to such number will arrive to one of my PBX extension. Is there some way to do this?
Thank you in advance.

Could someone help me?
I'm having problems with my Elastix (Asterisk) / Skype Connect Configuration. I always get "All circuits are busy now..." message. My configuration is:
Trunk Name: skype_in
type=friend
username=xxxxxxx
fromdomain=sip.skype.com
fromuser=xxxxxxx
realm=sip.skype.com
host=sip.skype.com
dtmfmode=rfc2833
secret=password
nat=no
insecure=invite
qualify=yes
disallow=all
allow=alaw&ulaw
amaflags=default
trustrpid=no
sendrpid=yes
context=skype_in
Register String: [email protected]/xxxxxx From asterisk I get:
elastix*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
skype_in/99051000147265 204.9.161.164 5060 OK (177 ms)
and
sip show registry
sip.skype.com:5060 990510001472 105 Registered Thu, 06 Oct 2011 20:46:30
Every looks good but when I try to make a call... I get the busy message:
SIP Debugging Enabled for IP: 10.168.16.115:5060
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-b0d83df763d87d56-1---d8754z-
Max-Forwards: 70
Contact: 6001>
To: 20551141256555>
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Allow-Events: presence, kpml
Content-Length: 255
v=0
o=Zoiper_user 0 0 IN IP4 10.168.16.115
s=Zoiper_session
c=IN IP4 10.168.16.115
t=0 0
m=audio 8000 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 10.168.16.115 : 5060 (no NAT)
Using INVITE request as basis request - MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
<--- Reliably Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-b0d83df763d87d56-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>;tag=as2f3cba10
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="60db2ce3"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.' in 32000 ms (Method: INVITE)
Found user '6001'
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
ACK sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-b0d83df763d87d56-1---d8754z-
Max-Forwards: 70
To: 20551141256555>;tag=as2f3cba10
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-
Max-Forwards: 70
Contact: 6001>
To: 20551141256555>
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Proxy-Authorization: Digest username="6001",realm="asterisk",nonce="60db2ce3",uri="sip:[email protected];transport=UDP",response="ed79aee161a6cc8a7e520cd011afe0bb",algorithm=MD5
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Allow-Events: presence, kpml
Content-Length: 255
v=0
o=Zoiper_user 0 0 IN IP4 10.168.16.115
s=Zoiper_session
c=IN IP4 10.168.16.115
t=0 0
m=audio 8000 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 10.168.16.115 : 5060 (NAT)
Using INVITE request as basis request - MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
Found user '6001'
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.168.16.115:8000
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.168.16.115:8000
Looking for 20551141256555 in from-internal (domain 10.168.16.3)
list_route: hop: 6001>
<--- Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: 20551141256555>
Content-Length: 0
<------------>
Audio is at 10.168.16.3 port 17408
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>;tag=as2650cb3c
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: 20551141256555>
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 14425 14425 IN IP4 10.168.16.3
s=session
c=IN IP4 10.168.16.3
t=0 0
m=audio 17408 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
<------------->
Scheduling destruction of SIP dialog 'MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 10.168.16.115:5060 --->
SIP/2.0 484 Address incomplete
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-;received=10.168.16.115
From: "Alexandre"6001>;tag=1a692d4e
To: 20551141256555>;tag=as2650cb3c
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------>
elastix*CLI>
<--- SIP read from 10.168.16.115:5060 --->
ACK sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.168.16.115:5060;branch=z9hG4bK-d8754z-99c9fceea6456942-1---d8754z-
Max-Forwards: 70
To: 20551141256555>;tag=as2650cb3c
From: "Alexandre"6001>;tag=1a692d4e
Call-ID: MDUyNWYyN2YxMjU5YjYwMWRmZGQ2NzBlNThmZGYwNzc.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
elastix*CLI> sip set debug off
SIP Debugging Disabled
Could someone help me?
Best regards
Eduardo6001>20551141256555>20551141256555>6001>20551141256555>20551141256555>6001>20551141256555>20551141256555>6001>6001>6001>20551141256555>6001>6001>20551141256555>20551141256555>6001>6001>20551141256555>6001>

MXP's and SIP Registration Failure Using a FQDN

A majority of our MXP's will not register to our SIP Server using a FQDN. They will register using the SIP Server IP Address. Why do they not like the FQDN? Our C Series, EX, and SX systems do not have this problem.

yes to both questions. some of the MXP's show that they are registered with the FQDN, but that the Active SIP Server Address is the FQDN. I was expecting to see the IP Address of the VCS it is registered to.The majority just won't register with the FQDN, but do with an IP Address.
All of our other non-MXP endpoints register just fine using the FQDN.
The same FQDN is used for Gatekeeper and SIP registration. The MXP's register to the gatekeeper as expected. It is just the SIP Registration.
The majority are running F9.3.3. Some others are running anywhere from F9.02 to F9.3.1, but all software releases are affected the same way.

SIP Registration is blocked by BT or Home Hub

Hi all,
I have a problem, between 7.02PM and 11.02PM my SIP Softphone looses registration - every night. I have isolated the issue to my BT Infinity connection as when I close my WIFI and use 3G the SIP telephone registers sucessfully.
I 'think' it's either my Home Hub 3 or BT themselves blocking SIP Registration
Anyone seen this before?
Regards
Matt

Is your SIP telephone connected by a cable, or does it use WiFi?
If it uses WiFi, then you may be getting wireless interference, and you would need to select a different wireless channel.
There are some useful help pages here, for BT Broadband customers only, on my personal website.
BT Broadband customers - help with broadband, WiFi, networking, e-mail and phones.

SPA-112 check SIP registration with SNMP

Which OID do I need to use to check the SIP registration with SNMP?

Which OID do I need to use to check the SIP registration with SNMP?

QSIG MWI asterisk SIP

Hi
I wanted to send MWI message from asterisk to old PBX,and use Asterisk as voicemail box.. the old pbx does not support SIP.
so it looks like these
Asterisk---sip trunk--Cisco2821---QSig---PBX--End points
I thought of two options, first I have media gateway, CISCO 2821 router in-between the PBX and the asterisk BOX, voicemail calls are sent from PBX to asterisk box which is the voicemail bank for pbx users..the problem is sending MWI notication to PBX end points
SIP MWI notify is sent to Cisco gateway, which is translated to QSIG, I had it working, but now facility info is missing? and before could get mwi activate/deactivate(80,81) message, but now the facility info is missing..
this is isdn debug output on the gateway for the MWI notification...
*Oct 15 01:15:38.759: ISDN Se0/1/0:15 Q931: TX -> RELEASE pd = 8 callref = 0xAE1B
*Oct 15 01:15:38.775: ISDN Se0/1/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x2E1B
*Oct 15 01:15:38.811: ISDN Se0/1/0:15 Q931: Sending SETUP callref = 0x0083 callID = 0x8004 switch = primary-qsig interface = User
*Oct 15 01:15:38.811: ISDN Se0/1/0:15 Q931: TX -> SETUP pd = 8 callref = 0x0083
Bearer Capability i = 0xA880
Standard = ISO/IEC
Transfer Capability = Unrestricted Digital
Transfer Mode = Circuit
Transfer Rate = Packet - not specified
Channel ID i = 0xAC
Exclusive, Channel 9
Called Party Number i = 0x80, 'xxxxxxxx'
Plan:Unknown, Type:Unknown
*Oct 15 01:15:38.827: ISDN Se0/1/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x8083
Cause i = 0x82E49804 - Invalid information element contents
*Oct 15 01:15:41.595: ISDN Se0/1/0:15 Q931: Sending SETUP callref = 0x0084 callID = 0x8005 switch = primary-qsig interface = User
*Oct 15 01:15:41.595: ISDN Se0/1/0:15 Q931: TX -> SETUP pd = 8 callref = 0x0084
Bearer Capability i = 0xA880
Standard = ISO/IEC
Transfer Capability = Unrestricted Digital
Transfer Mode = Circuit
Transfer Rate = Packet - not specified
Channel ID i = 0xAC
Exclusive, Channel 9
Called Party Number i = 0x80, 'xxxxxxxx'
Plan:Unknown, Type:Unknown
*Oct 15 01:15:41.611: ISDN Se0/1/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x8084
The second option i thought about was to connect PBX directly to Asterisk using diguim E1 cards.
Regards
Ali

If it was working before, try tracing back changes made that broke it.

7962 SCCP to SIP + Registration issue with FreePBX with asterisk engine CentOS Platform

Dear All,
     Hi,
     Please find attached file, have a look if you need so for troubleshooting. Thank you.
     I am with client, I am kind of stuck, let me know If I am wrong at any point.
     Skype ID: thewaleedkhan
Waleed
NE

There is no discussion of 3rd party call control in this forum. Try on websites like voip-info. org, or use CME that works even better.

E71 SIP REGISTRATION FAILED

I'm trying to register E71 to Brekeke SIP Server.
When password authentication is enabled on the server, the server returns "401 Unauthorized" for the UA's 1st REGISTER request to ask the UA to send user name and password. And when the UA sends the 2nd REGISTER request, the server accepts it.
The problem with E71 is after it receives "401 Unauthorized" from the server, it doesn't send the 2nd REGISTER request any more. And shows "Registration Failed" on the phone.
Does anyone have similiar problems? How can I work around this?
Thanks.

It will help you...
http://wiki.brekeke.com/wiki/InteroperabilityIP_Phones:Hard_Phones:Nokia_phone

Bug: E52/E55 SIP registration keeps WiFi on

On an E55 with firmware 022.009:
When you edit the SIP settings > Registration, and choose "On in home network" (which I think is supposed to be only "Only in home network" it doesn't save the setting and reverts to "When needed".
When it's in "When needed mode" it won't only connect when needed, but it will automatically and relentlessly connect, and continuously reconnect to the Internet all the time to register the SIP service.
You have to manually disconnect the SIP (either through the Contact menu or through Active Standby) after each call. If you don't manually disconnect it, WiFi will be turned on all the time, and it will eat up battery.
Message Edited by jj05 on 10-Jan-2010 11:52 AM

Hi there,
I have similar problem.
I also found that when I define a SIP profile, it works just for the first time that it registers. Once I disconnect it, and then connect it again, then while I call through "intenet call", it does noat dial.
Any suggestion?

Linksys SPA9000 and SIP registration with an E61

HI, I have my own IP telephony but I want to register my phone to be used with my Linksys SPA9000, anybody has the setup to do this with an E61?

Unfortunately my Nokia, an N95, can't register with the SPA9000 either. I managed to 'partially' register the phone. The phone registered itself an i saw it in the pbx status page.
(http:// - ip-adres of SPA9000 - /admin/status). The phone itself keeps notifying that it couldn't establish a connection and won't connect to the PBX.
What did the trick is realising that the PBX proxy listens to port 6060. So i added that port in the phone for the proxy and registrar. I know it's a part of the solution and i am working on it.
My plan
- setting up a syslog service on my linux server and search the logs for error mesages during the registration proces
- upgrading my firmware. The SPA9000 is up-to-date, but my N95 doesn't have the latest version yet. So i want to check the release notes from Nokia for information about solved SIP errors.
- checking the forums for tips
- ask Nokia for help

Cisco SX20 SIP Registration Failed

Hi,
I would like to share the issue we encountered on a Cisco SX20 endpoint that is failing to register on a VCS Control using SIP.
The endpoint is running TCNC6.2.0 version. The other endpoints running on TC5x.x, TC6.x.x, TC7.x.x has no issue.
The endpoint can register on the VCS but when we try to call one of the other endpoints, the call is not successful. The endpoint will ring and when we accept the call, there's no audio and video then call disconnects. After call disconnection, the endpoint failed to register on the VCS.
We tried to re-register again, restart the system then tried to do another call. Same issue encountered again.
On the VCS, we tried to create a subzone for that specific endpoint (subnet) with treat as authenticated policy. Problem still encountered.
We also tried to restore the codec to factory default settings but didn't resolve the issue.
Seeking for your advise.
Thank for help in advance.
Thanks & best regards,
Acevirgil

Since you are using the TCNC version of code, any encryption will not work for calls, so are you forcing encryption anywhere on the VCS or seeing the call try as TLS in the call history?

E65 Sip registration problem

hi
i am experiencing in registering my E65 with my internal sip server.
all other client are working fine using the same sip server.
my SIP config on e65 is below
- profile name Sensip
-service profile is IETF
- default access point is AtifAP2
- Public user name sip:150
-use compression No
-Registration: When needed
- Use security: No
-Proxy server
Proxy server address sip: 172.16.10.3
Realm None
user name: 150
password: blank
allow loose routing No
Transport type UDP
Port: 5060
- Registrar Server
Proxy server address sip: 172.16.10.3
Realm proxy: None
user name: 150
password: blank
allow loose routing No
Transport type UDP
Port: 5060
Can anybody please help.
Regards
Atif

I would get with your admin on the SIP host and verify...one tiny mistake and it does not work. It took us a lot of effort to get all the setting correct - the I documented them for everyone else. Our is of course different than yours.....
I will also add that the latest software for the E61i is now VERY hostile for the SIP phone....so you may be chasing a phantom
Ja Mata N80 | N80ie | N81 | N95 | E65 | E61 | E61i | E6i | E71| [ - NONE of them have ever worked as advertised]

My MacBook Pro Retina's Bluetooth chipset unknown/odd login message on the login screen states Login Window Authentication Login window Name edit text has keyboard focus. In addition, the login screen is not remembering me

I have been experiencing several issues with my MacBook Pro Retina mid 2012. My MBPR is scheduled to go into the depot. However, I am wondering if anyone may be able to shed light on a few issues as this is the third "official" time my MBPR is going back for service ("one depot" trip; "one authorized" dealer; several in-store visits).
My Bluetooth is stating that the Bluetooth Chipset is Unknown (0). I also have had Bluetooth Preferences mysteriously change on me. In addition, while Bluetooth is off there are two serial modems turning on. I have turned them off, but they continue to pop up.
In addition, when I log in, my MBPR is not remembering me and my login name is not appearing on the slate-gray screen. The name and password are blank and the following message appears in the lower left hand corner. "login window authentication login window Name edit text has keyboard focus." As a side note, I am the only user. The login issue is a recent occurrence as we just totally wiped it again via a Command + R, and I don't believe I have an accessibility setting set to anything that would cause this, but wanted to check.
Should I be concerned here? Has anyone else had issues like this? I don't want to worry if I don't have to. I have had so many issues over the course of nine months. 5-6 wipes. Airport card replaced and I am about to pull my hair out if my MBPR doesn't come back worldly like clock work this time. I just can't send my days trying to get a $2300 product to work for me any longer. No idea what is wrong with it, but it is driving me insane. Cross your fingers for me and any guidance you have or thoughts would be welcomed. Thank you. EMM

A few more issues...
In Console, the following is greyed out:
User and Diagnostic reports
Com.apple.launchd.peruser.0
Com.apple.launchd.peruser.88
Com.apple.launchd.peruser.89
Com.apple.launchd.peruser.92
Com.apple.launchd.peruser.97
Com.apple.launchd.peruser.200
Com.apple.launchd.peruser.201
Com.apple.launchd.peruser.202
Com.apple.launchd.peruser.212
*[user logs are accessible]
Krb5kdc
Radius
My guest files are locked, but again I am the administrator of MBPR.
I am worried about a keystroke logged or at least, trying to rule it out.
Also:
Mdworker32(225) [and other mdworker numbers] are sandboxing; stating deny Mach-lookup
Com.apple.Powermanagement.control, etc. long attachment with those files with version: ??? (???).
Postinstall: removing applications/Microsoft Office 2011/Microsoft Outlook.app
WARNINGS in Console include:
[NSImage compositeToPoint:fromRect:operation:fraction:] is deprecated in MacOSX 19.8 and later. Please use -[NSImage drawAtPoint:fromRect:operation:fraction] instead.
There are a ton of other warnings. Before I go through this again, can someone tell me if this is normal (all of it -- above too); or if these are symptoms is a keystroke logger or hardware issues?
I ask because originally, when my computer went in for diagnostics (more than once), Apple stated the hardware was fine (other than Airport Card -- finally). However, if I've done 5-6 total wipes; created new users; do not have sharing set-up; have not played around in Terminal; and am up-to-date with versions -- and various issues KEEP COMING BACK -- I am left wondering if a keystroke logger would be possible here?!? I thought maybe a faulty logic board, but why would diagnostics be okay, then? Not trying to be hyperbole, just desperate.
Please help me rule keystroke logger out or at least, tell me so I know, so I can take appropriate action. If you think it could be the logic board with symptoms above, that would be a great too.
All I want to do is use the computer as intended, but I can't seem to get a real answer, so after nine months -- I am turning to the communities to see if anyone -- anyone at all -- can help. The last thing I can do is have the MBPR come back from the depot and the same thing occur. Any guidance or advice would be so gratefully appreciated.

State of No Authentication in Asterisk SIP Registr...

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