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Asterisk@home and sipgate incoming calls

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Old August 24th, 2005, 02:16 AM
degsod degsod is offline
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Join Date: Aug 2005
Posts: 62
degsod
Default Asterisk@home and sipgate incoming calls
Hi,

I have setup an asterix@home server with my sipgate account.I can make external calls but not receive calls.

I followed the instructions as per user guide , when I dial in the phone appears dead but then disconnects after about aminute.

In the logs you can see my number coming in but asterisk is not routing.

Here is the details from the log in verbose mode:-



Aug 23 18:45:47 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 18:45:57 VERBOSE[1791]:

Sip read:


Aug 23 18:45:57 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 18:45:58 VERBOSE[1791]:

Sip read:
INVITE sip:MyUsername@172.203.247.115:5060 SIP/2.0
Record-Route:
Record-Route:
Max-Forwards: 8
Record-Route:
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKb759.d3df40a6.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKb759.a5b136f3.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKb759.95b136f3.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK242d9c23
From: "*****INCOMING*****NUMBER****" ;tag=as4d8c211d
To:
Contact:
Call-ID: 088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate.n et
CSeq: 102 INVITE
User-Agent: sipgate asterisk
Date: Tue, 23 Aug 2005 22:45:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 451

v=0
o=root 13711 13711 IN IP4 217.10.79.218
s=session
c=IN IP4 217.10.79.219
t=0 0
m=audio 35500 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSuppff - - - -
a=direction:active
a=nortpproxy:yes

Aug 23 18:45:58 VERBOSE[1791]: 19 headers, 20 lines
Aug 23 18:45:58 VERBOSE[1791]: Using latest request as basis request
Aug 23 18:45:58 VERBOSE[1791]: Sending to 217.10.79.219 : 5060 (non-NAT)
Aug 23 18:45:58 VERBOSE[1791]: Found peer 'sipgate'
Aug 23 18:45:58 DEBUG[1791]: Setting NAT on RTP to 0
Aug 23 18:45:58 VERBOSE[1791]: Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKb759.d3df40a6.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKb759.a5b136f3.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKb759.95b136f3.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK242d9c23
From: "*****INCOMING*****NUMBER****" ;tag=as4d8c211d
To: ;tag=as6b5de0d7
Call-ID: 088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate.n et
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="42f02d57"
Content-Length: 0


to 217.10.79.219:5060
Aug 23 18:45:58 VERBOSE[1791]: Scheduling destruction of call '088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate. net' in 15000 ms
Aug 23 18:45:58 VERBOSE[1791]:

Sip read:
ACK sip:MyUsername@172.203.247.115:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKb759.d3df40a6.0
From: "*****INCOMING*****NUMBER****" ;tag=as4d8c211d
Call-ID: 088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate.n et
To: ;tag=as6b5de0d7
CSeq: 102 ACK
User-Agent: sipgate ser
Content-Length: 0


Aug 23 18:45:58 VERBOSE[1791]: 8 headers, 0 lines
Aug 23 18:45:58 DEBUG[1791]: Stopping retransmission on '088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate. net' of Response 102: Found
Aug 23 18:45:58 VERBOSE[1791]:

Sip read:
INVITE sip:MyUsername@172.203.247.115:5060 SIP/2.0
Record-Route:
Record-Route:
Max-Forwards: 8
Record-Route:
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKc759.136328a3.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKc759.07d70c23.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKc759.f6d70c23.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK0f4df58c
From: "*****INCOMING*****NUMBER****" ;tag=as4d8c211d
To:
Contact:
Call-ID: 088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate.n et
CSeq: 103 INVITE
User-Agent: sipgate asterisk
Date: Tue, 23 Aug 2005 22:45:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 451

v=0
o=root 13711 13712 IN IP4 217.10.79.218
s=session
c=IN IP4 217.10.79.219
t=0 0
m=audio 35500 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSuppff - - - -
a=direction:active
a=nortpproxy:yes

Aug 23 18:45:58 VERBOSE[1791]: 19 headers, 20 lines
Aug 23 18:45:58 VERBOSE[1791]: Using latest request as basis request
Aug 23 18:45:58 VERBOSE[1791]: Sending to 217.10.79.219 : 5060 (non-NAT)
Aug 23 18:45:58 VERBOSE[1791]: Found peer 'sipgate'
Aug 23 18:45:58 DEBUG[1791]: Setting NAT on RTP to 0
Aug 23 18:45:58 VERBOSE[1791]: Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKc759.136328a3.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKc759.07d70c23.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKc759.f6d70c23.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK0f4df58c
From: "*****INCOMING*****NUMBER****" ;tag=as4d8c211d
To: ;tag=as6b5de0d7
Call-ID: 088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate.n et
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="52d4589e"
Content-Length: 0


to 217.10.79.219:5060
Aug 23 18:45:58 VERBOSE[1791]: Scheduling destruction of call '088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate. net' in 15000 ms
Aug 23 18:45:58 VERBOSE[1791]:

Sip read:
ACK sip:MyUsername@172.203.247.115:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bKc759.136328a3.0
From: "*****INCOMING*****NUMBER****" ;tag=as4d8c211d
Call-ID: 088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate.n et
To: ;tag=as6b5de0d7
CSeq: 103 ACK
User-Agent: sipgate ser
Content-Length: 0


Aug 23 18:45:58 VERBOSE[1791]: 8 headers, 0 lines
Aug 23 18:45:58 DEBUG[1791]: Stopping retransmission on '088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate. net' of Response 103: Found
Aug 23 18:45:58 VERBOSE[1791]:

Sip read:
INVITE sip:MyUsername@172.203.247.115:5060 SIP/2.0
Record-Route:
Record-Route:
Max-Forwards: 8
Record-Route:
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bK9759.77b64743.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK9759.66edc631.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK9759.56edc631.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK4ed020b0
From: "*****INCOMING*****NUMBER****" ;tag=as4d8c211d
To:
Contact:
Call-ID: 088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate.n et
CSeq: 104 INVITE
User-Agent: sipgate asterisk
Date: Tue, 23 Aug 2005 22:45:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 451

v=0
o=root 13711 13713 IN IP4 217.10.79.218
s=session
c=IN IP4 217.10.79.219
t=0 0
m=audio 35500 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:110 speex/8000
a=rtpmap:7 LPC/8000
a=rtpmap:10 L16/8000
a=silenceSuppff - - - -
a=direction:active
a=nortpproxy:yes

Aug 23 18:45:58 VERBOSE[1791]: 19 headers, 20 lines
Aug 23 18:45:58 VERBOSE[1791]: Using latest request as basis request
Aug 23 18:45:58 VERBOSE[1791]: Sending to 217.10.79.219 : 5060 (non-NAT)
Aug 23 18:45:58 VERBOSE[1791]: Found peer 'sipgate'
Aug 23 18:45:58 DEBUG[1791]: Setting NAT on RTP to 0
Aug 23 18:45:58 VERBOSE[1791]: Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bK9759.77b64743.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK9759.66edc631.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK9759.56edc631.0
Via: SIP/2.0/UDP 217.10.79.218:5060;branch=z9hG4bK4ed020b0
From: "*****INCOMING*****NUMBER****" ;tag=as4d8c211d
To: ;tag=as6b5de0d7
Call-ID: 088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate.n et
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="2b5f211e"
Content-Length: 0


to 217.10.79.219:5060
Aug 23 18:45:58 VERBOSE[1791]: Scheduling destruction of call '088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate. net' in 15000 ms
Aug 23 18:45:58 VERBOSE[1791]:

Sip read:
ACK sip:MyUsername@172.203.247.115:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.219;branch=z9hG4bK9759.77b64743.0
From: "*****INCOMING*****NUMBER****" ;tag=as4d8c211d
Call-ID: 088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate.n et
To: ;tag=as6b5de0d7
CSeq: 104 ACK
User-Agent: sipgate ser
Content-Length: 0


Aug 23 18:45:58 VERBOSE[1791]: 8 headers, 0 lines
Aug 23 18:45:58 DEBUG[1791]: Stopping retransmission on '088b3999483345a47bc517b25a272ff6@gw02.uk.sipgate. net' of Response 104: Found
Aug 23 18:46:02 VERBOSE[1791]:

Sip read:

Aug 23 18:46:02 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 18:46:07 VERBOSE[1791]:

Sip read:


Aug 23 18:46:07 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 18:46:07 VERBOSE[1791]: 0 headers, 0 lines


I have got the register command in the form myuserid:mypassword@sipgate.co.uk/myuserid

in the extensions.conf i have commented out the lines as suggested and added


exten => _.,1,Wait(1)
exten => _.,2,Goto(from-pstn,s,1)


If I change my conf to


context=sipgate.co.uk
allow=alaw
authuser=6012065
canreinvite=no
context=from-sipgate
disallow=all
dtmfmode=info
fromdomain=sipgate.co.uk
fromuser=6012065
host=sipgate.co.uk
insecure=very
nat=yes
secret=password
type=peer
username=6012065

now what happens is that i get a constant engaged number when i call my number.

Also there is no longer any sign of my inward number in the logs:-

Sip read:


Aug 23 21:14:57 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 21:15:03 VERBOSE[1791]:

Sip read:

Aug 23 21:15:03 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 21:15:07 VERBOSE[1791]:

Sip read:


Aug 23 21:15:07 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 21:15:17 VERBOSE[1791]:

Sip read:


Aug 23 21:15:17 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 21:15:25 VERBOSE[1791]:

Sip read:

Aug 23 21:15:25 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 21:15:27 VERBOSE[1791]:

Sip read:


Aug 23 21:15:27 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 21:15:37 VERBOSE[1791]:

Sip read:


Aug 23 21:15:37 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 21:15:40 VERBOSE[1791]:

Sip read:

Aug 23 21:15:40 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 21:15:47 VERBOSE[1791]:

Sip read:


Aug 23 21:15:47 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 21:15:56 VERBOSE[1791]:

Sip read:

Aug 23 21:15:56 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 21:15:57 VERBOSE[1791]:

Sip read:


Aug 23 21:15:57 VERBOSE[1791]: 0 headers, 0 lines
Aug 23 21:16:03 NOTICE[1791]: -- Re-registration for 6012065@sipgate.co.uk
Aug 23 21:16:03 DEBUG[1791]: Setting NAT on RTP to 4
Aug 23 21:16:03 DEBUG[1791]: Scheduled a registration timeout # 967
Aug 23 21:16:03 DEBUG[1791]: >>> Re-using Auth data for 6012065@sipgate.co.uk
Aug 23 21:16:03 VERBOSE[1791]: 12 headers, 0 lines
Aug 23 21:16:03 VERBOSE[1791]: Reliably Transmitting:
REGISTER sip:sipgate.co.uk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK21bd2411;rport
From: ;tag=as5d90ddb0
To:
Call-ID: 795e3afe01c6739902bd88d0512574a7@sipgate.co.uk
CSeq: 133 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="6012065", realm="sipgate.co.uk", algorithm=MD5, uri="sip:sipgate.co.uk", nonce="430bca438fd135c2c48b617997ddca6e199b3ac1", response="da0ab1ff71818b4ebcc183f6bb2acaee", opaque=""
Expires: 120
Contact:
Event: registration
Content-Length: 0

(NAT) to 217.10.79.219:5060
Aug 23 21:16:03 VERBOSE[1791]:

Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK21bd2411;rport=5060 ;received=172.203.247.115
From: ;tag=as5d90ddb0
To: ;tag=019efdbfb68dd123f382ae5ce73ea92d.aef5
Call-ID: 795e3afe01c6739902bd88d0512574a7@sipgate.co.uk
CSeq: 133 REGISTER
WWW-Authenticate: Digest realm="sipgate.co.uk", nonce="430bcb7ff0e531d36ca24085e14b07ff3e4a8902", stale=true
Server: sipgate ser
Content-Length: 0
Warning: 392 217.10.79.219:5060 "Noisy feedback tells: pid=1862 req_src_ip=172.203.247.115 req_src_port=5060 in_uri=sip:sipgate.co.uk out_uri=sip:sipgate.co.uk via_cnt==1"


Aug 23 21:16:03 VERBOSE[1791]: 10 headers, 0 lines
Aug 23 21:16:03 DEBUG[1791]: Stopping retransmission on '795e3afe01c6739902bd88d0512574a7@sipgate.co.uk' of Request 133: Found
Aug 23 21:16:03 VERBOSE[1791]: Responding to challenge, registration to domain/host name sipgate.co.uk
Aug 23 21:16:03 VERBOSE[1791]: 12 headers, 0 lines
Aug 23 21:16:03 VERBOSE[1791]: Reliably Transmitting:
REGISTER sip:sipgate.co.uk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK73025204;rport
From: ;tag=as5d90ddb0
To: ;tag=019efdbfb68dd123f382ae5ce73ea92d.aef5
Call-ID: 795e3afe01c6739902bd88d0512574a7@sipgate.co.uk
CSeq: 134 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="6012065", realm="sipgate.co.uk", algorithm=MD5, uri="sip:sipgate.co.uk", nonce="430bcb7ff0e531d36ca24085e14b07ff3e4a8902", response="4de5c2169aedf16ee2fe96cc61e2fa1b", opaque=""
Expires: 120
Contact:
Event: registration
Content-Length: 0

(NAT) to 217.10.79.219:5060
Aug 23 21:16:03 VERBOSE[1791]:

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK73025204;rport=5060 ;received=172.203.247.115
From: ;tag=as5d90ddb0
To: ;tag=019efdbfb68dd123f382ae5ce73ea92d.aef5
Call-ID: 795e3afe01c6739902bd88d0512574a7@sipgate.co.uk
CSeq: 134 REGISTER
Contact: ;q=0.00;expires=120
Server: sipgate ser
Content-Length: 0
Warning: 392 217.10.79.219:5060 "Noisy feedback tells: pid=1861 req_src_ip=172.203.247.115 req_src_port=5060 in_uri=sip:sipgate.co.uk out_uri=sip:sipgate.co.uk via_cnt==1"


Aug 23 21:16:03 VERBOSE[1791]: 10 headers, 0 lines
Aug 23 21:16:03 DEBUG[1791]: Stopping retransmission on '795e3afe01c6739902bd88d0512574a7@sipgate.co.uk' of Request 134: Found
Aug 23 21:16:03 DEBUG[1791]: Registration successful
Aug 23 21:16:03 DEBUG[1791]: Cancelling timeout 967
Aug 23 21:16:03 NOTICE[1791]: Outbound Registration: Expiry for sipgate.co.uk is 120 sec (Scheduling reregistration in 105000 ms)
Aug 23 21:16:03 VERBOSE[1791]: Destroying call '795e3afe01c6739902bd88d0512574a7@sipgate.co.uk'
Aug 23 21:16:07 VERBOSE[1791]:


any help would be appreciated
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  #2 (permalink)  
Old August 24th, 2005, 04:55 PM
mberlant's Avatar
mberlant mberlant is offline
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Join Date: Aug 2004
Location: USA or Japan
Posts: 5,013
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Default RE: Asterisk@home and sipgate incoming calls
Would you please post your sipgate "register=" line from sip.conf?

Also, when you do a "sip show registry", does sipgate show as registered?
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  #3 (permalink)  
Old August 24th, 2005, 08:17 PM
degsod degsod is offline
Member
  
Join Date: Aug 2005
Posts: 62
degsod
Default RE: Asterisk@home and sipgate incoming calls
I have sort of got it sorted now, my sip registry was:-

myuserid:mypassword@sipgate.co.uk/myuserid

Sipgate showed as registered.

my userid was 6012065, what I did was go into extensions in AMP and created an extension 6012065. I was then able to use DID to forward calls through the system.

However I would like to be able to use the hours function if this is possible.

Also I have setup some queues which work great, however how could I transfer a call through to a queue?

TIA
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