Gradwell SIP

[CoolJimy84]

Hello, there long time reader....

I've got an IP 500 on 4.0 and i'm trying to get some Gradwell SIP's up and running.

Now i've spoken to 3 guys at their helpdesk and they have never heard of Avaya or an IP Office (which never helps)

Now the only thing i've got from them is that they need the request to come from our static IP, which it is.
I have got a monitor trace of the call going through but i'm getting a packet back saying "PSTN access unavailable for current account" which they couldn't help me on.

I've left the authentication name and password blank (as they say they don't need this)
the ITSP domain name has their sip.gradwell.net in it
ITSP IP address i have resolved the name above to 193.111.200.56

Every thing else is as standard.

On the SIP URI tab i've also left every thing as default (use user data)

I've spent a total of an hour talking to people on their helpdesk and they can't help...

I've also logged this with my supplier to see if they can help....

 

[mattKnight]

>
On the SIP URI tab i've also left every thing as default (use user data)

So what have you got set on the SIP tab of the user trying to make teh call?

Try looking in monitor, with the SIP tab enabled.

A good start will be to post the INVITE packet sent from the IP and the respons epacket from gradwell

Take Care

Matt
I have always wished that my computer would be as easy to use as my telephone.
My wish has come true. I no longer know how to use my telephone.

 

[tlpeter]

How is your shortcode ?

Should be something like this

N
dial
N@"123.123.123.123" where 123.123.123.123 is the ipadras of the provider
line id siptrunk

ACA - Implement IP Office
ACS - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[Bas1234]

Ask them for the SIP trunk settings, i think they gave you the settings for SIP Phones.

You could try to use IAX-LB.gradwell.net (193.111.201.98)instead of sip.gradwell.net


Greetzzz....Bas

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[CoolJimy84]

Ok i have monitor trace of the system, i have also entered one of the numebers that we have bought to the from of the @sip.blah.net


178158mS StunInfo: Creating a STUN client to resolve ipaddr c0a8270a port 49152
178158mS PRN: Creating STUNUdpClient port 49152 address f59b8e38
178158mS StunInfo: StunClient: ResolveIPAndPort port:49152 ip:c0a8270a
178158mS StunInfo: Creating a STUN client to resolve ipaddr c0a8270a port 49153
178158mS PRN: Creating STUNUdpClient port 49153 address f59b8cdc
178159mS StunInfo: StunClient: ResolveIPAndPort port:49153 ip:c0a8270a
178160mS CD: CALL: 0.1008.0 BState=Idle Cut=0 Music=0.0 Aend="BoardroomConf(245)" (4.18) Bend="Line 19" [Line 19] (0.0) CalledNum=95441372731120# () CallingNum=245 (BoardroomConf) Internal=0 Time=6713 AState=Dialling
178258mS StunInfo: StunClient: ResponseTimeout in Resolve RTP, attempt 0
178259mS StunInfo: StunClient: ResponseTimeout in Resolve RTP, attempt 0
178260mS StunInfo: Response To Enquiry is: ip 57c2a079, port: 42930
178260mS SipDebugInfo: CMMediaSTUNFilter::callback_received addr f59b8e88 (rtp f59b8e88 rtcp f59b8d2c)
178262mS StunInfo: Response To Enquiry is: ip 57c2a079, port: 42931
178262mS SipDebugInfo: CMMediaSTUNFilter::callback_received addr f59b8d2c (rtp f59b8e88 rtcp f59b8d2c)
178263mS SipDebugInfo: CMMediaSTUNFilter substituting
178264mS SipDebugInfo: extension is dialing 441372731120
178264mS SipDebugInfo: CMSetup receive, ep f59c10d0, dialog f59bf4f0
178264mS SipDebugInfo: MZ extension is dialing 441372731120
178264mS SipDebugInfo: Registration Required is 0, Primary Status 0, Secondary Status 0
178266mS SipDebugInfo: TxInvite: to 193.111.200.56,5060, ep f59c10d0, dialog f59bf4f0
178266mS SipDebugInfo: Sip_sendToNetwork packet of length 813
178266mS SipDebugInfo: SIP Line (19): SendToTarget 193.111.200.56, 5060
178266mS SIP Trunk: 19:Tx
INVITE Tel:+441372731120 SIP/2.0
Via: SIP/2.0/UDP 87.194.160.121:64374;rport;branch=z9hG4bK8ad1355330c50f71bb25f9c2f5416a0a
From: BoardroomConf <sip:01212850280@sip.gradwell.net>;tag=a498089d6cce8762
To: Tel:+441372731120
Call-ID: b33e2e3379006384f6c1d2cc615a239e@87.194.160.121
CSeq: 1236846024 INVITE
Contact: BoardroomConf <sip:BoardroomConf@87.194.160.121:64374;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Type: application/sdp
Content-Length: 305

v=0
o=UserA 3539187482 1035641516 IN IP4 87.194.160.121
s=Session SDP
c=IN IP4 87.194.160.121
t=0 0
m=audio 42930 RTP/AVP 18 4 8 0 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=fmtp:18 annexb = no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
178267mS SIP Tx: UDP 192.168.39.10:5060 -> 193.111.200.56:5060
INVITE Tel:+441372731120 SIP/2.0
Via: SIP/2.0/UDP 87.194.160.121:64374;rport;branch=z9hG4bK8ad1355330c50f71bb25f9c2f5416a0a
From: BoardroomConf <sip:01212850280@sip.gradwell.net>;tag=a498089d6cce8762
To: Tel:+441372731120
Call-ID: b33e2e3379006384f6c1d2cc615a239e@87.194.160.121
CSeq: 1236846024 INVITE
Contact: BoardroomConf <sip:BoardroomConf@87.194.160.121:64374;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Type: application/sdp
Content-Length: 305

v=0
o=UserA 3539187482 1035641516 IN IP4 87.194.160.121
s=Session SDP
c=IN IP4 87.194.160.121
t=0 0
m=audio 42930 RTP/AVP 18 4 8 0 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=fmtp:18 annexb = no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
178267mS SipDebugInfo: SIPDialog Call State transition from 0 to New state 1
178268mS SipDebugInfo: SIPDialog SDP State transition from 0 to New state 1
178293mS SIP Rx: UDP 193.111.200.56:5060 -> 192.168.39.10:5060
SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL)
Via: SIP/2.0/UDP 87.194.160.121:64374;rport=64374;branch=z9hG4bK8ad1355330c50f71bb25f9c2f5416a0a
From: BoardroomConf <sip:01212850280@sip.gradwell.net>;tag=a498089d6cce8762
To: Tel:+441372731120;tag=95e55e7393f7733fb0d79381494abb70.8c9c
Call-ID: b33e2e3379006384f6c1d2cc615a239e@87.194.160.121
CSeq: 1236846024 INVITE
Server: Sip EXpress router (0.9.4 (i386/freebsd))
Content-Length: 0
Warning: 392 193.111.200.56:5060 "Noisy feedback tells: pid=8265 req_src_ip=87.194.160.121 req_src_port=64374 in_uri=Tel:+441372731120 out_uri=Tel:+441372731120 via_cnt==1"

178294mS SIP Trunk: 19:Rx
SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL)
Via: SIP/2.0/UDP 87.194.160.121:64374;rport=64374;branch=z9hG4bK8ad1355330c50f71bb25f9c2f5416a0a
From: BoardroomConf <sip:01212850280@sip.gradwell.net>;tag=a498089d6cce8762
To: Tel:+441372731120;tag=95e55e7393f7733fb0d79381494abb70.8c9c
Call-ID: b33e2e3379006384f6c1d2cc615a239e@87.194.160.121
CSeq: 1236846024 INVITE
Server: Sip EXpress router (0.9.4 (i386/freebsd))
Content-Length: 0
Warning: 392 193.111.200.56:5060 "Noisy feedback tells: pid=8265 req_src_ip=87.194.160.121 req_src_port=64374 in_uri=Tel:+441372731120 out_uri=Tel:+441372731120 via_cnt==1"

178296mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1
178296mS SipDebugInfo: Find End Point b33e2e3379006384f6c1d2cc615a239e@87.194.160.121
178296mS SipDebugInfo: Process SIP response dialog f59bf4f0, method INVITE,CodeNum 479 in state 1
178296mS SipDebugInfo: ExtractRouteFromRecord, entered
178297mS SipDebugInfo: ExtractContactFromMessage: cannot get From Header 2012
178297mS SipDebugInfo: SIPDialog Call State transition from 1 to New state 17
178297mS SipDebugInfo: SendSIPRequest: ACK SENT TO 193.111.200.56 5060
178298mS SipDebugInfo: Sip_sendToNetwork packet of length 432
178298mS SipDebugInfo: SIP Line (19): SendToTarget 193.111.200.56, 5060
178298mS SIP Trunk: 19:Tx
ACK Tel:+441372731120 SIP/2.0
Via: SIP/2.0/UDP 87.194.160.121:64374;rport;branch=z9hG4bK8ad1355330c50f71bb25f9c2f5416a0a
From: BoardroomConf <sip:01212850280@sip.gradwell.net>;tag=a498089d6cce8762
To: Tel:+441372731120;tag=95e55e7393f7733fb0d79381494abb70.8c9c
Call-ID: b33e2e3379006384f6c1d2cc615a239e@87.194.160.121
CSeq: 1236846024 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

178299mS SIP Tx: UDP 192.168.39.10:5060 -> 193.111.200.56:5060
ACK Tel:+441372731120 SIP/2.0
Via: SIP/2.0/UDP 87.194.160.121:64374;rport;branch=z9hG4bK8ad1355330c50f71bb25f9c2f5416a0a
From: BoardroomConf <sip:01212850280@sip.gradwell.net>;tag=a498089d6cce8762
To: Tel:+441372731120;tag=95e55e7393f7733fb0d79381494abb70.8c9c
Call-ID: b33e2e3379006384f6c1d2cc615a239e@87.194.160.121
CSeq: 1236846024 ACK
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

178299mS CMLineRx: v=0
CMReleaseComp
Line: type=IPLine 19 Call: lid=0 id=1010 in=0
IE CMIERespondingPartyNumber (230)(P:100 S:100 T:0 N:100 R:4) number=441372731120
IE CMIEDeviceDetail (231) LOCALE=eng HW=8 VER=4 class=CMDeviceSIPTrunk type=0 number=19 channel=0 rx_gain=32 tx_gain=32 ep_callid=0 ipaddr=0.0.0.0 apps=0
178300mS CMCallEvt: 0.1009.0 3 TargetingEP: RequestEnd 0.1010.0 3 SIPTrunk Endpoint
178300mS CMTARGET: 0.1008.0 3 BoardroomConf.0: CancelTimer CMTCNoAnswerTimeout
178300mS CMCallEvt: 0.1009.0 -1 BaseEP: DELETE CMEndpoint f59c5448 TOTAL NOW=2 CALL_LIST=1
178300mS CMCallEvt: 0.1010.0 3 SIPTrunk Endpoint: StateChange: END=B CMCSOffering->CMCSCompleted
178302mS CMLOGGING: CALL:2008/09/2516:19,00:00:00,000,245,O,441372731120,95441372731120#,BoardroomConf,,,0,,""n/a,0
178302mS CD: CALL: 0.1008.0 BState=Disconnecting Cut=0 Music=0.0 Aend="BoardroomConf(245)" (4.18) Bend="Line 19" [Line 19] (0.0) CalledNum=441372731120 () CallingNum=245 (BoardroomConf) Internal=0 Time=6855 AState=Dialling
178302mS CD: CALL: 0.1008.0 Deleted
178303mS CMExtnEvt: BoardroomConf: CALL LOST (CMCauseNormal)
178303mS CMExtnEvt: BoardroomConf: Extn(245) Calling Party Number(245) Type(CMNTypeInternal)
178303mS CMCallEvt: 0.1008.0 -1 BoardroomConf.0: StateChange: END=X CMCSDialling->CMCSCompletedTone
178303mS CMExtnEvt: v=2 State, new=CMESCompleted old=Dialling,0,0,BoardroomConf
178304mS SipDebugInfo: Terminating dialog f59bf4f0, state 17 for cause 16
178304mS CMCallEvt: END CALL:3 (f5f5393c)
178305mS SipDebugInfo: SIPDialog Call State transition from 17 to New state 40
178305mS SipDebugInfo: SIP Line (19): Cannot free Txn Key 2015
178305mS CMMap: a=21.39 b=0.0 DTMF::AllocateRAS allocated CMRTTonegen resource busy 6, total 8
178306mS CMMap: a=21.39 b=1.255 T
178306mS CMMap: a=21.39 b=4.18 M2
178354mS RES: Thu 25/9/2008 16:19:52 FreeMem=79555016(27) CMMsg=2 (2) Buff=200 952 999 7174 5 Links=21103
180051mS CMExtnRx: v=245, p1=0
CMReleaseComp
Line: type=AnalogueExtn 4 Call: lid=0 id=2 in=0
180051mS CMCallEvt: 0.1008.0 -1 BoardroomConf.0: StateChange: END=X CMCSCompletedTone->CMCSCompleted
180052mS CMExtnEvt: BoardroomConf: CALL LOST (CMCauseForceClear)
180052mS CMExtnEvt: BoardroomConf: Extn(245) Calling Party Number(245) Type(CMNTypeInternal)
180052mS CMExtnEvt: BoardroomConf: CMExtnHandler::SetCurrent( id: 1008->0 )
180052mS CMCallEvt: 0.1008.0 -1 BoardroomConf.-1: StateChange: END=X CMCSCompleted->CMCSDelete
180052mS CMExtnEvt: v=2 State, new=PortRecoverDelay old=CMESCompleted,0,0,BoardroomConf
180053mS CMTARGET: 0.1008.0 -1 BaseEP: ~CMTargetHandler
180053mS CMCallEvt: 0.1008.0 -1 BaseEP: DELETE CMEndpoint f59c63c0 TOTAL NOW=1 CALL_LIST=0
180054mS CMMap: a=21.39 b=4.18 M0
180054mS CMMap: a=21.39 b=0.0 T0
180054mS CMMap: a=21.39 b=0.0 DTMF::~DTMF freed CMRTTonegen resource busy 5, total 8

 

[Bas1234]

Are you sure they're using STUN?
If not turn it off.

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[CoolJimy84]

Thats the trouble, they don't know....

the guys i've spoken to have only configuraed asterxs system, so they can't tell me much apart from the ip address, even when i ask if there expecting it in internaltion format they say they are "fairly sure that they don't".....

 

[Bas1234]

When I do a ping to stun.gradwell.com i get a reply from 79.135.125.172

You could use that as your STUN ip address

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[dymip]

Hope to find here the sip solution we’ve looking for from long ago.
Facts:
This sip line is registered into an xlite softphone and its working right.
After checking many times the document Example SIP configurations for Release 4 by Mr IPO by our great poster MRIPO, and Checking my log
I found some unusual lines.
Quoting page 21 (the second ITSP example)
This LINES do never appear on my log.
2457742mS SipDebugInfo: SIPTrunks: active channels 0 overall number 10
this line is not present

788254mS SipDebugInfo: MZ extension is dialing 0296677152@byo.engin.com.au
instead of this I get
411808326mS SipDebugInfo: extension is dialing 0018007902029@64.94.36.152
note the lack of “MZ”

4166922mS SipDebugInfo: MZ: INVITE SENT TO 125.254.25.2 5060
instead of this I get
411808337mS SipDebugInfo: TxInvite: INVITE SENT TO 62.94.36.138 5060

4166923mS SipDebugInfo: MZ: Entered Sip_sendToNetwork packet destination is 125.254.25.2
instead of this I get
411808340mS SipDebugInfo: Sip_sendToNetwork packet of length 821

4166923mS SipDebugInfo: MZ SIPTrunk SendToTarget 7dfe1902, 5060
This line is not present

411808340mS SipDebugInfo: SIPTrunk SendToTarget 3e5e248a, 5060
From: Extn210 <sip:Extn210@soulaustralia.com.au>;tag=160a19c954b9aeb9
instead of this I get
From: 18007081111 <sip:18007081111@voip.telme.sg>;tag=932c83d3764a24b5



THIS LINES do appear on my log and DO NOT appear on the document
411808326mS SipDebugInfo: CMSetup receive, ep ffcb6b20, dialog ffcf4f50

411808328mS SipDebugInfo: Registration Required is 1, Primary Status 0, Secondary Status 0

411809131mS SIP Trunk: 10:Rx
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
In the document the log shows: trying and ringing but never “not authorized”

The system description is:
IPO SOE 4T 8A
Release 4.0 (14)
SIP channels 1 (its active)
ITSP voip.telme.sg
sip line id 10
Codec G729A

notice: this IPO is working as a gateway for Internet acces, LAN1 port is plugged to our LAN, and LAN2 si plugged to the ADSL modem (une.com.co) so we can assure the IPO is accesing internet.
system monitor has enabled all the SIP checkboxes
We think that I could be an error when sending the digits.
HERE IT’S THE LOG.

411808302mS CMARS: CMARSHandler::FindActiveARSByGroupID GroupID=249
411808303mS SipDebugInfo: SIPTrunks: Make Target voip, line group id is 249 and ip of 3e5e248a
411808304mS CMMap: a=3.1 b=0.0 Mapper::AllocateCodec allocated CMRTVocoder resource busy 1, total 3
411808305mS SipDebugInfo: License, Valid 1, Available 1, Consumed 0
411808326mS SipDebugInfo: extension is dialing 0018007902029@64.94.36.152
411808326mS SipDebugInfo: CMSetup receive, ep ffcb6b20, dialog ffcf4f50
411808327mS SipDebugInfo: MZ extension is dialing 0018007902029@64.94.36.152
411808327mS SipDebugInfo: *********************************************************
411808328mS SipDebugInfo: INVITE (method) SENT TO 62.94.36.138 5060
411808328mS SipDebugInfo: Registration Required is 1, Primary Status 0, Secondary Status 0
411808328mS SipDebugInfo: *********************************************************
411808337mS SipDebugInfo: *********************************************************
411808337mS SipDebugInfo: TxInvite: INVITE SENT TO 62.94.36.138 5060
411808338mS SipDebugInfo: *********************************************************
411808338mS SipDebugInfo: Sending INVITE, ep ffcb6b20, dialog ffcf4f50
411808340mS SipDebugInfo: Sip_sendToNetwork packet of length 821
411808340mS SipDebugInfo: SIPTrunk SendToTarget 3e5e248a, 5060
411808341mS SIP Trunk: 10:Tx
INVITE sip:0018007902029@64.94.36.152 SIP/2.0
Via: SIP/2.0/UDP 202.233.114.87:5060;rport;branch=z9hG4bK4526c0fd63e29de97c660c9ceff338ea
From: 18007081111 <sip:18007081111@voip.telme.sg>;tag=932c83d3764a24b5
To: <sip:0018007902029@64.94.36.152>
Call-ID: 839832baf42752ce699549f6d57b9092@202.233.114.87
CSeq: 611619632 INVITE
Contact: 18007081111 <sip:18007081111@202.233.114.87:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Type: application/sdp
Content-Length: 305

v=0
o=UserA 3508398016 1663733971 IN IP4 202.233.114.87
s=Session SDP
c=IN IP4 202.233.114.87
t=0 0
m=audio 49152 RTP/AVP 18 4 8 0 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=fmtp:18 annexb = no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
411808341mS SIP Tx: UDP 192.168.254.1:5060 -> 62.94.36.138:5060
INVITE sip:0018007902029@64.94.36.152 SIP/2.0
Via: SIP/2.0/UDP 202.233.114.87:5060;rport;branch=z9hG4bK4526c0fd63e29de97c660c9ceff338ea
From: 18007081111 <sip:18007081111@voip.telme.sg>;tag=932c83d3764a24b5
To: <sip:0018007902029@64.94.36.152>
Call-ID: 839832baf42752ce699549f6d57b9092@202.233.114.87
CSeq: 611619632 INVITE
Contact: 18007081111 <sip:18007081111@202.233.114.87:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Type: application/sdp
Content-Length: 305

v=0
o=UserA 3508398016 1663733971 IN IP4 202.233.114.87
s=Session SDP
c=IN IP4 202.233.114.87
t=0 0
m=audio 49152 RTP/AVP 18 4 8 0 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=fmtp:18 annexb = no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
411808343mS SipDebugInfo: initialising mTxnContext
411808345mS SipDebugInfo: *********************************************************
411808346mS SipDebugInfo: State Transtion form Old State 0 to New state 1
411808346mS SipDebugInfo: *********************************************************
411808347mS SipDebugInfo: SIPDialog::UpdateSDPState has just transitioned to state 1
411808361mS CD: CALL: 0.1738.0 BState=Idle Cut=1 Music=0.0 Aend="GERENCIA(202)" (0.10) Bend="" [Line 10] (0.0) CalledNum=0018007902029 () CallingNum=202 (GERENCIA) Internal=0 Time=10854 AState=Dialled
411809131mS SIP Rx: UDP 62.94.36.138:5060 -> 192.168.254.1:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 202.233.114.87:5060;rport=60524;branch=z9hG4bK4526c0fd63e29de97c660c9ceff338ea
To: <sip:0018007902029@64.94.36.152>
From: 18007081111 <sip:18007081111@voip.telme.sg>;tag=932c83d3764a24b5
Call-ID: 839832baf42752ce699549f6d57b9092@192.168.254.1
CSeq: 611619632 INVITE
Max-Forwards: 70

http://WWW-Authenticate:

Digest realm="62.94.36.138",nonce="2121778393",stale=FALSE,algorithm=MD5
Content-Length: 0

411809131mS SIP Trunk: 10:Rx
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 202.233.114.87:5060;rport=60524;branch=z9hG4bK4526c0fd63e29de97c660c9ceff338ea
To: <sip:0018007902029@64.94.36.152>
From: 18007081111 <sip:18007081111@voip.telme.sg>;tag=932c83d3764a24b5
Call-ID: 839832baf42752ce699549f6d57b9092@192.168.254.1
CSeq: 611619632 INVITE
Max-Forwards: 70

http://WWW-Authenticate:

Digest realm="62.94.36.138",nonce="2121778393",stale=FALSE,algorithm=MD5
Content-Length: 0

411809132mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget
411809138mS PRN: MZ stubs sip_cbk_fetchTxn no element found
411809138mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 2
411809139mS SipDebugInfo: SIP: ProcessInbound Message
411809139mS SipDebugInfo: SipTrunks: Cannot free Txn Key 2015
411810197mS SIP Rx: UDP 62.94.36.138:5060 -> 192.168.254.1:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 202.233.114.87:5060;rport=60524;branch=z9hG4bK4526c0fd63e29de97c660c9ceff338ea
To: <sip:0018007902029@64.94.36.152>
From: 18007081111 <sip:18007081111@voip.telme.sg>;tag=932c83d3764a24b5
Call-ID: 839832baf42752ce699549f6d57b9092@192.168.254.1
CSeq: 611619632 INVITE
Max-Forwards: 70

http://WWW-Authenticate:

Digest realm="62.94.36.138",nonce="2121778393",stale=FALSE,algorithm=MD5
Content-Length: 0

 

[IPOfficeIreland]

cooljimmy84, you are trying to call Tel:+00

Untick use Tel Uri in Sip, post new trace.

dymip, you should really post a new thread. 401 unathorised is simply either wrong uname/pw. What settings do you have under SIP - all settings, and what are your s/c. incoming OK?