Andrea Deltacom
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Posts
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Posts posted by Andrea Deltacom
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there are news about this feature? I need too to disable Automatic Flag with sip phone, or set both in boolean mode
thanks
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Thanks for Reply, I'm not sure how to make dial plan in Asterisk I tried:
exten => XXX,1,Dial(Zap/1)
[dialout]
exten => XXXXXXXXX,1,Dial(Zap/1,100,r)
but isn't working
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I'm trying to communicate a snom one installation with Asterisk. I have in the same machine, snom one with sip 5060, and asterisk with sip 5090. I configured in snom one a Sip Gateway with ip address 127.0.0.1:5090 and this seems ok. In Asterisk I configured an ISDN TE BRI pci card, there is 2 problem:
1. the incoming call, ring in Asterisk but don't go in Snom One extensions
2. the outgoing call, go in Asterisk but don't use idsn trunk
Can you help me?
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yup i know, if i can't find another driver maybe i do an asterisk trunk for isdn
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Hi, I'm trying to configure an OpenVOX BRI B100 ISDN PCI Card. I should installed it with zaptel drivers, but i'm not sure how configure to work with snom one. Specially what data i need to put in Trunk Configuration.
Thanks
Andrea
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we need 1 Port BRI, internal or external. The system is Centos 5 so if internal we need the drivers
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Hi, I need to use a ISDN trunk with Snom ONE. What device can I install?
Maybe an ISDN Moden PCI? I need a cheap devices.
Thanks
Andrea
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yes, it's ok. ty
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this file isn't working:
http://downloads.snom.net/snomONE/centos64/snomone-CentOS5-x64.bin
# snom m9 firmware update.
# Registrations to third-party devices (e.g., SIP phones, SIP cameras, softphones, etc.) have been increased: snom ONE free supports 5 third-party devices, snom ONE yellow supports 10, and blue supports 40.
# New addition of snom ONE is available: snom ONE green (includes call support for third-party devices).
this seem realy great!!
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ok tnx, i disabled this.
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My router is draytek 2820, it says 15.000 nat table entry. btw before i rebooted it, it was going good, after adsl went up again I can't anymore use my trunks
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Hi, I have 13 trunks (same provider) with 1 ISDN Adapter. Today after a router reboot, they aren't go on anymore with 408 error timeout message. So after many tries, I disabled some of them, and then registering one by one. So this make them work.
What could be happened? A SIP provider problem (flood deny?) or a Snom ONE problem?
I'm using
Version: 4.0.0.3343 (Linux)
License: snomONE - 15
Happy New Year
Andrea
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Hi, I'm using linksys ATA PAP2T to send and receives fax with snom one, but something isn't going well. Sometime I need to send 2-3 times before I got a positive response, what parameters I can tweak?
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thanks for your explanations, I used failover in old snom one version 4.0 (before goes free) and doesn't give me this problem.
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my trunk seems accept only G729 and G711 but thanks for reply
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ah ok i checked again and seems i didn't saved the changes. Without failover works good, I have written "For all codes error" so if a user don't reply in 10 seconds is a error? But why i can make a call against another twt number and it doesn't go in error?
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I had put it 10s, but then i tried removing it but nothing changed ... the phone rings but after some second it goes down before the calling reply
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Hi, i'm using Snom ONE Free with TWT Voip Trunks. It was going good, but from yesterday I can't no more dial. After 5 seconds it goes in timeout! I don't know what happened. It's making me crazy I didn't make any changes to my configuration
This is a part of Sip LOG
[7] 2010/12/16 10:20:05: SIP Rx udp:82.113.194.190:5060:
SIP/2.0 183 Call progress
v: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK-8065394e3b6d25dec7891a3c3b8933f7;rport=54084;received=78.7.88.171
Record-Route: <sip:5698784-192.168.0.133.dialog.cgatepro;lr>
Record-Route: <sip:192.168.0.133:5060;lr>
Record-Route: <sip:82.113.194.190:5060;lr>
f: "Deltacom Andrea" <sip:NUMTEL@localhost;user=phone>;tag=1302926275
t: <sip:NUMTEL@sip.twt.it;user=phone>;tag=B11E799E-387906-FBB87516_kmbcnrx-23A1
i: a2d6a018@pbx
CSeq: 19295 INVITE
m: <sip:signode-387906-FBB87516_kmbcnrx-23A1@82.113.194.190>
User-Agent: CommuniGatePro-callLeg/5.3.7
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,REFER
c: application/sdp
l: 267
v=0
o=CGPLeg387906 2482349901 1241174951 IN IP4 82.113.194.196
s=SIP Call
c=IN IP4 82.113.194.196
t=0 0
m=audio 20980 RTP/AVP 8 101
c=IN IP4 82.113.194.196
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcpping:F:381081:38108178
[6] 2010/12/16 10:20:05: Codec pcma/8000 is chosen for call id a2d6a018@pbx
[6] 2010/12/16 10:20:05: Sending RTP for a2d6a018@pbx to 82.113.194.196:20980, codec pcma/8000
[7] 2010/12/16 10:20:05: MjQ4MTZiOGEwYmNlZjJiMGMwMjlkYjNiZjhhZDgzYzY.: RTP pass-through mode
[7] 2010/12/16 10:20:05: a2d6a018@pbx: RTP pass-through mode
[6] 2010/12/16 10:20:05: Different Codecs (local pcmu/8000, remote pcma/8000), callid MjQ4MTZiOGEwYmNlZjJiMGMwMjlkYjNiZjhhZDgzYzY., falling back to transcoding
[7] 2010/12/16 10:20:09: SIP Tx udp:192.168.2.22:23646:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 192.168.2.22:23646;branch=z9hG4bK-d8754z-1006586e4773aa76-1---d8754z-;rport=23646
From: "999" <sip:999@192.168.2.254>;tag=3e27cd2c
To: "NUMTEL" <sip:NUMTEL@192.168.2.254>;tag=d8791cb712
Call-ID: MjQ4MTZiOGEwYmNlZjJiMGMwMjlkYjNiZjhhZDgzYzY.
CSeq: 2 INVITE
Contact: <sip:999@192.168.2.254:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/4.2.0.3958
Content-Length: 0
[7] 2010/12/16 10:20:09: SIP Tx udp:82.113.194.190:5060:
CANCEL sip:NUMTEL@sip.twt.it;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK-8065394e3b6d25dec7891a3c3b8933f7;rport
From: "Deltacom Andrea" <sip:NUMTEL@localhost;user=phone>;tag=1302926275
To: <sip:3928435208@sip.twt.it;user=phone>
Call-ID: a2d6a018@pbx
CSeq: 19295 CANCEL
Max-Forwards: 70
P-Asserted-Identity: "Consorzio Servizi Auto" <sip:NUMTEL@sip.twt.it>
Authorization: Digest realm="twtmail.twt.it",nonce="2EF7B69D7C18A493D722",response="1cc255a13c4901611ed21d8306a100b9",username="NUMTEL",uri="sip:NUMTEL@sip.twt.it;user=phone",qop="auth",nc=00000002,cnonce="cbd770c4",opaque="opaqueData",algorithm=MD5
Content-Length: 0
[7] 2010/12/16 10:20:09: SIP Rx udp:82.113.194.190:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK-8065394e3b6d25dec7891a3c3b8933f7;rport=54084;received=78.7.88.171
f: "Deltacom Andrea" <sip:NUMTEL@localhost;user=phone>;tag=1302926275
t: <sip:NUMTEL@sip.twt.it;user=phone>;tag=74D52A3E
i: a2d6a018@pbx
CSeq: 19295 CANCEL
Server: CommuniGatePro/5.3.7
l: 0
[7] 2010/12/16 10:20:09: Call a2d6a018@pbx: Clear last request
[7] 2010/12/16 10:20:09: SIP Rx udp:82.113.194.190:5060:
SIP/2.0 487 Request cancelled
v: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK-8065394e3b6d25dec7891a3c3b8933f7;rport=54084;received=78.7.88.171
f: "Deltacom Andrea" <sip:NUMTEL@localhost;user=phone>;tag=1302926275
t: <sip:NUMTEL@sip.twt.it;user=phone>;tag=B11E799E-387906-FBB87516_kmbcnrx-23A1
i: a2d6a018@pbx
CSeq: 19295 INVITE
Server: CommuniGatePro/5.3.7
l: 0
[7] 2010/12/16 10:20:09: Call a2d6a018@pbx: Clear last INVITE
[7] 2010/12/16 10:20:09: SIP Tx udp:82.113.194.190:5060:
ACK sip:signode-387906-FBB87516_kmbcnrx-23A1@82.113.194.190 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK-8065394e3b6d25dec7891a3c3b8933f7;rport
Route: <sip:82.113.194.190:5060;lr>
Route: <sip:192.168.0.133:5060;lr>
Route: <sip:5698784-192.168.0.133.dialog.cgatepro;lr>
From: "Deltacom Andrea" <sip:NUMTEL@localhost;user=phone>;tag=1302926275
To: <sip:NUMTEL@sip.twt.it;user=phone>;tag=B11E799E-387906-FBB87516_kmbcnrx-23A1
Call-ID: a2d6a018@pbx
CSeq: 19295 ACK
Max-Forwards: 70
Contact: <sip:NUMTEL@192.168.2.254:5060;transport=udp>
P-Asserted-Identity: "Consorzio Servizi Auto" <sip:509184860231@sip.twt.it>
Authorization: Digest realm="twtmail.twt.it",nonce="2EF7B69D7C18A493D722",response="b1e68d419c86b8de84ad041c2df03ccb",username="509184860231",uri="sip:signode-387906-FBB87516_kmbcnrx-23A1@82.113.194.190",qop="auth",nc=00000003,cnonce="49364538",opaque="opaqueData",algorithm=MD5
Content-Length: 0
[5] 2010/12/16 10:20:09: INVITE Response 487 Request cancelled: Terminate a2d6a018@pbx
[7] 2010/12/16 10:20:09: MjQ4MTZiOGEwYmNlZjJiMGMwMjlkYjNiZjhhZDgzYzY.: Media-aware pass-through mode
[7] 2010/12/16 10:20:09: SIP Rx udp:192.168.2.22:23646:
ACK sip:NUMTEL@192.168.2.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:23646;branch=z9hG4bK-d8754z-1006586e4773aa76-1---d8754z-;rport
To: "NUMTEL" <sip:3928435208@192.168.2.254>;tag=d8791cb712
From: "999"<sip:999@192.168.2.254>;tag=3e27cd2c
Call-ID: MjQ4MTZiOGEwYmNlZjJiMGMwMjlkYjNiZjhhZDgzYzY.
CSeq: 2 ACK
Content-Length: 0
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Tnx matt, I have read your faq days ago and it's great! I'm not sure if I'll buy a license. I have 7 snom phones (300 and m3) but my connection is a little bad so I would like to spare bandwith without spend 500$ for it.
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Hi, I can buy G729 Licenses and use them in Snom One Free as in Asterisk? Or i need to buy Snom Yellow / Blue?
Tnx
Andrea
Response 403 Unknown User/Endpoint Not Allowed
in Trunk Setup
Posted
Hi everyone,
my ISP changed his platform and I can't anymore call with snom one PBX. My configuration IS:
# Trunk 20 in domain localhost
Name: Deltacom_Hidden_1
Type: register
To: sip
RegPass: ********
Direction:
Disabled: false
Global: false
Display: 509091480XXX
RegAccount: 509091480XXX
RegRegistrar: sip.deltacomsrl.it
RegKeep:
RegUser: 509091480XXX
Icid:
Require:
OutboundProxy: sip.deltacomsrl.it
Ani: 509091480XXX
DialExtension: 15
Prefix:
Trusted: false
AcceptRedirect: false
RfcRtp: false
Analog: false
SendEmail:
UseUuid: false
Ring180: false
Failover: never
Privacy: pai
Glob:
RequestTimeout:
Codecs:
CodecLock: true
Expires: 3600
FromUser:
Tel: true
TranscodeDtmf: false
AssociatedAddresses:
InterOffice: false
DialPlan:
Colines:
DialogPermission:
And then log SIP
[5] 2011/07/20 12:11:47: SIP Rx udp:192.168.14.23:32256:
INVITE sip:090774XXX@192.168.14.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-016650393159f53f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:13@192.168.14.23:32256>
To: "090774XXX"<sip:090774XXX@192.168.14.254>
From: "13"<sip:13@192.168.14.254>;tag=ce18420b
Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 368
v=0
o=- 9 2 IN IP4 192.168.14.23
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.14.23
t=0 0
m=audio 52344 RTP/AVP 107 0 8 101
a=alt:1 3 : rxGTiS92 oGbOd6pG 192.168.76.1 52344
a=alt:2 2 : A35uDg1B 9Y0EP7mf 192.168.209.1 52344
a=alt:3 1 : KExkKgb7 WmV4o3jd 192.168.14.23 52344
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
[5] 2011/07/20 12:11:47: Last message repeated 2 times
[5] 2011/07/20 12:11:47: SIP Tx udp:192.168.14.23:32256:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-016650393159f53f-1---d8754z-;rport=32256
From: "13" <sip:13@192.168.14.254>;tag=ce18420b
To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83
Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM.
CSeq: 1 INVITE
Content-Length: 0
[5] 2011/07/20 12:11:47: SIP Tx udp:192.168.14.23:32256:
SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-016650393159f53f-1---d8754z-;rport=32256
From: "13" <sip:13@192.168.14.254>;tag=ce18420b
To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83
Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM.
CSeq: 1 INVITE
User-Agent: snomOne-PBX/2011-4.2.0.3981
WWW-Authenticate: Digest realm="192.168.14.254",nonce="595c3a4ced41b174ea57f9ef78dc6b1b",domain="sip:090774581@192.168.14.254",algorithm=MD5
Content-Length: 0
[5] 2011/07/20 12:11:47: SIP Rx udp:192.168.14.23:32256:
ACK sip:090774581@192.168.14.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-016650393159f53f-1---d8754z-;rport
To: "090774581" <sip:090774581@192.168.14.254>;tag=23f4d20a83
From: "13"<sip:13@192.168.14.254>;tag=ce18420b
Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM.
CSeq: 1 ACK
Content-Length: 0
[5] 2011/07/20 12:11:47: SIP Rx udp:192.168.14.23:32256:
INVITE sip:090774581@192.168.14.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-fe7b77615b04f54e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:13@192.168.14.23:32256>
To: "090774581"<sip:090774581@192.168.14.254>
From: "13"<sip:13@192.168.14.254>;tag=ce18420b
Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Authorization: Digest username="13",realm="192.168.14.254",nonce="595c3a4ced41b174ea57f9ef78dc6b1b",uri="sip:090774581@192.168.14.254",response="2b76fcecc75621c0f28c02fb0d59e682",algorithm=MD5
Content-Length: 368
v=0
o=- 9 2 IN IP4 192.168.14.23
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.14.23
t=0 0
m=audio 52344 RTP/AVP 107 0 8 101
a=alt:1 3 : rxGTiS92 oGbOd6pG 192.168.76.1 52344
a=alt:2 2 : A35uDg1B 9Y0EP7mf 192.168.209.1 52344
a=alt:3 1 : KExkKgb7 WmV4o3jd 192.168.14.23 52344
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
[6] 2011/07/20 12:11:47: Sending RTP for ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. to 192.168.14.23:52344, codec not set yet
[5] 2011/07/20 12:11:47: SIP Tx udp:192.168.14.23:32256:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-fe7b77615b04f54e-1---d8754z-;rport=32256
From: "13" <sip:13@192.168.14.254>;tag=ce18420b
To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83
Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM.
CSeq: 2 INVITE
Content-Length: 0
[7] 2011/07/20 12:11:47: set_codecs: for ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM. codecs "", codec_preference count 7
[5] 2011/07/20 12:11:47: Dialplan "Delta Test": Match 090774XXX@192.168.14.254 to <sip:090774XXX@sip.deltacomsrl.it;user=phone> on trunk Deltacom_Hidden_1
[7] 2011/07/20 12:11:47: Cannot convert number 50909148XXXX into global format
[7] 2011/07/20 12:11:47: Last message repeated 2 times
[7] 2011/07/20 12:11:47: set_codecs: for 47771b31@pbx codecs "", codec_preference count 7
[6] 2011/07/20 12:11:47: Codec pcma/8000 is chosen for call id ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM.
[5] 2011/07/20 12:11:47: SIP Tx udp:192.168.14.23:32256:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-fe7b77615b04f54e-1---d8754z-;rport=32256
From: "13" <sip:13@192.168.14.254>;tag=ce18420b
To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83
Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM.
CSeq: 2 INVITE
Contact: <sip:13@192.168.14.253:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomOne-PBX/2011-4.2.0.3981
Content-Type: application/sdp
Content-Length: 265
v=0
o=- 726816646 726816646 IN IP4 192.168.14.253
s=-
c=IN IP4 192.168.14.253
t=0 0
m=audio 51834 RTP/AVP 8 0 101
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2011/07/20 12:11:47: SIP Tx tcp:77.239.128.7:5060:
INVITE sip:090774XXX@sip.deltacomsrl.it;user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.14.253:35758;branch=z9hG4bK-1966c3fd6b5bdaa31520421ef93c3597;rport
From: "Andrea Ciccio'" <sip:509091480041@localhost;user=phone>;tag=1528124281
To: <sip:090774XXX@sip.deltacomsrl.it;user=phone>
Call-ID: 47771b31@pbx
CSeq: 11223 INVITE
Max-Forwards: 70
Contact: <sip:509091480041@192.168.14.253:35758;transport=tcp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomOne-PBX/2011-4.2.0.3981
P-Asserted-Identity: "509091480041" <sip:509091480041@sip.deltacomsrl.it>
Content-Type: application/sdp
Content-Length: 388
v=0
o=- 1142207063 1142207063 IN IP4 192.168.14.253
s=-
c=IN IP4 192.168.14.253
t=0 0
m=audio 57912 RTP/AVP 8 18 0 2 3 9 101
a=rtpmap:8 pcma/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 pcmu/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:9 g722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2011/07/20 12:11:47: SIP Rx tcp:77.239.128.7:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.14.253:35758;received=178.236.173.178;branch=z9hG4bK-1966c3fd6b5bdaa31520421ef93c3597;rport=51950
From: "Andrea Ciccio'" <sip:509091480041@localhost;user=phone>;tag=1528124281
To: <sip:090774XXX@sip.deltacomsrl.it;user=phone>
Call-ID: 47771b31@pbx
CSeq: 11223 INVITE
Content-Length: 0
[5] 2011/07/20 12:11:47: SIP Rx tcp:77.239.128.7:5060:
SIP/2.0 403 Unknown User/Endpoint Not Allowed
Via: SIP/2.0/TCP 192.168.14.253:35758;received=178.236.173.178;branch=z9hG4bK-1966c3fd6b5bdaa31520421ef93c3597;rport=51950
From: "Andrea Ciccio'" <sip:509091480041@localhost;user=phone>;tag=1528124281
To: <sip:090774XXX@sip.deltacomsrl.it;user=phone>;tag=aprqrjmtc-oulmvc300oaed
Call-ID: 47771b31@pbx
CSeq: 11223 INVITE
Content-Length: 0
[7] 2011/07/20 12:11:47: Call 47771b31@pbx: Clear last INVITE
[5] 2011/07/20 12:11:47: SIP Tx tcp:77.239.128.7:5060:
ACK sip:090774XXX@sip.deltacomsrl.it;user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.14.253:35758;branch=z9hG4bK-1966c3fd6b5bdaa31520421ef93c3597;rport
From: "Andrea Ciccio'" <sip:509091480041@localhost;user=phone>;tag=1528124281
To: <sip:090774XXX@sip.deltacomsrl.it;user=phone>;tag=aprqrjmtc-oulmvc300oaed
Call-ID: 47771b31@pbx
CSeq: 11223 ACK
Max-Forwards: 70
Contact: <sip:509091480041@192.168.14.253:35758;transport=tcp>
P-Asserted-Identity: "509091480041" <sip:509091480041@sip.deltacomsrl.it>
Content-Length: 0
[5] 2011/07/20 12:11:47: INVITE Response 403 Unknown User/Endpoint Not Allowed: Terminate 47771b31@pbx
[5] 2011/07/20 12:11:47: SIP Tx udp:192.168.14.23:32256:
SIP/2.0 403 Unknown User/Endpoint Not Allowed
Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-fe7b77615b04f54e-1---d8754z-;rport=32256
From: "13" <sip:13@192.168.14.254>;tag=ce18420b
To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83
Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM.
CSeq: 2 INVITE
Contact: <sip:13@192.168.14.253:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomOne-PBX/2011-4.2.0.3981
Content-Length: 0
[5] 2011/07/20 12:11:47: SIP Rx udp:192.168.14.23:32256:
ACK sip:090774XXX@192.168.14.254 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.23:32256;branch=z9hG4bK-d8754z-fe7b77615b04f54e-1---d8754z-;rport
To: "090774XXX" <sip:090774XXX@192.168.14.254>;tag=23f4d20a83
From: "13"<sip:13@192.168.14.254>;tag=ce18420b
Call-ID: ZDc0NTQzMmFhN2Y2ZjFiYmY3MDI0NDdiMjI1YmVkNmM.
CSeq: 2 ACK
Content-Length: 0
what can be?
Any idea?