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Dimitri

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Posts posted by Dimitri

  1. Trunk incoming call set-up problem

    in Trunk Setup

    Posted

    Hello snom One,

     

    Thanks for the reply ... My log levels are set on 9. But when I call in to a number of this trunk, notting happens. No extra logs ... Like I said, I hear a ring but the PBX gives notting (and de phone connected to the PBX stays dead). I don't understand why I'm seeing the call in my WireShark & firewall logs, and not in the PBX level 9 logs. It's like the PBX doesn't treat the incoming call although the trunk is correctly registered (outgoing calls are possible).

     

    The help desk @3StarsNet, although they don't have Snom One experience, told me to start a search for an option present in Asterisk. I don't have a name, but the gave me a description of this option. The option makes it possible to receive SIP invites coming from other servers then the registered proxy or address server. I think this option is similar to the Snom One option 'Associated Addresses'. I already completed this field with the 3 servers sending me the SIP invites 85.119.188.31, 85.119.188.67 & 85.119.188.2. I'm I correct?

     

    I've 2 other trunks on the Snom One. Those trunks have just 1 number and receive their SIP invites from the address & proxy server (85.119.188.3). Those trunks are working correctly. I see all activities of incoming calls in the log ... So the log is working.

     

    Thank you,

    Dimitri

  2. Trunk incoming call set-up problem

    in Trunk Setup

    Posted

    Thank you for the response SnomOne.

     

    I already checked these settings a few times. But it's possible that I'm missing out something. My accounts are like these 29 02880xx29 Extension (admin@....com). So everything is mentioned. I already tried to forward all traffic to extension 29, but notting changes. My country code is 32 and encoded in the domain settings (we don't have an area code, so this is left blank).

    On this same PBX, I've got a SIP registration account (one number), this one is coming in and matching the correct extension. When I try to contact one of the 10 numbers, I see traces in my firewall (ZyXel USG) and on my server (thanks to WireShark). I already called 3StarsNet (the VOIP provider) and the told me they see an error 404. Strange because my PBX log is on 9 and there is no mention of an incoming call being refused.

    If there is indeed a 404 error, I suppose it has to see something with not recognizing the called number ... I just don't get were.

     

    Thank you,

    Dimitri

  3. Trunk incoming call set-up problem

    in Trunk Setup

    Posted

    Hello everyone,

     

    i'm having some difficulties with a trunk (10 numbers). My trunk is registered and gives me status 200 OK. When I call one of the trunk numbers, my phone rings ... But the Snom One PBX gives me notting (no log, ...). This is strange, because I'm able to trace my incoming call in the firewall log and by a WireShark scan. One thing, the VOIP provider is using 3 different servers for incoming calls 85.119.188.67 - 85.119.188.31 - 85.119.188.2 and the address en proxy server is 85.119.188.3. Below I post a WireShark log and my trunk set-up. Some help would be greatly appreciated.

     

    Thank you,

    Dimitri

     

    Below you find my WireShark log of an incoming call:

    INVITE sip:02880xxxx@91.183.57.xxx:63221 SIP/2.0

    Via: SIP/2.0/UDP 85.119.188.67:5060;branch=z9hG4bK78913e88;rport

    From: "CreaVil" <sip:0288095xx@85.119.188.67>;tag=as568b832a

    To: <sip:02880xxxx@91.183.57.xxx:63221>

    Contact: <sip:028809582@85.119.188.67>

    Call-ID: 353cdb4f41830d4742f4e2ee699cf5c4@85.119.188.67

    CSeq: 102 INVITE

    User-Agent: Integrics Enswitch

    Max-Forwards: 70

    Date: Thu, 24 Mar 2011 13:31:18 GMT

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

    Supported: replaces

    Diversion: <sip:028809629@ast3>

    Content-Type: application/sdp

    Content-Length: 334

     

    v=0

    o=root 3737 3737 IN IP4 85.119.188.67

    s=session

    c=IN IP4 85.119.188.67

    t=0 0

    m=audio 18440 RTP/AVP 0 8 18 3 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:18 G729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:3 GSM/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=silenceSupp:off - - - -

    a=ptime:20

    a=sendrecv

     

    The Trunk config:

     

    # Trunk 7 in domain sip.xxx.com

    Name: FCD

    Type: register

    To: sip

    RegPass: ********

    Direction:

    Disabled: false

    Global: true

    Display: FCD

    RegAccount: 02880xxxx

    RegRegistrar: 85.119.188.3

    RegKeep: 60

    RegUser: 02880xxxx

    Icid:

    Require:

    OutboundProxy: 85.119.188.3

    Ani: 02880xxxx

    DialExtension:

    Prefix: 32

    Trusted: false

    AcceptRedirect: false

    RfcRtp: false

    Analog: false

    SendEmail:

    UseUuid: false

    Ring180: false

    Failover: never

    Privacy: rpi

    Glob:

    RequestTimeout:

    Codecs:

    CodecLock: true

    Expires: 360

    FromUser:

    Tel: true

    TranscodeDtmf: false

    AssociatedAddresses: 85.119.188.31 85.119.188.67 85.119.188.2

    InterOffice: false

    DialPlan:

    Colines: co10

    DialogPermission: *

  5. Can't get direct incoming calls

    in Extension Setup

    Posted

    Hi everyone,

     

    We are 2 months later and I still haven't found the hassle in the Snom One configuration. I adapted my settings to your advice (otherwise, notting changed since my previous post) ... but no change. Below, I post the log of an incoming call ... I try to call number 02xxxxx26 but I arrive on post 02xxxxx20 (central post).

     

    Thank you for your help.

     

     

    [5] 2011/02/04 22:49:29:

    SIP Rx udp:85.119.188.3:5060:

    INVITE sip:028809620@192.168.101.4:5060;transport=udp;line=c81e728d SIP/2.0

    Record-Route: <sip:85.119.188.3;lr=on;ftag=as3e559475>

    Via: SIP/2.0/UDP 85.119.188.3;branch=z9hG4bK4d58.fec7925.0

    Via: SIP/2.0/UDP 85.119.188.67:5060;received=85.119.188.67;branch=z9hG4bK06033434;rport=5060

    From: "CreaVil" <sip:02xxxxx82@85.119.188.3>;tag=as3e559475

    To: <sip:02xxxxx20@85.119.188.3>

    Contact: <sip:02xxxxx82@85.119.188.67>

    Call-ID: 0f476e9f1dc5aa837309cd7c46ea4050@85.119.188.3

    CSeq: 102 INVITE

    User-Agent: Integrics Enswitch

    Max-Forwards: 69

    Date: Fri, 04 Feb 2011 21:49:29 GMT

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

    Supported: replaces

    Diversion: <sip:02xxxxx26@ast3>

    Content-Type: application/sdp

    Content-Length: 336

    X-Enswitch-RURI: sip:02xxxxx20@85.119.188.3

    X-Enswitch-Source: 85.119.188.67:5060

     

    v=0

    o=root 28205 28205 IN IP4 85.119.188.67

    s=session

    c=IN IP4 85.119.188.67

    t=0 0

    m=audio 16194 RTP/AVP 0 8 18 3 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:18 G729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:3 GSM/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=silenceSupp:off - - - -

    a=ptime:20

    a=sendrecv

    [5] 2011/02/04 22:49:29:

    Identify trunk (line match) 2

    [5] 2011/02/04 22:49:29:

    SIP Rx udp:85.119.188.3:5060:

    INVITE sip:02xxxxx20@192.168.101.4:5060;transport=udp;line=c81e728d SIP/2.0

    Record-Route: <sip:85.119.188.3;lr=on;ftag=as3e559475>

    Via: SIP/2.0/UDP 85.119.188.3;branch=z9hG4bK4d58.fec7925.0

    Via: SIP/2.0/UDP 85.119.188.67:5060;received=85.119.188.67;branch=z9hG4bK06033434;rport=5060

    From: "CreaVil" <sip:02xxxxx82@85.119.188.3>;tag=as3e559475

    To: <sip:02xxxxx20@85.119.188.3>

    Contact: <sip:02xxxxx82@85.119.188.67>

    Call-ID: 0f476e9f1dc5aa837309cd7c46ea4050@85.119.188.3

    CSeq: 102 INVITE

    User-Agent: Integrics Enswitch

    Max-Forwards: 69

    Date: Fri, 04 Feb 2011 21:49:29 GMT

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

    Supported: replaces

    Diversion: <sip:02xxxxx26@ast3>

    Content-Type: application/sdp

    Content-Length: 336

    X-Enswitch-RURI: sip:02xxxxx20@85.119.188.3

    X-Enswitch-Source: 85.119.188.67:5060

     

    v=0

    o=root 28205 28205 IN IP4 85.119.188.67

    s=session

    c=IN IP4 85.119.188.67

    t=0 0

    m=audio 16194 RTP/AVP 0 8 18 3 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:18 G729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:3 GSM/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=silenceSupp:off - - - -

    a=ptime:20

    a=sendrecv

    [5] 2011/02/04 22:49:29:

    SIP Tx udp:85.119.188.3:5060:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 85.119.188.3;branch=z9hG4bK4d58.fec7925.0

    Via: SIP/2.0/UDP 85.119.188.67:5060;received=85.119.188.67;branch=z9hG4bK06033434;rport=5060

    Record-Route: <sip:85.119.188.3;lr=on;ftag=as3e559475>

    From: "CreaVil" <sip:02xxxxx82@85.119.188.3>;tag=as3e559475

    To: <sip:02xxxxx20@85.119.188.3>;tag=d328767900

    Call-ID: 0f476e9f1dc5aa837309cd7c46ea4050@85.119.188.3

    CSeq: 102 INVITE

    Content-Length: 0

     

    [6] 2011/02/04 22:49:29:

    Sending RTP for 0f476e9f1dc5aa837309cd7c46ea4050@85.119.188.3 to 85.119.188.67:16194, codec not set yet

    [5] 2011/02/04 22:49:29:

    Global trunk 3StarsNet@sip.somewhere.com sends call to 20 in domain sip.somewhere.com

    [5] 2011/02/04 22:49:29:

    SIP Tx udp:192.168.101.227:5060:

    INVITE sip:20@192.168.101.227:5060;line=b9vidtev SIP/2.0

    Via: SIP/2.0/UDP 192.168.101.4:5060;branch=z9hG4bK-ec930f6b4be97ba3e6f72349f0b3816e;rport

    From: "CreaVil" <sip:02xxxxx82@sip.somewhere.com;user=phone>;tag=11221

    To: "info@somewhere.com" <sip:20@sip.somewhere.com>

    Call-ID: 95089ebe@pbx

    CSeq: 15824 INVITE

    Max-Forwards: 70

    Contact: <sip:20@192.168.101.4:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.0.3981

    Alert-Info: <http://127.0.0.1/Bellcore-dr3>

    Content-Type: application/sdp

    Content-Length: 329

     

    v=0

    o=- 12390 12390 IN IP4 192.168.101.4

    s=-

    c=IN IP4 192.168.101.4

    t=0 0

    m=audio 57282 RTP/AVP 0 8 9 2 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    [5] 2011/02/04 22:49:30:

    SIP Rx udp:192.168.101.227:5060:

    SIP/2.0 180 Ringing

    Via: SIP/2.0/UDP 192.168.101.4:5060;branch=z9hG4bK-ec930f6b4be97ba3e6f72349f0b3816e;rport=5060

    From: "CreaVil" <sip:02xxxxx82@sip.somewhere.com;user=phone>;tag=11221

    To: "info@somewhere.com" <sip:20@sip.somewhereco.m>;tag=6rmzq06iwa

    Call-ID: 95089ebe@pbx

    CSeq: 15824 INVITE

    Contact: <sip:20@192.168.101.227:5060;line=b9vidtev>;reg-id=1

    Require: 100rel

    RSeq: 1

    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE

    Allow-Events: talk, hold, refer, call-info

    Content-Length: 0

     

    [5] 2011/02/04 22:49:30:

    SIP Tx udp:192.168.101.227:5060:

    PRACK sip:20@192.168.101.227:5060;line=b9vidtev SIP/2.0

    Via: SIP/2.0/UDP 192.168.101.4:5060;branch=z9hG4bK-09b1b968faa0b20732bce56abebe4291;rport

    From: "CreaVil" <sip:028809582@sip.somewhere.com;user=phone>;tag=11221

    To: "info@somewhere.com" <sip:20@sip.somewhere.com>;tag=6rmzq06iwa

    Call-ID: 95089ebe@pbx

    CSeq: 15825 PRACK

    Max-Forwards: 70

    Contact: <sip:20@192.168.101.4:5060;transport=udp>

    RAck: 1 15824 INVITE

    Content-Length: 0

     

    [6] 2011/02/04 22:49:30:

    Codec pcmu/8000 is chosen for call id 0f476e9f1dc5aa837309cd7c46ea4050@85.119.188.3

    [5] 2011/02/04 22:49:30:

    SIP Tx udp:85.119.188.3:5060:

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/UDP 85.119.188.3;branch=z9hG4bK4d58.fec7925.0

    Via: SIP/2.0/UDP 85.119.188.67:5060;received=85.119.188.67;branch=z9hG4bK06033434;rport=5060

    Record-Route: <sip:85.119.188.3;lr=on;ftag=as3e559475>

    From: "CreaVil" <sip:02xxxxx82@85.119.188.3>;tag=as3e559475

    To: <sip:02xxxxx20@85.119.188.3>;tag=d328767900

    Call-ID: 0f476e9f1dc5aa837309cd7c46ea4050@85.119.188.3

    CSeq: 102 INVITE

    Contact: <sip:028809620@192.168.101.4:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: snom-PBX/2011-4.2.0.3981

    Content-Type: application/sdp

    Content-Length: 290

     

    v=0

    o=- 46234 46234 IN IP4 192.168.101.4

    s=-

    c=IN IP4 192.168.101.4

    t=0 0

    m=audio 53932 RTP/AVP 0 8 3 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:3 gsm/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    [5] 2011/02/04 22:49:30:

    SIP Rx udp:192.168.101.227:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.101.4:5060;branch=z9hG4bK-09b1b968faa0b20732bce56abebe4291;rport=5060

    From: "CreaVil" <sip:02xxxxx82@sip.somewhere.com;user=phone>;tag=11221

    To: "info@somewhere.com" <sip:20@sip.somewhere.com>;tag=6rmzq06iwa

    Call-ID: 95089ebe@pbx

    CSeq: 15825 PRACK

    Contact: <sip:20@192.168.101.227:5060;line=b9vidtev>;reg-id=1

    Content-Length: 0

  6. Can't get direct incoming calls

    in Extension Setup

    Posted

    Yes indeed. I'm using the trunk for incoming and outgoing sessions ... So my outbound proxy is activated and using the IP from the provider 3StarsNet. This is the config of mu trunk:

    # Trunk 2 in domain sip.test.com

    Name: 3StarsNet

    Type: register

    To: sip

    RegPass: ********

    Direction:

    Disabled: false

    Global: true

    Display: Test

    RegAccount: 02xxxx620

    RegRegistrar: 85.119.188.3

    RegKeep:

    RegUser: 02xxxx620

    Icid:

    Require:

    OutboundProxy: 85.119.188.3

    Ani:

    DialExtension:

    Prefix:

    Trusted: false

    AcceptRedirect: true

    RfcRtp: true

    Analog: false

    SendEmail:

    UseUuid: false

    Ring180: false

    Failover: never

    Privacy: rpi

    Glob:

    RequestTimeout:

    Codecs:

    CodecLock: true

    Expires: 360

    FromUser:

    Tel: false

    TranscodeDtmf: false

    AssociatedAddresses:

    InterOffice: false

    DialPlan:

    Colines: co1 co2 co3 co4 co5

    DialogPermission:

     

    What do you mean with 'specify the associated addresses'?

     

    I activated log level 9 (till now I only have messages of level 6. These look like this:

    [5] 2010/12/14 15:11:50: Identify trunk (line match) 2

    [6] 2010/12/14 15:11:50: Sending RTP for 6e7f7a60151ad3cb4640b2fa057bd6d6@85.119.188.3 to 85.119.188.31:10634, codec not set yet

    [5] 2010/12/14 15:11:50: Global trunk 3StarsNet@sip.test.com sends call to 20 in domain sip.test.com

    [6] 2010/12/14 15:11:50: Codec pcmu/8000 is chosen for call id 6e7f7a60151ad3cb4640b2fa057bd6d6@85.119.188.3

     

    Thanks for your feedback.

    Dimitri

  7. Can't get direct incoming calls

    in Extension Setup

    Posted

    Hello everyone,

     

    I'm having a problem with the configuration of 10 extensions. I'm having 10 DID numbers in the style of 028809920 - 028809929. 3StarsNet (a Belgian provider) has given me 1 trunk for those 10 numbers. I've configured 10 extensions 20 till 29. On every extension I attached the phone number for example 29 028809929 (section: account number(s)). When making a call to one of those 10 numbers, the call is coming in and routed to extension 20. In Trunk, my send call to extension is empty ... I'm getting nuts. Can someone help me a bit further ... O yeah, I already read thousand times the old wiki.

     

    Thanks,

    Dimitri

  8. After 1 correct incoming call, all next calls fail.

    in Trunk Setup

    Posted

    Ah, the forum is back ;-) Thx for the reply.

     

    Last 48h I tried different set-ups. I changed the firmware version of my ZyXel ZyWall USG 20. I connected the server (with Snom One on it) directly on one of the routers LAN ports. I tried different set-ups in Snom One versions 2011-4.2.0.3950 and 2011-4.2.0.3958. Last but not least, I made one change in the trunk. I changed 'Interpret SIP URI always as telephone number' to no.

     

    At the moment, the service is up and running for over 12 hours. I hope I'm on the right trail ...

     

    There is one thing left. All incoming calls (on one of the 10 numbers) are routed to my head extension. Direct calling is impossible. I've got all accounts set-up like 20 02xxxxx20 / 21 02xxxxx21/ ... In my trunk settings, 'send call to extension' is blank. What am I doing wrong?

  9. After 1 correct incoming call, all next calls fail.

    in Trunk Setup

    Posted

    Thx pbxnsip for the reply.

     

    I verified every parameter in my ZyXel ZyWall USG20. I've no keep alive in my NAT settings. There are SIP settings ... Those are SIP media timeout = 120s and Signaling inactivity timeout = 1800s.

     

    But I'm afraid, the problem lies elsewhere. The firewall log is showing me that a connection is entering and forwarded to my server (PBX Snom One). But on the PBX software, I see no movement. Like I said before, when I start the system. The first call comes in on 1 of my 10 phone numbers. From the second call on, notting happens in the Snom One log. I see the providers IP connecting and my firewall forwards all the calls (the first, the second, ...) to the server IP (a Windows Server 2003 32-bit). I'm able to get an other incoming call when I click the save button (without changing a thing) in the trunk setup. What is restarted or reset when I click the save button?

     

    To test the firewall settings, I configured a phone (I configured it with the same IP as the server) directly to my provider (3StarsNet) and tried numerous calls. Every call passed ...

     

    Any other ideas? I'm willing to test every possible solution.

     

    Thx, Dimitri

  10. After 1 correct incoming call, all next calls fail.

    in Trunk Setup

    Posted

    Hello,

     

    I'm new to SNOM One and after a week of testing (I installed a 32bit version for Windows on a Windows Server 2003 Standard), I need to call in your help. I've 10 numbers with 3StarsNet (a belgian SIP provider). I've set-up a trunk and a dial-plan. I'm able to make outbound calls, but with the incoming calls I'm experiencing difficulties. I'm able to receive 1 call (see log below). After this call, when I tried to call one of the 10 numbers ... I hear a tone, but the PBX doesn't react. No extra lines in the log and no incoming call. If I reopen my saved trunk settings and click 'save' (without changing a thing). I'm again able to receive one call ... I don't understand. Thanks for the help

     

    The log

    [5] 2010/11/11 21:35:41: Identify trunk (line match) 2

    [6] 2010/11/11 21:35:41: Sending RTP for 76b4228141d389812b7d74ba0d35c782@85.119.188.3 to 85.119.188.67:16706, codec not set yet

    [5] 2010/11/11 21:35:41: Global trunk 3StarsNet@sip.test.com sends call to 20 in domain sip.test.com

    [6] 2010/11/11 21:35:41: Codec pcmu/8000 is chosen for call id 76b4228141d389812b7d74ba0d35c782@85.119.188.3

     

    My trunk settings

    Name:3StarsNet

    Type: SIP registration

    Direction: inbound and outbound

    Trunk Destination: generic sip server

    State: enabled

     

    Display Name:Test

    Account: 02880xxxx

    Domain: 85.119.188.3

    Username: 02880xxxx

    Password: xxxxxx

    Password (repeat): xxxxxx

    Proxy Address: 85.119.188.3

    CO Lines: co1 co2 co3 co4 co5

    Permissions to monitor this account: empty

     

    Lock codec during conversation: Yes

    Proposed Duration (s): 3600

    Keepalive Time: 3600

    Send email on status change: don't send email

    Strict RTP Routing: Yes

    Avoid RFC4122 (UUID): No

    Generate unique extension identifier: No

    Accept Redirect: Yes

    Interpret SIP URI always as telephone number: Yes

    Requires busy tone detection: No

    Trunk requires out of band-DTMF tones: No

     

    Prefix: empty

    Global: Yes

    Trunk ANI: empty

    Remote Party/Privacy Indication: Remote-Party-ID

    Rewrite global numbers: Check domain country code

    Failover Behavior: No failover

    Is Secure: No

    Inter-Office Trunk: No

    ICID (RFC 3455): empty

     

    Explicitly list addresses for inbound traffic: empty

    Send call to extension: empty

    Assume that call comes from user: empty

    Ringback: Media

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