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Fisher Networks

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Posts posted by Fisher Networks

  1. Phone requires + to dial out

    in Dial Plan Setup

    Posted

    I would not enable ENUM on the phone just to be able to dial regular numbers. ENUM is a long long topic that you probably don't want to touch just to solve that problem. I would solve the problem with that dial plan on the PBX above.

     

    I hooked up a SNOM 320 at my home since it would be easier to test than going back and forth to the office, and it worked almost immediately. There is no + requirement, the dial plan setup on the SIP server works great, and strangely, the phone doesn't list anything in it's own dial plan field. I'm not sure no whether there is some strange legacy setting on the 360 causing this, a firmware version issue or what. I'll wipe out the settings and try and set it up again.

     

    I'm very excited that the 320 works though, thanks for the help!

  2. Phone requires + to dial out

    in Dial Plan Setup

    Posted

    I just enabled ENUM on the phone, put in 1 for country code and 206 for the area code and it automatically added the +1206 when dialing a seven digit number. Now the only problem is when I put in an area code (as in to dial a different number) it doesn't put a +1xxx in front, it just dials the number as I put it in and this is what PBXNSIP says:

     

    SIP/2.0 404 Number not in e164 format, example +12125551212

    Via: SIP/2.0/UDP 67.xxx.xxx.xxx:2054;branch=z9hG4bK-lic0jur614u1;rport=2054

    From: "Tanya" <sip:501@domain.com>;tag=7pss29b2s0

    To: <sip:4255551212@domain.com;user=phone>;tag=e95c35967e

    Call-ID: 3c27ab34c484-13ayr4lhwsm5

    CSeq: 2 INVITE

    Contact: <sip:501@67.xxx.xxx.xxx:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.5.2357

    Content-Length: 0

  5. Phone requires + to dial out

    in Dial Plan Setup

    Posted

    I am using the * wildcard in my Dial Plans setup with a single trunk. However the phone complains about needing me to dial in this fashion: +1xxxxxxxxxx. I need no plus, and for local calls I'd prefer being able to dial out without putting the 1 in, and just dial areacode + 7. Is this possible?

     

    Would the + requirement be the SNOM phone?

  7. Setup issues

    in Trunk Setup

    Posted

    Now we're getting somewhere. The logs now appear to be dial plan issues? I'm not too sure what the error is here besides the e164 message.

     

    [7]2008/02/07 13:43:17: SIP Tx tls:10.0.1.109:3337:

    SIP/2.0 404 Number not in e164 format, example +12125551212

    Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-d67yw065242a;rport=3337

    From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi

    To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96

    Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB

    CSeq: 1 INVITE

    Contact: <sip:501@10.0.1.3:5061;transport=tls>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.5.2357

    Content-Length: 0

     

     

    [7] 2008/02/07 13:43:17: SIP Rx tls:10.0.1.109:3337:

    PRACK sip:501@10.0.1.3:5061;transport=tls SIP/2.0

    Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-8mic0lb2vaj9;rport

    From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi

    To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96

    Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB

    CSeq: 2 PRACK

    Max-Forwards: 70

    Contact: <sip:501@10.0.1.109:3337;transport=tls;line=uexh66e7>;flow-id=1

    RAck: 1 1 INVITE

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

    Allow-Events: talk, hold, refer

    Content-Length: 0

     

     

    [7] 2008/02/07 13:43:17: SIP Tx tls:10.0.1.109:3337:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-8mic0lb2vaj9;rport=3337

    From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi

    To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96

    Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB

    CSeq: 2 PRACK

    Contact: <sip:501@10.0.1.3:5061;transport=tls>

    User-Agent: pbxnsip-PBX/2.1.5.2357

    Content-Length: 0

     

     

    [7] 2008/02/07 13:43:18: SIP Rx tls:10.0.1.109:3337:

    ACK sip:1206xxxxxxx@domain.com;user=phone SIP/2.0

    Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-d67yw065242a;rport

    From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi

    To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96

    Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB

    CSeq: 1 ACK

    Max-Forwards: 70

    Contact: <sip:501@10.0.1.109:3337;transport=tls;line=uexh66e7>;flow-id=1

    Content-Length: 0

  8. Setup issues

    in Trunk Setup

    Posted

    This is all I get when I issue a call:

     

    [5] 2008/02/05 09:09:29: Dialplan External: Match 206xxxxxxx@domain.com to <sip:206xxxxxxx@216.82.xxx.xxx;user=phone> on trunk BWGW1

     

    The phone says "Forbidden".

     

    It doesn't appear the PBX is doing anything.

  10. Setup issues

    in Trunk Setup

    Posted

    The outbound proxy should be the Trunking service IP or my gateway?

     

    "SIP Logging" was on, but under that I enabled every type of SIP logging. However, the same (lack of) errors appear. However the SIP trace on the phone yields more info:

     

    Sent to tls:10.0.1.3:5061 at 5/2/2008 01:00:28:050 (1254 bytes):

     

    INVITE sip:206xxxxxxx@domain.com;user=phone SIP/2.0

    Via: SIP/2.0/TLS 10.0.1.109:4967;branch=z9hG4bK-dktogirkbrph;rport

    From: "User" <sip:501@domain.com>;tag=9yc0cxo949

    To: <sip:206xxxxxxx@domain.com;user=phone>

    Call-ID: 3c272286c0df-b8iwzmbz2a5g@snom360-00041323C1DB

    CSeq: 1 INVITE

    Max-Forwards: 70

    Contact: <sip:501@10.0.1.109:4967;transport=tls;line=uexh66e7>;flow-id=1

    P-Key-Flags: resolution="31x13", keys="4"

    User-Agent: snom360/6.2.3

    Accept: application/sdp

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

    Allow-Events: talk, hold, refer

    Supported: timer, 100rel, replaces, callerid

    Session-Expires: 3600;refresher=uas

    Min-SE: 90

    Content-Type: application/sdp

    Content-Length: 471

     

    v=0

    o=root 1927523315 1927523315 IN IP4 10.0.1.109

    s=call

    c=IN IP4 10.0.1.109

    t=0 0

    m=audio 51080 RTP/AVP 0 8 9 2 3 18 4 101

    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XUR1mT6ftFtPm4go7v37e/vtfmrfHzWmQ4OxofLW

    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:18 g729/8000

    a=rtpmap:4 g723/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=encryption:optional

    a=sendrecv

     

     

     

    --------------------------------------------------------------------------------

     

    Received from tls:10.0.1.3:5061 at 5/2/2008 01:00:28:140 (323 bytes):

     

    SIP/2.0 100 Trying

    Via: SIP/2.0/TLS 10.0.1.109:4967;branch=z9hG4bK-dktogirkbrph;rport=4967

    From: "Tanya " <sip:501@domain.com>;tag=9yc0cxo949

    To: <sip:206xxxxxxx@domain.com;user=phone>;tag=0e7ad1ccc9

    Call-ID: 3c272286c0df-b8iwzmbz2a5g@snom360-00041323C1DB

    CSeq: 1 INVITE

    Content-Length: 0

     

     

     

     

    --------------------------------------------------------------------------------

     

    Received from tls:10.0.1.3:5061 at 5/2/2008 01:00:28:150 (685 bytes):

     

    NOTIFY sip:501@10.0.1.109:4967;transport=tls;line=uexh66e7 SIP/2.0

    Via: SIP/2.0/TLS 10.0.1.3:5061;branch=z9hG4bK-0d5b74cb3e5a5bdc447bbbe02c21b327;rport

    From: <sip:501@domain.com;user=phone>;tag=f7b6300e03

    To: <sip:501@domain.com>;tag=oyj484iitx

    Call-ID: 3c267009c832-wjx5tt2ysc3u@snom360-00041323C1DB

    CSeq: 19600 NOTIFY

    Max-Forwards: 70

    Contact: <sip:10.0.1.3:5061;transport=tls>

    Event: dialog

    Subscription-State: active;expires=187

    Content-Type: application/dialog-info+xml

    Content-Length: 158

     

    <?xml version="1.0"?>

    <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="36" state="full" entity="sip:501@domain.com"></dialog-info>

     

    <snip>

     

     

    Sent to tls:10.0.1.3:5061 at 5/2/2008 01:00:28:380 (318 bytes):

     

    SIP/2.0 200 Ok

    Via: SIP/2.0/TLS 10.0.1.3:5061;branch=z9hG4bK-24329e1bc69a2121a722c9dd45bdb3e5;rport=5061

    From: <sip:501@domain.com;user=phone>;tag=f7b6300e03

    To: <sip:501@domain.com>;tag=oyj484iitx

    Call-ID: 3c267009c832-wjx5tt2ysc3u@snom360-00041323C1DB

    CSeq: 19601 NOTIFY

    Content-Length: 0

     

     

     

     

    --------------------------------------------------------------------------------

     

    Received from tls:10.0.1.3:5061 at 5/2/2008 01:00:28:390 (402 bytes):

     

    SIP/2.0 200 Ok

    Via: SIP/2.0/TLS 10.0.1.109:4967;branch=z9hG4bK-wweqrjre6ksm;rport=4967

    From: "Tanya " <sip:501@domain.com>;tag=9yc0cxo949

    To: <sip:206xxxxxxx@domain.com;user=phone>;tag=0e7ad1ccc9

    Call-ID: 3c272286c0df-b8iwzmbz2a5g@snom360-00041323C1DB

    CSeq: 2 PRACK

    Contact: <sip:501@10.0.1.3:5061;transport=tls>

    User-Agent: pbxnsip-PBX/2.1.5.2357

    Content-Length: 0

  11. Setup issues

    in Trunk Setup

    Posted

    Hey, I started our settings from scratch and appear to be having issues getting the ability to make outbound calls (and inbound, really). We use bandwidth.com SIP trunking and with their IP in the domain field, the same IP in the outbound gateway and our number as the DID in a SIP Gateway we can't get out. The only log I get is the following.

     

    [5] 2008/02/04 17:52:21: Identify trunk (domain name match) 1

    [5] 2008/02/04 17:52:21: Dialplan External: Match 206xxxxxxx@domain.com to <sip:2067690931@216.82.x.x;user=phone> on trunk BWGW1

     

    It does not appear to be getting out at all at this point. The phone says this:

     

    [5]4/2/2008 18:54:08: Dialog 7/6 going to trying

    [5]4/2/2008 18:54:08: Dialog 7/6 going to early

    [5]4/2/2008 18:54:08: Dialog 7/6 going to terminated

    [5]4/2/2008 18:54:08: timeout::callback: Registering with timeout of 0 ms

    [5]4/2/2008 18:54:08: timeout::callback: Registering with timeout of 0 ms

    [2]4/2/2008 18:54:36: Registered at registrar as 501@domain.com

    [0]4/2/2008 18:54:37: Webclient: Could not find host snom360.htm:80

    [0]4/2/2008 18:54:37: Webclient: Could not find host snom360-00041323C1DB.htm:80

     

    I think the [5]s are the only important logs there.

     

    Anyway, the dial plan is a *, supposedly to accept any input and pass it along.

     

    Any ideas what I am missing? I simplified the internal setup, so there are no hunt groups, attendants or anything but a single, registered extension.

     

    Also, what ports do I need to make sure are open? I know of 5060 and 5061.

     

    Thanks for your help!

  13. Incoming calls get a busy signal

    in Trunk Setup

    Posted

    Is "localhost" your domain name on the PBX? What did you put into "Send call to extension" in the trunk?

     

    As for your first question, do you mean on the software itself or the actual windows domain name?

     

    Your second question was right on. I missed it. It says "Extension" and in it was an incorrect extension. I renamed the alias on the Hunt Group to 900 and changed that field in the trunk to 900 and it rang through.

  14. Incoming calls get a busy signal

    in Trunk Setup

    Posted

    This is a new setup and not going too well. All incoming calls get are a busy signal. The log shows the attempt and displays a 404: Not Found error. Outgoing calls work fine. We are running pbxnsip 2.0.3.1715. The server has an external and internal IP. Ports 5060 and 5061 are open. Here is a snippet of my call (I masked some numbers):

     

     

    [7] 2007/10/16 21:41:42: SIP Rx udp:4.xx.xxx.236:5060:

     

    INVITE sip:+1206xxx4108@xx.xxx.173.140:5060;transport=udp SIP/2.0

     

    Record-Route: <sip:4.xx.xxx.236;lr;ftag=VPSF506071629460>

     

    Record-Route: <sip:4.xx.xxx.229;lr;ftag=VPSF506071629460>

     

    Via: SIP/2.0/UDP 4.xx.xxx.236;branch=z9hG4bK347b.a1776e47.0

     

    Via: SIP/2.0/UDP 4.xx.xxx.229;branch=z9hG4bK347b.127161b4.0

     

    Via: SIP/2.0/UDP 4.xx.xxx.148:5060;branch=z9hG4bK506071629460-1192517067514

     

    From: "PCS Phone WA" <sip:+1206xxx0931@4.xx.xxx.148;isup-oli=62>;tag=VPSF506071629460

     

    To: <sip:+1206xxx4108@4.xx.xxx.229:5060>

     

    Call-ID: SEAMGC0120071017044305059728@xxx.xxx.63.11

     

    CSeq: 1 INVITE

     

    Contact: <sip:+1206xxx0931@4.xx.xxx.148:5060;transport=udp>

     

    Max-Forwards: 67

     

    Content-Type: application/sdp

     

    Content-Length: 173

     

    Remote-Party-ID: "PCS Phone WA" <sip:+1206xxx0931@4.xx.xxx.148>;party=calling;screen=yes;privacy=off

     

     

     

    v=0

     

    o=- 1192596185 1192596186 IN IP4 xxx.xxx.31.53

     

    s=-

     

    c=IN IP4 xxx.xxx.31.53

     

    t=0 0

     

    m=audio 60724 RTP/AVP 0 18 101

     

    a=rtpmap:101 telephone-event/8000

     

    a=fmtp:101 0-15

     

     

     

    [7] 2007/10/16 21:41:42: UDP: Opening socket on port 50952

     

    [7] 2007/10/16 21:41:42: UDP: Opening socket on port 50953

     

    [5] 2007/10/16 21:41:42: Identify trunk 4

     

    [7] 2007/10/16 21:41:42: SIP Tx udp:4.xx.xxx.236:5060:

     

    SIP/2.0 100 Trying

     

    Via: SIP/2.0/UDP 4.xx.xxx.236;branch=z9hG4bK347b.a1776e47.0

     

    Via: SIP/2.0/UDP 4.xx.xxx.229;branch=z9hG4bK347b.127161b4.0

     

    Via: SIP/2.0/UDP 4.xx.xxx.148:5060;branch=z9hG4bK506071629460-1192517067514

     

    Record-Route: <sip:4.xx.xxx.236;lr;ftag=VPSF506071629460>

     

    Record-Route: <sip:4.xx.xxx.229;lr;ftag=VPSF506071629460>

     

    From: "PCS Phone WA" <sip:+1206xxx0931@4.xx.xxx.148;isup-oli=62>;tag=VPSF506071629460

     

    To: <sip:+1206xxx4108@4.xx.xxx.229:5060>;tag=d7cd18744c

     

    Call-ID: SEAMGC0120071017044305059728@xxx.xxx.63.11

     

    CSeq: 1 INVITE

     

    Content-Length: 0

     

     

     

     

     

    [7] 2007/10/16 21:41:42: SIP Tx udp:4.xx.xxx.236:5060:

     

    SIP/2.0 404 Not Found

     

    Via: SIP/2.0/UDP 4.xx.xxx.236;branch=z9hG4bK347b.a1776e47.0

     

    Via: SIP/2.0/UDP 4.xx.xxx.229;branch=z9hG4bK347b.127161b4.0

     

    Via: SIP/2.0/UDP 4.xx.xxx.148:5060;branch=z9hG4bK506071629460-1192517067514

     

    Record-Route: <sip:4.xx.xxx.236;lr;ftag=VPSF506071629460>

     

    Record-Route: <sip:4.xx.xxx.229;lr;ftag=VPSF506071629460>

     

    From: "PCS Phone WA" <sip:+1206xxx0931@4.xx.xxx.148;isup-oli=62>;tag=VPSF506071629460

     

    To: <sip:+1206xxx4108@4.xx.xxx.229:5060>;tag=d7cd18744c

     

    Call-ID: SEAMGC0120071017044305059728@xxx.xxx.63.11

     

    CSeq: 1 INVITE

     

    Contact: <sip:+1206xxx4108@xx.xxx.173.140:5060;transport=udp>

     

    Supported: 100rel, replaces, norefersub

     

    Allow-Events: refer

     

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

     

    Accept: application/sdp

     

    User-Agent: pbxnsip-PBX/2.0.3.1715

     

    Content-Length: 0

     

     

     

     

     

    [7] 2007/10/16 21:41:42: SIP Rx udp:4.xx.xxx.236:5060:

     

    ACK sip:+1206xxx4108@xx.xxx.173.140:5060;transport=udp SIP/2.0

     

    Via: SIP/2.0/UDP 4.xx.xxx.236;branch=z9hG4bK347b.a1776e47.0

     

    From: "PCS Phone WA" <sip:+1206xxx0931@4.xx.xxx.148;isup-oli=62>;tag=VPSF506071629460

     

    Call-ID: SEAMGC0120071017044305059728@xxx.xxx.63.11

     

    To: <sip:+1206xxx4108@4.xx.xxx.229:5060>;tag=d7cd18744c

     

    CSeq: 1 ACK

     

    Max-Forwards: 70

     

    User-Agent: Bandwidth.com TRM (gold.13)

     

    Content-Length: 0

     

     

     

    Any idea what I'm looking at? It appears that everything is getting past the firewall, but no calls are accepted. I have a hunt group setup with the name 1206xxx4106 with an alias to 4107 (I know 4108 is mentioned in this log but the same happens to all three numbers).

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