Fisher Networks

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!

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

The dial plan in PBXnSIP is just a *. The dial plan on the phone is sip:\1@\r;user=phone.

It is the phone. It says it needs to be dialed using the e164 dialing or something along those lines when you dial without the + sign. It is a new installation.

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?

I just tested outgoing and got another number to ring. I think this thread is done. Thanks!

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.

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.

I set the outbound proxy to the server's internal NIC (that the clients all funnel through) and it said "Connection refused".

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

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!

I think I'll rename it back to localhost. Should the SIP phones be set to localhost as well (registrar)?

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.

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).

I'm actually attempting to setup my pbxnsip software with Bandwidth.com. So far I can make outgoing calls and internal calls, but nothing internal. I have a feeling my issue is server related.