nathans

Everything was working fine on the boxes. Only the emails where not being send for missed calls/cdr/etc. My other installation of snomONE that use the same GMAIL account for emails, continued working fine. the only difference between the working and non-working is that the non-working were all snomONE plus appliances. The working ones were either WIN32 or MacOS installation that were not affected by this strange behavior.

Something very strange happened on May 10th to three different snomONE plus boxes we have in 3 separate installations. At about 8:00pm EST all of the stopped sending email. We use GMAIL a the smtp server for all. The log showed a refused connection the the imap server. These boxes have been up and running for more than 1 year (some of them) with no problems on sending emails./ After a lot of frustration, we reloaded the security GMAIL certificates and emails started to flow again. Any ideas why? Does the GMAIL certificate expires or is there some setting we are missing? Thanks Nathan

Thanks Katerina! I was going crazy trying to find the obscure setting I was missing to take it out

HI, After upgrading a 870 to the latest 8.7.3.7 we now have a "(srtp/tls)' string showing under the time timer for in/out calls instead of the extension number as we had before. This is with a snomone plus. Any ideas how to get rid of it and replace it with either the incoming extension number or dialed one? We are showing the name correctly on top, we just want to replace this strange string with the extension number. Thanks Nathan

Thanks. We have not touched any of those files. Looking at the line you mentioned we have the following right now: <user_dp_str idx="{lc}" perm="RW">{dialplan snom}</user_dp_str> So 2 questions: 1) Where does the {dial plan snom} value come from? 2) Just erase all? including the brackets? Thanks! Nathan BTW, this is what the phones are getting in the dial plan string field: !^(1[0-9]{10})!sip:\1@\d;user=phone!d !^([2-7][0-9]{3})!sip:\1@\d;user=phone!d !^(8[2-7][0-9]{3})!sip:\1@\d;user=phone!d !^([2-9]11)!sip:\1@\d;user=phone!d

Setup: snomone+ running with a bunch of 300s, 821s and 870s We are having some issues with some of the phones that when restarting they get a "standard" string in their Dial-Plan String for each identity (under SIP menu) On the snomone one they are using the correct dial plan to dial out but since they have these defaults on the phone itself, they mess up the dial out plan. so the questions: where or how do we set via PnP to clear out the dial plan string for the phones so that I do not have to go one by one via the web interface to clean it up?

Thanks. I also thought this was the problem. But you see, the PBX is indeed with a public and routable IP address with the correct port forwarding. I have about 5 extensions registering remotely to it and being able to make and received calls from It is only one remote extension that is showing this problems of not being able to make outgoing calls. Are you referring to the fact that this one extension needs something special besides being either on a DMZ or with the correct SIP and UDP ports forwarded?

Here is the log from the snomone on an attempted call from extension 7788 (the external problem one to extension 9999. 192.168.30.2 is the LAN IP address of the snomone+ [7] 2012/02/23 21:58:03: SIP Rx udp:189.25.204.17:22928: INVITE sip:9999@192.168.30.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 189.25.204.17:22928;branch=z9hG4bK-w7plx9ppz0hw;rport From: "Jacobo " <sip:7788@192.168.30.2:5060>;tag=bpgfum53tr To: <sip:9999@192.168.30.2:5060;user=phone> Call-ID: 3c2677382783-7fnzl9w6qqo7 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:7788@189.25.204.17:22928;line=7ybp1ezv>;reg-id=1 X-Serialnumber: 0004133B3E16 P-Key-Flags: keys="3" User-Agent: snom300/8.4.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:187.174.100.120>;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 533 v=0 o=root 642205070 642205070 IN IP4 189.25.204.17 s=call c=IN IP4 189.25.204.17 t=0 0 m=audio 22972 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Uu3rCyl9ghfeMZrR2Z18tEwKwzutivBHnjdRfbMR 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=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-22973xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [7] 2012/02/23 21:58:03: SIP Tx udp:189.25.204.17:22928: SIP/2.0 100 Trying Via: SIP/2.0/UDP 189.25.204.17:22928;branch=z9hG4bK-w7plx9ppz0hw;rport=22928 From: "Jacobo " <sip:7788@192.168.30.2:5060>;tag=bpgfum53tr To: <sip:9999@192.168.30.2:5060;user=phone>;tag=49c5c1f64d Call-ID: 3c2677382783-7fnzl9w6qqo7 CSeq: 1 INVITE Content-Length: 0 [7] 2012/02/23 21:58:03: SIP Tx udp:189.25.204.17:22928: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 189.25.204.17:22928;branch=z9hG4bK-w7plx9ppz0hw;rport=22928 From: "Jacobo " <sip:7788@192.168.30.2:5060>;tag=bpgfum53tr To: <sip:9999@192.168.30.2:5060;user=phone>;tag=49c5c1f64d Call-ID: 3c2677382783-7fnzl9w6qqo7 CSeq: 1 INVITE User-Agent: snom-PBX/2011-4.2.1.4025 WWW-Authenticate: Digest realm="192.168.30.2",nonce="b8977264cfb4066cfecd45ea0d085090",domain="sip:9999@192.168.30.2:5060;user=phone",algorithm=MD5 Content-Length: 0 [7] 2012/02/23 21:58:03: SIP Rx udp:189.25.204.17:22928: ACK sip:9999@192.168.30.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 189.25.204.17:22928;branch=z9hG4bK-w7plx9ppz0hw;rport From: "Jacobo " <sip:7788@192.168.30.2:5060>;tag=bpgfum53tr To: <sip:9999@192.168.30.2:5060;user=phone>;tag=49c5c1f64d Call-ID: 3c2677382783-7fnzl9w6qqo7 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:7788@189.25.204.17:22928;line=7ybp1ezv>;reg-id=1 Proxy-Require: buttons Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Rx udp:189.25.204.17:22928: CANCEL sip:9999@192.168.30.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 189.25.204.17:22928;branch=z9hG4bK-tnlw528ee5y9;rport From: "Jacobo " <sip:7788@192.168.30.2:5060>;tag=bpgfum53tr To: <sip:9999@192.168.30.2:5060;user=phone> Call-ID: 3c2677382783-7fnzl9w6qqo7 CSeq: 2 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Proxy-Require: buttons Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Tx udp:189.25.204.17:22928: SIP/2.0 100 Trying Via: SIP/2.0/UDP 189.25.204.17:22928;branch=z9hG4bK-tnlw528ee5y9;rport=22928 From: "Jacobo " <sip:7788@192.168.30.2:5060>;tag=bpgfum53tr To: <sip:9999@192.168.30.2:5060;user=phone>;tag=49c5c1f64d Call-ID: 3c2677382783-7fnzl9w6qqo7 CSeq: 2 CANCEL Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Tx udp:216.19.16.77:3072: INVITE sip:9999@216.19.16.77:3072;line=msef87x4 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-1e9e57e84c9ce2bcdf8b6a220d2d25c0;rport From: "Jacobo " <sip:7788@pbx.snom.com>;tag=514136832 To: "Nathan" <sip:9999@pbx.snom.com> Call-ID: 15293691@pbx CSeq: 17146 INVITE Max-Forwards: 70 Contact: <sip:9999@192.168.30.2:5060;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/2011-4.2.1.4025 Alert-Info: <http://127.0.0.1/Bellcore-dr2> Content-Type: application/sdp Content-Length: 384 v=0 o=- 1576670420 1576670420 IN IP4 192.168.30.2 s=- c=IN IP4 192.168.30.2 t=0 0 m=audio 59018 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no 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 [7] 2012/02/23 21:58:09: SIP Rx udp:189.25.204.17:22928: CANCEL sip:9999@192.168.30.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 189.25.204.17:22928;branch=z9hG4bK-tnlw528ee5y9;rport From: "Jacobo " <sip:7788@192.168.30.2:5060>;tag=bpgfum53tr To: <sip:9999@192.168.30.2:5060;user=phone> Call-ID: 3c2677382783-7fnzl9w6qqo7 CSeq: 2 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Proxy-Require: buttons Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Tr udp:189.25.204.17:22928: SIP/2.0 100 Trying Via: SIP/2.0/UDP 189.25.204.17:22928;branch=z9hG4bK-tnlw528ee5y9;rport=22928 From: "Jacobo " <sip:7788@192.168.30.2:5060>;tag=bpgfum53tr To: <sip:9999@192.168.30.2:5060;user=phone>;tag=49c5c1f64d Call-ID: 3c2677382783-7fnzl9w6qqo7 CSeq: 2 CANCEL Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Tx udp:75.149.181.121:3072: INVITE sip:9999@192.168.88.107:3072;line=ad4e2gld SIP/2.0 Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-64737b79c64e7dda8e1071014447f2da;rport From: "Jacobo " <sip:7788@pbx.snom.com>;tag=1112862256 To: "Nathan" <sip:9999@pbx.snom.com> Call-ID: 19bbc33d@pbx CSeq: 16740 INVITE Max-Forwards: 70 Contact: <sip:9999@192.168.30.2:5060;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/2011-4.2.1.4025 Alert-Info: <http://127.0.0.1/Bellcore-dr2> Content-Type: application/sdp Content-Length: 384 v=0 o=- 1241561074 1241561074 IN IP4 192.168.30.2 s=- c=IN IP4 192.168.30.2 t=0 0 m=audio 64382 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no 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 [7] 2012/02/23 21:58:09: SIP Tx udp:189.25.204.17:22928: SIP/2.0 200 Ok Via: SIP/2.0/UDP 189.25.204.17:22928;branch=z9hG4bK-tnlw528ee5y9;rport=22928 From: "Jacobo " <sip:7788@192.168.30.2:5060>;tag=bpgfum53tr To: <sip:9999@192.168.30.2:5060;user=phone>;tag=49c5c1f64d Call-ID: 3c2677382783-7fnzl9w6qqo7 CSeq: 2 CANCEL Contact: <sip:7788@192.168.30.2:5060> User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Tx tls:192.168.30.106:3117: SIP/2.0 487 Request Terminated From: <> To: <> Call-ID: CSeq: 0 Contact: <sip:7788@192.168.30.2:5061> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Tx udp:216.19.16.77:3072: CANCEL sip:9999@216.19.16.77:3072;line=msef87x4 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-1e9e57e84c9ce2bcdf8b6a220d2d25c0;rport From: "Jacobo " <sip:7788@pbx.snom.com>;tag=514136832 To: "Nathan" <sip:9999@pbx.snom.com> Call-ID: 15293691@pbx CSeq: 17146 CANCEL Max-Forwards: 70 Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Tx udp:75.149.181.121:3072: CANCEL sip:9999@192.168.88.107:3072;line=ad4e2gld SIP/2.0 Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-64737b79c64e7dda8e1071014447f2da;rport From: "Jacobo " <sip:7788@pbx.snom.com>;tag=1112862256 To: "Nathan" <sip:9999@pbx.snom.com> Call-ID: 19bbc33d@pbx CSeq: 16740 CANCEL Max-Forwards: 70 Content-Length: 0 [3] 2012/02/23 21:58:09: Could not connect to 74.208.67.196:8888 [1] 2012/02/23 21:58:09: Could not send via TCP: 36 bytes [7] 2012/02/23 21:58:09: SIP Rx udp:75.149.181.121:3072: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-64737b79c64e7dda8e1071014447f2da;rport=5060 From: "Jacobo " <sip:7788@pbx.snom.com>;tag=1112862256 To: "Nathan" <sip:9999@pbx.snom.com>;tag=l34qs3dgkb Call-ID: 19bbc33d@pbx CSeq: 16740 INVITE Contact: <sip:9999@192.168.88.107:3072;line=ad4e2gld>;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] 2012/02/23 21:58:09: Call 19bbc33d@pbx: Last request not finished [7] 2012/02/23 21:58:09: SIP Tx udp:75.149.181.121:3072: PRACK sip:9999@192.168.88.107:3072;line=ad4e2gld SIP/2.0 Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-80a284f90c91b350b17fa32bd6ed2db5;rport From: "Jacobo " <sip:7788@pbx.snom.com>;tag=1112862256 To: "Nathan" <sip:9999@pbx.snom.com>;tag=l34qs3dgkb Call-ID: 19bbc33d@pbx CSeq: 16741 PRACK Max-Forwards: 70 Contact: <sip:9999@192.168.30.2:5060;transport=udp> RAck: 1 16740 INVITE Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Rx udp:75.149.181.121:3072: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-64737b79c64e7dda8e1071014447f2da;rport=5060 From: "Jacobo " <sip:7788@pbx.snom.com>;tag=1112862256 To: "Nathan" <sip:9999@pbx.snom.com>;tag=l34qs3dgkb Call-ID: 19bbc33d@pbx CSeq: 16740 CANCEL Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Rx udp:75.149.181.121:3072: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-64737b79c64e7dda8e1071014447f2da;rport=5060 From: "Jacobo " <sip:7788@pbx.snom.com>;tag=1112862256 To: "Nathan" <sip:9999@pbx.snom.com>;tag=l34qs3dgkb Call-ID: 19bbc33d@pbx CSeq: 16740 INVITE Contact: <sip:9999@192.168.88.107:3072;line=ad4e2gld>;reg-id=1 Content-Length: 0 [7] 2012/02/23 21:58:09: Call 19bbc33d@pbx: Clear last INVITE [7] 2012/02/23 21:58:09: SIP Tx udp:75.149.181.121:3072: ACK sip:9999@192.168.88.107:3072;line=ad4e2gld SIP/2.0 Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-64737b79c64e7dda8e1071014447f2da;rport From: "Jacobo " <sip:7788@pbx.snom.com>;tag=1112862256 To: "Nathan" <sip:9999@pbx.snom.com>;tag=l34qs3dgkb Call-ID: 19bbc33d@pbx CSeq: 16740 ACK Max-Forwards: 70 Contact: <sip:9999@192.168.30.2:5060;transport=udp> Content-Length: 0 [5] 2012/02/23 21:58:09: INVITE Response 487 Request Terminated: Terminate 19bbc33d@pbx [7] 2012/02/23 21:58:09: SIP Rx udp:216.19.16.77:3072: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-1e9e57e84c9ce2bcdf8b6a220d2d25c0;rport=5060 From: "Jacobo " <sip:7788@pbx.snom.com>;tag=514136832 To: "Nathan" <sip:9999@pbx.snom.com>;tag=dt3ujuapv2 Call-ID: 15293691@pbx CSeq: 17146 INVITE Contact: <sip:9999@216.19.16.77:3072;line=msef87x4>;reg-id=1 Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Rx udp:216.19.16.77:3072: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-1e9e57e84c9ce2bcdf8b6a220d2d25c0;rport=5060 From: "Jacobo " <sip:7788@pbx.snom.com>;tag=514136832 To: "Nathan" <sip:9999@pbx.snom.com>;tag=dt3ujuapv2 Call-ID: 15293691@pbx CSeq: 17146 INVITE Contact: <sip:9999@216.19.16.77:3072;line=msef87x4>;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] 2012/02/23 21:58:09: Call 15293691@pbx: Last request not finished [7] 2012/02/23 21:58:09: SIP Tx udp:216.19.16.77:3072: PRACK sip:9999@216.19.16.77:3072;line=msef87x4 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-e37fd0d92c9114104c622a8936e89505;rport From: "Jacobo " <sip:7788@pbx.snom.com>;tag=514136832 To: "Nathan" <sip:9999@pbx.snom.com>;tag=dt3ujuapv2 Call-ID: 15293691@pbx CSeq: 17147 PRACK Max-Forwards: 70 Contact: <sip:9999@192.168.30.2:5060;transport=udp> RAck: 1 17146 INVITE Content-Length: 0 [7] 2012/02/23 21:58:09: SIP Rx udp:75.149.181.121:3072: SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-80a284f90c91b350b17fa32bd6ed2db5;rport=5060 From: "Jacobo " <sip:7788@pbx.snom.com>;tag=1112862256 To: "Nathan" <sip:9999@pbx.snom.com>;tag=l34qs3dgkb Call-ID: 19bbc33d@pbx CSeq: 16741 PRACK User-Agent: snom870/8.4.33 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0 [3] 2012/02/23 21:58:09: Could not connect to 74.208.67.196:8888 [1] 2012/02/23 21:58:09: Could not send via TCP: 36 bytes [7] 2012/02/23 21:58:09: SIP Tr udp:216.19.16.77:3072: CANCEL sip:9999@216.19.16.77:3072;line=msef87x4 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-1e9e57e84c9ce2bcdf8b6a220d2d25c0;rport From: "Jacobo " <sip:7788@pbx.snom.com>;tag=514136832 To: "Nathan" <sip:9999@pbx.snom.com> Call-ID: 15293691@pbx CSeq: 17146 CANCEL Max-Forwards: 70 Content-Length: 0 [7] 2012/02/23 21:58:10: SIP Rx udp:216.19.16.77:3072: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-1e9e57e84c9ce2bcdf8b6a220d2d25c0;rport=5060 From: "Jacobo " <sip:7788@pbx.snom.com>;tag=514136832 To: "Nathan" <sip:9999@pbx.snom.com>;tag=dt3ujuapv2 Call-ID: 15293691@pbx CSeq: 17146 CANCEL Content-Length: 0 [7] 2012/02/23 21:58:10: SIP Rx udp:216.19.16.77:3072: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-1e9e57e84c9ce2bcdf8b6a220d2d25c0;rport=5060 From: "Jacobo " <sip:7788@pbx.snom.com>;tag=514136832 To: "Nathan" <sip:9999@pbx.snom.com>;tag=dt3ujuapv2 Call-ID: 15293691@pbx CSeq: 17146 INVITE Contact: <sip:9999@216.19.16.77:3072;line=msef87x4>;reg-id=1 Content-Length: 0 [7] 2012/02/23 21:58:10: Call 15293691@pbx: Clear last INVITE [7] 2012/02/23 21:58:10: SIP Tx udp:216.19.16.77:3072: ACK sip:9999@216.19.16.77:3072;line=msef87x4 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-1e9e57e84c9ce2bcdf8b6a220d2d25c0;rport From: "Jacobo " <sip:7788@pbx.snom.com>;tag=514136832 To: "Nathan" <sip:9999@pbx.snom.com>;tag=dt3ujuapv2 Call-ID: 15293691@pbx CSeq: 17146 ACK Max-Forwards: 70 Contact: <sip:9999@192.168.30.2:5060;transport=udp> Content-Length: 0 [5] 2012/02/23 21:58:10: INVITE Response 487 Request Terminated: Terminate 15293691@pbx [3] 2012/02/23 21:58:10: Could not connect to 74.208.67.196:8888 [1] 2012/02/23 21:58:10: Could not send via TCP: 36 bytes [7] 2012/02/23 21:58:10: SIP Rx udp:216.19.16.77:3072: SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.30.2:5060;branch=z9hG4bK-e37fd0d92c9114104c622a8936e89505;rport=5060 From: "Jacobo " <sip:7788@pbx.snom.com>;tag=514136832 To: "Nathan" <sip:9999@pbx.snom.com>;tag=dt3ujuapv2 Call-ID: 15293691@pbx CSeq: 17147 PRACK User-Agent: snom821/8.4.35 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Content-Length: 0

Hi, I have a snomone+ with a bunch of internal (in the lan) and external (via the internet) phones connecting no problems....except in one external location. This is a regular DSL connection house were I have connected a 300 (and tried also with a 821) both running the latest .35 FW. WIth any of the phones I can get incoming calls perfectly with excellent audio. The problem is this phone can NOT dial to any of the other extensions. On it you do not even get a dialing or busy signal and on the remote extension you get 1ring and then nothing more. Any ideas on what to start looking for? On the DLS modem I have even tried putting the phone in a DMZ zone to see if this helps. Nothing. I have the required ports open on the snomone+ location firewall and have some 5 others phones registering remotely from other locations with no problem. It is only this one location. Any ideas? Tkx Nathan

Are the G729 licenses recycled? I'm guessing the error happens if I have X number of simultaneous calls going on and I try to pick up one?

some log on the snomone of a failed *83 call pick attempt: [7] 2012/02/22 13:49:17: SIP Rx tls:192.168.30.77:4422: INVITE sip:*877121@pbx.bondojito.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay To: <sip:*877121@pbx.bondojito.com;user=phone> Call-ID: 17c0263cd8d7-yvp7g9nueny5 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1 X-Serialnumber: 000413457659 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 524 v=0 o=root 367130050 367130050 IN IP4 192.168.30.77 s=call c=IN IP4 192.168.30.77 t=0 0 m=audio 64274 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:UKfZKYOjMYhdvmxL3Uk8T2mUEmznx5sPoPLZ0/HE a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [7] 2012/02/22 13:49:17: SIP Tx tls:192.168.30.77:4422: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport=4422 From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=f9b72edc54 Call-ID: 17c0263cd8d7-yvp7g9nueny5 CSeq: 1 INVITE Content-Length: 0 [7] 2012/02/22 13:49:17: SIP Tx tls:192.168.30.77:4422: SIP/2.0 404 Not Found Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport=4422 From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=f9b72edc54 Call-ID: 17c0263cd8d7-yvp7g9nueny5 CSeq: 1 INVITE Contact: <sip:7120@192.168.30.2:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0 [7] 2012/02/22 13:49:17: SIP Rx tls:192.168.30.77:4422: ACK sip:*877121@pbx.bondojito.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=f9b72edc54 Call-ID: 17c0263cd8d7-yvp7g9nueny5 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1 Proxy-Require: buttons Content-Length: 0 [7] 2012/02/22 13:49:29: SIP Rx tls:192.168.30.77:4422: INVITE sip:*877121@pbx.bondojito.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi To: <sip:*877121@pbx.bondojito.com;user=phone> Call-ID: 25c0263c5251-7ubabgx00lt8 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1 X-Serialnumber: 000413457659 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.32 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 526 v=0 o=root 1685612402 1685612402 IN IP4 192.168.30.77 s=call c=IN IP4 192.168.30.77 t=0 0 m=audio 56568 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+gqsKVDkqLc0eaydfsXtEcnTYUf1dCTo+h/mTo34 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [7] 2012/02/22 13:49:29: SIP Tx tls:192.168.30.77:4422: SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport=4422 From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=fa33140c8c Call-ID: 25c0263c5251-7ubabgx00lt8 CSeq: 1 INVITE Content-Length: 0 [7] 2012/02/22 13:49:29: SIP Tx tls:192.168.30.77:4422: SIP/2.0 404 Not Found Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport=4422 From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=fa33140c8c Call-ID: 25c0263c5251-7ubabgx00lt8 CSeq: 1 INVITE Contact: <sip:7120@192.168.30.2:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0 [7] 2012/02/22 13:49:29: SIP Rx tls:192.168.30.77:4422: ACK sip:*877121@pbx.bondojito.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=fa33140c8c Call-ID: 25c0263c5251-7ubabgx00lt8 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1 Proxy-Require: buttons Content-Length: 0

Try this: http://wiki.wireshark.org/SIP

screen shoot

HI, Actually most of the codes requiring an extension like *83 and *77 are the ones not working anymore. I just found out also that the call forwarding also is not working any more. Is this the screen status you wanted? Let me know if you need more infer before rebooting the system. I will try to get a a log later on .

Hi, is the upgrade procedure for snomone+ tested and working via the web interface? Last time we had a few issues but I see my boxes now showing the following are available: snom ONE 2011-4.2.1.4025 >>> 2011-4.5.0.1030 Sangoma NBE 4.1.4-1 >>> 4.1.6-1 Are the upgrades warranted? needed? recommended? We also have some specific sangoma E1 configurations in one of the boxes. Can we apply this and not loose it? Thanks Nathan

And after 3 weeks the same problem again. The call redirect or pick ceased to work. No changes done to the system. Any ideas? It is very frustrating having to reboot the system every 3 weeks.

Hi Stephen, Glad to hear it is working. And I agree, Sangoma support are awesome. Our outbound caller id is setup in the domain setting tab at the "Default ANI:" field. I believe you can also setup a different outbound caller ID in the ANI field for each extension (to match individual DIDs) but we have not tested this. I also think whomever gives you the trunk service has a saying on how or what you can control on the outgoing. I know we have a customer on which I can put any number I want on the ANI and it will show up on the other side and another customer is limited to the real DID number of the service they are getting. Hope this helps. Nathan

Hi stephen, As far as we understand, the CO lines is just more of "label" to be used by park & orbits and such things (I think). It is basically any string and we have as many as PSTN analog lines and you can call them whatever you what. Each one get associated to each of the analog lines somehow by the snomone. As for them doing something useful, I will defer to someone at snom to let us know. We just configure them always just in case We do nothing about them in the NBE setup.

sorry! here it goes

Hi Stephen, We have a few snomone+ FXOs working outside of the US with no problem, but agree that the manila could do a better job of giving examples. As somebody who learns by looking, here is a snapshot of our sangoma trunk. Just make sure you replace your boxes IP. our is a 16fxo that is why we have all of this CO lines in there. Hope this helps Nathan

Thanks Moishe for the answer. The snomone is indeed behind a firewall but it has all of the ports forwarded to it (kind of DMZ) We are using a Peplink router. For the NAT issue, we have the local/public IP in the port part of the configuration. Playing around with some settings in the trunk I managed to make outgoing calls. The incoming ones are still failing. Form the log it seems like the snomone is getting and recognizing the incoming call, even sending it to the right AA account but nothing happens: [5] 2012/02/12 22:39:19: Domain trunk vitelity@sxxxx.beta-brain.com sends call to 70 in domain snom.beta-brain.com [6] 2012/02/12 22:39:19: SIP Tx udp:64.2.142.xxx:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK3f21e420;rport=5060 From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.xx>;tag=as44e70cd8 To: <sip:361xxxxxxx@75.149.181.xxx:5060>;tag=288c11fec4 Call-ID: 50499c6c0b573e8b62cdae0201ab3d25@64.2.142.15 CSeq: 102 CANCEL Contact: <sip:nsandler@75.149.181.xxx:5060;transport=udp> User-Agent: snom-PBX/2011-4.3.0.5021 Content-Length: 0 [3] 2012/02/12 22:39:19: Via and source address are empty for SIP/2.0, cannot send reply [1] 2012/02/12 22:39:21: Timeout: Call 44 not found Any chance you can share with me your trunk definitions for any a Vitelity line? Also the LAG your are referring is in the router or in the snomone? Thanks a lot!! Nathan

Hi, I'm having a tough time getting a Vitelity trunk to work. It used to work but something happen and it no longer receives calls or makes them. Vitelity has checked it thousands of time but no luck. Tried both Sip registration and getaway. Now i'm in the point where calls out ring the phone but with only 1 way audio (snom to remote phone) Incoming calls do nothing. This is the log of an incoming one: 5] 2012/02/11 01:05:28: Identify trunk (IP address/port and domain match) 6 [6] 2012/02/11 01:05:28: SIP Tx udp:64.2.142.15:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060 From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 INVITE Content-Length: 0 [6] 2012/02/11 01:05:36: SIP Rx udp:64.2.142.15:5060: CANCEL sip:36144423xx@192.168.88.127:5060 SIP/2.0 Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport From: "+305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060> Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 [6] 2012/02/11 01:05:36: SIP Tx udp:64.2.142.15:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060 From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 CANCEL Content-Length: 0 [5] 2012/02/11 01:05:36: Domain trunk vitelity@snom.beta-brain.com sends call to 70 in domain snom.beta-brain.com [6] 2012/02/11 01:05:36: SIP Tx udp:64.2.142.15:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060 From: "+305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 CANCEL Contact: <sip:nsandler@75.149.181.125:5060;transport=udp> User-Agent: snom-PBX/2011-4.3.0.5021 Content-Length: 0 [3] 2012/02/11 01:05:36: Via and source address are empty for SIP/2.0, cannot send reply The call should route to extension 70 (as it is correctly trying to) but nothing happens. Any ideas what "Via and source address are empty for SIP/2.0, cannot send reply" means? Thanks Nathan

We are having the same issue for months now with more than 80 phones...no Spanish regardless of settings. The phones are all 300, 370 and 821.

ok. If changing the format is not possible, is there a way to include the full display of the admin call log into the daily call log email? As an example, the daily call log email show this infer for a call: 18:52 I 5558760384 6101 5558760384 18:57 An this is the same call as displayed by the admin call log on the web interface: 2012/02/05 6:52P 5558760384 6101(8912) 18:57 Trunk Analog The last one includes the Trunk and way much more important for our clients, the directed extension the call was transfers (ext 6101 is a hunt group) From the first one, the information is really useless as our client only sees an incoming call to the hunt group where if we had the full info, they can see that extension 8912 handled this incoming call. Anyway we can include this info the in the daily calls email? Thanks Nathan

We rebooted both systems and the *codes are working again. Very, very strange as nothing was changed. Just a reboot and now it work. Strange.