Vodia support

1. Is there a way that they can do that from the phone? Yes this can done on the phone or on the User Web interface under "When caller reaches mailbox, play this greeting:" http://wiki.snomone.com/index.php?title=Mailbox 2. Is there a way that this can be done automatically? No

Hi, Carlos Update the ticket with the IP of the box user/pass for ssh. Best regards

You will have to .tar the whole directory. here is example sudo tar -zcvf snomONE.tar.gz /usr/local/snomONE you can be in the root directory when you tar the file as well the tar will be stored there btw

Hi, Tobymac can you check if the certificates are available on the PBX?

The only way to call other ACD groups is by using the "Caller in queue" but it's not possible to call multiple ACD. Anymore specifics?

switch to admin---> scroll down to status general to check if the license has been activated If the issue is that you don't see any of the domains or user please upgrade your chrome browser to the latest or firefox. you can verify if your web browser support html5 or websocket as well. http://www.websocket.org/echo.html

Thanks for the info, We will test this item in our lab and see if it's reproducible. Best regards

Can you PM us the log files usually there are some clues there. Best regards

What do the phone logs say? if it's not fully provisioning then usually the phone will report an error code of 500 internal server error also check if there has been any changes on the snom templates this could also cause some issues.

GENE: means general

The best defense for these types of scenarios is using the access list on the PBX by allowing the users whom are tied to the system. The SIP scanners only send request to the PBX and if a SIP request is answer then they will start guessing sip password ect or just simply by sending a bogus invite to the PBX which result in the trunk provider trusting the source. The access list will automatically block these request by automatically blk listing these external IP.

Please check if the alert playback is turned off. under advance-->audio--->Alert Info playback: on or off if it's off then the phone will ignore the Alert-Info sent by the PBX. http://wiki.snom.com/wiki/index.php/Settings/alert_info_playback

I have tried to reproduce this item and was unsuccessful . Does the extension "Call forward all calls to" or "Call forward when not registered" set in the redirection? or perhaps there is a timeout on the AA that send the call elsewhere? Maybe you can post a log on the event or PM me a PCAP trace http://wiki.snomone.com/index.php?title=Trunk_PCAP_Generation

Just for testing purpose can you create a new AA , I just tested here and it waits for me to input the extensions at 3?

Hi, Stefano here is an article on extension PCAP tracing. http://wiki.snomone.com/index.php?title=Extension_PCAP also are you incoming calls coming into a hunt group or agent group? Best regards

Did you setup the "Extension input" for 3 Digits under the behavior section of the AA?

That could cause some confusion but have you tried creating the col lines for trunk 2? colines 7-8. The pbx should be able to determine the line parameter that will help determine which trunk is actually calling into the system. http://wiki.snomone.com/index.php?title=Inbound_Calls

Based on your question you want to know when a hunt group is being called by looking at the co-line button? 1. Add the DID to the hunt group 2. Create a coline1 account against the trunk you want to monitored. 3. Assign or create a button for that extension using the shared line option and name the parameter coline 1 the user should see the LED light up when an incoming call is coming into the hunt group. http://www.youtube.com/watch?v=6A5ZJFnjPwg&feature=youtu.be

We will re-test this item. Keep you updated.

This will also depend if the carrier as well, some will not show CID and will just advertise a number. Not sure which one will work but it's worth a shot. or If non work you may want to try different combinations. http://wiki.snomone.com/index.php?title=Trunk_Custom_Headers

You will have to create that specific co-line on a button.

Can you post the full SIP trace? Thx

Have you tried adding the Asterisk IP address in the trunk setting "Explicitly list addresses for inbound traffic" on the Vodia PBX?

Looks like the phone is remote? The phone send it's SDP with 192.168.1.182 and the PBX is sending it's SDP with 24.119.220.155. The PBX cannot send RTP traffic to a local IP. Check if you have a firewall set at the server level and on the router you can turn them off for testing services and try your the VM test. i this case if the phone was remote it would be sending SDP based on its Dynamic IP. INVITE sip:*97@voip.barrettsys.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rport From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone> Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1 X-Serialnumber: 000413244C69 P-Key-Flags: keys="3" User-Agent: snom320/8.7.3.25 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-snom320 Content-Type: application/sdp Content-Length: 502 v=0 o=root 584979493 584979493 IN IP4 192.168.1.182 s=call c=IN IP4 192.168.1.182 t=0 0 m=audio 50780 RTP/AVP 0 8 3 9 99 18 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6Dwq4Jkn2h+aiF4YkKmZnmu9wwiBVgl01psqpScw a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rport=2066;received=24.119.220.154 From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 1 INVITE Content-Length: 0 SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rport=2066;received=24.119.220.154 From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 1 INVITE Contact: <sip:112@24.119.220.155:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Vodia-PBX/5.1.3 Content-Type: application/sdp Content-Length: 348 v=0 o=- 677813057 677813057 IN IP4 24.119.220.155 s=- c=IN IP4 24.119.220.155 t=0 0 m=audio 19170 RTP/AVP 9 0 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:KIovh3hlKqgGlVvULirklht4JS79+jXzdtOGGeMr a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ACK sip:112@24.119.220.155:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-xbj4q151mgkb;rport From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1 Proxy-Require: buttons-snom320 Content-Length: 0 BYE sip:112@24.119.220.155:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-x0tq5z5vo8lp;rport From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 2 BYE Max-Forwards: 70 Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1 User-Agent: snom320/8.7.3.25 RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=502,Tx_Pkts=502,Remote_Tx_Pkts=0 Proxy-Require: buttons-snom320 Content-Length: 0 SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-x0tq5z5vo8lp;rport=2066;received=24.119.220.154 From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64taz To: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7a Call-ID: 52b0f3c5d925-dsalfsaxbvip CSeq: 2 BYE Contact: <sip:112@24.119.220.155:5061;transport=tls> User-Agent: Vodia-PBX/5.1.3 Content-Length: 0

Under status there is a setting called phones, here you can see all the IPs that are remote to the system, you can also add them to the access list. It's advisable to set up the email notification so that you can be updated when a IP has been blacklisted from the system there is no clean way of doing this but the sad reality is there are SIP scanner and SIP vicious programs built on hacking the PBX that's why the access list is your best bet in protecting the integrity of the PBX.