Vodia support
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Can you post the full SIP trace?
Thx
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Have you tried adding the Asterisk IP address in the trunk setting "Explicitly list addresses for inbound traffic" on the Vodia PBX?
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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/2.0Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rportFrom: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64tazTo: <sip:*97@voip.barrettsys.com;user=phone>Call-ID: 52b0f3c5d925-dsalfsaxbvipCSeq: 1 INVITEMax-Forwards: 70Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1X-Serialnumber: 000413244C69P-Key-Flags: keys="3"User-Agent: snom320/8.7.3.25Accept: application/sdpAllow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATEAllow-Events: talk, hold, refer, call-infoSupported: timer, 100rel, replaces, from-changeSession-Expires: 3600;refresher=uasMin-SE: 90Proxy-Require: buttons-snom320Content-Type: application/sdpContent-Length: 502v=0o=root 584979493 584979493 IN IP4 192.168.1.182s=callc=IN IP4 192.168.1.182t=0 0m=audio 50780 RTP/AVP 0 8 3 9 99 18 101a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6Dwq4Jkn2h+aiF4YkKmZnmu9wwiBVgl01psqpScwa=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:9 G722/8000a=rtpmap:99 G726-32/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=ptime:20a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitta=sendrecvSIP/2.0 100 TryingVia: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rport=2066;received=24.119.220.154From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64tazTo: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7aCall-ID: 52b0f3c5d925-dsalfsaxbvipCSeq: 1 INVITEContent-Length: 0SIP/2.0 200 OkVia: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-543qhk05dj5z;rport=2066;received=24.119.220.154From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64tazTo: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7aCall-ID: 52b0f3c5d925-dsalfsaxbvipCSeq: 1 INVITEContact: <sip:112@24.119.220.155:5061;transport=tls>Supported: 100rel, replaces, norefersubAllow-Events: referAllow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATEAccept: application/sdpUser-Agent: Vodia-PBX/5.1.3Content-Type: application/sdpContent-Length: 348v=0o=- 677813057 677813057 IN IP4 24.119.220.155s=-c=IN IP4 24.119.220.155t=0 0m=audio 19170 RTP/AVP 9 0 8 101a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:KIovh3hlKqgGlVvULirklht4JS79+jXzdtOGGeMra=rtpmap:9 G722/8000a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=sendrecvACK sip:112@24.119.220.155:5061;transport=tls SIP/2.0Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-xbj4q151mgkb;rportFrom: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64tazTo: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7aCall-ID: 52b0f3c5d925-dsalfsaxbvipCSeq: 1 ACKMax-Forwards: 70Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1Proxy-Require: buttons-snom320Content-Length: 0BYE sip:112@24.119.220.155:5061;transport=tls SIP/2.0Via: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-x0tq5z5vo8lp;rportFrom: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64tazTo: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7aCall-ID: 52b0f3c5d925-dsalfsaxbvipCSeq: 2 BYEMax-Forwards: 70Contact: <sip:112@192.168.1.182:2066;transport=tls;line=ihfiyuz5>;reg-id=1User-Agent: snom320/8.7.3.25RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0RTP-TxStat: Total_Tx_Pkts=502,Tx_Pkts=502,Remote_Tx_Pkts=0Proxy-Require: buttons-snom320Content-Length: 0SIP/2.0 200 OkVia: SIP/2.0/TLS 192.168.1.182:2066;branch=z9hG4bK-x0tq5z5vo8lp;rport=2066;received=24.119.220.154From: "Steve Butterfield" <sip:112@voip.barrettsys.com>;tag=fndlv64tazTo: <sip:*97@voip.barrettsys.com;user=phone>;tag=24b5840c7aCall-ID: 52b0f3c5d925-dsalfsaxbvipCSeq: 2 BYEContact: <sip:112@24.119.220.155:5061;transport=tls>User-Agent: Vodia-PBX/5.1.3Content-Length: 0
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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.
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Can you select external call under behavior--->Ring melody, then make an inbound call and see if that helps.
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You can utilize the access list to allow remote user and ITSP into the system then you can lock the system down.The access list will act as a firewall withing the vodia PBX.
http://wiki.snomone.com/index.php?title=Access_List -
Hi, Steve can you check this article and run the PCAP setting on the extension level when you using TLS
http://wiki.snomone.com/index.php?title=Extension_PCAP
You can also activate the PCAP setting on the trunk level for inbound and outbound call.
http://wiki.snomone.com/index.php?title=Trunk_PCAP_Generation
Best regards
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Hi, Please refer to out online documentation on custom headers.
http://wiki.snomone.com/index.php?title=Trunk_Custom_Headers
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Also try to start from scratch remove all the information from the settings save the UC, close the application, make sure the firewall allows the application as well then try to register the account again. You can always use the PCAP on the pbx to get more data.
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it's working fine on my end.... using 5.1.3
[8] 0:07:00.917 TFTP: HTTP: Received request for file snom-m9-settings-000413000000.xml from 10.0.0.3 [8] 0:07:00.917 TFTP: Provisioning file snom-m9-settings-000413000000.xml looking for MAC 000413000000 [8] 0:07:00.917 TFTP: PnP: Using the credentials of 40@10.0.0.3 for file snom_m9_settings.xml [8] 0:07:00.925 TFTP: HTTP: file snom-m9-settings-000413000000.xml based on template snom_m9_settings.xml is sent to 10.0.0.3 [8] 0:07:00.930 SIP: Packet authenticated by transport layerWhat does your logs say?
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The user agent VaxSIPUserAgent/3.1 reports the change in ports and the pbx is just reporting it, This happens on xlite smart phone application as well the Xlite keeps changing ports there nothing that can done on the PBX side, maybe VAX as some information on this item?
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We will test this item for you, It might just be a some setting on the trunk header.
Can you also tell us which version of software the snom units are running
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if you set both "Allow access for extensions"and "Extensions that may access this mailbox:" then the output will send 2 MSG try using only Allow access for extensions and the msg that are deleted by the user should reflect on the other units as well.
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Please sign up here to open a ticket.
https://snomone.zendesk.com/registration
if you have individual items open individual tickets.
Best regard.
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Hi, Can you provide us access to your machines so that we can sort this out, You can PM me the credentials and I will be able to look at the log files.
Thanks
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Luis, please send me a PM with the PBX credentials also attach your header logo as well, so I can add this for you, we will continue with other topics afterward.
Best regards
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Sie haben, um den Anruf zu übertragen * 52 dann wird das System den Anruf auf Ihr Handy übertragen.
tried my best here
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On your Bria app you will have to transfer the call to *52 then the system will call to your cell phone.
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1st create folders html and in the html folder create img folder in the snom ONE working directory. Place your image in the img folder and name the file logo_snom_small.jpg
this will take care of the log in screen.
Are you looking to change the top header as well?
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please follow this article and post the log file here.
http://wiki.snomone.com/index.php?title=How_to_Take_SIP_log_file.
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Can you double check if web sockets is supported http://www.websocket.org/echo.html
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Please check if you web browser support web socket.
http://www.websocket.org/echo.html
you can use the latest chrome or IE
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Please check if you web browser support web socket.
http://www.websocket.org/echo.html
you can use the latest chrome or IE
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Not possible, how would the call to terminate? What is the +extension for?
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Hi, please check this article on how to add on.
http://wiki.snomone.com/index.php?title=Assign_an_add_on_to_your_license.
co lines
in General Setup
Posted
You will have to create that specific co-line on a button.