Vodia support

We will test this scenario in our lab and see if it's a firmware issue on the phone. Thanks

Navigate to the "Routing Rules" on the Sangoma and copy paste the rules on note pad and replace all localhost with @127.0.0.1 after your done copy paste it back into Sangoma and validate the change and save, then restart the gateway. an example would look like this. sip:dnis@127.0.0.1:5060

Do you have the country code set on the "Domain Setting" "Country code"?

There are still some issue with the "show to test" Area sometime it shows the correct output, this will have to revised at some point. You can check this article on dial plans. http://wiki.snomone.com/index.php?title=Dial_Plan_Samples

You will have to open a ticket with sangoma to solve this configuration issue on NBE, On the snomONE side of things there isn't much we can do about NBE not disconnecting the call. http://www.sangoma.com/support/contact_support.html

Do you have any firewalls in the middle that may be blocking ports on the pbx? it sounds like your having a one-way-audio issue. Can you provide with a sip-trace from the pbx? follow these instruction so you capture the trace. http://wiki.snomone.com/index.php?title=Retrieving_SIP_logging

We are sending the User/Pass on the invite so somethings up? Can you recreate the trunk and they try again. [5] 20120131151306: SIP Rx udp:91.121.129.20:5060: SIP/2.0 407 authentication required Call-ID: f5324e4f@pbx Contact: <sip:0486688001@91.121.129.17:5060;user=phone> CSeq: 20957 INVITE From: "LAURENT:45" <sip:45@pbx.company.com;user=phone>;tag=38247 Proxy-Authenticate: Digest realm="sip.ovh.net",nonce="085d90ab445fd6f22fd5ca59158a95d0",opaque="085c3e0d6890b34",stale=false,algorithm=MD5 Record-Route: <sip:91.121.129.20:5060;transport=udp;lr> To: "0486688001" <sip:0486688001@pbx.company.com;user=phone>;tag=00-07383-085d97f4-13d15b8e3 Via: SIP/2.0/UDP 192.168.1.200:5060;received=213.41.241.100;rport=50628;branch=z9hG4bK-476f609b2f865e83ac8b945cd6635f5a Allow: UPDATE,REFER,INFO Server: Cirpack/v4.42a (gw_sip) Content-Length: 0 [5] 20120131151306: SIP Tx udp:91.121.129.20:5060: INVITE sip:0486688001@sip.ovh.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-c5908628c5b8af81f250d9b01af49dec;rport From: "LAURENT:45" <sip:45@pbx.company.com;user=phone>;tag=38247 To: "0486688001" <sip:0486688001@pbx.company.com;user=phone> Call-ID: f5324e4f@pbx CSeq: 20958 INVITE Max-Forwards: 70 Contact: <sip:0033411930139@192.168.1.200:5060;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/2011-4.5.0.1016 Alpha Monocerotids P-Asserted-Identity: <sip:0033411930139@sip.ovh.net> Privacy: id P-Charging-Vector: icid-value=;icid-generated-at=192.168.1.200;orig-ioi=pbx.company.com Proxy-Authorization: Digest realm="sip.ovh.net",nonce="085d90ab445fd6f22fd5ca59158a95d0",response="4b60c5a793172824776c0766fd32c914",username="0033411930139",uri="sip:0486688001@sip.ovh.net;user=phone",opaque="085c3e0d6890b34",algorithm=MD5 Content-Type: application/sdp Content-Length: 231 v=0 o=- 44713 44713 IN IP4 192.168.1.200 s=- c=IN IP4 192.168.1.200 t=0 0 m=audio 55210 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 20120131151306: SIP Rx udp:91.121.129.20:5060: SIP/2.0 100 Trying Call-ID: f5324e4f@pbx CSeq: 20958 INVITE From: "LAURENT:45" <sip:45@pbx.company.com;user=phone>;tag=38247 To: "0486688001" <sip:0486688001@pbx.company.com;user=phone> Via: SIP/2.0/UDP 192.168.1.200:5060;received=213.41.241.100;rport=50628;branch=z9hG4bK-c5908628c5b8af81f250d9b01af49dec Content-Length: 0 [5] 20120131151306: SIP Rx udp:91.121.129.20:5060: SIP/2.0 403 Wrong login or password Call-ID: f5324e4f@pbx Contact: <sip:91.121.129.17:5060> CSeq: 20958 INVITE From: "LAURENT:45" <sip:45@pbx.company.com;user=phone>;tag=38247 Record-Route: <sip:91.121.129.20:5060;transport=udp;lr> To: "0486688001" <sip:0486688001@pbx.company.com;user=phone>;tag=00-07369-085d9807-6be51daf5 Via: SIP/2.0/UDP 192.168.1.200:5060;received=213.41.241.100;rport=50628;branch=z9hG4bK-c5908628c5b8af81f250d9b01af49dec Allow: UPDATE,REFER,INFO Reason: q.850;cause=1 Server: Cirpack/v4.42a (gw_sip) Content-Length: 0

tested both scenarios *90 without the extension is prompted by the system "audio_en/aa_enter_extension_number.wav" *9044 with the extension just intercom into the extension. snom-PBX/2011-4.5.0.1016 Alpha Monocerotids

This item is on our ticket system, we will continue to debug from there. Thank you

Can you revert back to 2011-4.5.0.1016 and change your sip password? Maybe somethings has changed after the update.

You should be able to set this parameter at the extension level. Check the following items IVR language: Web language: After you have set the correct language, reboot the phone.

You will have to run quick setup to configure the A200 card, are you able to see the Sangoma GUI? Another way to reach the sangoma GUI is through port 7783 http://pbx adddress:7783

By placing a pound sign after the direct destination (e.g., “1#”), the system will wait 3 seconds before dialing the direct destination. This is useful if you have extension numbers in the 200 range (201, 200, etc.). The 3-second delay ensures that the caller’s complete input (e.g., 212) will be processed rather than just the first digit. This is a classic scenario.

sure PM me the portal information. I should be able to look at your settings.

looks like the pbx detected the DTMF from the sangoma card. [7] 2012/01/28 08:56:00: Received RFC4733 DTMF on codec 101 [6] 2012/01/28 08:56:00: Received DTMF 2 [8] 2012/01/28 08:56:00: call port 30: state code from 200 to 200 [8] 2012/01/28 08:56:00: Play audio_moh/noise.wav [6] 2012/01/28 08:56:01: Received DTMF 1 [8] 2012/01/28 08:56:01: Packet authenticated by transport layer [6] 2012/01/28 08:56:01: Received DTMF 2 [8] 2012/01/28 08:56:03: Packet authenticated by transport layer Can you add 212 to the Direct Destinations options and dial 1. Lets see if the extension can be reached.

Can you send us the invite to both scenarios? Set the following. Log level 9 Log general events:Yes Log SIP events:Yes Log other messages:Yes Log media events:Yes everything else you can set to no then save. Clear the log-file 1. make a inbound call to the AA, dial the extension 2. Call internal from an extension to the AA and dial an extension.

Navigieren Sie zu Domain-Einstellungen, Provisioning_Parameters Sie können die Authentifizierungs-PIN dort zu ändern. Starten Sie das Telefon, nachdem Sie die Änderung vorgenommen hat. http://de.wiki.snomone.com/index.php?title=Administering_the_Domain#Provisioning_Parameters

Navigate to "server console" --> Networking-->Network Configuration--->Network Interfaces You will see 2 nic available eth1 and eth2 if you choose "Activated at Boot" you can choose eth1 and add your static IP there. Save and reboot the unit. If you want WAN access you can use the 2nd nic and do the same procedure.

You can do this on the web page control. 1. Search for snom_3xx_phone.xml 2. open the file and add this syntax anywhere on the file make sure it does overlap with anything else. <user_xml_screen_url idx="INDEX" perm="RW">You will add you URL here</user_xml_screen_url> 3. Save the file and reboot the phone for any body that wants to try this you can use this link to generate an image. http://wiki.snom.com/Category:Idle_Screen

You should be able to put a * on each extension so they will provision in order, what version are you running?

Are you using "Ad-hoc-Conference" or Schedule conference room? If you're using Ad hoc you should be able to remove the Moderator Access Code and the Participant Access Code. Have you tried calling into the conference room to comfirm?

We do have a failover mechanism on the pbx but this only works if you have more then one trunk on the system. http://wiki.snomone.com/index.php?title=Configuring_Trunks#Outbound_Settings

Based on your setting your pnp will be "WAN" based and because you're a hosting solution there are certain criteria that have to be done on the phone in order to pnp to your server. Make sure that all of the http, https ports are open on your router. This article should help you get started. WAN BASED PROVISIONING

You will just have to create more park orbit accounts on the system and assigned the buttons to the receptionist. You have to make sure that the star codes are the same has step 2 mentions, one button can park a call and retrieve it as well. Steps: 1. Create a park orbit account (this is the new account type), say 555 2. Domain->Settings->Star codes page - set both park and retrieve as the same star code, say *85 3. Create a button profile with "Name=7, Type=Park orbit, Parameter=555" 4. Assign this button profile to the snom phone/extension of your choice, say 200. 5. Log in to phone and see if the "Function Keys" page show this button set 6. Place a call to this extension 200 7. Answer the call 8. Put the call on hold and Press the park button 7. 9. You will hear the message that the call is parked on orbit 555 10. Caller will be hearing the music on hold 11. Press the retrieve button (same button 7) 12. You should be connected back to the call. 0

The ticket system has changed some of its criteria a bit , if you would like to resolve this send me a PM.