[call log - missed, received & dial not displayed ver 7]

I have upgraded snom 300 & 320 to version 7 (7.1.30 ), after upgrades I no longer see call history - missed calls, received calls and dial calls. Afer an incoming or outgoing calls when I go into the call history to review it display "no data available". I further upgrades to 7.1.33 and the results are the same - still no missed calls, received calls and dial calls data

[Audiocodes Standalone Survivability issue]

If you have a Static Public IP Address (WAN) at the MP114 location, put it in the NAT IP Address on the Quick Setup Page.

 

This will identify where the MP114 (and your extensions) are located.

 

Bill H

 

 

Tried this before and again, but the problem is still there.

[Audiocodes Standalone Survivability issue]

You said: "ip phone configired with domain name of pbx as sip proxy and LAN ip address of MP114 as outbound proxy."

 

Enable the SAS feature in the MP 114 and give it an IP Address.

 

The phone should only register to the MP114 on port 5080 as the Registar and you need to create a Proxy Name for the gateway.

 

Once you make these changes try it and let me know if it still does not work.

 

There may be several things you need to do.

 

This is not an easy gateway to setup for the first few times.

 

Bill H

 

The MP114 is configured as indicated. The issue is that extension(s) registered behind the MP114 call call remote extension (extensions registered to the hosted proxy from other networks) and extension registered to the proxy from the LAN network - two way audio works fine. When an extension which is registered from the LAN or an extension that is registered to the hosted proxy from other network calls an extension which is registered through the MP114, the pbx ivr plays 'the extension is busy at the moment ....' my thoughts are that part of the problem is that the exts behind the MP114 are registered to pbxnsip with their LAN IP address 192.168 xxx and pbxnsip is does not know where to find the exts because the MP114 nor the exts sends any reference to their WAN address, so since pbxnsip is initiating the call out it can not fine the public IP of the exts behind the MP114. just my thoughts.

[Audiocodes Standalone Survivability issue]

The senerio is as follows: Pbxnsip as hosted pbx, MP114 on different network with ip phone (ext 101) registered to hosted pbx through MP114. ip phone configired with domain name of pbx as sip proxy and LAN ip address of MP114 as outbound proxy. PBX shows ext 101 registered w/ its LAN ip address and MP114 ports registered with its own LAN ip address. MP114 is registered to pbx as extension- dial plan is set to route pstn calls out 'call extension" configuration. Wheb another remote ext (102) calls ext 101 - ivr play 'the extension is busy at the moment ..." When extension 101 dials extensions 102 call fails.

[SPA3000 configuration]

PROBLEM: Does not work consistantly when trying to make call from Hosted PBXnSIP via SPA3000 FXO where Pbxnsip and the SPA3000 are on different LAN/Network

 

Here is comfiguration.

 

1. Configured SPA to communicate via FXO in both directions

2. SPA3000 is behind NAT

3. Hosted Pbxnsip is not behind NAT

 

Result

 

1. SPA3000 line 1 works well as extension receive calls and make calls to other extensions.

2. SPA3000 PSTN works consistantly only in 1 direction- receives calls from PSTN and onpasses to Pbxnsip for processing.

3. PROBLEM: Does not work consistantly when trying to make call out from pbxnsip to spa3000 FXO.

 

I have the PSTN line registering, and dial plan set to call extension, but if I try to call it from another extension, I sometimes get 503 Service Unavailable and sometimes I get second stage dial tone for FXO.

 

Any advise?

[Custom Logo setup]

I think the trick here is that the new logo must be located in the html/img directory.

 

I tried that and still did not work.

[PBX will not register ip devices on other than port 5060]

I changed SIP UDP Port - SIP TCP Port - SIP TLS Port to other than 5060 could not get devices to reqister. note, I also changed the sip port in the device(s) to correspond with the ports in the pbx - Devices used snom 320 (set network identity port to sip port of pbx - will not register_, Linksys PAP2T-NA - set sip port to that of pbx - will not register.

[How to setup GSM gateway as trunk when registered as extension]

I have a GSM SIP gateway for routing mobile calls to and from the pbx. I have relocated the GSM gateway to a different network, in the different country (no longer on the same LAN and the pbx). How can I configure the pbx to see the GSM SIP gateway as a trunk for routing calls outbound thru the cellular network and forwarding inbound calls from the GSM SIM to the AA of the pbx? I have the GSM gateway regiseter as an extension on the pbx, the second stage is where I am stuck. I would also like the call process to be as follows: when extension dial any number beginning with 56xxxxxxxxx call is routed to the GSM gateway. When someone dials the SIM number call is routed to the pbx AA, then caller can dial extension etc.

[Custom Logo setup]

I have installed version 2.1.0.2112 (Win32), as a new installation (no upgrade) I have followed the information on the wiki and admin manual for changing the appearance (logo) yet i am unable to change the logo. I removed all information in the fields under appearance restarted the service and the 'pbxnsip' logo still appears.