[call Disconnects after timeouts]

See attached trace as requested.

Calllog.txt

[Dial Plan replacement]

Can someone advise on the best way to achieve the following

 

 

0011643XXXXXXX replace with 03XXXXXXX

so basically match 001164 and repalce with 0

 

 

Any advice is appreciated as this is rather urgent.

 

Thanks.

[call Disconnects after timeouts]

Disconnect reasons are:

 

• Call duration reached. For example, when the 3-minute demo key is used, the PBX disconnects the call
  
• Same thing happens after the maximum call duration, which is by default 2 hours
  
• One way audio. That means the PBX does send media, but does not receive anything. Classic is when someone in a conf call hits the mute button and the phone stops sending media (which is obviously a bug). Or when a phone crashed and restarted.
  
• Call never connects. This is a nother classic when using analog FXO gateways that have buggy polarity detection and they never connect the call (though media flows both ways).
  

Thanks for the response but im more concerned about the "timeout" issue.

Can you explain why this would occur? Outbound calls through an FXO port disconnect at a 3-5 min internal when there is media for no apparent reason and we see a timeout error.

Please explain this.

I have a customer who is getting very impatient.

[call Disconnects after timeouts]

Most of the cases it is a genuine disconnect because of no-media end-to-end. I think it is a good thing that PBX disconnects the call after detecting there is not media in either direction. If you experiencing the disconnects even if the media is present, then it is a different case.

The issue does occur when there is media. If it is a different case, can you elaborate?

Can you explain the "timeout" error when there is media?

This is occuring to a few guys as posted here.

[call Disconnects after timeouts]

We are eperiencing the same problem on the firmware 3.4.0.3194. Is there any fix or work around for this as client is not very happy

 

Can we get a response on this issue?

Is starting to occur more and more and we have not had a response.

Is this issue being looked into?

Will there be a f/w fix that will address this issue?

 

Timeout occurs at random intervals on both inbound and outbound calls.

 

Please advise asap.

[Inbound PSTN trunk call]

Hi,

Can anyone elaborate on the below and perhaps give an example of what the below should look like in the invite

 

"The incoming call matches the IP address of the outbound proxy of that trunk and a DID number in the domain of the trunk " referenced from http://kiwi.pbxnsip.com/index.php/Inbound_Calls_on_Trunk

I have set this up before using different dial strings like the ones at the bottom of the referenced wiki page however for a large scale deployment (up to 500) what is the best way?

 

Any help is appreciated.

 

Thanks.

[call Disconnects after timeouts]

I also have a customer with the same issue running f/w 3.4.0.3194, outbound calls only are disconnecting at different call duration's, typically 3-5 minutes.

Also getting the timeout error.

 

The f/w version 3.4.0.3201 did resolve this issue but introduced other problems with DTMF detection.

Will there be a f/w upgrade/fix for these issues.

 

 

 

We only install the version below. Of the 872 calls during the last 34 days, we have seen 5 to 10 of the disconnected calls...resetting a box each night isn't something we would do.

 

 

Version: 3.4.0.3194 (Linux)

License Status: pbxnsip CS410

License Duration: Permanent

Additional license information: Extensions: 7/10 Accounts: 17/20

Working Directory: /pbx

MAC Addresses: 0019156BF914 00191XXXXXX

Calls: 872/112 (CDR: 316) 0/0 Calls

SIP packet statistics: Tx: 578617 Rx: 579546

Emails: Successful sent: 160 Unsuccessful attempts: 10

Available file system space: 69%

Uptime: 2009/8/15 16:12:51 (uptime: 34 days 05:58:06) (22622 23714944-0) WAV cache:

[CS410 not detecting disconnect tone on PSTN]

Did you try to use the tone detection of the trunk? This is a generic tone detector built into the PBX (not the gateway). It is pretty generic, should also work in Australia.

Thank you for that after enabling "Requires busy tone detection" (In the PSTN g/w trunk settings) the call is cleared out after approx 5 seconds.

[CS410 not detecting disconnect tone on PSTN]

Scenario is very simple to reproduce.

Inbound PSTN call routed to AA, if calling party disconnects call at this time the CS410 does not detect the disconnect tone and keeps the line active and in use.

Clearing the call out from the web m/ment and disconnecting the PSTN line both obviously resolve the issue but this is not a feasible option.

Also occurs if call is terminated at an extensions VM box.

Current f/w is 3.4.0.3201 but also occurs on earlier f/w versions.

 

Any advice would be appreciated.

[Trunk Limit in CS410]

Hi,

I have a customer who is having issues adding in a 3rd trunk to his CS410.

I have run the license through a decoder which clearly states "Trunks = 3".

There is only 2 trunks configured! 1 trunk to an ITSP and 1 to a SIP g/w.

Upon trying to add a 3rd trunk the system gives an error stating that trunk limit exceeded.

 

Currently running f/w v.3.3.0.3165, the same error occured on f/w v.3.1.2.3120 & 3.2.0.3143.

 

In my experience with CS410's i have seen inconsistencies with this which can allow more than 3 trunks to be configured without any additional trunk licenses.

 

Anyone else come across this issue?

Does the CS410 count the PSTN g/w as a trunk even if it is deleted from the trunks list?

 

Some feedback on this issue asap would be appreciated.

Thanks.

[Direct SIP URI Dialling]

Does V.3.X in pbxnsip support Direct SIP URI dialling for outbound and inbound calls?

 

e.g. 30@<IP Address>

 

I really need it for outbound for example:

 

I’m logged into Salesforce and i keep a table of my phones IP address and ext number, in Salesforce can i highlight and dial a number from there.

Is the possible to be configured at the PBX level and not on the handset?

 

 

Please advise in detail as this is fairly urgent.

[Automatically passing a sip call out an FXO port]

Well you can add a static registration to an extension that looks like this: "sip:0@127.0.0.1:5062;line=4" (where 4 would be the FXS 4).

I have added a static registration with the following "sip:0@127.0.0.1:5062;line=1". Its still not working and all faxes are failing.

Does the "sip:0" need to be changed or is that what is needed for a static route to work?

[Automatically passing a sip call out an FXO port]

Is there anyway to automatically pass an incoming sip call to a CS410 and have the dial plan configured to pass the call out an FXO port?

 

Reason is i am testing faxing s/w on a Windows PC and have configured a registration in the CS410 with the IP address of the PC which allows the call to come into the CS410 (which i can see from the system logs) but from there the CS410 does not know what to do with the call, i dont want to send it to an extension but rather send it out an FXO port as a T.38 fax.

 

Is this possible?

[Voicemail greeting on f/w 3.0.0.2998]

Nevermind, i missed this setting.

 

Mailbox Explanation Prompt: Always or Not on personal recording

[Voicemail greeting on f/w 3.0.0.2998]

Ive noticed that after recording a personal greeting on the CS410, when a call is answered by an extensions VM box, my personal greeting is played and then the default message of "please leave you message, for further options press the pound key etc..." is also played.

 

Is there anyway to remove this message once a personal greeting has been recorded?

[Detecting Caller Id]

Is there anyway to tell the CS410 not to try and detect Caller ID?

When dialling in through FXO the CS410 has a 2 ring delay (generally) before the handsets start ringing, presumably because it takes that long to detect the Caller ID on the line, but when Caller ID is disabled on the PSTN line i would like the phones to not have this delay before they start ringing.

 

It then also rings for 2 rings once the calling end has already hang up, this can lead to users answering there phones but nobody is there because the calling end has already hung up.

[SIP Routing in v3]

Has the SIP IP Replacement list and IP Routing List been improved in v3 f/w?

I can see in your release notes it says that "The SIP routing subsystem was improved. Especially in cases when the system was using advanced routing mechanisms like IP tables, the old way could lead to wrong IP addresses", has this issue been resolved in the new f/w?

[Problem with v3 f/w]

Seems that v3 f/w has a bug when running more than 4+ domains in a hosted setup.

All outbound calls remain ok but inbound calls were getting dropped and sometimes even stuck in a loop.

Backdating to v2 f/w removed the issue.

 

Anyone else experienced this?

[Music On Hold]

Steve,

I'm not sure what to tell, I just upgrade f/w v3.0.0.2975, and re-test the MOH. It is working fine for my CS410 in my lab.

Hi Pats,

How are you streaming the music? Are you using Wave Input or RTP stream?

[Music On Hold]

I am able to make the MOH to work by following this wiki; http://wiki.pbxnsip.com/index.php/Applianc...#Setting_up_MoH

Hi Pats,

I have it configured correctly, theres not much to it.

Have you tested it using the same f/w though?

i am running f/w v3.0.0.2975.

[Music On Hold]

Anyone have MoH working through the line in port on the CS410?

I have set the source as:

 

Name:Music In Connector

Type: Wave Input

Domain:localhost

 

and have also set "Music In Connector" in the domain settings for music on hold source.

If i put both an internal or external call on hold i only get silence.

 

Default hold music works ok.

 

Also once i have made these changes to stream music through the line in port, once i reboot the CS410 all the settings are lost and revert back to the default Music on Hold source settings.

[Registering ITSP accounts on the CS410]

Okay. Because you have only one IP address, the "Separation by IP Address" does not work. So you must use the "Separation by Route" method.

 

So you should use the setting "192.168.0.0/255.255.0.0/192.168.1.1 0.0.0.0/0.0.0.0/123.124.125.126". Fill in you public IP address here.

 

Also, you must make sure that your router does not change the ports for outbound traffic. That may be a problem. For example, if there is another device in the LAN (e.g. a regular SIP phone) sending traffic to the public Internet from port 5060, then this device might take NAT port 5060 on the router - which is then not available any more for the DMZ. Unfortunately most SOHO routers have very limited routing capabilities.

 

That makes sense and i have tried it but i am still getting the same result, (one way audio on inbound call, outbound is ok) and i have now tried 2 different NAT routers. Is there any types/models of routers that you are aware of that can handle this type of routing?

Do you have a recommended setup for situations like mine?

 

Also what is the reason for provisioning phones on the WAN port and not on the LAN?

[Registering ITSP accounts on the CS410]

Maybe you have a network diagram, then we can discuss settings.

I have attached the image of my network setup, it is very basic.

 

thanks.

[Registering ITSP accounts on the CS410]

Generally, http://wiki.pbxnsip.com/index.php/One-way_Audio should be a interesting checklist. This setup is described in http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses.

 

The problem here should be that the PBX does not know what the public IP address is. So you will have to work with the IP replacement list. Generally, this is very complicated, tricky, error-prone and everything. It is recommended to use a transparent router (no NAT) and configure the eth2 interface with a routable IP address.

 

Also, make sure that the CS410 does not run DHCP on both ports. This screws the routing table up in Linux (that is a general Linux problem), at least if both DHCP servers give a IP gateway.

 

I have tried the IP replacement list and it infact made the problem worse, i only had one way voice when dialling out through my ITSP.

I dont have the option of using a routable IP address and i need to get this working with a NAT router. It is also not a codec problem as i have double checked this.

 

The CS410 is also not running DHCP on both ports, im only using the WAN port at the moment.

There must be some way of getting this to work behind NAT.

 

Any other suggestions?

[Registering ITSP accounts on the CS410]

Anyone had any issues with one-way voice on the CS410.

I have the CS410 sitting behind a NAT router with all ports forwarded to the CS410 from the router.

 

Outbound calls out my 2 different ITSP accounts are ok, but inbound calls i get one way voice! The external (Calling party) cannot hear me but i can hear them ok.

 

Any suggestions? am i missing something in the CS410?