Author Topic: 4 x GXP2100 - Incoming audio drops (Randomly), Calls in don't always connect  (Read 1701 times)

wooshman

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Hi

I have a client who has just purchased 4 of the above phones from Sipgate and has a Sipgate Team 5 account.  I have setup according to Sipgate, which is basically Open Box, plug in and enter user credentials, no need to change settings, set to DHCP client.  I have the above problems.

The setup is as follows:

A brand new Netgear DGN2200 modem/router, plugged into 16 port switch.
3 different makes of modem/router do the same so not a modem/router issue?? 
In the switch are 2 phones which are dedicated Ethernet cables, 2 are pass-through to the PC.  4 PCs in total.  2 with phone pass-through, 2 just PC's.  To be a pain, the two phones which are dedicated ethernet cables are having the second problem.

Windows Small Business Server 2011 - Setup as DHCP server and DNS server.  Phone IPs are reseved. x.x.x.18, 19, 20 and 21. (arp -a reports correct devices on network)

SIP and RTP ports all open in router and setup in the phones.  Sipgate shows correct external IP and SIP ports for all four phones.
5160/5104, 5260/5204, 5360/6304, 5460/5404

After a factory reset on each phone I ran a 5 minutes test to a cell phone which worked without problems, all 4 phones.  I then called the office from my cell phone and answered the call on one of the phones.  I then transferred the call to each phone one after another and back to the original phone, all without a problem.

I then called the boss to tell him about fixing it and it was the end of his problems.  Calling his cell, on the call for about 1m 30s and it went quiet.  I hung up and called him back.  Same again, about 1m20s.  He called the office and again the same happened.  He could hear me swearing at the loss of incoming audio but I could not hear him. (maybe a good thing).  I had even set the Qos in the router for each device as high priority.

Please, what is causing this issue?  I am pulling my hair out.  It also happens on calls to land lines during office hours.  Not good for a business.

The second problem is that I have 4 calling groups and the standard Sipgate welcome message "If you know the extension......" (Bear with me)

Option 1 - All phones ring
Option 2 - phones 1 & 2 should ring
Option 3 - phone 3 rings
Option 4 - phone 4 rings.

I use option 1 and all phones ring.  I try option 2 and the phones ring.  Then leave it some time and do option 2 and I get unobtainable.  Call again and use option 2, the phones ring.  At 4am this morning, Option 2 rings first time, 8am, the phones did not ring, call a second time, they ring.  It is as if inactivity kills the phone until an attempt to wake them.  But at 4am I know 100% that they will have been inactive for hours so this makes me scratch my head more.

Are there any settings you would suggest changing as Sipgate basically say don't change anything, yet the GXP 2010 needs things changing... makes no sense to me how some settings need to be changed on one phone model, but the same option on the 2100 should not be changed??

Any suggestions would be great.

Many many thanks in advance.

Woosh


 
« Last Edit: February 26, 2012, 09:11:25 AM by wooshman »

GSSupport65

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Hi wooshman,

    For the first issue--one way audio after 90 seconds connected to cell phone, is it possible to run a wireshark capture on the phone?

    For the second issue--does the random "inactive" issue happen to all 4 options or only option 2?

    Also, please make sure the GXP21xx are running the latest firmware 1.0.3.18.

Thanks!

wooshman

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Hi Support

I have been onsite today and audio inwards started to stop around 1 minute.  Both incoming and outgoing calls to both cell and landlines.  I have installed wireshark on one of the network PCs.  Now I have not used Wireshark before other than testing it lastnight on a home network but I do know that in either the source or destination fields are any of the voip phone IP addresses.

Above I mentioned that the Windows SBS 2011 is both DHCP and DNS server, would this stop Wireshark from seeing the IP's of the phones?
 (I appreciate your are not Wireshark support)
The second issue: Due to client demands I have removed all calling groups at Sipgate except one, so now when the main telephone number is called, all four phones ring.  This seems to be working fine at present.

(INFO: I have remote access to this site and can connect to all phones if needed as well as install software on server etc)

« Last Edit: February 27, 2012, 05:28:04 AM by wooshman »

wooshman

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A simple question.

As Sipgate believe the phones will work out of the box, is there a setting on the phones to keep the RTP port open for longer?

Call has now gone to 60 seconds before incoming audio stops.

Thanks in advance.
Woosh

GSSupport85

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Hi there,

What firmware are you using right now? Could you please send syslog to us? You may do this in your "Advanced Settings", "Syslog Server", put in your pc's ip address down here, and in "Syslog Level", select "debug"; if you are using 1.0.1.110, there should be another option "Send SIP Log", please select Yes.

Once things are all set, you can open your Wireshark, make a call, wait for 1 min until the issue has reproduced. Then, you could see something in Wireshark. Last, we will all very appreciate it if you save this .pcap file and send to us.

Thanks,

wooshman

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Thanks support,

The Firmware is indeed 1.0.1.110 and I should have put this in my first post, sorry.  I know you have asked for the firmware to be updated, but should this issue not be resolved by "Sipgate" (hopefully you can resolve it) with the firmware upgrade it gives Sipgate another excuse not to help.

I have set the phone syslog up as directed above but due to time I am cannot get access to site until UK time 8.30am Tuesday.  I will then be able to carry out the wireshark tests.

Many thanks again.
Woosh / Ben

INFO: UK Current Time 17:45

wooshman

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I received an email Monday from Abdel, part of Grandstream support saying he would be assisting me.

So this thread does not die without a resolution I thought I would post and update.

After me learning how to use Wireshark and getting the logs from the Grandstream GXP2100 the following change was suggested:

NAT Traversal - set this to "Keep-Alive"
Random Port - Set to "Yes"

Keep alive as most people know send empty RTP packets.
Random Port makes the phone chose a random SIP port.

Now, incoming audio has been fine for over 24 hours.  Why?

One possible reason is SIP ALG. It is suggested that this is turned off as it can cause more trouble than it is worth and can mess up packets etc.

Google Disabling SIP ALG for more info.

GSSupport85

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Hi wooshman,

It's great that your phone is back to normal. Please let us know if you have any questions in future.

wooshman

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Oh I sure will.

It is still in the testing process at the moment but things are looking a lot brighter than last week and Monday.

Also, thank you for your initial help.  Not sure how Abdel ended up with the problem, but your initial help was key.

Thanks again.

wooshman

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Well 2 weeks later and the dropping of audio after 1 minute has gone.  My client can now use the phones as they should be able to.

We have had two drops in audio.  One after 18 minutes and another after 22 minutes, not a problem really.

The routing was a Sipgate problem.  I got them to set it up remotely whilst on the telephone.  The way routing needs to be setup is not perfect but it works and the phones ring as they should.

Many thanks GS for the support and advice.