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Grandstream Phones and ATA are locking up...


VADOOM

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We are using for testing:

 

Grandstream GXP2000,GXP2020 (firmware 1.1.6.16) and BudgeTone 102 (firmware 1.0.8.33), BudgeTone 200 (1.1.14.2) and one pair of GXV3000 IP Video Phones.

Grandstream adapters HandyTone 486, HandyTone 488 HandyTone 496.

Netgear WGR613V and Netgear WGR615V.

Linksys SPA2102-NA, Linksys SPA3102-NA ATA.

Linksys SPA941-NA, Linksys SPA962 Phones.

Snom 320, Snom 360 phones.

Old Sipura phones and adapters.

 

We do not have any problems with Snom and Linksys phones and adapters.Also no problems with old Sipura devices,- all of them are working perfect, NP.

But we have some problems with Grandstream and Netgear.

For example:

 

On GXP2000 we have a few lines already registred on Asterisk PBX and they are working fine. When next line is defined for PBXnsip; after reboot this line on some phones is registering OK but whe you trying to establish the call,- phone freezes and only after reboot it operates again. Some of them during registration are locking up. BT102 are locking immediately and only if you unplug the network cable it is possible to reset the configuration. To be honest depending on hardware revision,- the composition is different (old ones are locking instantly and latest ones only when you initiate media stream). Grandstream adapters working very similar.

 

GXV3000 with latest firmware isn't able to stream video when call is established (if to use Asterisk or Communigate,- video is coming)

 

Netgear Router WGR613V is registering, but only works for a while and loses registartion. The registration process is too long (sometimes more than 12min)

Netgear 615 is the same but second line is never going in to the registred status.

 

Another issue:

 

When we are using PBXnsip v.2.1.8 from gateways connected through LAN our extensions do not hear one participant (only one way audio stream). If we replace the executable file with v.2.0.3.1715 and restart the PBXnsip service,- both sides are working well. If we stop the service and to put back latest release,- the situation is going back to one way. This situation is the same with any kind of devices.

 

We are going to test with Nortel in the end of the week and will post our comments.

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Are you using outbound proxy? Some devices have problems is the domain name is "localhost", that may be the problem with the Grandstream devices as well.

 

No, I am not using "localhost" domain. Also I've tried to use "Outbound proxy", STUN, different set of codecs etc. and to not use...

Why Grandstreams and Netgears only?

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No, I am not using "localhost" domain. Also I've tried to use "Outbound proxy", STUN, different set of codecs etc. and to not use...

Why Grandstreams and Netgears only?

 

Well, a device should not crash if there is something unexpected. Did you contact Grandstream? I think if the device is crashing it should be a high priority trouble ticket and you will get a new firmware soon.

 

We are using an ATA in one office (Grandstream HT-502 V1.1B 1.0.0.44), and it was so far pretty stable and also T.38 is working fine after we disabled UPDATE support on the PBX.

 

BTW don't use STUN, it is instable and IMHO not suitable for enterprise communication.

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Well, a device should not crash if there is something unexpected. Did you contact Grandstream? I think if the device is crashing it should be a high priority trouble ticket and you will get a new firmware soon.

 

We are using an ATA in one office (Grandstream HT-502 V1.1B 1.0.0.44), and it was so far pretty stable and also T.38 is working fine after we disabled UPDATE support on the PBX.

 

BTW don't use STUN, it is instable and IMHO not suitable for enterprise communication.

 

I've contacted Ovislink, they represent Grandstream in Canada and their answer is:

if phones are working fine with Asterisk, Communigate, Brekeke, SER and other application and only one which is not working is PBXnsip,- the problem is in this application. To be honest we did not test other GS device then we have.

How about second issue? Why 2.0.3.175 is working and 2.1.8.. is not (one way audio)?

This night were tested two Techwell SIP gateways (2FXO and 4FXO),- both are working well, a little bit longer the usual registration time but NP.

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If phones are working fine with Asterisk, Communigate, Brekeke, SER and other application and only one which is not working is PBXnsip,- the problem is in this application. To be honest we did not test other GS device then we have.

 

I don't know... I don't think it is okay to say the device is crashing and they don't have any plan to take care about it. Other vendors take crashes pretty serious.

 

I think the only thing that we can do is trying out another version that is not crashing the phone.

 

How about second issue? Why 2.0.3.175 is working and 2.1.8.. is not (one way audio)?

 

What you can try is disabling the pass-through RTP. There is a global setting called "allow_pass_through", if you set it to false then it will not simply pass the RTP through, but run it through the jitter buffer. See http://wiki.pbxnsip.com/index.php/Global_Configuration_File on how to set this setting.

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