Linksys New Zealand specific configuration

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#323230 26-Apr-2010 20:28

sbiddle:
tanivula: Hey Steve,
A couple of times when we've made a call, the other end could not hear us. First time we encountered the problem we did a complete restart of everything (modem, router, ata, phone). Second time a simple hang up and redial was all that was required.

There is nothing in there that should cause any effect like that. What device and router are you using?

While it shouldn't be necessary port forwarding the ports that are being used for RTP traffic by the device may be necessary, NAT issues are normally the cause of one way audio.

Using a WRT54GL running Tomato. Only port 5060 (UDP) is forwarded to the SPA2102. What other ports do you think I need to forward?

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#323237 26-Apr-2010 20:34

tanivula:
sbiddle:
tanivula: Hey Steve,
A couple of times when we've made a call, the other end could not hear us. First time we encountered the problem we did a complete restart of everything (modem, router, ata, phone). Second time a simple hang up and redial was all that was required.

There is nothing in there that should cause any effect like that. What device and router are you using?

While it shouldn't be necessary port forwarding the ports that are being used for RTP traffic by the device may be necessary, NAT issues are normally the cause of one way audio.

Using a WRT54GL running Tomato. Only port 5060 (UDP) is forwarded to the SPA2102. What other ports do you think I need to forward?

There are two parts to a VoIP call. Port 5060 (typically UDP) carries the SIP signalling for the call setup. The multimedia traffic is carried via RTP and normally uses two additional ports for this. Under SIP settings there is an option for the minimum and maximum RTP ports. You could try forwarding these ports just to see whether it fixes the issue.

I'd be surprised if Tomato had issues though, I haven't heard any other bad things about it.

polglase

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#323275 26-Apr-2010 22:09

sbiddle:

What numbers? THe only issue I'm aware of is the delay after the 3rd digit if you're dialling a local number. If you do pause for more than (from memory) 3 seconds between digit 3 and 4 it will pass the call as a 3 digit local number.

If this is causing issues I can tell you how to modify the dialplan to remove this issue.

A variety of numbers all with the same result. I think it is the pause after the 3rd digit that is causing the problem. If I modify the dialplan manually then I will have to stop the provisioning service or it will revert correct?

tanivula

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#323285 26-Apr-2010 22:23

sbiddle:
There are two parts to a VoIP call. Port 5060 (typically UDP) carries the SIP signalling for the call setup. The multimedia traffic is carried via RTP and normally uses two additional ports for this. Under SIP settings there is an option for the minimum and maximum RTP ports. You could try forwarding these ports just to see whether it fixes the issue.

I'd be surprised if Tomato had issues though, I haven't heard any other bad things about it.

Ok, thanks for that, I will try this and see if we have any more occurences of this. Is the RTP values updated as part of your config or are they outside that scope?

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#323361 27-Apr-2010 07:21

polglase:
sbiddle:

What numbers? THe only issue I'm aware of is the delay after the 3rd digit if you're dialling a local number. If you do pause for more than (from memory) 3 seconds between digit 3 and 4 it will pass the call as a 3 digit local number.

If this is causing issues I can tell you how to modify the dialplan to remove this issue.

A variety of numbers all with the same result. I think it is the pause after the 3rd digit that is causing the problem. If I modify the dialplan manually then I will have to stop the provisioning service or it will revert correct?

You would need to disable provising so it won't update again and then modify the dialplan

Remove xxxS1 and replace it with |018|1xx| which will now pattern match 018 and pattern match all 3 digit numbers that begin with 1 such as 111,123 etc

sbiddle

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#323362 27-Apr-2010 07:22

tanivula:
sbiddle:
There are two parts to a VoIP call. Port 5060 (typically UDP) carries the SIP signalling for the call setup. The multimedia traffic is carried via RTP and normally uses two additional ports for this. Under SIP settings there is an option for the minimum and maximum RTP ports. You could try forwarding these ports just to see whether it fixes the issue.

I'd be surprised if Tomato had issues though, I haven't heard any other bad things about it.

Ok, thanks for that, I will try this and see if we have any more occurences of this. Is the RTP values updated as part of your config or are they outside that scope?

I change nothing to do with RTP settings except for changing the packet size to 20ms away from 30ms

tanivula

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#326032 3-May-2010 14:22

Hmmmm, we're still having the odd phone call where people can't hear us - a quick hang up and redial and it's working like it should.

Any other suggestions about what could be causing this?

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sbiddle

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#326047 3-May-2010 14:46

Nope. Do you have the RTP ports that are set in the ATA forwarded in your router? While this shouldn't be necessary it should fix any such issues.

tanivula

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#327142 5-May-2010 22:13

sbiddle: Nope. Do you have the RTP ports that are set in the ATA forwarded in your router? While this shouldn't be necessary it should fix any such issues.

I had a look around the tomato config and it didn't show that there were any conflicts (in both the manual and triggered lists). Strange!

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#327210 6-May-2010 07:57

sbiddle: Nope. Do you have the RTP ports that are set in the ATA forwarded in your router? While this shouldn't be necessary it should fix any such issues.

Do you have the ports forwarded though? Lack of audio on some calls is a commonly reported fault with 2talk. Since I don't use the service I have no idea where the issue is but port ensuring the ports are forwarded is the first thing I'd look at to resolve it.

tanivula

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#327221 6-May-2010 08:50

Yep, checked RTP ports on ATA, created new TCP & UDP forwading entry in Tomato in line with these numbers. Checked again last night and the port range on both devices was correct.

The reason I thought this was provisioning related is because we've never encountered this issue using the settings that 2talk have on their set up guides.

Sorry to be a pain but appreciate all the help!

sbiddle

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#327231 6-May-2010 09:16

tanivula: Yep, checked RTP ports on ATA, created new TCP & UDP forwading entry in Tomato in line with these numbers. Checked again last night and the port range on both devices was correct.

The reason I thought this was provisioning related is because we've never encountered this issue using the settings that 2talk have on their set up guides.

Sorry to be a pain but appreciate all the help!

I don't have any ideas - there is nothing I've changed that would affect this. If it doesn't work for you I suggest you go back to 2talks config and adding in any of my settings as you see appropiate (ie 2talk provide incorrect time and DST settings on their website).

The only setting that could affect RTP traffic is the change to 20ms packets. 20ms is correct and the preferred setting for both ulaw/alaw and g729 on the vast majority of IP PBX's and VoIP providers. The Linksys default is 30ms.

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#334692 25-May-2010 18:38

Hey Steve,

We have had two phone calls today where we have dialed out a local Pukekohe number 238**** and the phone has connected to 423**** (added a 4 to the front and dropped the last digit). This is the first time this has happened since we started using the new dialplan. The redial recall on the phone shows the correct 238****. A redial works no problem and connects correctly. Any ideas?

Cheers, Matt.

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sbiddle

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#334694 25-May-2010 18:43

To be honest I don't really have any idea how something like that could happen.

hairy1

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#334697 25-May-2010 18:47

Okey dokey. It was my wife both times and I didn't have the courage to suggest that it could be her dialing (I suspect if I did it would result in me taking the 2talk line to a small 1 bedroom flat!). I just thought I would eliminate the dial plan as potential problem.

Cheers.

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