[Freeswitch-users] 30 Second call drop.

[Paul]

Thanks Tim!

I'm not convinced it's firewall because of the VPN, and I allow and
masquerade all traffic to and from the local and vpn subnets. All my
Asterisk boxes work fine, and so does NGCP, I'm just having this issue
with FreeSwitch.. most likely some noob mistake as this is my first
attempt at freeswitch :)

Paul


On Thu, July 18, 2013 10:29 am, Tim Meade wrote:
> I have found in the past that a 30 second call drop is normally a
> firewall issue.
>
> If you have bypass media on:
>
>
> <action application="set" data="bypass_media=true"/>
>
>
> Then at almost exactly 30 seconds FS will take you out of the media path.
> If your 2 endpoints cannot see each other due to a firewall (or nat) then
> the media will drop.
>
> In our case it's been firewall settings every time.
>
>
> Hope that helps.
>
>
> Tim
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul
> Sent: Thursday, July 18, 2013 11:54 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] 30 Second call drop.
>
>
> Hi guys,
>
>
> After hours of googling and trying every different config options I need
> some help!
>
> My scenario is as follows:
>
>
> Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote
> Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
>
>
> Main Network (where Freeswitch resides): 10.0.0.0/24
>
>
> My sip gateway/proxy: 10.0.0.40 (kamailio)
> Freeswitch: 10.0.0.34
>
>
> I can fully use the VPN from both networks (ssh, web gui control, etc)
>
>
> phones can register, however here are my problems:
>
> When calling incoming DID none of the extensions ever get ring, it skips
> to voicemail... when leaving a message voicemail cuts of after 30-31
> seconds exactly. I am wondering if freeswitch isn't detecting my client
> sending audio and interprets it as silence and therefore hangs up. The
> voicemail message itsel does contain everything said right up to the hang
> up.
>
> Sometimes I can call between extensions if they are on the same side of
> the VPN subnet (say 101 calling 101 and they both are coming from
> 192.168.1.0/24 via 10.8.0.0/24)
> If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user
> is unavailable.
>
> I'm suspecting this is a NAT issue, all the reading I have done talks
> about "external sip and rtp ip" in my case my trunk hooks up from private
> ip to private ip, it's the kamailio that's supposed to do the NAT (which
> I have verified works fine, I can place outgoing calls through it from
> other PBXs no problem with two way audio and working as it should be).
>
> I'm not sure if there is a way to enable nat or not with freeswitch. One
> thing I have done (as I've found this worked for some people) is set all
> references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my
> freeswitch eth0). There are no other adapters on the box.
>
> I am hoping that as this is my first time setting up freeswitch that one
> of you freeswitch experts can point me in the right direction :)
>
> Thanks in advance guys!
>
>
>
>
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>
>
> 
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>
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>
>
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