[Freeswitch-users] 30 Second call drop.

[Kaleem]

Hi Paul,

As u mention port 5060, can u also check for 5080 in your firewall logs and sip traces,
Just as curiously I would allow 5080 and see if it helps then remove

Sent from my iPhone

On 19 Jul 2013, at 15:49, "Paul" <pasha at prosperity4ever.com> wrote:

> Hi Kaleem,
> 
> I will double check those settings.
> 
> I have my firewall setup very specific (it's not wide open to the world)..
> I h accept port 5060 UDP and all the RTP ports but only if the source IP
> is of my SIP Trunk provider proxy, otherwise it's invisible.
> 
> Thanks for the tip!
> 
> Paul
> 
> On Thu, July 18, 2013 1:06 pm, Kaleem wrote:
>> For some wired reason my last mail is not showing any text,
>> 
>> 
>> It was "Hi Paul,
>> 
>> 
>> Had similar issues myself few months back, it ended up port blocked on
>> external firewall.
>> 
>> To confirm if its same issue, please test inbound and outbound call,
>> outbound was disconnecting after 30 sec when free switch was sending 200
>> ok and not receiving response."
>> 
>> Don't know my emails are going blank :(
>> 
>> 
>> Sent from my iPhone
>> 
>> 
>> On 18 Jul 2013, at 20:55, Kaleem <k4kaleem at gmail.com> wrote:
>> 
>> 
>>> Hi Paul,
>>> 
>>> 
>>> Had similar issues myself few months back, it ended up port blocked on
>>> external firewall.
>>> 
>>> To confirm if its same issue, please test inbound and outbound call,
>>> outbound was disconnecting after 30 sec when free switch was sending
>>> 200 ok and not receiving response.
>>> 
>>> 
>>> Sent from my iPhone
>>> 
>>> 
>>> On 18 Jul 2013, at 16:54, "Paul" <pasha at prosperity4ever.com> wrote:
>>> 
>>> 
>>>> 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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>> consulting at freeswitch.org http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> 
>> 
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>> 
> 
> 
> 
>