[SR-Users] Audio stops after resuming call from hold

[Sergiu Pojoga]

OMG, what are the odds, a client reported the same problem today! Edge
proxy running same 4.2.3, requests are forwarded to a farm of Asterisks v13
in a similar way based on $rd, far-end NAT traversal is handled by Kamailio.

I've had only an hour or so to debug today. Re-invites containing SDP are
handled the same way as invites in terms of SDP mangling, all looks good in
that sense. There's nothing special to be done about re-invites.

Preliminary clue is that this happens (or not) depending on the type of
firewall/NAT behind which the phone is located. In the case with the
trouble, it's a Sonicwall, probably a Symmetric NAT. Is doesn't happen to a
phone behind a Full/Restricted Cone NAT.

What nat= are you setting for Asterisk peers?
Do you engage rtpproxy/rtpengine?
Any far-end NAT traversal manipulations involved such as SIP ALG or STUN?

Cheers.

On Thu, Mar 22, 2018 at 3:55 PM, gerry kernan <gerry.kernan at infinityit.ie>
wrote:

> Hi
>
>
>
> Hoping someone can point me in the right direction.
>
> I have a Kamailio Ver: 4.2.3-1.1  running in front of a few asterisk
> servers Ver: 13.17.2  sip is routed to an asterisk server depending the
> domain name in the sip request, all working as expected . but if a call is
> put on hold  after resuming the call the party that placed the call on hold
> can’t hear any audio. The other party can hear . do I need to do anything
> special to handle re-invites for calls put on hold?
>
>
>
>
>
> *Gerry Kernan*
>
>
>
> [image: cid:image001.jpg at 01D105A5.2701B0E0]
>
>
>
> *Infinity IT   |   17 The Mall   |   Beacon Court   |   Sandyford   |
> Dublin D18 E3C8   |   Ireland*
>
> *Tel:  +353 - (0)1 - 293 0090   |   E-Mail:  *gerry.kernan at infinityit.ie
>
>
>
> *Managed IT Services       Infinity IT* - www.infinityit.ie
>
> *IP Telephony                    Asterisk Consulting* –
> www.asteriskconsulting.com
>
> *Contact Centre                Total Interact* – www.totalinteract.com
>
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