Hi Paul,
I had similar issue while testing Freeswitch and not by using
gateways and cisco phones.Please check the issue XX-9428.But for me call
is disconnected while resuming the call(2nd Time).
Thanks,
Kumaran T
Paul Curtis wrote:
> I am having an issue with version sipXconfig (4.4.0-
> 2011-03-21EDT13:50:34 swift) where the call is disconnected when put
> on hold twice.
>
> Scenario:
> We have a cisco 2811 ver IOS15.1 with and ISDN PRI and using it as a
> sip gateway to SipXecs.
>
> User calls in from PSTN
> Call is answered.
> Answering user places PSTN call on hold.
> (MOH is invited but not heard by PSTN user)
> Answering user resumes call, now MOH is heard along with the voice.
> Answering user places PSTN call on hold again. MOH is not invited again.
> 4 seconds later, the call is disconnected.
>
> It seems that cisco 2811 disconnects the call because it thinks its a
> mute call after the second on hold.
> Also, in SipXecs and on the cisco. The ~~mh~~@ is still active for
> about 5 minutes.
> This did not occur on version 4.2.1.
>
> Here is the pertinent cisco config:
> voice call carrier capacity active
> voice rtp send-recv
> !
> voice service voip
> allow-connections sip to sip
> no supplementary-service sip moved-temporarily
> no supplementary-service sip refer
> fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback
> pass-through g711ulaw
> sip
> bind control source-interface FastEthernet0/0
> bind media source-interface FastEthernet0/0
> registrar server expires max 3600 min 3600
> !
> voice class codec 101
> codec preference 1 g711ulaw
>
> dial-peer voice 5551212 voip
> huntstop
> destination-pattern 5035551212
> progress_ind setup enable 3
> session protocol sipv2
> session target sip-server
> voice-class codec 101
> dtmf-relay rtp-nte
> ip qos dscp cs5 media
> no vad
>
>
> sip-ua
> max-forwards 15
> retry invite 3
> retry response 3
> retry bye 3
> retry cancel 3
> sip-server dns:uws.edu
>
>
>
> Does anyone have any ideas?
>
>
> --
> Paul Curtis
>
>
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>
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