Similar, I did look at that issue. But I don't get the issue if calls stay within SipXecs. Could be related.
-- Paul Curtis On 3/23/11 11:10 PM, "Kumaran" <[email protected]> wrote: >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 >> >> > > > >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> sipx-dev mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > >_______________________________________________ >sipx-dev mailing list >[email protected] >List Archive: http://list.sipfoundry.org/archive/sipx-dev/ _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev/
