Hi Paul,
Thanks for your update.Give a try by user in sipXecs calls 100
or 101.
Regards
Kumaran T
Paul Curtis wrote:
> 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
>>>
>>>
>>>
>>
>>
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