Re: [sipxtapi-dev] EVENT_CATEGORY_CALLSTATE::TRANSFER_EVENT::CAUSE_TRANSFER_INITIATED

[Roman Rusconi]

Forgive me if my question is stupid, but I’m still new at this.   You said: “One thing that I can point out is that transfer works only when both UA's are sipXtapi based. It fails for me when I try some other softphone.”   Why you talk about both UA when in a transfer there are 3 parties involved? One party (the one making the transfer) establishes two different calls with two other parties.   I’ve been testing my code against VoIP service providers and so far I haven’t been able to replicate what I can easily do with any softphone that’s in the market (X-Lite, Express Talk, etc.). I tried initially using BroadVoice and now I got a Vonage softphone line.   Why you say sipX doesn’t work unless the UAs involved use sipX? Is there a problem with sipX or is this a design issue?                   Roman Rusconi From: logan [mailto:[EMAIL PROTECTED] Sent: Friday, September 22, 2006 11:21 AM To: Roman Rusconi Subject: Re: [sipxtapi-dev]EVENT_CATEGORY_CALLSTATE::TRANSFER_EVENT::CAUSE_TRANSFER_INITIATED   Hi,   I have been able to transfer calls. But, I haven't tested my code on VoIP service providers. It works on with Brekeke OnDO SIP server and across NAT.   If you want then I can send you some binaries and see if it works for you. If it works then I would be happy to share the codes.   One thing that I can point out is that transfer works only when both UA's are sipXtapi based. It fails for me when I try some other softphone.   Best Wishes, Hitesh ----- Original Message ----- From: Roman Rusconi To: sipxtapi-dev@list.sipfoundry.org Sent: Friday, September 22, 2006 7:25 PM Subject: [sipxtapi-dev]EVENT_CATEGORY_CALLSTATE::TRANSFER_EVENT::CAUSE_TRANSFER_INITIATED   I have my UA with two calls established and try to make a consultative transfer but this fails. I get these events when I call sipxCallTransfer(g_hCall, g_hCall2):     EVENT_CATEGORY_CALLSTATE::TRANSFER_EVENT::CAUSE_TRANSFER_INITIATED     hCall=1, hAssociatedCall=0 EVENT_CATEGORY_CALLSTATE::DISCONNECTED::CAUSE_RESOURCE_LIMIT     hCall=1, hAssociatedCall=0     What exactly means RESORCE_LIMIT?                   Roman Rusconi     _______________________________________________ sipxtapi-dev mailing list sipxtapi-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/

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