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LanScape VOIP Media Engine™ - Technical Support
 LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
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mfitzgerald
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Posted: October 10 2006 at 5:31pm | IP Logged Quote mfitzgerald


Transfer capabilities


I saw the post by Jalal at http://www.lanscapecorp.com/forum/forum_posts.asp?TID=176

It noted the only type of call transfer as of Aug 2006 LanScape would handle would be Unattended Call Transfers.

I looked in the documentation for LanScape and saw nothing regarding any additional transfer types supported such as
Attended Call Transfers:

I am using LanScape version 5.12.3.7.

I am in need of a solution or work-around for the attended call transfer.

Is there such a work-around this type of transfer, such as:
Note: Phone 1 is a traditional phone, phone 2 & 3 are LanScape phones
1. Phone 1 calls phone 2
2. Phone 2 calls phone 3 in a 3-way conference call
3. Phone 2 drops out of the conference call leaving Phone 1 and Phone 3 connected
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support
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Posted: October 11 2006 at 8:57am | IP Logged Quote support

Hi Fitz,

Your problem description was very good. We think we understand what you are attempting to do.

Your description above in steps 1, 2 and 3 can be done this way:

1) Phone 1 calls phone 2.
2) Phone 2 then calls phone 3.
3) Phone 2 places the call from phone 1 and to phone 3 in a conference session.

All parties can now hear each other.

When phone 2 wants to leave the conference and allow phone 1 to communicate directly with phone 3, this is what currently has to happen:

1) Phone 2 hangs up on phone 3.
2) Phone 2 uses unattended call transfer (the SIP Bye-Also method) to inform phone 1 that it must now call phone 3 directly.

3) When the call terminates at phone 1, phone 1 established a new call session to phone 3. Phone 3 again rings for the incoming call, the user of phone 3 answers and media is exchanged between phone 1 and 3.


Note: Some of the issues are the following:

1) VOIP phones in this deployment scenario must support the SIP Bye-Also unattended call transfer capability from RFC2543.

2) Because phone 1 is on the PSTN side and calls in to the VOIP domain via a VOIP gateway, the gateway must support unattended call transfer so that phone 1 can be reestablished with phone 3 after the transfer.

3) Internal VOIP phones cannot be transferred to a PSTN phone unless the PSTN phone is multi-lined.

3) The unattended call transfer from a conference session is not as seamless as the attended call transfer. It forces the user of phone 3 to answer another incoming call from phone 1. However, other customers are using capability as described.


The only other simple way a seamless interaction might occur is to simply mute phone 2 out of the call and allow media to continue to be exchanged at phone 2 the entire time the call is active. This has obvious ramifications though. This would also require an update from us to do this, but it’s a possibility. In this case, phone 2 remains in the call path as a simple media relay until either phone 1 or phone 3 terminated the call. This method also eats up VOIP phone lines of the LanScape phones.


Note:
We are working on the next release of the media engine. It will support attended call transfer operations (using the REFER/NOTIFY mechanism of RFC3261) and a bunch of other user requested capabilities. Attended call transfer will allow you to do what you originally described in your initial steps 1 to 3 above. We are working hard to get this new version released before the end of 2006. We do not have a firm release date yet however.

Repost additional questions as needed.


Support
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mfitzgerald
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Posted: October 11 2006 at 5:11pm | IP Logged Quote mfitzgerald

I have checked with our PBX provider (Emergent) they do not support BYE-ALSO messages. The SIP Server only supports RFC 3261.

According to their Development Team RFC 3261’s implementation replaces RFC 2543. They do support REFER/NOTIFY. They went further to explain REFER/NOTIFY would be the preferred method of doing transfers.

It would appear at this time I am at a bit of a stand-still. Unless there is some solution to generate the REFER/NOTIFY SIP commands myself. Is there some embedded way in which to format SIP Headers with LanScape directly? If so then I could hard-code the desired SIP Header and transmit as needed.

Thanks as always for your prompt response.

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support
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Posted: October 12 2006 at 10:49am | IP Logged Quote support

Hi Fitz,

You are correct when you say that REFER/NOTIFY would be the preferred method of doing call transfers. We know this fact and have this new functionality scheduled for upcoming development sometime in the near future.

Unfortunately there is no simple or elegant way to manually generate or receive REFER/NOTIFY SIP packets by an application. It would be better from a software development perspective to wait until we get this capability officially release instead of trying to hack something into your application and the media engine. However, we know this is not a great option if you are in need right now.

One thing you might want your team to consider is having us perform the required work under a short term contract. If your company can cover the cost of one of our developers full time for the duration of the work, we could start the work anytime and get you an updated product version. When the official product release is then offered, you would be updated at no charge.

We are open to further suggestion you may have so post additional question if needed.


Support
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mfitzgerald
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Posted: October 12 2006 at 11:24am | IP Logged Quote mfitzgerald

Quote:

One thing you might want your team to consider is having us perform the required work under a short term contract. If your company can cover the cost of one of our developers full time for the duration of the work, we could start the work anytime and get you an updated product version. When the official product release is then offered, you would be updated at no charge.


Would you email me directly with some details on this proposal?


Thanks.
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support
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Posted: October 12 2006 at 11:31am | IP Logged Quote support

Fitz,

Ok, We will gather the needed info and email it to you.
Hang on...

Support
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