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juice
Vetran
Joined: December 05 2006
Location: United States
Posts: 139
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| Posted: September 05 2007 at 2:36pm | IP Logged
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Hi there,
we find very difficult to match lines with sip messages,
maybe you can suggest how to do it reliable way.
Or maybe you can consider extending the SIP_INCOMING_CALL_INFO, SIP_OUTGOING_CALL_INFO structures with CallID.
Suppose it's not a big deal for you to add it, but it would be for sure great help for sdk developers.
Thanks
Andrew
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Jalal
Vetran
Joined: April 24 2006
Location: Iran
Posts: 188
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| Posted: September 06 2007 at 6:32am | IP Logged
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Hi,
This is what we had also requested on http://www.lanscapecorp.com/forum/forum_posts.asp?TID=359&PN =1 topic.
Regards,
Jalal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: September 06 2007 at 7:13am | IP Logged
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Hi Andrew,
You are correct. We have the same issues when debugging high line count SIP logs. It can be made easier. We will be addressing this issue – so your input will be very helpful. We have noted your suggestions and this posting. We will probably ping you when we actually perform the product changes.
Jalal is correct. We need to allow VOIP apps to access critical SIP dialog (“call leg”) parameters via the API. You are also correct in your assumption that it will be relatively easy to do.
There will be additional upcoming changes released that will help us all in this area. Post other ideas as needed so that they may be incorporated into the product.
We have recently added call state logging support for individual phone lines and individual phone line SIP logging to the product. This is useful when tracking down possible call state sequence issues and SIP issues for high line density VOIP apps. These capabilities will be made available in the near future.
You also had a few other good ideas regarding the format of the SIP logs and the information they contain. Make sure to pass this information along to us so that the changes can be considered.
Thanks Andrew. Thanks Jalal.
Support
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: October 26 2007 at 10:00am | IP Logged
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Andrew and Jalal,
A follow up note regarding this thread topic.
We have added call ID values to the following call structures: SIP_INCOMING_CALL_INFO, SIP_OUTGOING_CALL_INFO and SIP_ACTIVE_CALL_INFO. This will help correlate SIP messages with phone lines.
Regarding other "call leg" or call dialog values such as "From tags", "To tags" and branch values:
(See thread http://www.lanscapecorp.com/forum/forum_posts.asp?TID=359&PN =1)
We will wait to add these later if required. At the moment, parse out of available SIP messages what ever information you may require to create your own custom SIP messages.
Support
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juice
Vetran
Joined: December 05 2006
Location: United States
Posts: 139
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| Posted: October 31 2007 at 5:37pm | IP Logged
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Hi Support,
Sounds great that we can have access to the CallID, will be very useful for tracing calls and all other uses. We do not currently need access to the branch, tag, etc, as anything else from the SIP message we can parse out as needed.
When should we expect a release to test out this much needed feature?
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: November 01 2007 at 7:08am | IP Logged
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You and a bunch of other customers are waiting for updated product. What we will do is put together a small temp distribution image that all of you can download via your support FTP accounts. We will time bomb the product image. That way you can just use the new media engine version and build against the latest API header file. This will make it easier for all of us and you won’t have to go through all the reinstall hassles.
We are expecting to release soon (possibly next few days). As soon as we can take the product out of QA, we will release it.
Support
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juice
Vetran
Joined: December 05 2006
Location: United States
Posts: 139
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| Posted: November 26 2007 at 7:14pm | IP Logged
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Sorry for not posting sooner, but we have been very busy, and well you know things always get busy during the holidays :)
The .16 build (and we just noticed there is a .18 build with RTP enhancements, nice!) we have been testing for a week now and seems very good indeed. With the ability to easily get call ids we have been very happy to more quickly debug issues.
Plus, the user allowed line assignment of incoming lines is a very nice feature and so far has worked flawlessly. We are still running it through its paces, but so far looks really good.
Thanks ;-)
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: Phone lines with sip messages matching
