SoundPoint IP 650 Ring Answer Attendant Call Address Issue

[Anonymous]

Hello,

I am having an issue with our Asterisk PBX telephone system which utilizes a mixture of SoundPoint IP 650 (all with 1-2 Attendant Modules) and SoundPoint IP 335.  I created an extension in Asterisk which utilizes the Asterisk Module "Page" which calls all of the SIP phones using their registered extension.  I embedded the SIPAddHeader (Alert-Info: Ring Answer) to my extension prior to calling the phones so that they auto answer the intercom request. 

When the extension is dialed all of the 335 phones answer the request however none of the 650's do (they use the same configuration files); however the 650's do not have a problem auto answering a call if they are directly dialed.  

What I noticed is when I do the page all of the 650's seem to begin to call at the exact time the page is occurring everyone on their attendant module as I see sip:5101@192.168.3.11 on each phone as well as when I use the arrow keys I can see them trying to call everyone on their module.  If I remove the configuration for the attendant module on the 650 (so it just has its own extension) it is able to answer the page.    What is making the attendant module want to dial all of the phones?  Asterisk should be doing this not each phone doing it themselves.

Here are some example setups:

attendant.resourceList.1.address="sip:6101@192.168.3.11" attendant.resourceList.1.callAddress="5101" attendant.resourceList.1.label="NAME HERE" attendant.resourceList.1.proceedingIsRecipient="0" attendant.resourceList.1.type="normal"

[global]

PAGE_ALL=SIP/6101&SIP/6102&SIP/6103

[intercom]
exten => s,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => s,n,Page(${PAGE_ALL})
exten => s,n,Hangup

Thanks!

[SteffenBaierUK · Accepted solution]

Hello Aeudian,

welcome to the Polycom Community.

You kind o have already given yourself the answer. The SPIP650 are simply to busy to answer the call.

Asterisk actually sends the amount of monitored phones as SIP NOTIFY messages as Asterisk see's no difference than a normal call and the Phone's simply display this via the BLF LED's.

The maximum amount of simultaneous calls signaling on a phone is around 6.

Simply upgrade to UCS 4.x.x and utilize the PTT functionality where a Multicast message is send to the phones which will not affect their performance.

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

[Anonymous]

Steffen,

Thank you for your response.  I read the article from Polycom on PTT and understand that the PTT comes from the phone directly rather then Asterisk (so its not dependent on the phone system).  My only question is assuming someone is on the phone and the page comes across; I assume the PTT knows this and will ignore that phone? 

Also is PTT configurable via configuration files; the article I found only shows how to do it from the phone directly (as a user).

I am going to setup the PTT tomorrow when onsite and let you know the outcome.

Thanks again.

[Anonymous]

Scratch my above post.  I answered my own quetions.  Thanks for the help this worked.