SPA 942 Paging/Intercom

Hello,

We have a customer that would like to paging an extension using SPA942’s. When they page currently, it rings the phone until they pick up and then the bridge is created. We would like be able to dial the extension and break through the speaker “if” they aren’t on the phone. Is this considered Intercom at that point? If so, how can we establish an intercom 2 way session with that user? Currently when we try it and add an answer on 0 into the context, it kicks the person off their current phone call without checking first. Any ideas?

I see other people viewing this. It’s been over a year without a reply so i’m assuming there is no fix for this. Please let me know if anyone has anything on this, as we are desperate. Thanks.

-Rick

Sorry, I can’t speak for the SPA942. I haven’t worked with that phone.

We do what your talking about with the Aastra 6757i. It has the ability in the phone to turn on AutoAnswer.

these are the settings in the aastra.cfg we use…

sip intercom mute mic: 1
sip allow auto answer: 1
sip intercom type: 2
sip intercom allow barge in: 0

Then I use the following dialplan…

exten => _**5XXXX,1,NoOp(Intercom)
exten => _**5XXXX,n,SIPAddHeader(Alert-Info: info=alert-autoanswer)
exten => _**5XXXX,n,set(CALLERID(name)=[INTERCOM] ${CALLERID(name)})
exten => _**5XXXX,n,Dial(SIP/${EXTEN:3},12,TtrA(beep))

When they are not on the phone…it will auto answer and the caller can speak out which will be heard out the speaker. We have the phone’s microphone muted by default so that a caller can not just listen in. The called person has to press the mute to be able to speak back.

If they are on the phone, one of the other lines rings as it would if they are receiving a call, the word INTERCOM was added to the Caller ID so they can distiguish that it is an intercom call.

Perhaps some of this will help you find a compariable solution for the SPA942…

I will try this in our test enviornment and give it a shot. Below is from the configuration page of the phones. These are our current settings. Do you see anything that may be of use changing before I make the Call Plan adjustments? Thanks.

                           Supplementary Services  

Conference Serv: yes
Attn Transfer Serv: yes
Blind Transfer Serv: yes
DND Serv: yes
Block ANC Serv: yes
Call Back Serv: yes
Block CID Serv: yes
Secure Call Serv: yes
Cfwd All Serv: yes
Cfwd Busy Serv: yes
Cfwd No Ans Serv: yes
Paging Serv: yes
Call Park Serv: yes
Call Pick Up Serv: yes
ACD Login Serv: no
Group Call Pick Up Serv: yes
ACD Ext: 1

                           Supplementary Services  

Block CID Setting: sno
Block ANC Setting: no
DND Setting: no
Secure Call Setting: no
Dial Assistance: no
Auto Answer Page: yes
Preferred Audio Device: Speaker
Headset Send Audio To Speaker: no
Time Format: 12hr
Date Format: month/dayday/month

Google search comes up with this …

http://homecommunity.cisco.com/t5/VoIP-Phones/SPA942-auto-answer-page-intercom-beeping-loudly/td-p/215064

note they they use this sip header:

Thanks. I see in the extensions.conf they already have something in there for the SPA and I now have it configured to Auto-Answer in 0. That works fine. It’s just when we intercom that phone, it auto answers if the person is on the phone, and throws the line on hold to pickup the intercom. I see in your config you have SIP Intercom Barge disabled. This is exactly what i need to find on our phones to disable it, but not quite sure how. :frowning:

Hi, I see this old post. You can write a short AGI to loop through the hints of those phones and see the current state, and then return any phone that is not being used.

I can’t believe there is still life out there haha. I still have not resolved this. Any help is welcome. Thank you kindly for responding. I purchased books at Borders & barnes and Noble to help with code but nothing worked. Thanks again, and looking forward to your response. Thanks.

P.s. If you need any files, let me know. Thanks.

What if you limit the ammount of concurrent calls for that phone in Asterisk itself? You can do that with defining call-limit=1 in the sip.conf for the extension.
That would mean, that when a person is on an active call, a second call to the phone would stop in Asterisk. The down side of limiting the ammount of concurrent calls for an extension to 1 is, that the user of that extension can not do Call Transfer.