QUESTION Aastra ICOM Question

[timg]

Running PIAF Bronze 32 bit version and applied Asterisk update 1.6.2.16. This is with a Sangoma A200 card with 7 Aastra 6757i phones, POTS lines only.

When I page someone with the ICOM button on my Aastra 57i, I no longer hear a beep of any kind to let me know the call is going through and the person is listening, it was before with an older version. Having to guess when to start talking to the person at the other end. Sometimes I start too soon.

It beeps on the receiving end to let them know the page is coming. Is there a way to have it beep on my end so I can know it's OK to talk??

 

[randy7376]

Sounds like the phones do not have that feature programmed.

Here is what we have in aastra.cfg for the intercom to function.

sip intercom play warning tone: 1
sip intercom allow auto answer: 1
sip intercom allow barge in: 0
sip intercom type: 2
sip intercom prefix code: *80
sip intercom mute mic: 0

The above plays a tone, automatically answers, does not allow intercom to work if they're currently on the phone and makes the microphone hot/live. You may want to review these directives in the Aastra adminstration guide. This is working on a 1.6.2.14 installation. I don't know of a way to have it beep on the caller's side, however. I don't remember that being a feature on the phone, either. Ours have always behaved this way.

Are you using the XML scripts? We're not - so that may be difference.

EDIT: There is an xml beep notification: mentioned in the administration guide. Maybe that is set to 0 (disabled) on your phones?

 

[timg]

Randy,

Yes, I am using the xml scripts.

I thought there was a beep right before the connection was made to the dest extension. I could be wrong tho.

 

[timg]

Now that I think about it, I think I was hearing the end of the warning tone on the phone I was calling.

Our phones are no longer playing the warning tone when we ICOM to one. It just answers silently until the person starts talking.

I have play warning tone selected in the preferences for each phone. Is that the only place it should be set?

From the log file it appears it's trying to play beep.gsm but for whatever reason we aren't hearing the beep any more since upgrading the hardware and loading piaf from scratch.

[2011-01-27 10:30:55] VERBOSE[10754] netsock.c: == Using SIP RTP TOS bits 184
[2011-01-27 10:30:55] VERBOSE[10754] netsock.c: == Using SIP RTP CoS mark 5
[2011-01-27 10:30:55] VERBOSE[10754] pbx.c: -- Executing [*80211@ext-intercom:21] Dial("SIP/212-000002e5", "SIP/211,5,A(beep)") in new stack
[2011-01-27 10:30:55] VERBOSE[10754] netsock.c: == Using SIP RTP TOS bits 184
[2011-01-27 10:30:55] VERBOSE[10754] netsock.c: == Using SIP RTP CoS mark 5
[2011-01-27 10:30:55] VERBOSE[10754] app_dial.c: -- Called 211
[2011-01-27 10:30:55] VERBOSE[10754] app_dial.c: -- SIP/211-000002e7 answered SIP/212-000002e5
[2011-01-27 10:30:55] VERBOSE[10754] file.c: -- <SIP/211-000002e7> Playing 'beep.gsm' (language 'en')
[2011-01-27 10:30:55] VERBOSE[10754] rtp.c: -- Packet2Packet bridging SIP/212-000002e5 and SIP/211-000002e7
[2011-01-27 10:31:01] VERBOSE[10754] pbx.c: == Spawn extension (ext-intercom, *80211, 21) exited non-zero on 'SIP/212-000002e5'
[2011-01-27 10:31:01] VERBOSE[10760] manager.c: == Manager 'aastra-xml' logged on from 127.0.0.1
[2011-01-27 10:31:01] VERBOSE[10760] manager.c: == Manager 'aastra-xml' logged off from 127.0.0.1
[2011-01-27 10:31:02] VERBOSE[10764] manager.c: == Manager 'aastra-xml' logged on from 127.0.0.1
[2011-01-27 10:31:02] VERBOSE[10764] manager.c: == Manager 'aastra-xml' logged off from 127.0.0.1
[2011-01-27 10:31:12] VERBOSE[10768] manager.c: == Manager 'aastra-xml' logged on from 127.0.0.1
[2011-01-27 10:31:12] VERBOSE[10768] manager.c: == Manager 'aastra-xml' logged off from 127.0.0.1
[2011-01-27 10:31:12] VERBOSE[8539] netsock.c: == Using SIP RTP TOS bits 184
[2011-01-27 10:31:12] VERBOSE[8539] netsock.c: == Using SIP RTP CoS mark 5
[2011-01-27 10:31:13] VERBOSE[10769] pbx.c: -- Executing [71@from-internal:1] ParkedCall("SIP/211-000002e8", "71") in new stack

File is there:

-rwxrwxr-x 1 asterisk asterisk 726 Jul 5 2010 beep.gsm

 

[Derrick32]

Hi All,
I know this is an old post but somebody may catch it. Per the settings mentioned above:

sip intercom play warning tone: 1
sip intercom allow auto answer: 1
sip intercom allow barge in: 0
sip intercom type: 2
sip intercom prefix code: *80
sip intercom mute mic: 0

Is there a way to manually configure these settings on the server/phone rather than using a configuration file? One user is reporting that when he gets an intercom call, his phone comes up on mute and he has to un mute to answer the intercom. I was hoping to mess around with the settings above.

Thanks,
Derrick