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  #1 (permalink)  
Old June 14th, 2005, 08:58 PM
amigliora amigliora is offline
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Default Asterisk and Polycom Phones Caller ID

Hi,
I have an A@H with some Polycom Phones and some Sipura 841.
Sipura's phone show me both Caller Name and Caller Num while they are ringing, but Polycom phone only show Caller Name while they are ringing and when I off hook the hand set, then show me the Caller ID num.
I was trying to change this parameter in the phone, using the web but I couldn't find anything related to Caller ID, I've also sent an email to Polycom, they reply with a case number and that 's it.

Anyone know how to change the configuation in the Polycom phone to show the Caller ID number?

Thanks
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  #2 (permalink)  
Old June 16th, 2005, 05:23 PM
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mberlant mberlant is offline
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Default RE: Asterisk and Polycom Phones Caller ID

I don't know how to change it in the Polycom, but you can configure the Asterisk to make a substitution to put the Caller Number in the Caller Name field before presenting the call to the Polycom.

I don't know how to get AMP to do this, so maybe someone else can contribute to that part of the solution. In your appropriate [in-provider] context block you can add a line like "exten => s,3,SetCIDName(${CALLERIDNUM})" This command will take the vallue of the CALLERIDNUM field and paste it into the CALLERIDNAME field before passing it on to your Polycom.
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Old June 19th, 2005, 01:32 AM
amigliora amigliora is offline
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Default

Thanks for your help.
I did what you said, but AMP still change my info and show me the Caller Name, probably I'm doing something wrong but don't know yet.
Regards
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Old June 20th, 2005, 08:47 PM
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mberlant mberlant is offline
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Someone who knows the peculiarities of AMP will have to teach us where to make the change so that AMP won't undo your modification.
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Old August 2nd, 2005, 08:34 PM
Northeasternat Northeasternat is offline
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To nest the CALLERIDNUM in the CALLERIDNAME variable while manging Asterisk with AMP you will need to edit the following:

Use the "Config Edit" link on the Maintenience tap within AMP to edit the /etc/asterisk/extensions.conf file.
1. scroll to the macro-dial context.
2. reassign exten => s,1,AGI,dialparties.agi to exten => s,2,AGI,dialparties.agi
4. add this line above the "exten => s,2,AGI,dialparties.agi" line: exten => s,1,SetCIDName(${CALLERIDNAME}-${CALLERIDNUM:1})

should read as follows:
[macro-dial]
exten => s,1,SetCIDName(${CALLERIDNAME}-${CALLERIDNUM:1})
exten => s,2,AGI,dialparties.agi
exten => s,10,Dial(${ds})
exten => s,20,Wait(1)
exten => s,21,Voicemail(b${ARG3})
exten => o,1,Background(one-moment-please)
exten => o,2,goto(from-pstn,s,1)

If you have any call queues defined you will have to make some additional changes to invoke this activity when recieving a call through a queue.

perform the following:
1. Use the phpMyAdmin link on the Maintenience tab within AMP to access the asterisk database. (use the drop down box and select asterisk)
2. click on the extensions table.
3. click on browse.
4. look for every occorence of ${CALLERIDNAME}
5. Click on the edit button for that field and replace the above value with ${CALLERIDNAME}-${CALLERIDNUM:1}.
6. Remember to click "GO" to save your changes.
7. Restart AMP to enable changes. (your .conf file will be automaticlly updated)
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Old August 2nd, 2005, 08:34 PM
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Old December 9th, 2005, 09:32 PM
isepic isepic is offline
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Quote:
Originally Posted by mberlant
Someone who knows the peculiarities of AMP will have to teach us where to make the change so that AMP won't undo your modification.
I tell you what I do, but I would not recommend it to many - I go into the actucal PHP database using the portal via AAH - in there, I get to the pulldown of the databases, look in asterisk, browse around to find the actual line item in the x.conf file match it up (e.g. change a 7sec to 3sec delay for instance).. then I edit x.conf to change the 7 to a 3 also.. so now when AMP rewrites the x.conf file, it rewrites it using the 3 instead of a 7 .... for this example I'm talking about the delay on a Auto Attendant... so I see other items in the DB, line for line.. it's worked 100% for me so far, but I've been told this is not a good idea.
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Old March 4th, 2006, 03:27 AM
ManyCom ManyCom is offline
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Both calling name and number should be displayed when using SIP 1.5.1 and later. Earlier versions only display the name, if available, or the number, if the name is not available.
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