Connected Party ID

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Hi,

Has anyone running Asterisk 1.8 been able to configure CallerID for attended transfers (aka Connected Caller Identification)?

With Asterisk 1.8, I see the release notes state Connected Caller ID support. And this post linked below from Kevin@Digium indicates that it is meant to provide for CallerID for Attended Transfers. That would be groovy. But I can't find any reference to it in the application.

Here's the link to a Kevin@Digium explaining what Connected Caller Identification is.

http://www.mail-archive.com/[email protected]/msg238484.html

Here's a link to to the release notes for 1.8, notice the features for the reference to Connected Caller Identification.

http://lists.digium.com/pipermail/asterisk-announce/2010-October/000277.html
 

kenn10

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I believe I read something on the FreePBX site saying they need to do some work on their code before this will work.
 

malcolmd

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Howdy,

There are macros that you can configure for complex stuffs, but for most people, the simple method's going to suffice.

The simple method utilizes two parameters in sip.conf:

trustrpid
sendrpid

You need to set trustrpid and sendrpid to "yes" in the general section.

You also need to set trustrpid to "yes" for each sip peer you'd like to be able to receive the updates.

sendrpid should be set for each peer you'd like to send the updates. There are two options for it:

"rpid" to use Remote Party Identification
"pai" to use P-Asserted Identity

It's SIP phone specific though. Your phone may not support it. If your phone does support it, it might only support one type.

The phones that support it include Siemens phones, Polycom phones (rpid and pai), and Snom phones (pai only).

Cheers
 
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Thanks for the replies Kenn10 and Malcolm.

Malcolm, that is great info! I was able to make it work with my Aastra 6739i endpoints. I set sendrpid=pai, and I now have CallerID set correctly for Attended Transfers and for directed call pickup. Man this is sooo cool guys, thanks a lot!

For documentation purposes for anyone else that wants to try, here are the files I modified on my PIAF Purple w/ Asterisk 1.8.0

root@pbx:/ $ cat /etc/asterisk/sip_general_custom.conf
trustrpid=yes
sendrpid=yes
root@pbx:/ $
root@pbx:/ $
root@pbx:/ $ cat /etc/asterisk/sip_custom_post.conf
[2478](+)
trustrpid=yes
sendrpid=pai
[2479](+)
trustrpid=yes
sendrpid=pai
root@pbx:/ $
After making the changes, I reloaded Asterisk, and restarted the Aastra 6739i endpoints.


┌────────────────────────System Information───────────────────────────┐
│ Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE │
│ SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE │
│ Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE │
│ BlueTooth = ONLINE | Hidd = ONLINE | NTPD = ONLINE │
│ SendMail = ONLINE | Samba = OFFLINE | Webmin = ONLINE │
│ Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A │
│ │
│ PBX in a Flash Version = 1.7.5.5 │
│ FreePBX Version = 2.8.0.4 │
│ Running Asterisk Version = Asterisk 1.8.0 │
│ Asterisk Source Version = 1.8.0 │
│ Dahdi Source Version = 2.4.0+2.4.0 │
│ Libpri Source Version = 1.4.11.5 │
│ Addons Source Version = 1.6.2.2 │
│ IP Address = 10.0.1 on eth0 │
│ Operating System = CentOS release 5.5 (Final) │
│ Kernel Version = 2.6.18-194.26.1.el5 - 64 Bit │
└─────────────────────────────────────────────────────────────────────┘
 
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Thanks for the info and testing. As a reminder, please consider sending this sort of information over to FreePBX (usually best in the form of a ticket), e.g.

freepbx.org/trac/ticket/4790

That helps to make sure that we capture this and make accommodations as such functionality comes out. For now that's just been added to 2.9 per the ticket so you won't have to hand edit "complicated" (for most users) configuration files to get things working as you like.

It would also be a great thing for tm1000 to look at facilitating in the endpoint manager to address it based on the knowledge of what endpoint is configured and what it needs.
 

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