Solved: How to pass CallerID from cisco fxo to asterisk?

General Help
1

Updated 04/03/2009

After deleting “clid strip” from cisco configuration, it works now. It’s that simple…

So the working config would be:

translation-rule 1
Rule 1 null null

voice-port 1/0/0
translate calling 1
translate called 1
input gain 14
no comfort-noise
connection plar opx 1510357XXXX
caller-id enable

dial-peer voice 100 voip
destination-pattern 1510357XXXX
session protocol sipv2
session target ipv4:192.168.0.10:5060
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad

Hi, guys,

I have a cisco 2651xm and a vic-2fxo-m1 card, I follow the instruction this article (http://www.voip-info.org/wiki-Asterisk+cisco+FXO) in voip-info.org but still cannot make the caller id work, does anybody know how to do it?

Any help would be appreciated!

2

what did you do exactly? That config will work for a Pure asterisk only install but if you did everything file for file on a FreePBX system it will break things.

  1. you don’t edit extensions.conf or sip.conf when using FreePBX. See: http://freepbx.org/configuration_files
  2. they have that example coded to make calls go to a non-Freepbx aware context in the sip.conf. change the context they have to from-trunk instead.
3

Hi, fskrotzki,

Thanks for your input, but I still dont get it, sorry.

What I did is to create a trunk in FreePBX and configure the router to forward calls to the PBX.

Even I change the context from “pstn-incoming” to “from-trunk”, it still doesn’t work, and I don’t know what else in extensions.conf or sip.conf I should edit, please tell me the detail.

Thanks

4

howlym,

DO NOT EDIT the extensions.conf or sip.conf files. The page you pointed to is for a pure asterisk only setup and adding these items that they say in those files is wrong, will not work and possibly break FreePBX.

That’s why I asked what exactly did you do? Do you also realize that the base writing for that was done in Jan. 2004, that’s 5 years ago.

You are asking for help but have not provided any details for us to go on. First off what version of everything are you using? (Look here to understand: http://freepbx.org/forum/freepbx/installation/so-you-have-a-problem-and-want-help)

Please look at the previous link I provided above, it tells you what files you can, should not or CAN’T edit when using FreePBX. When you do not follow these guidelines/rules you break things at the worst, or at the best, the moment you update a setting with FreePBX your changes will get lost as the files will get re-written.

5

Sorry, fskrotzki, I am not get used to the forum rules yet.

I am using FreePBX 2.5.1.2, and I didn’t edit any original file. (Actually I do not know how to edit them yet)

here is my PIAF Status:
PBX in a Flash Version 1.4 Daemon Status

  • Asterisk * ONLINE * Zaptel * ONLINE * MySQL * ONLINE *
  • SSH * ONLINE * Apache * ONLINE * Iptables * ONLINE *
  • Fail2ban * ONLINE * IP Connect* ONLINE * Ip6tables * ONLINE *
  • BlueTooth * ONLINE * Hidd * ONLINE * NTPD * ONLINE *
  • Sendmail * ONLINE * Samba * OFFLINE * Webmin * LOADING *
  • Ethernet0 * ONLINE * Ethernet1 * N/A * Wlan0 * N/A *
  • Running Asterisk Version : LOADING
  • Asterisk Source Version : 1.4.21.2
  • Zaptel Source Version : 1.4.12.1
  • Libpri Source Version : 1.4.9
  • Addons Source Version : 1.4.7

pbx.local on 192.168.0.10 - eth0
CentOS release 5.2 (Final) :32 Bit Kernel: 2.6.18-92.1.22.el5

For help on PBX commands than you can run type help-pbx *

Cisco Configuration:

voice-port 1/0/0
translate calling 1
translate called 1
input gain 14
no comfort-noise
connection plar opx 1510357XXXX
caller-id enable

dial-peer voice 100 voip
destination-pattern 1510357XXXX
progress_ind setup enable 3
session protocol sipv2
session target ipv4:192.168.0.10:5060
session transport udp
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
clid strip
no vad

and my trunk configuration:

[VIC-2FXO-M1]
disallow=all
host=192.168.0.1
type=friend
context=from-trunk
dtmfmode=rfc2833
allow=ulaw

Thanks!

6

Did you read the entire post on voi-info.org?

There is a discussion of FXO revisions and caller ID.

7

Hi, SkykingOH,

Did I miss something there? I’ve been searching for the configuration for a long time.

Please offer me the links, Thanks!

8

The link you listed in your first post, half way down the page.

It explains the different VIC’s and Caller-ID

9

Hi, SkykingOH,

Mine is vic-2fxo-m1, it supports callerid, I can see it does get the callerid info when debug it.

Thanks

10

Any further help?

Thanks