Page issue (sound dropping in 28-30 seconds)

Hi all

I use a web browser to hit a page as seen below

<?php $socket = fsockopen("",5038, $errno, $errstr,30); fputs($socket, "Action: Login\r\n"); fputs($socket, "UserName: admin\r\n"); fputs($socket, "Secret: mypass\r\n\r\n"); fputs($socket, "Action: Originate\r\n"); fputs($socket, "Channel: Local/230@from-internal-custom\r\n"); fputs($socket, "WaitTime: 60\r\n"); fputs($socket, "Exten: 13111\r\n"); fputs($socket, "Priority: 1\r\n"); fputs($socket, "Callerid: !!ocanada!!\r\n\r\n"); $wrets=fgets($socket,128); echo 'calling'; ?>

This then fires off

exten => 230,1,Page(LOCAL/PAGE161@testing)

This in turn fires off

include => ext-paging-custom
exten => PAGE161,1,Set(_FORCE_PAGE=1)
exten => PAGE161,n,Set(TIMEOUT(absolute) = 125)
exten => PAGE161,n,SIPAddHeader(Call-Info:answer-after=0)
exten => PAGE161,n,SIPAddHeader(Alert-Info:Ring Answer)
exten => PAGE161,n,SIPAddHeader(Alert-Info:alert_info=RA)
exten => PAGE161,n,Set(__SIP_URI_Options=intercom=true)
exten => PAGE161,n,Dial(${DB(DEVICE/7000/dial)},150,A(ocanada))
exten => PAGE161,n,Hangup

On asterisk 1.4.26 it will play the song in full

On asterisk it plays for only 28-30 seconds as showen on the phone

What I see happening is that on the 1.6 system, the channels don’t appear to be staying open for the page long enough to complete

I’m a little lost to the reason so any advice would be great


James Mackinnon

You need to provide SIP debugging information as detailed on’s reporting guidelines.

A SIP drop in this timeframe an be due to a broken implementation of re-invites.


I found and resolved

in my call, I added timeout for originate which wasn’t needed when using 1.4 but 1.6 has a 30000ms timeout by default

Works like a charm now. :wink:

would you please upload your new code

Hi, Im sorry my english is not too good to understand somethings. I also have this problem. I have installed 52 Polycom IP 450 phones with switchvox aa355 and when we page to all phones, caller id stays about 30 seconds on screens. How and from where I reduce this time so that after paging, caller id will disappear immediately. I dont know the Linux / Unix programming.
Please help, thank you in advance.

Your problem sounds to be completely different. You seem to one all trace of the call to disappear very quickly. The original issue was about making the call last longer.

I suspect this is a function of the phone’s firmware, but, if not, you need to provide the SIP debugging information and the relevant verbose CLI output from Asterisk.

I have found the problem and fixed it. I have tested with various options but failed to reduce the disconnection time after paging. Caller ID stayed for 30 seconds while I am using TOKENS or provisioning phones. I configured some phones manually (without tokens) and they were working fine. Caller ID was disappeard immediately after paging. So I open the web interfaces of both manuallly / provisioning phones and check the different and here is the solution for provisioning phones.

When you log in to the web interface of a phone, go to General tab then go to Application and here you will see “Telephony Notification URL” and IP of your PBX and under this row you will see “Incoming Call” which is ENABLED in provisioning but DISABLED in manual configuration. I just Disabled this and it is working for me now. Even if you just DISABLE AND THEN RE_ENABLE and submit, it will work and caller ID will remove immediately after paging or calling.

The above is for your information, testing and further investigating to figure out a more better result.
Thank you to you all.

Jamal Sarwar
Great Computer Systems
Tel: 847-763-0763