We recently switched phone providers and went from a POTS line to a T1. Our FreePBX config pretty much stayed the same. Dialing out and dialing an extension works fine. However, when we call a conference call bridge and enter the con call code, numbers will get randomly dropped. I did notice that if you dial fast enough, it will usually pick up all of the numbers. Any input would be apreciated.
lack of details results in lack of responses…
What version of asterisk, FreePBX? who’s card are you using, what phones, etc. Details please otherwise we can only guess…
http://freepbx.org/forum/freepbx/installation/so-you-have-a-problem-and-want-help
FreePBX version: 2.4.1.2
using a pix of Cisco 7970 phones and Polycom SoundPoint 601’s
What card are you referring to?
Thanks.
To quote you “went from a POTS line to a T1” so are you using a t1 card? who’s? or did you really mean that you dropped POTS for a sip connection.
Being specific in your information and providing accurate details is how we can figure your situation out, otherwise we have to try and become telepathic due to my weegee board not working correctly on Friday the 13th.
Ah, sorry. We are using a Cisco 2845 with a T1 multiflex trunk voice card. Before our switch, we had 4 analog lines going into the voice gateway which would send the calls to asterisk via a SIP trunk. Now we have our voice coming over a T1 into the same Cisco device.
ok that helps.
Sounds like a misconfiguration between the 2845 and sip settings you have setup for DTMF.
Is this using a external Conference call bridge? Or are you refering to the conference rooms you have setup in FreePBX/Asterisk? (again it’s all in the details).
I’m going to assume you meant FreePBX conference rooms unless you say otherwise.
You didn’t say specifically if it is a issue with internal phone having the issue, external calls coming in or both that have dtmf issues.
How a phone works to dial a call and how it transmits DTMF while on a call are two different matters. To actually place a call DTMF is not transmitted for the sip protocol. The phone collects the digits pressed then sends a sip packet that says dial xxxxxx.
When a call is in progress and you press a key then it sends dtmf. It can be done using one of several methods based on how you have it configured. in-band, out of band, etc…
So again I’m going to assume it’s external coming in that does not work. So a T1 normally provides out of band dtmf (where POTS can only do in-band) and your configuration on the 2845 might need to be adjusted for this change. IT is also possible that you had it configured for in-band but when you switched to the T1 the 2845 side switched to out of band by default and the issue is both sides are out of sync.
Hope that give you some ideas as to where to look and why one side of the other might have changed.
The conference number is an external number. We are calling from our office to an external conference bridge. Thanks for all of your input. If there are any other details you need, I will be glad to provide them.
ok then take a look at the setup on the 2845 for dtmf settings. Something is configured wrong on how it is taking the signals from SIP and sending them out. Sip on the server was probably configured for in-band and now in the 2845 it is defaulting to the T1’s setting which would be out-of-band.
What you are describing is exactly what happens if it was a SIP trunk coming in to a conference room on the phone system and one side didn’t match the other.
Please post the output of ‘show run’ from the 2845 and the contents of the peer and user details from the associated trunk in FreePBX.
With this information I can help you.
###user details from the trunk in FreePBX###
type=user
port=5060
nat=no
insecure=very
host=xxx.xxx.xxx.110
disallow=all
context=from-pstn
canreinvite=no
allow=ulaw
###show run output from 2845#####
version 12.4
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
service sequence-numbers
!
hostname xxxxxxxxxxxxxx
!
boot-start-marker
boot system flash:c3845-adventerprisek9-mz.124-15.T6.bin
!
card type t1 0 2
! card type command needed for slot/vwic-slot 1/0
!
!
!
aaa session-id common
memory-size iomem 10
clock timezone Chicago -6
clock summer-time Chicago date Mar 9 2008 2:00 Nov 2 2008 2:00
network-clock-participate wic 2
dot11 syslog
ip cef
!
!
!
!
isdn switch-type primary-4ess
voice-card 0
no dspfarm
!
!
!
trunk group !!!
!
!
trunk group !!!
!
!
trunk group !!!fax
!
!
!
voice service voip
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
!
!
!
!
!
!
!
!
!
!
!
!
voice register global
max-dn 720
max-pool 480
!
voice register pool 1
!
!
voice translation-rule 1
!
!
voice translation-profile filter_512
!
!
controller T1 0/2/0
shutdown
framing esf
linecode b8zs
!
controller T1 0/2/1
framing esf
linecode b8zs
pri-group timeslots 1-24
trunk-group globalres timeslots 5,9
trunk-group cdi timeslots 1-4,6-8
trunk-group cdifax timeslots 10
!
ip tcp synwait-time 10
!
!
!
!
interface Null0
no ip unreachables
!
!
interface GigabitEthernet0/1
ip address xxx.xxx.xxx.2
ip nbar protocol-discovery
ip flow egress
ip route-cache flow
duplex full
speed 1000
media-type rj45
no keepalive
!
interface Serial0/2/1:23
no ip address
encapsulation hdlc
isdn switch-type primary-4ess
isdn incoming-voice voice
no cdp enable
!
!
ip default-gateway xxx.xxx.xxx.1
ip forward-protocol nd
!
voice-port 0/1/0
description Fax_line
!
voice-port 0/1/1
!
voice-port 0/2/1:23
description CDI_ISDN_Lines
!
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
!
!
!
dial-peer voice 94 pots
trunkgroup globalres
destination-pattern 8T
!
dial-peer voice 15 pots
preference 1
incoming called-number 6021
no digit-strip
direct-inward-dial
!
dial-peer voice 16 pots
preference 2
incoming called-number 6022
no digit-strip
direct-inward-dial
!
dial-peer voice 17 pots
preference 3
incoming called-number 6023
no digit-strip
direct-inward-dial
!
dial-peer voice 18 pots
preference 4
incoming called-number 6024
no digit-strip
direct-inward-dial
!
dial-peer voice 20 pots
preference 6
incoming called-number 3722
no digit-strip
direct-inward-dial
!
dial-peer voice 21 pots
preference 7
incoming called-number 3713
no digit-strip
direct-inward-dial
!
dial-peer voice 22 pots
preference 8
incoming called-number 3700
no digit-strip
direct-inward-dial
!
dial-peer voice 24 pots
incoming called-number 5952
no digit-strip
direct-inward-dial
!
dial-peer voice 80 pots
preference 2
incoming called-number 0011
no digit-strip
direct-inward-dial
!
dial-peer voice 95 pots
trunkgroup cdi
destination-pattern 9T
!
dial-peer voice 25 pots
destination-pattern 5952
port 0/1/0
!
dial-peer voice 70 pots
trunkgroup cdifax
preference 1
destination-pattern 7T
!
dial-peer voice 81 pots
preference 1
incoming called-number 6025
no digit-strip
direct-inward-dial
!
dial-peer voice 102 voip
destination-pattern 602.
progress_ind setup enable 3
session protocol sipv2
session target ipv4:xxx.xxx.xxx.110:5060
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 103 voip
destination-pattern 0011
progress_ind setup enable 3
session protocol sipv2
session target ipv4:xxx.xxx.xxx.110:5060
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 104 voip
destination-pattern 37…
progress_ind setup enable 3
session protocol sipv2
session target ipv4:xxx.xxx.xxx.110:5060
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
!
gateway
timer receive-rtp 1200
!
sip-ua
sip-server ipv4:xxx.xxx.xx.110:5060
!
!
ntp logging
ntp clock-period 17179751
ntp update-calendar
ntp server 192.168.10.1 prefer
ntp server 192.168.9.5
!
end
Also, here are the relevant sections of a config file from both a Cisco 7970 phone and a Polycom Sound Series.
###Polycom sip.cfg###
###Cisco phone config###
false
g711ulaw
101
3
avt_always
false
false
3
John Doe
2
false
15000
10
true
16384
32766
5060
184
0
dialplan.xml