Mp-104 FXO no outgoing calls / Caller ID issue?

NickDC · December 27, 2006, 8:03pm

Anyone using this FXO? It is probably a simple configuration but I can’t get outgoing calls to work, incoming are setup through autodialing to go to an extension but when I call out it has no idea what todo with the phone number, it assumes it will know an endpoint for it but it’s a PSTN number, so it should send it out the PSTN line. Bell here wants to use the 10 digit dialing even for local calls, would be cool to get the FXO to pretend the area code if its not included to the local one.

here’s my sip debug from asterisk

asterisk1*CLI>
<-- SIP read from 192.168.1.211:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK536ac3bb;rport
From: “1xxx5911526” sip:1xxx5911526@192.168.1.136;tag=as7c81df4f
To: sip:8662018@192.168.1.211;tag=1c74683406
Call-ID: 64ae18a526515b362ca29c526655a1db@192.168.1.136
CSeq: 102 INVITE
Contact: sip:192.168.1.211
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-104 FXO/v.4.60A.023.003
Content-Length: 0

i check a syslog output on the FXO and it reports that it doesn’t know the endpoint for that number… ?

also, calls coming in , even with caller ID detection always show the number of the endpoint setup in the FXO on my other phones. since I didn’t put any number (only has one line) it is set to 1000, so it’s showing 1000 on every other phone…

any help would be great
Nick

mbruni · December 28, 2006, 11:47am

Hello Nick,
I used this gw with Asterisk in the past and it worked well.
To route a call coming from ip to a pstn channel you have to set the “IP to Hunt Group Routing Table”, you can find this table in this section of the web interface: Protocol Management -> Routing Table -> IP to Hunt Group Routing.
To manipulate a number before using a pstn channel you have to set the table “Destination Phone Number Manipulation Table for IP->Tel Calls”, you can find it under Protocol Management -> Manipulation Table.
To obtain the caller id set the caller id protocol used in the section Advanced Configuration -> Channel Settings -> Fax / Modem / CID Settings and then “allow” the caller id through the section Protocol Management -> Endpoint Settings.

Hope this helps.

Regards.

Marco Bruni

NickDC · December 28, 2006, 5:19pm

Hi Marco,

I tried some of these settings but now It won’t answer any incoming calls either, and outgoing still don’t work

perhaps you can help with my config , this is my BOARD.ini file:

;**************
;** Ini File **
;**************

;Board: MP-104 FXO
;Serial Number: 405916
;Slot Number: 1
;Software Version: 4.60A.023.003
;Board IP Address: 192.168.1.211
;Board Subnet Mask: 255.255.255.0
;Board Default Gateway: 192.168.1.1
;Ram size: 16M Flash size: 4M
;Num DSPs: 1 Num DSP channels: 4
;Profile: NONE
;------------------------------

[SYSTEM Params]

DNSPriServerIP = 192.168.1.1
SyslogServerIP = 192.168.1.122
EnableSyslog = 1
VXMLFIleName = ‘’

[BSP Params]

LocalOAMIPAddress = 192.168.1.211

[ATM Params]

[Analog Params]

FXOLoopCharacteristicsFilename = 'MP1xx12-1-16khz-fxo.dat’
CallProgressTonesFilename = ‘usa_tones_11.dat’

[ControlProtocols Params]

[MGCP Params]

[MEGACO Params]

[SS7 Params]

[Voice Engine Params]

IdlePCMPattern = 255

[WEB Params]

LogoWidth = ‘339’

[SIP Params]

MAXDIGITS = 32
ALWAYSUSEROUTETABLE = 1
ISPROXYUSED = 1
AUTHENTICATIONMODE = 1
CHANNELSELECTMODE = 1
PROXYNAME = '192.168.1.136’
SIPGATEWAYNAME = '192.168.1.211’
USERNAME = 'Audiotest’
PASSWORD = '220’
ALWAYSSENDTOPROXY = 1
SENDINVITETOPROXY = 1
GWREGISTRATIONNAME = '192.168.1.211’
WAITFORDIALTIME = 350
CODERNAME = g711Ulaw64k,20
CODERNAME = g711Alaw64k,20
CODERNAME = g729,20
CODERNAME = g7231,30
PREFIX = ,192.168.1.136,,0
NUMBERMAPIP2TEL = ,0,$$,$$,$$,$$,,$$,*
PSTNPREFIX = ,0,,*,0
TARGETOFCHANNEL0 = 200,1
TRUNKGROUP = 1-1,0
PROXYIP = 192.168.1.136
ENABLECALLERID_0 = 1
ENABLECALLERID_1 = 1
ENABLECALLERID_2 = 1
ENABLECALLERID_3 = 1

[VXML Params]

[Audio Staging Params]

[PSTN-SDH Params]

my trunk in * is

[Audiotest]
disallow=all
host=192.168.1.211
secret=220
type=friend
username=Audiotest

thanks
Nick

SuperB · December 28, 2006, 10:23pm

While I have only configured this gateway for use with sipx, I am able to dial out, by configuring just an entry in End Point Numbers (to create the trunk group) and making sure the dialing mode is set to 1 under the fxo settings.

I did not have to configure the huntgroup or manipulation tables.

Incoming is just a matter of defining the SIP number to be automatically dialed when ringing is detected.

SuperB · December 28, 2006, 10:37pm

Also on your incoming caller id; Detect Caller ID for Tel should be enabled for each port with a connected line and the fxo setting “Rings before Detecting Caller ID” should be set to 1.

NickDC · December 29, 2006, 4:51am

not sure what you mean, an entry in the endpoint numbers to create a trunk group? where? now i’m not able to receive incoming calls anymore, i check the status when the phone rings and it detects nothing… i will try to reset it to factory default and play with it again. does my asterisk trunk look ok though? does it have to authenticate with the FXO?

mbruni · December 29, 2006, 9:47am

Every endpoint number should be part of a trunk group, in this way you can select a trunk group and use the first free channel of that group to dial out.
The hunt group of a channel is pecified in the column “Hunt Group ID” of the the table found under “Protocol Management -> Endpoint Phone Numbers”; also check how a channel is selected for dialing out, “Protocol Management -> Hunt Group Settings”.
In the * trunk comment out “disallow=all”, or after it allow at least one audio codec.
The mp gw doesn’t work as a proxy so * can’t authenticate with it, you may register the mp gw to * using * as its registrar and proxy server and this should be the preferred configuration where applicable.

Regards.

Marco Bruni

NickDC · January 6, 2007, 6:26pm

Hello, I managed to get this to work. the only thing I cannot get to work at the moment is the caller ID name resolution. I’m not sure if this FXO supports the Name of the Caller ID when it comes in but it doesn’t get passed to Asterisk, only the number. Would there be a way to make it resolve the name? or I’d have to setup some sort of caller ID lookup system on asterisk for all external numbers?

NickDC · January 6, 2007, 10:28pm

Also, it takes 4 rings before it actually sends the call to asterisk and rings my extensions… thats pretty slow. it almost hits the voicemail on my PSTN line.

I have the automatic dialing setup and if i watch the syslog, it waits 1 ring for caller ID detection, then it goes right away and starts constructing the call, then 2 rings later it goes through and starts ringing. I’d like to get an IVR setup to pick up the call so it never hits voicemail but it would be nice if it could answer it sooner then 4 rings. How do I get an IVR to have an extension to use in the automatic dialing?

mbruni · January 8, 2007, 9:24am

I think you can only pass the caller id number, from pstn to *, so if you want to have the caller id name you have to do an external lookup.

Regards.

Marco Bruni