asterisk users - Dec 2005 - Problems with multiple outbound calls going to PSTN - Wildcard TE405P

Hello everyone,

 

I'm having an outbound calling issue with our SIP phones. When one call is
made to the PSTN another person trying to call receives a 404 error on the
SIP phone. If we call the PSTN using SIP phone A and also calling from SIP
phone B to SIP phone C everything works. The only problem we're seeing is
multiple calls going to the PSTN. Please let me know if anyone has any
suggestions or recommendations. 

 

Here are the specifications of our server.

 

1 Digium Wildcard TE405P

Asterisk 1.0.9

 

 

zapata.conf

 

; Zapata telephony interface

;

; Configuration file

 

[trunkgroups]

;

; Trunk groups are used for NFAS or GR-303 connections.

;

; Group: Defines a trunk group.

;        group => <trunkgroup>,<dchannel>[,<backup1>...]

;

;        trunkgroup  is the numerical trunk group to create

;        dchannel    is the zap channel which will have the

;                    d-channel for the trunk.

;        backup1     is an optional list of backup d-channels.

;

;trunkgroup => 1,24,48

;

; Spanmap: Associates a span with a trunk group

;        spanmap => <zapspan>,<trunkgroup>[,<logicalspan>]

;

;        zapspan     is the zap span number to associate

;        trunkgroup  is the trunkgroup (specified above) for the mapping

;        logicalspan is the logical span number within the trunk group to
use.

;                    if unspecified, no logical span number is used.

;

;spanmap => 1,1,1

;spanmap => 2,1,2

;spanmap => 3,1,3

;spanmap => 4,1,4

 

[channels]

;

; Default language

;

;language=en

 

signalling = pri_cpe

Switchtype=dms100

group=1

context=default

channel => 1-23

 

;busydetect=1

;busycount=5

;relaxdtmf=yes

;callwaiting=yes

usecallerid=yes

 

hidecallerid=no

 

callwaiting=yes

 

usecallingpres=yes

 

callreturn=yes

 

callwaitingcallerid=yes

 

threewaycalling=yes

 

transfer=yes

 

cancallforward=yes

 

echocancel=yes

 

echocancelwhenbridged=yes

 

group=1

 

callgroup=1

pickupgroup=1

 

 

immediate=no

; SOS context

;context=SOS

;usecallerid=yes

;group=1

;callerid="SOS" <XXXXXX-5000>

;channel => 14-18

 

 

; D-D context

context=D-D

usecallerid=yes

group=1

callerid="D2D" <XXXXXX5010>

;CHANNELs may be associated with account codes 4 billing

;accountcode=DD5010

channel => 1-23

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20051228/de86247e/attachment.htm

In article <20051228194014.E4B484BC8@lists.digium.com>, 
sdale@neighborhoodinternet.com says...
> I'm having an outbound calling issue with our SIP phones. When one call
is
> made to the PSTN another person trying to call receives a 404 error on the
> SIP phone. If we call the PSTN using SIP phone A and also calling from SIP
> phone B to SIP phone C everything works. The only problem we're seeing
is
> multiple calls going to the PSTN. Please let me know if anyone has any
> suggestions or recommendations. 
I haven't use PRI line, and from your explanation I guess that you need 
to define that every next calls needs to use next free line.


-- 

Tomislav Parcina
ime.prezime@email.t-com.hr