asterisk users - Jun 2006 - Asterisk x Siemens HiPath 4000

Hello all. I have installed and functioning asterisk-1.2.9.1 where I
effected one upgrade in asterisk-1.0.9, is interconnected with a PABX
Siemens HiPath 4000 in ISDN PRI with protocol QSIG, the one that is
happening he is that the calls originated for PABX Siemens and
destined to SIP phones asterisk are being without audio, nor Ring, is dumb.
They could help in this case me?
Best Regards

Josu?
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Hi,

Could you post your /etc/zaptel.conf and zapata.conf?

Also, is everything OK the other way round (i.e., from the SIP phones to the
PBX)?

Silviu

----
Hello all.
I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade
in asterisk-1.0.9, is interconnected with a PABX Siemens HiPath 4000 in ISDN PRI
with protocol QSIG, the one that is happening he is that the calls originated
for PABX Siemens and destined to SIP phones asterisk are being without audio,
nor Ring, is dumb. They could help in this case me?
Best Regards
 
Josu?
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Hello,
 
The main differences I can see:
 
- in zaptel.conf
you have span=1,0,0,ccs,hdb3, which means you ask Asterisk to serve as a timer
for the PBX - on my setup the PBX is the master clock and Asterisk is the
secondary one, so I have span=1,1,0,ccs,hdb3 (in fact, as I use CRC4 error
correction, my setup is span=1,1,0,ccs,hdb3,crc4)
 
- in zapata.conf
I have switchtype=EuroISDN. Generally speaking, try using other switchtypes.
 
Regards,
 
Silviu
________________________________

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Josu? Conti
Sent: 27 June 2006 14:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk x Siemens HiPath 4000


Silviu, thank's will be this attention. Below my configurations of
zapata.conf and zaptel.conf
#zapte.conf
span=1,0,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
loadzone=us
defaultzone=us
 
#zapata.conf

[trunkgroups]

[channels]
language=pt_BR
context=default
switchtype=qsig
pridialplan=private
prilocaldialplan=private
facilityenable = yes
signalling=pri_cpe
;rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
restrictcid=no
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes 
rxgain=0.0
txgain=0.0
group=1
callgroup=1
immediate=no
callerid=asreceived
musiconhold=default
group=1
channel=>1-15
channel=>17-31


 

Best Regards
 
Josu?


 
2006/6/27, Herchi Silviu <Silviu.Herchi@arcelor.com>: 

	Hi, 

	Could you post your /etc/zaptel.conf and zapata.conf? 

	Also, is everything OK the other way round (i.e., from the SIP phones to the
PBX)?

	Silviu 

	---- 

	
	Hello all. 
	I have installed and functioning asterisk-1.2.9.1 where I effected one upgrade
in asterisk-1.0.9 , is interconnected with a PABX Siemens HiPath 4000 in ISDN
PRI with protocol QSIG, the one that is happening he is that the calls
originated for PABX Siemens and destined to SIP phones asterisk are being
without audio, nor Ring, is dumb. They could help in this case me?

	

	Best Regards 
	  
	Josu? 


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