Asterisk 1.6 and ISDN

[suPerb0b]

Hey guys

I'm doing something wrong and need a nudge in the right direction.
It Asterisk 1.6 installed from the Ubuntu (lucid) repo. The card is a Digium Wildcard B410P. I think I installed everything I needed to (went according to the card's manual). I can phone sip to sip (or iax) and all are ok, but phoning from sip to cell/landline over the card only have audio working properly on the cell/landline side. On the sip side I hear nothing over the headset, although voice does transmit. Running dahdi_monitor also confirms tx, but not rx.

Here's the bits of my configs:

/etc/dahdi/system.conf:

# Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" (MASTER) AMI/CCS 
span=1,1,0,ccs,ami
# termtype: te
bchan=1-2
hardhdlc=3
echocanceller=mg2,1-2

# Span 2: B4/0/2 "B4XXP (PCI) Card 0 Span 2" RED
span=2,2,0,ccs,ami
# termtype: te
bchan=4-5
hardhdlc=6
echocanceller=mg2,4-5

# Span 3: B4/0/3 "B4XXP (PCI) Card 0 Span 3" RED
span=3,3,0,ccs,ami
# termtype: te
bchan=7-8
hardhdlc=9
echocanceller=mg2,7-8

# Span 4: B4/0/4 "B4XXP (PCI) Card 0 Span 4" RED
span=4,4,0,ccs,ami
# termtype: te
bchan=10-11
hardhdlc=12
echocanceller=mg2,10-11

# Global data

loadzone        = za
defaultzone     = za

chan_dahdi.conf:

[trunkgroups]

[channels]
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
internationalprefix = 00
nationalprefix = 0
signalling=bri_cpe_ptmp
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
channel => 1,2,4,5,7,8,10,11
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
dahdichan=1-2
callgroup=1
pickupgroup=1
callerid = asreceived
callprogress=yes
progzone=za
tonezone=-1

Let me know if I need to post more info. Any help will be appreciated.

 

[morkhans]

Did you purchase a HW echo canceller with your B410P? If so you don't need the echocanceller=mg2 settings.

In chan_dahdi.conf in [channels] you seem to be missing things like group and context (e.g group=1 and context=from-pstn). Also what codec are you using on your sip device? I would suggest ulaw or alaw. Not sure if you need the dahdichan=1-2 flags.

 

[suPerb0b]

Thanks morkhans for your suggestions.
I actually do have a context in chan_dahdi.conf (context=incoming) and group=1. I just failed to copy/paste it here.
I tried your suggestion about the echocancel, but unfortunately that did not change anything.
The sip phone was using alaw, but changing to ulaw didn't help.

I'm really at a loss... first time doing isdn :\

 

[morkhans]

So did you decide on a context of incoming or does the distro you installed create dialplan code with that context? I would change that to from-pstn (which is generally the standard). What versions of asterisk and dahdi did you install?

 

[suPerb0b]

This probably makes no sense, but I changed all the context from incoming to from-pstn and it works now. (Yes, I did change the context to incoming when I started configuring)

Thanks a mil!

 

[morkhans]

This probably makes no sense, but I changed all the context from incoming to from-pstn and it works now. (Yes, I did change the context to incoming when I started configuring)

Thanks a mil!

It does make sense because contexts in Asterisk dial-plan have a lot to do with what an extension or trunk can or can't do. Telco trunks are generally in from-pstn and SIP extensions are in from-internal unless of course you have specific requirements and code your own contexts.