asterisk users - Aug 2006 - lost packets when bridging zap and iax

We have a machine with a TE410P in it acting as a client to route calls 
via iax2 to our central server,

caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called
                 client                  server

often the called can't hear the caller (both machines on public ip)
'iax2 show netstats" on client machine shows more and more dropped 
packets on the local side

if we use sip as the "entering point" for the calls all works well :

caller --> ( sip -> iax ) ---> ( iax -> whatever ) --> called
                 client                  server

seems something in the bridging between zap and iax screws up, but I 
don't know if it's a bug or a misconfiguration, my conf files follows, 
someone has similar experiences to share ?

/etc/asterisk# cat iax.conf

[general]
bindport=4569
bindaddr=xxx.xx.xx.xxx

disallow=all
allow=alaw

jitterbuffer=yes
forcejitterbuffer=no

tos=lowdelay
autokill=yes

language=it
notransfer=yes


/etc/asterisk# cat sip.conf

[general]

context=invalid

bindport=5060
bindaddr=xxx.xx.xx.xxx

srvlookup=no

disallow=all
allow=alaw

progressinband=no
canreinvite=no

language=it

[authentication]

[some-ip]
type=friend
context=ip
host=some-ip


/etc/asterisk# cat zapata.conf

[channels]
language=it
context=default

switchtype=euroisdn
pridialplan=national
prilocaldialplan=national
signalling=pri_cpe
immediate=no

callerid=asreceived
usecallingpres=yes

echocancel=yes
echocancelwhenbridged=no
;echotraining=yes

rxgain=0.0
txgain=0.0

group=1
callgroup=1
pickupgroup=1

group = 1
channel => 1-15
channel => 17-31

channel => 32-46
channel => 48-62

channel => 63-77
channel => 79-93

channel => 94-108
channel => 110-124


/etc/asterisk# cat /etc/zaptel.conf

span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31

span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62

span=3,1,0,ccs,hdb3
bchan=63-77
dchan=78
bchan=79-93

span=4,1,0,ccs,hdb3
bchan=94-108
dchan=109
bchan=110-124

loadzone=it
defaultzone=it

Simone Cittadini wrote:
> We have a machine with a TE410P in it acting as a client to route calls 
> via iax2 to our central server,
> 
> caller --> ( zap -> iax ) ---> ( iax -> whatever ) -->
called
>                 client                  server
> 
> often the called can't hear the caller (both machines on public ip)
> 'iax2 show netstats" on client machine shows more and more dropped
> packets on the local side
> 
> if we use sip as the "entering point" for the calls all works
well :
> 
> caller --> ( sip -> iax ) ---> ( iax -> whatever ) -->
called
>                 client                  server
> 
> seems something in the bridging between zap and iax screws up, but I 
> don't know if it's a bug or a misconfiguration, my conf files
follows,
> someone has similar experiences to share ?
> 
> /etc/asterisk# cat iax.conf
> 
> [general]
> bindport=4569
> bindaddr=xxx.xx.xx.xxx
> 
> disallow=all
> allow=alaw
> 
> jitterbuffer=yes
> forcejitterbuffer=no
> 
> tos=lowdelay
> autokill=yes
> 
> language=it
> notransfer=yes
> 
> 
> /etc/asterisk# cat sip.conf
> 
> [general]
> 
> context=invalid
> 
> bindport=5060
> bindaddr=xxx.xx.xx.xxx
> 
> srvlookup=no
> 
> disallow=all
> allow=alaw
> 
> progressinband=no
> canreinvite=no
> 
> language=it
> 
> [authentication]
> 
> [some-ip]
> type=friend
> context=ip
> host=some-ip
> 
> 
> /etc/asterisk# cat zapata.conf
> 
> [channels]
> language=it
> context=default
> 
> switchtype=euroisdn
> pridialplan=national
> prilocaldialplan=national
> signalling=pri_cpe
> immediate=no
> 
> callerid=asreceived
> usecallingpres=yes
> 
> echocancel=yes
> echocancelwhenbridged=no
> ;echotraining=yes
> 
> rxgain=0.0
> txgain=0.0
> 
> group=1
> callgroup=1
> pickupgroup=1
> 
> group = 1
> channel => 1-15
> channel => 17-31
> 
> channel => 32-46
> channel => 48-62
> 
> channel => 63-77
> channel => 79-93
> 
> channel => 94-108
> channel => 110-124
> 
> 
> /etc/asterisk# cat /etc/zaptel.conf
> 
> span=1,1,0,ccs,hdb3
> bchan=1-15
> dchan=16
> bchan=17-31
> 
> span=2,1,0,ccs,hdb3,crc4
> bchan=32-46
> dchan=47
> bchan=48-62
> 
> span=3,1,0,ccs,hdb3
> bchan=63-77
> dchan=78
> bchan=79-93
> 
> span=4,1,0,ccs,hdb3
> bchan=94-108
> dchan=109
> bchan=110-124
> 
> loadzone=it
> defaultzone=it
Only "one" of the above four "span" entries should have a
"1" as the
second digit. That second digit is telling the digium card which span to 
sync its on-board clock to. Pick the span that goes to a central office 
and specify it as "1" and all other spans should be either
"0" or
increasing numerical digits (eg, 2,3,4).

If none of the spans go to a central office, its still a problem.

You'll have to reload the drivers for the change to take effect.

Hi all,

  I have got same problem - bridging between IAX and IAX goes fine without lost
packets. ZAP to IAX - one lag show lost packets. Any ideas and/or solutions?

Best regards,
Dmitry

-----Oprindelig meddelelse-----
Fra: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Simone Cittadini
Sendt: 28. august 2006 13:58
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [asterisk-users] lost packets when bridging zap and iax

We have a machine with a TE410P in it acting as a client to route calls 
via iax2 to our central server,

caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called
                 client                  server

often the called can't hear the caller (both machines on public ip)
'iax2 show netstats" on client machine shows more and more dropped 
packets on the local side

if we use sip as the "entering point" for the calls all works well :

caller --> ( sip -> iax ) ---> ( iax -> whatever ) --> called
                 client                  server

seems something in the bridging between zap and iax screws up, but I 
don't know if it's a bug or a misconfiguration, my conf files follows, 
someone has similar experiences to share ?

/etc/asterisk# cat iax.conf

[general]
bindport=4569
bindaddr=xxx.xx.xx.xxx

disallow=all
allow=alaw

jitterbuffer=yes
forcejitterbuffer=no

tos=lowdelay
autokill=yes

language=it
notransfer=yes


/etc/asterisk# cat sip.conf

[general]

context=invalid

bindport=5060
bindaddr=xxx.xx.xx.xxx

srvlookup=no

disallow=all
allow=alaw

progressinband=no
canreinvite=no

language=it

[authentication]

[some-ip]
type=friend
context=ip
host=some-ip


/etc/asterisk# cat zapata.conf

[channels]
language=it
context=default

switchtype=euroisdn
pridialplan=national
prilocaldialplan=national
signalling=pri_cpe
immediate=no

callerid=asreceived
usecallingpres=yes

echocancel=yes
echocancelwhenbridged=no
;echotraining=yes

rxgain=0.0
txgain=0.0

group=1
callgroup=1
pickupgroup=1

group = 1
channel => 1-15
channel => 17-31

channel => 32-46
channel => 48-62

channel => 63-77
channel => 79-93

channel => 94-108
channel => 110-124


/etc/asterisk# cat /etc/zaptel.conf

span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31

span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62

span=3,1,0,ccs,hdb3
bchan=63-77
dchan=78
bchan=79-93

span=4,1,0,ccs,hdb3
bchan=94-108
dchan=109
bchan=110-124

loadzone=it
defaultzone=it
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I have an Asterisk servers (recent SVN version 1.2) and two Sipura ATAs
(one 2000 and one 1001).

I have Three-way Conf Serv and Three-way Call Serv enabled on both ATAs.

When I make a SIP call from phone 1 to phone 2 on my Asterisk box, it
works fine, then when I press the hookflash on phone 1, the caller on
phone 2 is placed on hold (Asterisk MoH plays). I get the "second dial
tone" on the Sipura and dial another extension (phone 3) from phone 1
and it rings and I am connected. Now, if I press hookflash again on
phone 1, it switches the connection to phone 2, but phone 3 is placed on
hold rather than getting conferenced in.

Anyone got any idea why the three-way call might not be working in these
circumstances?

The message type for setting of Hookflash (none, AVT, INFO) seems to
make no difference. I normally set it to "none" as it should be
handled
locally rather than get sent to Asterisk.

Thanks
Peter


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