Strange issue

Asterisk Asterisk Support
1

Hi everyone!

I have this strange issue going on and I’d be very grateful if someone could give me any kind of help. So in advance, thanks a lot :wink:

The config is the following :

CentOs 4.4
Asterisk 1.2.24
FreePBX 2.3.1
AVM C2 with c2-suse8.1-03.09.1
chan_capi 1.0.2

some Cisco 7905 phones
and for the moment testing a Linksys SPA921
in the same LAN with asterisk

When calling from the Cisco 7905 to either landline or mobile, everything is fine. Being called is also ok.
On the other hand, when calling from the Linksys phone, everything works fine except
when calling a number being part of a PBX, then I only have an one-way audio communication. I can hear the called part but they can’t hear me.

So, I ran into the logs and these are the errors that appear:

(NCCI=0x10102) (error=0x1103 The message could not be accepted because of a queue full condition !! The error code does not imply that CAPI cannot receive messages directed to another controller, PLCI or NCCI)
Nov 20 17:16:09 ERROR[9437] chan_capi_utils.c: CAPI error sending DATA_B3_REQ ID=002 #0x5966 LEN=0022
Controller/PLCI/NCCI = 0x10102
Data32 = 0x896b60c
DataLength = 0xa0
DataHandle = 0x43c9
Flags = 0x0
Data64 = 0x0
(NCCI=0x10102) (error=0x1103 The message could not be accepted because of a queue full condition !! The error code does not imply that CAPI cannot receive messages directed to another controller, PLCI or NCCI)
Nov 20 17:16:09 DEBUG[9437] channel.c: Didn’t get a frame from channel: SIP/612-08a16b90
Nov 20 17:16:09 DEBUG[9437] channel.c: Bridge stops bridging channels SIP/612-08a16b90 and CAPI/ISDN2#02/071600123-7

I took a walk on google where someone suggested to change the following parameter in the chan_capi.h
#define CAPI_MAX_B3_BLOCK_SIZE 160 -> 320

then compile again and reload the module in asterisk

I don’t know if I got lucky, but the first time I tested it, it worked. Unfortunately, this was the only and unique one :frowning:

I then thought that it might be a time/synchro issue as time was not setup on the phone. Still not lucky :frowning:

I finally tested all codecs but nothing. The config between the SPA and the 7905 are basically the same.

I’m stuck. So, any ideas would be highly appreciated :wink:

Many thx,
Elena

2

Hi
i think it is Nating issue
what is your user data in sip.conf plz mention nat=yes & qualify=yes
& check

Amit

3

Hi,

This is the sip conf for the specific user:

[612]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=612@default
host=dynamic
dtmfmode=rfc2833
disallow=
dial=SIP/612
context=from-internal
canreinvite=no
callgroup=
callerid=device <612>
allow=
accountcode=

in the sip_nat.conf (as we use it for another device connected from the outside):

nat=yes
externip=
localnet=
externrefresh=120

the device is in the same lan with the asterisk server

Elena

4

HI
plz mention
username=6024 ;-------> means ur user extension
Also my suggestiion remove caller id parameters & set caller id for out going calls in extensuion.conf dail plan in out bound call context b4 Dial()
Set(CALLERID(num)=646xxxxxx) ur 10 or 11 digit number

plz mention only username & all ther parameters for type- firned are ok

Amit’

5

Hi

I’m not sure I undestand what u’re asking for. Could u plz be more precise?

Many thx,
Elena

6

hi
plz mention username=612
in sip.conf type=friend for ur each users

also dont mention caller id in sip.conf mention it at out dail context in extension.conf
i think your not mention username =612
so mention it under[612] same in each user

Amit

7

Hi

users’ config is in sip_additional.conf and for all type friend

Elena