Incoming Calls Problem with

Asterisk Asterisk Support
1

Dear all,
im using asterisk 1.4.2 with mISDN and i have a problem with the incoming calls in the sip channel. Specificaly, the phone is ringing to any sip device i choose in extensions.conf but when i pick up the phone the communication is one way. The caller can hear me, but the sip device cant hear the caller. For example:

*CLI> P[ 0] maxnum:3 – Executing [s@from_isdn:1] Dial(“mISDN/1-u0”, “SIP/bedroom|15”) in new stack
– Called bedroom
– SIP/bedroom-095bd970 is ringing
– Call on SIP/bedroom-095bd970 left from hold
– SIP/bedroom-095bd970 answered mISDN/1-u0
[Apr 6 12:13:18] WARNING[4832]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw) 8 read/write = 0x8 (alaw) 8/0x8 (alaw) 8

sip.conf
[general]
static=yes
srvlookup=yes
disallow=all
allow=alaw
allow=gsm

[bedroom]
type=friend
callerid=“Bedroom” <603>
username=bedroom
secret=12345
;dtmfmode=rfc2833
dtmfmode=info
qualify=yes
nat=never
host=dynamic
canreinvite=no
disallow=all
;allow=slin
allow=alaw
;allow=ulaw
allow=gsm
context=birdome

misdn.conf

[general]
debug=0
bridging=no

[default]
context=misdn
language=en
senddtmf=yes
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
method=standard
dialplan=0
localdialplan=0
cpndialplan=0

[outgoing]
ports=1
context=from_isdn
msns=*
callgroup=1
pickupgroup=1
always_immediate=yes

extensions.conf

[birdome]
include => eswterika
include => to_isdn
include => from_isdn

[eswterika]

exten => 600,1,Dial(SIP/biris,15)
exten => 600,2,Hangup

exten => 601,1,Dial(SIP/joint,15)
exten => 601,2,Hangup

exten => 602,1,Dial(SIP/newage,15)
exten => 602,2,Hangup

exten => 603,1,Dial(SIP/bedroom,15)
exten => 603,2,Hangup

exten => biris,1,goto(600,1)
exten => joint,1,goto(601,1)
exten => newage,1,goto(602,1)
exten => bedroom,1,goto(603,1)

[to_isdn]
exten => _9.,1,Dial(misdn/1/${EXTEN:1},60,th)

[from_isdn]
exten => s,1,Dial(SIP/bedroom,15)
exten => s,2,Hangup

Any help will be appreciated,

BiRiS

2

You haven’t indicated if you have looked at voip-info.org/wiki/view/One-way+Audio.

3

Comment the allow=gsm lines and allow only the alaw codec and try again. Please give me some feedback :smile:

4

i did the comments but still The same:

*CLI> – Saved useragent “SJphone/1.60.289a (SJ Labs)” for peer biris
P[ 0] maxnum:3 – Executing [s@from_isdn:1] Answer(“mISDN/1-u0”, " ") in new stack
– Executing [s@from_isdn:2] Dial(“mISDN/1-u0”, “SIP/bedroom|15”) in new stack
– Called bedroom
– SIP/bedroom-09808ce0 is ringing
[Apr 9 19:48:45] WARNING[2236]: channel.c:2450 ast_indicate_data: Unable to handle indication 3 for ‘mISDN/1-u0’
– Call on SIP/bedroom-09808ce0 left from hold
– SIP/bedroom-09808ce0 answered mISDN/1-u0
[Apr 9 19:48:46] WARNING[2236]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Apr 9 19:48:46] WARNING[2236]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Apr 9 19:48:46] WARNING[2236]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Apr 9 19:48:47] WARNING[2236]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Apr 9 19:48:47] WARNING[2236]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Apr 9 19:48:47] WARNING[2236]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Apr 9 19:48:47] WARNING[2236]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Apr 9 19:48:47] WARNING[2236]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Apr 9 19:48:47] WARNING[2236]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
[Apr 9 19:48:47] WARNING[2236]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)
== Spawn extension (from_isdn, s, 2) exited non-zero on ‘mISDN/1-u0’

5

I cant visit the url, but i did this:
in indications.conf i choose my default language (gr) and in misdn.conf i choose language=gr. Still the same :
[Apr 9 20:12:19] WARNING[2470]: chan_sip.c:3486 sip_write: Asked to transmit frame type 64, while native formats is 0x8 (alaw)(8) read/write = 0x8 (alaw)(8)/0x8 (alaw)(8)

6

What happens if you only allow the gsm codec?

7

I cant visit the url, but i did this:
[/quote]

Try voip-info.org/wiki/view/One-way+Audio

This forum messes up the URL with the trailing . :cry:

8

Same think, i mean the message disappeared but the problem persists. One way Audio. If i choose two sip channels everything is ok.
The problem appears only from misdn (external call) to sip.
Any suggestions please?

9

Googling about this I saw a post (old one) from Mark Spencer about a bug on the code. Are you using the latest versions of the misdn drivers? It’s something about the transcoding of audio between the two channels. Try changing the codec allowed to alaw.

10

when i change the codec allow=alaw the same message, if i allow any other codec the message disappeared but the same problem.

11

I have the same problem, but only in 1.4.2, in other versions all works fine. Anyone knows how to fix this?

12

i currently have the same problem, with incomming audio via isdn
the interesting thing is that sometimes it works fine.

i read somewhere that switching kernel back to 2.6.16 would solve this - I’ll try this in the evening.

do you guys get the extension dialed by the caller into the $exten variable?
misdn debug output displays it in the DID field, but the dialplan only gets triggered with the s extension. Exchanging the s extension to a catch all extension (_X.) the dialplan resutls in immediate hangup with the message “extension can never match”

my config: ast 1.4.2; misdn 1.1.2; kernel 2.6.18; debian

13

The same symptom may have very different causes. Better start your own thread and publish relavant config info.