[Home]

Summary:ASTERISK-13668: one way voice on incoming isdn calls
Reporter:fenareta (fenareta)Labels:
Date Opened:2009-02-28 19:58:46.000-0600Date Closed:2011-06-07 14:03:13
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_misdn
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) isdn_to_sip_trace.txt
( 1) rdsi_sip.pcap
Description:When i call from sip_phone to isdn phone all works fine
When i make calls betwen sip_phones everything its ok.
But whe i call from Isdn_phone to sip_phone the sip phone can´t hear the other side.

I have tried with sip and iax protocolos and the same result.
Asterisk dont sent data voice to clients(SIP o IAX)

****** ADDITIONAL INFORMATION ******

RTP debug only show "got packets.."

I have a Centos  2.6.18...
I try with asterisk 1.4.21.2,1.4.23.1,1.4.17.2 and the same result
mISDN 1_1_7_2
mISDNuser_1_1_7_2
zaptel 1.4.12.1
libpri latest

Extensions.conf

[general]
static=yes
writeprotect=no
;autofallthrough=no
clearglobalvars=no
;priorityjumping=yes
;userscontext=default
;#include "filename.conf"

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest                                   ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/G2                                    ; Trunk interface
TRUNKMSD=1                                      ; MSD digits to strip (usually 1
or 0)
;TRUNK=IAX2/user:pass@provider

;ignorepat => 9

[phones]
include => from_misdn
include => to_misdn
include => internas_sip

[to_misdn]
exten => _91626XXXX,1,Dial(misdn/4/${EXTEN})

[from_misdn]
exten => s,1,NoOP(estado ${EXTEN})
exten => s,n,NoOp("DNID = " ${DNID})
exten => s,n,NoOp("CALLERID = " ${CALLERID(all)})
exten => s,n,NoOp("EXTEN = " ${EXTEN})
exten => s,n,NoOp("RDNIS = " ${RDNIS})
exten => s,n,Dial(SIP/10001)

[internas_sip]
exten => _1000X,1,Dial(SIP/${EXTEN})

[default]

misdn.conf

login as: root
root@<SERVER>'s password:
Last login: Sat Feb 28 22:44:49 2009 from <REMOTE_IP>
[root@localhost ~]#
[root@localhost ~]#
[root@localhost ~]# more /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
;autofallthrough=no
clearglobalvars=no
;priorityjumping=yes
;userscontext=default
;#include "filename.conf"

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest                                   ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/G2                                    ; Trunk interface
TRUNKMSD=1                                      ; MSD digits to strip (usually 1
or 0)
;TRUNK=IAX2/user:pass@provider

;ignorepat => 9

[phones]
include => from_misdn
include => to_misdn
include => internas_sip

[to_misdn]
exten => _91626XXXX,1,Dial(misdn/4/${EXTEN})

[from_misdn]
exten => s,1,NoOP(estado ${EXTEN})
exten => s,n,NoOp("DNID = " ${DNID})
exten => s,n,NoOp("CALLERID = " ${CALLERID(all)})
exten => s,n,NoOp("EXTEN = " ${EXTEN})
exten => s,n,NoOp("RDNIS = " ${RDNIS})
exten => s,n,Dial(SIP/10001)

[internas_sip]
exten => _1000X,1,Dial(SIP/${EXTEN})

[default]

misdn.conf

[general]
misdn_init=/etc/misdn-init.conf
debug=4
ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
ntkeepcalls=no
tracefile=/var/log/asterisk/misdn.log
bridging=no
;l1watcher_timeout=0
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh

[default]
context=misdn
language=en
musicclass=default
senddtmf=yes
far_alerting=no
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=yes
pmp_l1_check=no
reject_cause=16
need_more_infos=no
nttimeout=no
method=standard
overlapdial=yes
dialplan=0
localdialplan=0
cpndialplan=0
early_bconnect=yes
incoming_early_audio=no
always_immediate=yes
nodialtone=no
;immediate=no
;hold_allowed=yes
;callgroup=1
;pickupgroup=1
presentation=-1
screen=-1
;echocancel=no
;echotraining=no
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
max_incoming=-1
max_outgoing=-1

[from_misdn]
;immediate=yes
ports=4
context=from_misdn
msns=*




Comments:By: Leif Madsen (lmadsen) 2009-06-29 14:36:49

Removed IP addresses from description.
By: Russell Bryant (russell) 2009-10-07 08:41:35

Is this still a problem for you?  Have you tried the latest version of Asterisk?

So many people use chan_misdn all the time, I'm not sure how we would reproduce this...
By: Leif Madsen (lmadsen) 2009-10-26 10:20:38

Closed due to lack of feedback. If the reporter is able to provide the requested information, please reopen this issue. Thanks!