No rtp until i press hold and release

Asterisk Asterisk Support
1

I have a really weird problem. when I call from my personal phone to sip number the sound works well. but when I call from internal phone (extension) lets say from 100 to 200 the sound not working until I press and release the hold. when I look in the asterisk log it looks that the rtp not start before the hold although the phones bridge correctly. I google the problem and see a lot of info about the problem but none of those fixed the problem.

the point is that the rtp not even start when someone answering but when I press hold or transfer the call to another ext then the rtp start and everything works correct

btw all the phones connect to asterisk from another network (from my home network to asterisk server)

2

Please post Asterisk version, sip config, and logging.

3

asterisk version 13
sip settings (I have freepbx installed)

accept_outofcall_message=yes
auth_message_requests=no
outofcall_message_context=dpma_message_context
faxdetect=no
vmexten=97
useragent=FPBX-13.0.190.11(13.12.1)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
allow=g723
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
limitonpeers=yes
context=from-sip-external
rtpend=20000
rtpstart=10000
tcpenable=no
callevents=yes
bindport=5160
jbenable=no
notifyringing=yes
allowguest=yes
tlsbindaddr=[::]:5161
tlsclientmethod=sslv2
g726nonstandard=no
srvlookup=no
tlsenable=no
defaultexpiry=120
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
notifyhold=yes
checkmwi=10
nat=force_rport,comedia
externip=**.***.***.***
ALLOW_SIP_ANON=no
callerid=Unknown
localnet=10.0.0.0/24 ;my local net
localnet=10.251.0.0/24 ; my vlan
language=en

I add a little comments like my local net and my vlan to explain what is and of course hide the ip
which part of the log you need (its big)

4

You need to enable SIP debugging and include all requests and responses with the Call-ID for the failing call.

5

before I post all that data (I need to remove the ips. security reasons). maybe u can guide me to right point.

thks

6

We have seen this on external calls http://www.cyber-cottage.eu/?p=1625 and also on yealink handsets. http://www.cyber-cottage.eu/?p=1507 but never internal between handsets, are they on teh same subnet ? or across teh lan/vlan ?

Ian

7

try setting internal_timing = yes in asterisk.conf