[Linphone-users] connecting linphone to asterisk

[a . ahmad]

Hello 
 
I am trying to connect linphone 0.12.1 to an asterisk 0.9.1 box over a LAN 
using the console version of linphone. both boxs are using alsa on a LFS 
kernal 2.4. I am running into errors with respect to "no decoder available for 
payload 101" despite tweeking the codec settings on both sides. I have 
attached the linephonec debug output, .linphonec config and my asterisk 
sip.conf file below. 
 
A point to note is that I am able to connect to asterisk using other sip 
phones noteably sjphone however linephone is giving me some problems. 
 
having followed some of the advice in the archives i have tried to get both 
sides to use the same decoder but linephone still complains. Any other advice 
is greatly appreciated. thanks in advance. Amjad 
  
output from linphonc..... 
 
sh-2.05b$ linphonec -d 6 
INFO: no logfile, logging to stdout 
| INFO1 | <osipua.c: 65> Starting osip stack and osipua layer. 
 
| INFO1 | <udp.c: 112> Entering osipua thread. 
 
MediaStreamer-Message: Found /dev/dsp. 
MediaStreamer-Message: Found ALSA device: VIA 8235 
MediaStreamer-Message: alsa_set_params:  blocksize=512. 
| INFO1 | <osipmanager.c: 148> port already listened 
 
| INFO1 | <osipmanager.c: 148> port already listened 
 
LinphoneCore-Message: Adding new codec PCMU/8000 
LinphoneCore-Message: Adding new codec GSM/8000 
LinphoneCore-Message: Adding new codec PCMA/8000 
LinphoneCore-Message: Adding new codec speex/8000 
LinphoneCore-Message: Adding new codec speex/16000 
LinphoneCore-Message: Adding new codec 1015/8000 
Ready. 
 
Command ? c address@hidden 
Contacting  sip:address@hidden 
 
| INFO1 | <utils.c: 409> Outgoing interface to reach 192.168.10.20 is 
192.168.10.24. 
 
adding sdp body... 
| INFO2 | <transaction.c: 38> allocating transaction ressource 1 1043145758 
| INFO2 | <ict.c: 34> allocating ICT context 
Executing transaction... 
 
Command ? | INFO1 | <udp.c: 295> Sending message: 
INVITE sip:address@hidden SIP/2.0 
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK1765101061 
From: <sip:address@hidden>;tag=2495366955 
To: <sip:address@hidden> 
Call-ID: address@hidden 
CSeq: 20 INVITE 
Contact: <sip:address@hidden> 
max-forwards: 10 
user-agent: oSIP/Linphone-0.12.1 
Content-Type: application/sdp 
Content-Length:   242 
 
v=0 
o=aa 123456 654321 IN IP4 192.168.10.24 
s=A conversation 
c=IN IP4 192.168.10.24 
t=0 0 
m=audio 7078 RTP/AVP 110 115 101 
b=AS:8 
a=rtpmap:110 speex/8000/1 
a=rtpmap:115 1015/8000/1 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-11 
 
| INFO1 | <udp.c: 295> Sending message: 
INVITE sip:address@hidden SIP/2.0 
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK1765101061 
From: <sip:address@hidden>;tag=2495366955 
To: <sip:address@hidden> 
Call-ID: address@hidden 
CSeq: 20 INVITE 
Contact: <sip:address@hidden> 
max-forwards: 10 
user-agent: oSIP/Linphone-0.12.1 
Content-Type: application/sdp 
Content-Length:   242 
 
v=0 
o=aa 123456 654321 IN IP4 192.168.10.24 
s=A conversation 
c=IN IP4 192.168.10.24 
t=0 0 
m=audio 7078 RTP/AVP 110 115 101 
b=AS:8 
a=rtpmap:110 speex/8000/1 
a=rtpmap:115 1015/8000/1 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-11 
 
| INFO1 | <udp.c: 186> info: Message from 192.168.10.20:5060 
 
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE: 
SIP/2.0 100 Trying 
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK1765101061 
From: <sip:address@hidden>;tag=2495366955 
To: <sip:address@hidden>;tag=as3b81e5d4 
Call-ID: address@hidden 
CSeq: 20 INVITE 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 
Contact: <sip:address@hidden> 
Content-Length: 0 
 
 
 
| INFO1 | <ict_callbacks.c: 41> OnEvent_New_Incoming1xxResponse! 
 
| INFO1 | <udp.c: 186> info: Message from 192.168.10.20:5060 
 
| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE: 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK1765101061 
From: <sip:address@hidden>;tag=2495366955 
To: <sip:address@hidden>;tag=as3b81e5d4 
Call-ID: address@hidden 
CSeq: 20 INVITE 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 
Contact: <sip:address@hidden> 
Content-Type: application/sdp 
Content-Length: 265 
 
v=0 
o=root 26656 26656 IN IP4 192.168.10.20 
s=session 
c=IN IP4 192.168.10.20 
t=0 0 
m=audio 13906 RTP/AVP 3 0 8 101 
a=rtpmap:3 GSM/8000 
a=rtpmap:0 PCMU/8000 
a=rtpmap:8 PCMA/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-16 
a=silenceSupp:off - - - - 
 
 
| INFO1 | <ict_callbacks.c: 71> OnEvent_New_Incoming2xxResponse! 
 
| INFO1 | <ict_callbacks.c: 122> Found body application/sdp 
 
 
(process:10816): LinphoneCore-WARNING **: payload GSM is not usable or 
enabled. 
 
(process:10816): LinphoneCore-WARNING **: This remote sip phone did not answer 
properly to my sdp offer! 
 
(process:10816): LinphoneCore-WARNING **: payload PCMU is not usable or 
enabled. 
 
(process:10816): LinphoneCore-WARNING **: This remote sip phone did not answer 
properly to my sdp offer! 
 
(process:10816): LinphoneCore-WARNING **: payload PCMA is not usable or 
enabled. 
 
(process:10816): LinphoneCore-WARNING **: This remote sip phone did not answer 
properly to my sdp offer! 
Connected. 
MediaStreamer-Message: alsa_set_params:  blocksize=512. 
 
MediaStreamer-ERROR **: mediastream.c: No decoder availlable for payload 101. 
aborting... 
Aborted 
sh-2.05b$ 
 
.linphonec config settings.... 
 
[net] 
if_name=rhine 
con_type=1 
use_nat=0 
 
[sip] 
sip_port=5060 
use_registrar=0 
username=aa 
hostname=192.168.10.24 
registrar= 
passwd= 
as_proxy=0 
expires=900 
 
 
[rtp] 
audio_rtp_port=7078 
video_rtp_port=0 
audio_jitt_comp=120 
video_jitt_comp=120 
 
my sip.conf file..... 
 
[general] 
port=5060                       ; Port to bind to 
bindaddr=192.168.10.20          ; Address to bind SIP channel to 
context=default                 ; Default context for incoming calls 
 
;srvlookup = yes                ; Enable DNS SRV lookups on outbound calls 
;pedantic = yes                 ; Enable slow, pedantic checking for Pingtel 
;tos=lowdelay                   ; IP QoS parameter, either keyword or value 
 
;maxexpirey=3600                ; Max length of incoming registration we allow 
;defaultexpirey=120             ; Default length of incoming/outoing 
registration 
;notifymimetype=text/plain      ; Allow overriding of mime type in NOTIFY 
;videosupport=yes               ; Turn on support for SIP video 
 
;disallow=all                   ; Disallow all codecs 
;allow=gsm 
;allow=ulaw                     ; Allow codecs in order of preference 
;allow=ilbc 
... 
 
[aa] 
type=friend 
host=192.168.10.24 
defaultip=192.168.10.24 
;username=blah 
;secret=blah 
dtmfmode=rfc2833                ; Choices are inband, rfc2833, or info 
context=default 
callerid="aa" <1234> 
 
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