[Linphone-users] Fwd: connecting linphone to asterisk

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[Linphone-users] Fwd: connecting linphone to asterisk

From:	a . ahmad
Subject:	[Linphone-users] Fwd: connecting linphone to asterisk
Date:	Fri, 9 Jul 2004 10:25:00 +0100
User-agent:	PIPEX NetMail (IMP3.1)
 
 
----- Forwarded message from address@hidden ----- 
    Date: Fri,  2 Jul 2004 13:00:21 +0100 
    From: address@hidden 
Reply-To: address@hidden 
 Subject: connecting linphone to asterisk 
      To: address@hidden 
 
Hello   
    
 I am trying to connect linphone 0.12.2 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 "Error sending rtp 
packet: Bad file descriptor" despite tweeking the codec settings on both 
sides. I have attached the linephonec debug output, .linphonec config and my 
asterisk sip.conf file below. Can someone provide any advice as to the correct 
settings to the dtmfmode in sip.conf for linphone as I am not entirely sure of 
the correct settings 
    
 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.   
 
I can even connect from the linphone console to asterisk and hear the default 
message however the following output is given: 
 
  | INFO1 | <ict_callbacks.c: 30> Transaction 1 killed. 
 
    
 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 .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 
 
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 3387977064 
| 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=z9hG4bK3291790997 
From: <sip:address@hidden>;tag=3488500359 
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=z9hG4bK3291790997 
From: <sip:address@hidden>;tag=3488500359 
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=z9hG4bK3291790997 
From: <sip:address@hidden>;tag=3488500359 
To: <sip:address@hidden>;tag=as123b40ae 
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=z9hG4bK3291790997 
From: <sip:address@hidden>;tag=3488500359 
To: <sip:address@hidden>;tag=as123b40ae 
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: 233 
 
v=0 
o=root 6546 6546 IN IP4 192.168.10.20 
s=session 
c=IN IP4 192.168.10.20 
t=0 0 
m=audio 15206 RTP/AVP 0 8 97 3 
a=rtpmap:0 PCMU/8000 
a=rtpmap:8 PCMA/8000 
a=rtpmap:97 iLBC/8000 
a=rtpmap:3 GSM/8000 
a=silenceSupp:off - - - - 
 
 
| INFO1 | <ict_callbacks.c: 71> OnEvent_New_Incoming2xxResponse! 
 
| INFO1 | <ict_callbacks.c: 122> Found body application/sdp 
 
 
(process:4860): LinphoneCore-WARNING **: payload PCMU is not usable or 
enabled. 
 
(process:4860): LinphoneCore-WARNING **: This remote sip phone did not answer 
properly to my sdp offer! 
 
(process:4860): LinphoneCore-WARNING **: payload PCMA is not usable or 
enabled. 
 
(process:4860): LinphoneCore-WARNING **: This remote sip phone did not answer 
properly to my sdp offer! 
 
(process:4860): LinphoneCore-WARNING **: This remote sip phone did not answer 
properly to my sdp offer! 
 
(process:4860): LinphoneCore-WARNING **: payload GSM is not usable or enabled. 
 
(process:4860): LinphoneCore-WARNING **: This remote sip phone did not answer 
properly to my sdp offer! 
Connected. 
MediaStreamer-Message: alsa_set_params:  blocksize=512. 
 
(process:4860): oRTP-WARNING **: Fail to bind rtp socket to port 0: Permission 
denied. 
MediaStreamer-Message: ms_filter_add_link: OssRead,0 -> MULAWEncoder,0 
MediaStreamer-Message: ms_filter_add_link: MULAWEncoder,0 -> RTPSend,0 
MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 -> MULAWDecoder,0 
MediaStreamer-Message: ms_filter_add_link: MULAWDecoder,0 -> OssWrite,0 
MediaStreamer-Message: Opening sound card in capture mode with 
stereo=0,rate=8000,bits=16 
MediaStreamer-Message: alsa_set_params:  blocksize=512. 
MediaStreamer-Message: Opening sound card in playback mode with 
stereo=0,rate=8000,bits=16 
MediaStreamer-Message: alsa_set_params:  blocksize=512. 
| INFO1 | <udp.c: 295> Sending message: 
ACK sip:address@hidden SIP/2.0 
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK1498941086 
From: <sip:address@hidden>;tag=3488500359 
To: <sip:address@hidden>;tag=as123b40ae 
Call-ID: address@hidden 
CSeq: 20 ACK 
max-forwards: 10 
user-agent: oSIP/Linphone-0.12.1 
Content-Length: 0 
 
 
| INFO1 | <ict_callbacks.c: 30> Transaction 1 killed. 
 
| INFO2 | <transaction.c: 191> free transaction ressource 1 3387977064 
| INFO2 | <ict.c: 117> free ict ressource 
 
(process:4881): oRTP-WARNING **: Error sending rtp packet: Bad file 
descriptor. 
 
(process:4881): oRTP-WARNING **: Error sending rtp packet: Bad file 
descriptor. 
 
(process:4881): oRTP-WARNING **: Error sending rtp packet: Bad file 
descriptor. 
 
..... 
  
*********************     asterisk debug output ********************** 
 
Sip read: 
INVITE sip:address@hidden SIP/2.0 
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK3291790997 
From: <sip:address@hidden>;tag=3488500359 
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:8LI> 
a=rtpmap:110 speex/8000/1 
a=rtpmap:115 1015/8000/1 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-11 
 
11 headers, 11 lines 
Using latest request as basis request 
Sending to 192.168.10.24 : 5060 (non-NAT) 
Found audio format UNKN 
Found audio format UNKN 
Found audio format UNKN 
Found description format speex 
Found description format 1015 
Found description format telephone-event 
Capabilities: us - 525326, them - 512/0, combined - 0 
Non-codec capabilities: us - 1, them - 1, combined - 1 
Jul  8 19:43:37 WARNING[196621]: chan_sip.c:2116 process_sdp: No compatible 
codec 
s! 
isis*CLI> 
 
Sip read: 
INVITE sip:address@hidden SIP/2.0 
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK3291790997 
From: <sip:address@hidden>;tag=3488500359 
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:8LI> 
a=rtpmap:110 speex/8000/1 
a=rtpmap:115 1015/8000/1 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-11 
 
11 headers, 11 lines 
Ignoring this request 
Looking for 1000 in default 
list_route: hop: <sip:address@hidden> 
Transmitting (no NAT): 
SIP/2.0 100 Trying 
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK3291790997 
From: <sip:address@hidden>;tag=3488500359 
To: <sip:address@hidden>;tag=as123b40ae 
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 
 
 
 to 192.168.10.24:5060 
We're at 192.168.10.20 port 15206 
Answering with preferred capability 4 
Answering with preferred capability 8 
Answering with preferred capability 1024 
Answering with preferred capability 2 
Reliably Transmitting (no NAT): 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK3291790997 
From: <sip:address@hidden>;tag=3488500359 
To: <sip:address@hidden>;tag=as123b40ae 
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: 233 
 
v=0 
o=root 6546 6546 IN IP4 192.168.10.20 
s=session 
c=IN IP4 192.168.10.20 
t=0 0 
m=audio 15206 RTP/AVP 0 8 97 3 
a=rtpmap:0 PCMU/8000 
a=rtpmap:8 PCMA/8000 
a=rtpmap:97 iLBC/8000 
a=rtpmap:3 GSM/8000 
a=silenceSupp:off - - - - 
 
 to 192.168.10.24:5060 
isis*CLI> 
 
Sip read: 
ACK sip:address@hidden SIP/2.0 
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK1498941086 
From: <sip:address@hidden>;tag=3488500359 
To: <sip:address@hidden>;tag=as123b40ae 
Call-ID: address@hidden 
CSeq: 20 ACK 
max-forwards: 10 
user-agent: oSIP/Linphone-0.12.1 
Content-Length: 0 
 
 
9 headers, 0 lines 
Jul  8 19:43:39 WARNING[376847]: file.c:874 ast_waitstream: Select failed (Bad 
file descriptor 
 
**************** asterisk sip.conf setting ************************** 
 
[general] 
port = 5060                     ; Port to bind to 
bindaddr = 0.0.0.0              ; 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=ulaw                      ; Allow codecs in order of preference 
allow=alaw 
allow=gsm 
allow=ilbc 
allow=gsm 
 
.. 
 
[aa] 
type=friend 
host=192.168.10.24 
defaultip=192.168.10.24 
dtmfmode=info                   ; Choices are inband, rfc2833, or info 
 
thanks in advance 
 
Amjad 
 
**************** .linphonec config settings.... *********************  
    
[net] 
if_name=rhine 
con_type=1 
use_nat=0 
 
[sip] 
username=aa 
hostname=192.168.10.24 
sip_port=5060 
use_registrar=0 
as_proxy=0 
expires=900 
 
[sound] 
dev_id=1 
rec_lev=80 
play_lev=80 
source=m 
local_ring=/usr/share/sounds/linphone/rings/oldphone.wav 
remote_ring=/usr/share/sounds/linphone/ringback.wav 
 
[rtp] 
audio_rtp_port=7078 
video_rtp_port=0 
audio_jitt_comp=60 
video_jitt_comp=60 
 
[video] 
enabled=0 
show_local=0 
 
[audio_codec_0] 
mime=PCMU 
rate=8000 
enabled=1 
 
[audio_codec_1] 
mime=GSM 
rate=8000 
enabled=1 
 
[audio_codec_2] 
mime=PCMA 
rate=8000 
enabled=1 
 
[audio_codec_3] 
mime=speex 
rate=8000 
enabled=1 
 
[audio_codec_4] 
mime=speex 
rate=16000 
enabled=1 
 
[audio_codec_5] 
mime=1015 
rate=8000 
enabled=1 
 
[address_book] 
entry_count=0 
 
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