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Re: [Linphone-users] problems with asterisk

From: Jason A. Pattie
Subject: Re: [Linphone-users] problems with asterisk
Date: Wed, 14 Jan 2004 17:03:36 -0600
User-agent: Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.5) Gecko/20031107 Debian/1.5-3 
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Tom Poindexter wrote:
| Note that the Asterisk folks have just released 0.7.0, and there are a
number
| of SIP related fixes.  You probably want to download and try out the newer
| version.

That's exactly what I just did (actually, upgraded to the CVS version).
~ For the first time in a long time, I was actually able to hear scratchy
sounds again from linphone-gpe!  It worked the first time I ever tried
it, about a month ago, and then didn't ever work after that.  Now that I
upgraded Asterisk, it worked again.  Here's what happened when I
attempted to check voicemail (ext. 8500) from linphone-gpe (hope it's
not too much information):

(As you will notice at the very end, it gives a SIGSEGV and quits)

~ $ linphone-gpe
linphone_gpe_init
| INFO1 | <osipua.c: 65> Starting osip stack and osipua layer.

| INFO1 | <udp.c: 112> Entering osipua thread.

MediaStreamer-Message: Found /dev/dsp.
MediaStreamer-Message: alsa_card_manager_init

MediaStreamer-Message: Found ALSA device: H3600 UDA1341TS
MediaStreamer-Message: alsa_set_params:  blocksize=512.
| INFO1 | <osipmanager.c: 148> port already listened

| INFO1 | <utils.c: 405> Outgoing interface to reach 192.168.1.28 is
192.168.1.17.

| ERROR | <osipdialog.c: 2173> generating_request_out_of_dialog: setting
ua->ua_family=2 from localip 192.168.1.17

| INFO1 | <utils.c: 405> Outgoing interface to reach 192.168.1.28 is
192.168.1.17.

| ERROR | <osipdialog.c: 2173> generating_request_out_of_dialog: setting
ua->ua_family=2 from localip 192.168.1.17

| INFO1 | <osipmanager.c: 148> port already listened

payload_type_check_usable: pt=GSM pt->type=1 normal_bitrate=13500
bandwidth=1e+07 usable=1
payload_type_check_usable: pt=PCMU pt->type=0 normal_bitrate=64000
bandwidth=1e+07 usable=1
payload_type_check_usable: pt=PCMA pt->type=0 normal_bitrate=64000
bandwidth=1e+07 usable=1
payload_type_check_usable: pt=iLBC pt->type=1 normal_bitrate=15200
bandwidth=1e+07 usable=1
payload_type_check_usable: pt=1015 pt->type=1 normal_bitrate=2400
bandwidth=1e+07 usable=1
| INFO1 | <udp.c: 295> Sending message:
REGISTER sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK601701693
From: <sip:address@hidden>;tag=2777994288
To: <sip:address@hidden>;tag=2777994288
Call-ID: address@hidden
CSeq: 0 REGISTER
Contact: <sip:address@hidden>
max-forwards: 10
expires: 900
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0


| INFO1 | <udp.c: 295> Sending message:
SUBSCRIBE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK1646452004
From: <sip:address@hidden>;tag=4025914474
To: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 20 SUBSCRIBE
Contact: <sip:address@hidden>
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Event: presence.winfo
Accept: application/watcherinfo+xml
Content-Length: 0


presence_validate=0xaa000
presence_box_changed done
| INFO1 | <udp.c: 186> info: Message from 192.168.1.28:5060

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK601701693
From: <sip:address@hidden>;tag=2777994288
To: <sip:address@hidden>;tag=2777994288
Call-ID: address@hidden
CSeq: 0 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Content-Length: 0



| INFO1 | <nict_callbacks.c: 116> OnEvent_New_Incoming4xxResponse!

| INFO1 | <nict_callbacks.c: 136> User need to authenticate to REGISTER!

| INFO1 | <utils.c: 405> Outgoing interface to reach 192.168.1.28 is
192.168.1.17.

| ERROR | <osipdialog.c: 2173> generating_request_out_of_dialog: setting
ua->ua_family=2 from localip 192.168.1.17

| INFO1 | <udp.c: 295> Sending message:
REGISTER sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK3792981598
From: <sip:address@hidden>;tag=223114929
To: <sip:address@hidden>;tag=223114929
Call-ID: address@hidden
CSeq: 1 REGISTER
Contact: <sip:address@hidden>
max-forwards: 10
expires: 900
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0


| INFO1 | <udp.c: 186> info: Message from 192.168.1.28:5060

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK1646452004
From: <sip:address@hidden>;tag=4025914474
To: <sip:address@hidden>;tag=as1f321d70
Call-ID: address@hidden
CSeq: 20 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Proxy-Authenticate: Digest realm="asterisk", nonce="6ab28a51"
Content-Length: 0



| INFO1 | <nict_callbacks.c: 116> OnEvent_New_Incoming4xxResponse!

| INFO1 | <nict_callbacks.c: 136> User need to authenticate to REGISTER!

| ERROR | <osipdialog.c: 618>
osip_dialog_register_with_authentification: no password, aborting
| INFO1 | <udp.c: 186> info: Message from 192.168.1.28:5060

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK3792981598
From: <sip:address@hidden>;tag=223114929
To: <sip:address@hidden>;tag=223114929
Call-ID: address@hidden
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Content-Length: 0



| INFO1 | <nict_callbacks.c: 116> OnEvent_New_Incoming4xxResponse!

| INFO1 | <nict_callbacks.c: 144> Authentification aborted.

| INFO1 | <nict_callbacks.c: 30> Transaction 1 killed.

| INFO1 | <nict_callbacks.c: 30> Transaction 2 killed.

| INFO1 | <nict_callbacks.c: 30> Transaction 3 killed.

| INFO1 | <osipdialog.c: 2411> Dialog is removed. It remains 1 dialog(s)
in the ua list.

url=8500 <sip:address@hidden:5060> strstr=(null)
| INFO1 | <utils.c: 405> Outgoing interface to reach 192.168.1.28 is
192.168.1.17.

| ERROR | <osipdialog.c: 2173> generating_request_out_of_dialog: setting
ua->ua_family=2 from localip 192.168.1.17

| INFO1 | <udp.c: 295> Sending message:
INVITE sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK360288805
From: <sip:address@hidden>;tag=989919275
To: 8500 <sip:address@hidden:5060>
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:   343

v=0
o=pattieja 123456 654321 IN IP4 192.168.1.17
s=A conversation
c=IN IP4 192.168.1.17
t=0 0
m=audio 7078 RTP/AVP 3 0 8 97 115 101
b=AS:97 15
a=rtpmap:3 GSM/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:97 iLBC/8000/1
a=fmtp:97 mode=20
a=rtpmap:115 1015/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

| INFO1 | <udp.c: 186> info: Message from 192.168.1.28:5060

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK360288805
From: <sip:address@hidden>;tag=989919275
To: 8500 <sip:address@hidden:5060>;tag=as082ad9d2
Call-ID: address@hidden
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Proxy-Authenticate: Digest realm="asterisk", nonce="25817c71"
Content-Length: 0



| INFO1 | <udp.c: 295> Sending message:
ACK sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK360288805
From: <sip:address@hidden>;tag=989919275
To: 8500 <sip:address@hidden:5060>;tag=as082ad9d2
Call-ID: address@hidden
CSeq: 20 ACK
Content-Length: 0


| INFO1 | <ict_callbacks.c: 249> OnEvent_New_Incoming4xxResponse!

| INFO1 | <ict_callbacks.c: 277> User need to authenticate to INVITE!

| INFO1 | <authentication.c: 377> Response in proxy_authorization
|1dc2fd29037cc4aea7ef44dd9dac3a38|

| INFO1 | <udp.c: 295> Sending message:
INVITE sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK2946050642
From: <sip:address@hidden>;tag=989919275
To: 8500 <sip:address@hidden:5060>
Call-ID: address@hidden
CSeq: 21 INVITE
Contact: <sip:address@hidden>
Proxy-Authorization: Digest username="pattieja", realm="asterisk",
nonce="25817c71", uri="sip:address@hidden:5060",
response="1dc2fd29037cc4aea7ef44dd9dac3a38", algorithm=MD5
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Type: application/sdp
Content-Length:   343

v=0
o=pattieja 123456 654321 IN IP4 192.168.1.17
s=A conversation
c=IN IP4 192.168.1.17
t=0 0
m=audio 7078 RTP/AVP 3 0 8 97 115 101
b=AS:97 15
a=rtpmap:3 GSM/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:97 iLBC/8000/1
a=fmtp:97 mode=20
a=rtpmap:115 1015/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

| INFO1 | <udp.c: 186> info: Message from 192.168.1.28:5060

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
ACK sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.1.28:5060;branch=z9hG4bK3fb58abc
From: 8500 <sip:address@hidden:5060>;tag=as4f68b4cd
To: <sip:address@hidden>;tag=989919275
Contact: <sip:address@hidden>
Call-ID: address@hidden
CSeq: 21 ACK
User-Agent: Asterisk PBX
Content-Length: 0



| INFO1 | <udp.c: 186> info: Message from 192.168.1.28:5060

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK2946050642
From: <sip:address@hidden>;tag=989919275
To: 8500 <sip:address@hidden:5060>;tag=as4f68b4cd
Call-ID: address@hidden
CSeq: 21 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 17677 17677 IN IP4 192.168.1.28
s=session
c=IN IP4 192.168.1.28
t=0 0
m=audio 12346 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


| INFO1 | <ict_callbacks.c: 71> OnEvent_New_Incoming2xxResponse!

| INFO1 | <ict_callbacks.c: 132> Found body application/sdp

osip_dialog_get_body_context: Comparing application/sdp <>
application/sdp , context=fffd0, handler=38528
MediaStreamer-Message: alsa_set_params:  blocksize=512.
MediaStreamer-Message: ms_filter_add_link: OssRead,0 -> GSMEncoder,0
MediaStreamer-Message: ms_filter_add_link: GSMEncoder,0 -> RTPSend,0
MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 -> GSMDecoder,0
MediaStreamer-Message: ms_filter_add_link: GSMDecoder,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.
| ERROR | <osipdialog.c: 1642>
osip_dialog_generate_request_within_dialog: setting ua->ua_family=2 from
localip 192.168.1.17

| INFO1 | <udp.c: 295> Sending message:
ACK sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK2762269855
From: <sip:address@hidden>;tag=989919275
To: 8500 <sip:address@hidden:5060>;tag=as4f68b4cd
Call-ID: address@hidden
CSeq: 21 ACK
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0


| INFO1 | <ict_callbacks.c: 30> Transaction 5 killed.

ALSA lib pcm_hw.c:494:(snd_pcm_hw_start) SNDRV_PCM_IOCTL_START failed:
Broken pipe
ALSA lib pcm_hw.c:494:(snd_pcm_hw_start) SNDRV_PCM_IOCTL_START failed:
File descriptor in bad state

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_write: Error
writing sound buffer (size=512):Resource temporarily unavailable

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_write: Error
writing sound buffer (size=512):Resource temporarily unavailable

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_write: Error
writing sound buffer (size=512):Resource temporarily unavailable

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_write: Error
writing sound buffer (size=512):Resource temporarily unavailable

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_write: Error
writing sound buffer (size=512):Resource temporarily unavailable

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_write: Error
writing sound buffer (size=512):Resource temporarily unavailable

(linphone-gpe:1962): MediaStreamer-WARNING **: MSTimer: must catchup 7
ticks.

(linphone-gpe:1962): MediaStreamer-WARNING **: MSTimer: must catchup 9
ticks.

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_write: Error
writing sound buffer (size=512):Resource temporarily unavailable

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_write: Error
writing sound buffer (size=512):Resource temporarily unavailable

(linphone-gpe:1962): MediaStreamer-WARNING **: MSTimer: must catchup 6
ticks.

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_write: Error
writing sound buffer (size=512):Resource temporarily unavailable

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_write: Error
writing sound buffer (size=512):Resource temporarily unavailable

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_read:
snd_pcm_writei() failed:Resource temporarily unavailable.
ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

(linphone-gpe:1962): MediaStreamer-WARNING **: alsa_card_write: Error
writing sound buffer (size=512):Resource temporarily unavailable

~   Global statistics :
~ packet_sent=417
~ sent=18765 bytes
~ packet_recv=67
~ hw_recv=3904 bytes
~ recv=3904 bytes
~ unavaillable=415 bytes
~ outoftime=0
~ bad=0
~ discarded=0
| ERROR | <osipdialog.c: 1642>
osip_dialog_generate_request_within_dialog: setting ua->ua_family=2 from
localip 192.168.1.17

MediaStreamer-Message: Mediastreamer processing thread is exiting.
| INFO1 | <udp.c: 295> Sending message:
BYE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK596235532
From: <sip:address@hidden>;tag=989919275
To: 8500 <sip:address@hidden:5060>;tag=as4f68b4cd
Call-ID: address@hidden
CSeq: 22 BYE
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0


| INFO1 | <udp.c: 186> info: Message from 192.168.1.28:5060

| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK596235532
From: <sip:address@hidden>;tag=989919275
To: 8500 <sip:address@hidden:5060>;tag=as4f68b4cd
Call-ID: address@hidden
CSeq: 22 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:address@hidden>
Content-Length: 0



| INFO1 | <nict_callbacks.c: 30> Transaction 6 killed.

| INFO1 | <osipdialog.c: 2411> Dialog is removed. It remains 1 dialog(s)
in the ua list.

| INFO1 | <ict_callbacks.c: 30> Transaction 4 killed.

| ERROR | <osipdialog.c: 2399> Could not remove dialog from list.

| ERROR | <osipdialog.c: 2405> Could not remove dialog from ua list.

SIGSEGV

- --
Jason A. Pattie
address@hidden
Xperience, Inc. (http://www.xperienceinc.com)
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