Updating the configuration did not help.
I'm not telling FS to use any other codecs other than what I have
specified below, so I'm not sure what I have done wrong. Below is the
full trace:
------------------------------------------------------------------------
INVITE
sip:1020@192.168.1.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae
From: 1000 <
sip:1000@192.168.1.120>;tag=2c7a518d12f9370eo0
To: <
sip:1020@192.168.1.120>
Call-ID:
316156b9-f6b413d2@192.168.1.121 CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="1000",realm="192.168.1.120",nonce="77836a7a-8447-11df-93a6-d9ad5b204ca2",uri="
sip:1020@192.168.1.120",algorithm=MD5,response="c4a5e5cb60676366b8bcfb1329f3fc08",qop=auth,nc=00000001,cnonce="6a0220d0"
Contact: 1000 <
sip:1000@192.168.1.121:5060>
Expires: 240
User-Agent: Linksys/PAP2T-5.1.6(LS)
Content-Length: 451
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 17789112 17789112 IN IP4 192.168.1.121
s=-
c=IN IP4 192.168.1.121
t=0 0
m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
------------------------------------------------------------------------
send 343 bytes to udp/[192.168.1.121]:5060 at 13:00:32.882684:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae
From: 1000 <
sip:1000@192.168.1.120>;tag=2c7a518d12f9370eo0
To: <
sip:1020@192.168.1.120>
Call-ID:
316156b9-f6b413d2@192.168.1.121 CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19
14-49-15 -0500
Content-Length: 0
------------------------------------------------------------------------
2010-06-30 23:00:32.899423 [DEBUG] sofia.c:5975 IP 192.168.1.121
Rejected by acl "domains". Falling back to Digest auth.
2010-06-30 23:00:32.914917 [NOTICE] switch_channel.c:776 New Channel
sofia/internal/1000@192.168.1.120[77a3ab82-8447-11df-93a7-d9ad5b204ca2]
2010-06-30 23:00:32.943135 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000@192.168.1.120) Running State Change CS_NEW
2010-06-30 23:00:32.945232 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/1000@192.168.1.120) State NEW
2010-06-30 23:00:32.989778 [DEBUG] sofia.c:4293 Channel
sofia/internal/1000@192.168.1.120 entering state [received][100]
2010-06-30 23:00:32.997500 [DEBUG] sofia.c:4304 Remote SDP:
v=0
o=- 17789112 17789112 IN IP4 192.168.1.121
s=-
c=IN IP4 192.168.1.121
t=0 0
m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
2010-06-30 23:00:33.002369 [DEBUG] sofia_glue.c:3877 Audio Codec
Compare [PCMU:0:8000:30]/[G729:18:8000:20]
2010-06-30 23:00:33.005967 [DEBUG] sofia_glue.c:3877 Audio Codec
Compare [PCMU:0:8000:30]/[PCMU:0:8000:20]
2010-06-30 23:00:33.008248 [DEBUG] sofia_glue.c:3877 Audio Codec
Compare [PCMU:0:8000:30]/[GSM:3:8000:20]
2010-06-30 23:00:33.010721 [DEBUG] sofia_glue.c:3924 Substituting
codec PCMU@30i@8000h
2010-06-30 23:00:33.019002 [DEBUG] sofia_glue.c:2462 Set Codec
sofia/internal/1000@192.168.1.120 PCMU/8000 30 ms 240 samples
2010-06-30 23:00:33.038714 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf
send/recv payload to 101
2010-06-30 23:00:33.040992 [DEBUG] sofia.c:4451
(sofia/internal/1000@192.168.1.120) State Change CS_NEW -> CS_INIT
2010-06-30 23:00:33.043182 [DEBUG] switch_core_session.c:1027 Send
signal sofia/internal/1000@192.168.1.120 [BREAK]
2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000@192.168.1.120) Running State Change CS_INIT
2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1000@192.168.1.120) State INIT
2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:83
sofia/internal/1000@192.168.1.120 SOFIA INIT
2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:117
(sofia/internal/1000@192.168.1.120) State Change CS_INIT -> CS_ROUTING
2010-06-30 23:00:33.048752 [DEBUG] switch_core_session.c:1027 Send
signal sofia/internal/1000@192.168.1.120 [BREAK]
2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1000@192.168.1.120) State INIT going to sleep
2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000@192.168.1.120) Running State Change CS_ROUTING
2010-06-30 23:00:33.048752 [DEBUG] switch_channel.c:1474
(sofia/internal/1000@192.168.1.120) Callstate Change DOWN -> RINGING
2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/1000@192.168.1.120) State ROUTING
2010-06-30 23:00:33.055456 [DEBUG] switch_channel.c:1333
(sofia/internal/1000@192.168.1.120) Callstate Change RINGING -> ACTIVE
2010-06-30 23:00:33.055456 [DEBUG] mod_sofia.c:140
sofia/internal/1000@192.168.1.120 SOFIA ROUTING
2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:77
sofia/internal/1000@192.168.1.120 Standard ROUTING
2010-06-30 23:00:33.055456 [INFO] mod_dialplan_xml.c:331 Processing
1000->1020 in context default
Dialplan: sofia/internal/1000@192.168.1.120 parsing
[default->Local_1000_1019] continue=false
Dialplan: sofia/internal/1000@192.168.1.120 Regex (FAIL)
[Local_1000_1019] destination_number(1020) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/internal/1000@192.168.1.120 parsing
[default->Mobile_1020s] continue=false
Dialplan: sofia/internal/1000@192.168.1.120 Regex (PASS)
[Mobile_1020s] destination_number(1020) =~ /^(102[0-9])$/
break=on-false
Dialplan: sofia/internal/1000@192.168.1.120 Action set(dialed_extension=1020)
Dialplan: sofia/internal/1000@192.168.1.120 Action export(codec_string=GSM)
Dialplan: sofia/internal/1000@192.168.1.120 Action set(codec_string=GSM)
Dialplan: sofia/internal/1000@192.168.1.120 Action
bridge(user/${dialed_extension}@${domain})
Dialplan: sofia/internal/1000@192.168.1.120 Action set_user(1000@${domain})
Dialplan: sofia/internal/1000@192.168.1.120 Action answer()
Dialplan: sofia/internal/1000@192.168.1.120 Action sleep(1000)
Dialplan: sofia/internal/1000@192.168.1.120 Action
system(/usr/local/freeswitch/scripts/sms.pl ${smsaccount}
${smspassword} ${smsnumber} 'You have one new voicemail from ${effec$
<action application=)
2010-06-30 23:00:33.110926 [DEBUG] switch_core_state_machine.c:119
(sofia/internal/1000@192.168.1.120) State Change CS_ROUTING ->
CS_EXECUTE
2010-06-30 23:00:33.114917 [DEBUG] switch_core_session.c:1027 Send
signal sofia/internal/1000@192.168.1.120 [BREAK]
2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/1000@192.168.1.120) State ROUTING going to sleep
2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000@192.168.1.120) Running State Change CS_EXECUTE
2010-06-30 23:00:33.128360 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/1000@192.168.1.120) State EXECUTE
2010-06-30 23:00:33.129661 [DEBUG] mod_sofia.c:233
sofia/internal/1000@192.168.1.120 SOFIA EXECUTE
2010-06-30 23:00:33.134566 [DEBUG] switch_core_state_machine.c:157
sofia/internal/1000@192.168.1.120 Standard EXECUTE
EXECUTE sofia/internal/1000@192.168.1.120 set(dialed_extension=1020)
2010-06-30 23:00:33.157698 [DEBUG] mod_dptools.c:843
sofia/internal/1000@192.168.1.120 SET [dialed_extension]=[1020]
EXECUTE sofia/internal/1000@192.168.1.120 bridge(user/1020@mydns.dyndns.org)
2010-06-30 23:00:33.248458 [DEBUG] switch_ivr_originate.c:1956
variable string 0 = [presence_id=1020@mydns.dyndns.org]
2010-06-30 23:00:33.260573 [NOTICE] switch_channel.c:776 New Channel
sofia/internal/sip:1020@192.168.1.123:5060
[77d7e3b6-8447-11df-93a8-d9ad5b204ca2]
2010-06-30 23:00:33.303487 [DEBUG] mod_sofia.c:3883
(sofia/internal/sip:1020@192.168.1.123:5060) State Change CS_NEW ->
CS_INIT
2010-06-30 23:00:33.304564 [DEBUG] switch_core_session.c:1027 Send
signal sofia/internal/sip:1020@192.168.1.123:5060 [BREAK]
2010-06-30 23:00:33.327507 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1020@192.168.1.123:5060) Running State Change
CS_INIT
2010-06-30 23:00:33.332010 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1020@192.168.1.123:5060) State INIT
2010-06-30 23:00:33.334621 [DEBUG] mod_sofia.c:83
sofia/internal/sip:1020@192.168.1.123:5060 SOFIA INIT
send 1183 bytes to udp/[192.168.1.123]:5060 at 13:00:33.351944:
------------------------------------------------------------------------
INVITE sip:1020@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bKFDpKe5v6j2QcQ
Max-Forwards: 69
From: "1000" <sip:1000@192.168.1.120>;tag=Z51Qve55SUHta
To: <sip:1020@192.168.1.123:5060>
Call-ID: 4f442091-feea-122d-448a-00e04c0312e9
CSeq: 132833080 INVITE
Contact: <sip:mod_sofia@192.168.1.120:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19
14-49-15 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
sla, include-session-description, presence.winfo, message-summary,
refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 246
X-FS-Support: update_display
Remote-Party-ID: "1000"
<sip:1000@192.168.1.120>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1277875447 1277875448 IN IP4 192.168.1.120
s=FreeSWITCH
c=IN IP4 192.168.1.120
t=0 0
m=audio 27386 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:30
It fails after this with Not Acceptable Here / INCOMPATIBLE_DESTINATION
On Mon, Jun 28, 2010 at 5:15 PM, David Ponzone <david.ponzone@gmail.com> wrote:
Please, retry with a genuine config (the default one would be a wise
choice).
David Ponzone Direction Technique
email: david.ponzone@ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autorisée est interdite. Tout message électronique est susceptible
d'altération. IPeva décline toute responsabilité au titre de ce message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 28/06/2010 à 07:37, Mark Campbell-Smith a écrit :
Hi All,
I'm not really sure if I got a firm answer for this one. Is the only
way to ensure that transcoding is performed is by using
late-negotiation? Why aren't all my codecs sent in the INVITE message
to the B-leg (extension 1020)?
Thanks!
On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith
<mcampbellsmith@gmail.com> wrote:
FS version and codecs are shown below, but my config file are probably
quite old. But I guess they should still work?
All codecs are loaded, and the call works if late negotiation is set
on profile internal.
As I wrote above:
The call setup is extension 1000 calls extension 1020
1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
FS as the A-leg codec
2. Extension 1020 only supports GSM codec. The call fails with Not
Acceptable Here.
I forgot to write that Extension 1000 does not support GSM (I want to
force transcoding). Is that why FS is filtering out GSM on the b-leg?
freeswitch@internal> version
FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500)
freeswitch@internal> show codecs
type,name,ikey
codec,ADPCM (IMA),mod_voipcodecs
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.722,mod_voipcodecs
codec,G.723.1 6.3k,mod_g723_1
codec,G.726 16k,mod_voipcodecs
codec,G.726 16k (AAL2),mod_voipcodecs
codec,G.726 24k,mod_voipcodecs
codec,G.726 24k (AAL2),mod_voipcodecs
codec,G.726 32k,mod_voipcodecs
codec,G.726 32k (AAL2),mod_voipcodecs
codec,G.726 40k,mod_voipcodecs
codec,G.726 40k (AAL2),mod_voipcodecs
codec,G.729,mod_com_g729
codec,GSM,mod_voipcodecs
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,LPC-10,mod_voipcodecs
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Speex,mod_speex
25 total.
On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone <david.ponzone@gmail.com>
wrote:
Mark,
I confirm that, as I wrote that wiki page (the early negotiation part) :)
Can you really confirm your FS version ?
The parameter you showed is old.
codec-prefs has been replaced in SIP profiles by:
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
David Ponzone Direction Technique
email: david.ponzone@ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autorisée est interdite. Tout message électronique est susceptible
d'altération. IPeva décline toute responsabilité au titre de ce message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 23/06/2010 à 14:43, Mark Campbell-Smith a écrit :
Check this good wiki page for how FS negotiates codecs (early
negotiation default):
http://wiki.freeswitch.org/wiki/Codec_Negotiation
I have this set in my internal profile:
<param name="codec-prefs" value="$${global_codec_prefs}"/>
and as stated before vars.xml:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
Setting late negotiation works (thanks Sergey), but reading the wiki
page, I see the following sentence, which I interpret that GSM should
still be sent:
When FS calls leg B, the list of codecs in outbound-codec-prefs for
the SIP profile is reorganized by pushing the codec negotiated above
for leg A at the top . If B does not accept any of the codecs, the
calls fails, obviously.
On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano
<tgraziano@myitdepartment.net> wrote:
I'm a newb to fs, but doesn't codec get neogtiated by the endpoints?
Wouldn't fs only get involved when its media server is referred to?
If the "other endpoint" will only accept G729, doesn't that mean you
need to change that endpoint to also accept G711 or also license G729
in FS?
On 6/23/10, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Test Setup:
vars.xml:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
The call setup is extension 1000 calls extension 1020
1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
FS as the A-leg codec
2. Extension 1020 only supports GSM codec. The call fails with Not
Acceptable Here.
FS only offers G729 and PCMU to 1020. How do I change the number of
codecs that are offered to an extension? I know I can change the
order in the codec_prefs, but would prefer FS to offer all three
codecs to an extension.
m=audio 23662 RTP/AVP 0 18 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:30
Thanks
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--
Sent from my mobile device
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgraziano@voice.myitdepartment.net
Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk@voice.myitdepartment.net
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
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