FreeSWITCH version: 1.0.head, sofia.c:6033 Reinvite Codec Error!
Hello, On a fresh 1.0.head (FreeSWITCH version: 1.0.head (git-303de52 2012-02-14 18-59-37 -0600), outgoing calls never succeed and the following error is logged in loop :
2012-02-15 11:34:52.894480 [DEBUG] sofia.c:5525 Channel sofia/external/0251685292 entering state [received][100] 2012-02-15 11:34:52.894480 [DEBUG] sofia.c:5536 Remote SDP: v=0 o=root 14865 14867 IN IP4 192.168.22.13 s=session c=IN IP4 83.136.163.36 t=0 0 m=audio 59418 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=nortpproxy:yes
2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [PCMA:8:8000:20:64000]/[GSM:3:8000:20:13200] 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [PCMU:0:8000:20:64000]/[GSM:3:8000:20:13200] 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4915 Set 2833 dtmf send payload to 101 2012-02-15 11:34:52.894480 [ERR] sofia.c:6033 Reinvite Codec Error! 2012-02-15 11:34:52.894480 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] 2012-02-15 11:34:52.894480 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] 2012-02-15 11:34:52.894480 [DEBUG] sofia.c:5525 Channel sofia/external/0251685292 entering state [ready][488]
Codecs prefs are the following : <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263"/> I don't really understand the codec negociation issue, any idea, anyone ? Kind Regards Éric Masson
you only have GSM and its trying to change it to PCMA/PCMU On Wed, Feb 15, 2012 at 6:01 AM, Eric Masson <emss.mail@gmail.com> wrote:
Hello,
On a fresh 1.0.head (FreeSWITCH version: 1.0.head (git-303de52 2012-02-14 18-59-37 -0600), outgoing calls never succeed and the following error is logged in loop :
2012-02-15 11:34:52.894480 [DEBUG] sofia.c:5525 Channel sofia/external/0251685292 entering state [received][100] 2012-02-15 11:34:52.894480 [DEBUG] sofia.c:5536 Remote SDP: v=0 o=root 14865 14867 IN IP4 192.168.22.13 s=session c=IN IP4 83.136.163.36 t=0 0 m=audio 59418 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=nortpproxy:yes
2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [PCMA:8:8000:20:64000]/[GSM:3:8000:20:13200] 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [PCMU:0:8000:20:64000]/[GSM:3:8000:20:13200] 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4915 Set 2833 dtmf send payload to 101 2012-02-15 11:34:52.894480 [ERR] sofia.c:6033 Reinvite Codec Error! 2012-02-15 11:34:52.894480 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] 2012-02-15 11:34:52.894480 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] 2012-02-15 11:34:52.894480 [DEBUG] sofia.c:5525 Channel sofia/external/0251685292 entering state [ready][488]
Codecs prefs are the following : <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263"/>
I don't really understand the codec negociation issue, any idea, anyone ?
Kind Regards
Éric Masson
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Le 15/02/2012 19:47, Anthony Minessale a écrit : Hello Anthony,
you only have GSM and its trying to change it to PCMA/PCMU
Thanks for your answer but I don't think I'm getting the point here From cli : show codec type,name,ikey codec,ADPCM (IMA),mod_spandsp codec,AMR,mod_amr codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_spandsp codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_spandsp codec,G.726 16k (AAL2),mod_spandsp codec,G.726 24k,mod_spandsp codec,G.726 24k (AAL2),mod_spandsp codec,G.726 32k,mod_spandsp codec,G.726 32k (AAL2),mod_spandsp codec,G.726 40k,mod_spandsp codec,G.726 40k (AAL2),mod_spandsp codec,G.729,mod_g729 codec,GSM,mod_spandsp 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_spandsp codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,iLBC,mod_ilbc 28 total. It seems to me that PCMA & PCMU are available (G711 alaw, G711 ulaw) Or am I missing something ? Kind Regards Éric Masson
On Wed, Feb 15, 2012 at 11:18 AM, Eric Masson <emss.mail@gmail.com> wrote:
Le 15/02/2012 19:47, Anthony Minessale a écrit :
Hello Anthony,
you only have GSM and its trying to change it to PCMA/PCMU
Thanks for your answer but I don't think I'm getting the point here From cli : show codec
show codec shows the codecs that FreeSWITCH can handle, but it does *not* show specific codecs your SIP profile is using. Look in your SIP profile (internal.xml or external.xml or whatever) and check out the codec_prefs. The reason Anthony knew you only had GSM is because of the "audio codec compare" lines in the log:
2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [PCMA:8:8000:20:64000]/[GSM:3: 8000:20:13200] 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [PCMU:0:8000:20:64000]/[GSM:3:8000:20:13200] 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
Note that the only codec that FS tries to match against is GSM. If other codecs were set in your inbound codec prefs then you'd see those being listed in "Audio Codec Compare" as well. -MC
Also, your doing a reinvite. You probably want proxy_media=true and bypass_media=false to transcode? Sent from my BlackBerry® smartphone on O2 -----Original Message----- From: Michael Collins <msc@freeswitch.org> Sender: freeswitch-users-bounces@lists.freeswitch.org Date: Wed, 15 Feb 2012 11:29:49 To: FreeSWITCH Users Help<freeswitch-users@lists.freeswitch.org> Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] FreeSWITCH version: 1.0.head, sofia.c:6033 Reinvite Codec Error! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Le 15/02/2012 20:29, Michael Collins a écrit : Hello Michael,
show codec shows the codecs that FreeSWITCH can handle, but it does *not* show specific codecs your SIP profile is using.
Ok, thanks.
Look in your SIP profile (internal.xml or external.xml or whatever) and check out the codec_prefs.
I've checked in freeswitch.xml.fsxml for profiles external, internal-ipv6 & internal and PCMA/PCMU codecs seem to be enabled (see profiles section at the end, I can provide the full file if necessary).
The reason Anthony knew you only had GSM is because of the "audio codec compare" lines in the log:
2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [PCMA:8:8000:20:64000]/[GSM:3: 8000:20:13200] 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [PCMU:0:8000:20:64000]/[GSM:3:8000:20:13200] 2012-02-15 11:34:52.894480 [DEBUG] sofia_glue.c:4801 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
Note that the only codec that FS tries to match against is GSM. If other codecs were set in your inbound codec prefs then you'd see those being listed in "Audio Codec Compare" as well.
Ok, understood. Kind Regards Éric Masson
<profiles> <profile name="external"> <gateways> <gateway name="keyyo.net"> <param name="username" value="myusername"/> <param name="password" value="mypassword"/> <param name="realm" value="keyyo.net"/> <param name="proxy" value="keyyo.net"/> <param name="register-proxy" value="keyyo.net"/> <param name="register" value="true"/> <param name="ping" value="5"/> <param name="ping-max" value="10"/> <param name="ping-min" value="3"/> <param name="retry-seconds" value="5"/> </gateway> </gateways>
<aliases> </aliases>
<domains> <domain name="all" alias="false" parse="true"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5080"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263"/> <param name="outbound-codec-prefs" value="PCMU,PCMA,GSM"/> <param name="hold-music" value="local_stream://moh"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-ip" value="192.168.85.1"/> <param name="sip-ip" value="192.168.85.1"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="tls" value="false"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5081"/> <param name="tls-cert-dir" value="/opt/freeswitch/conf/ssl"/> <param name="tls-passphrase" value=""/> <param name="tls-verify-date" value="true"/> <param name="tls-verify-policy" value="none"/> <param name="tls-verify-depth" value="2"/> <param name="tls-verify-in-subjects" value=""/> <param name="tls-version" value="tlsv1"/> </settings> </profile> <profile name="internal-ipv6"> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263"/> <param name="outbound-codec-prefs" value="G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="2a01:240:fe6a::1"/> <param name="sip-ip" value="2a01:240:fe6a::1"/> <param name="hold-music" value="local_stream://moh"/> <param name="apply-inbound-acl" value="domains"/> <param name="record-template" value="/opt/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5061"/> <param name="tls-cert-dir" value="/opt/freeswitch/conf/ssl"/> <param name="tls-version" value="tlsv1"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="force-register-domain" value="srvbsdfenssv"/> <param name="force-register-db-domain" value="srvbsdfenssv"/> </settings> </profile> <profile name="internal"> <aliases> </aliases> <gateways> </gateways>
<domains> <domain name="all" alias="true" parse="false"/> </domains>
<settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="watchdog-enabled" value="no"/> <param name="watchdog-step-timeout" value="30000"/> <param name="watchdog-event-timeout" value="30000"/> <param name="log-auth-failures" value="false"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="public"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263"/> <param name="outbound-codec-prefs" value="G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,H263"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="192.168.85.1"/> <param name="sip-ip" value="192.168.85.1"/> <param name="hold-music" value="local_stream://moh"/> <param name="apply-nat-acl" value="nat.auto"/> <param name="apply-inbound-acl" value="domains"/> <param name="local-network-acl" value="localnet.auto"/> <param name="record-path" value="/opt/freeswitch/recordings"/> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="presence-hosts" value="srvbsdfenssv,192.168.85.1"/> <param name="presence-privacy" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5061"/> <param name="tls-cert-dir" value="/opt/freeswitch/conf/ssl"/> <param name="tls-passphrase" value=""/> <param name="tls-verify-date" value="true"/> <param name="tls-verify-policy" value="none"/> <param name="tls-verify-depth" value="2"/> <param name="tls-verify-in-subjects" value=""/> <param name="tls-version" value="tlsv1"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="force-register-domain" value="srvbsdfenssv"/> <param name="force-subscription-domain" value="srvbsdfenssv"/> <param name="force-register-db-domain" value="srvbsdfenssv"/> <param name="challenge-realm" value="auto_from"/> </settings> </profile>
</profiles>
Le 15/02/2012 20:29, Michael Collins a écrit : Hello Michael,
show codec shows the codecs that FreeSWITCH can handle, but it does *not* show specific codecs your SIP profile is using. Look in your SIP profile (internal.xml or external.xml or whatever) and check out the codec_prefs.
I've fully reinstalled fs and sample configuration files (rm -rf $prefix, gmake install, gmake samples, gmake sounds/moh) The following files have been added to the base config. gateway definition file in $prefix/conf/sip_profiles/external :
<include> <gateway name="keyyo.net"> <param name="username" value="myname"/> <param name="password" value="mypw"/> <param name="realm" value="keyyo.net"/> <param name="proxy" value="keyyo.net"/> <param name="register-proxy" value="keyyo.net"/> <param name="register" value="true"/> <param name="ping" value="5"/> <param name="ping-max" value="10"/> <param name="ping-min" value="3"/> <param name="retry-seconds" value="5"/> </gateway> </include>
user definition file in $prefix/conf/directory/default
<include> <user id="1501"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1501"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1501"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="XLite Test"/> <variable name="effective_caller_id_number" value="1501"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="test"/> </variables> </user> </include>
extension for outgoing calls in £prefix/conf/diaplan/default
<include> <extension name="outbound_calls" continue="false"> <condition field="destination_number" expression = "^0(.*)$"> <action application="set" data="effective_caller_id_name=test"/> <action application="set" data="effective_caller_id_number=0980406292"/> <action application="bridge" data="sofia/gateway/keyyo.net/0$1"/> </condition> </extension> </include>
Except these three files, everything's out of the box setup. When dialing any number, the same codec negotiation issue pops up :
2012-02-16 19:50:40.735750 [DEBUG] sofia.c:5525 Channel sofia/external/0251685292 entering state [received][100] 2012-02-16 19:50:40.735750 [DEBUG] sofia.c:5536 Remote SDP: v=0^M o=root 14865 14867 IN IP4 192.168.22.103^M s=session^M c=IN IP4 83.136.164.34^M t=0 0^M m=audio 58298 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=ptime:20^M a=nortpproxy:yes^M
2012-02-16 19:50:40.735750 [DEBUG] sofia_glue.c:4820 Audio Codec Compare [PCMA:8:8000:20:64000]/[GSM:3:8000:20:13200] 2012-02-16 19:50:40.735750 [DEBUG] sofia_glue.c:4820 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2012-02-16 19:50:40.735750 [DEBUG] sofia_glue.c:4934 Set 2833 dtmf send payload to 101 2012-02-16 19:50:40.735750 [ERR] sofia.c:6036 Reinvite Codec Error! 2012-02-16 19:50:40.735750 [DEBUG] sofia.c:5525 Channel sofia/internal/1501@srvbsdfenssv.interne.associated-bears.org entering state [ready][200] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:791 Send signal sofia/external/0251685292 [BREAK] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:791 Send signal sofia/internal/1501@srvbsdfenssv.interne.associated-bears.org [BREAK]
So, any help solving this issue would be gladly received. Éric Masson
Somehow your codec prefs are getting messed up. Go to your fs_cli and type this: sofia status profile external Put the output here. Hopefully it will have a clue for us. -MC On Thu, Feb 16, 2012 at 10:59 AM, Eric Masson <emss.mail@gmail.com> wrote:
Le 15/02/2012 20:29, Michael Collins a écrit :
Hello Michael,
show codec shows the codecs that FreeSWITCH can handle, but it does *not* show specific codecs your SIP profile is using. Look in your SIP profile (internal.xml or external.xml or whatever) and check out the codec_prefs.
I've fully reinstalled fs and sample configuration files (rm -rf $prefix, gmake install, gmake samples, gmake sounds/moh)
The following files have been added to the base config.
gateway definition file in $prefix/conf/sip_profiles/external :
<include> <gateway name="keyyo.net"> <param name="username" value="myname"/> <param name="password" value="mypw"/> <param name="realm" value="keyyo.net"/> <param name="proxy" value="keyyo.net"/> <param name="register-proxy" value="keyyo.net"/> <param name="register" value="true"/> <param name="ping" value="5"/> <param name="ping-max" value="10"/> <param name="ping-min" value="3"/> <param name="retry-seconds" value="5"/> </gateway> </include>
user definition file in $prefix/conf/directory/default
<include> <user id="1501"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1501"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1501"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="XLite Test"/> <variable name="effective_caller_id_number" value="1501"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="test"/> </variables> </user> </include>
extension for outgoing calls in £prefix/conf/diaplan/default
<include> <extension name="outbound_calls" continue="false"> <condition field="destination_number" expression = "^0(.*)$"> <action application="set" data="effective_caller_id_name=test"/> <action application="set" data="effective_caller_id_number=0980406292"/> <action application="bridge" data="sofia/gateway/keyyo.net/0$1"/> </condition> </extension> </include>
Except these three files, everything's out of the box setup.
When dialing any number, the same codec negotiation issue pops up :
2012-02-16 19:50:40.735750 [DEBUG] sofia.c:5525 Channel sofia/external/0251685292 entering state [received][100] 2012-02-16 19:50:40.735750 [DEBUG] sofia.c:5536 Remote SDP: v=0^M o=root 14865 14867 IN IP4 192.168.22.103^M s=session^M c=IN IP4 83.136.164.34^M t=0 0^M m=audio 58298 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=ptime:20^M a=nortpproxy:yes^M
2012-02-16 19:50:40.735750 [DEBUG] sofia_glue.c:4820 Audio Codec Compare [PCMA:8:8000:20:64000]/[GSM:3:8000:20:13200] 2012-02-16 19:50:40.735750 [DEBUG] sofia_glue.c:4820 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2012-02-16 19:50:40.735750 [DEBUG] sofia_glue.c:4934 Set 2833 dtmf send payload to 101 2012-02-16 19:50:40.735750 [ERR] sofia.c:6036 Reinvite Codec Error! 2012-02-16 19:50:40.735750 [DEBUG] sofia.c:5525 Channel sofia/internal/ 1501@srvbsdfenssv.interne.associated-bears.org entering state [ready][200] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:875 Send signal sofia/external/0251685292 [BREAK] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:791 Send signal sofia/external/0251685292 [BREAK] 2012-02-16 19:50:40.735750 [DEBUG] switch_core_session.c:791 Send signal sofia/internal/1501@srvbsdfenssv.interne.associated-bears.org [BREAK]
So, any help solving this issue would be gladly received.
Éric Masson
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Le 21/02/2012 00:01, Michael Collins a écrit : Hello Michael, Sorry, didn't see your answer (subscribed via GMane).
Somehow your codec prefs are getting messed up. Go to your fs_cli and type this: sofia status profile external
I'll check asa I rebuild FS git-head on my box (maybe this weekend) After these mails, I switched back to FreeBSD port version (seems to be based on 1.0.7). At first run, I got the same problem. I then wiped both the port version and my local head build, rebuilt the port version and first run gave the same codec negotiation issue. I then removed references to GSM in global_codec_prefs & outbound_codec_prefs, then it started working fine :
2012-02-21 09:11:41.944103 [DEBUG] sofia_glue.c:4727 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-02-21 09:11:41.944103 [DEBUG] sofia_glue.c:4727 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-02-21 09:11:41.944103 [DEBUG] sofia_glue.c:4727 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2012-02-21 09:11:41.944103 [DEBUG] sofia_glue.c:2848 Set Codec sofia/external/0251804822 PCMA/8000 20 ms 160 samples 64000 bits
Sorry, this is really vague, and I'm not even sure the problem was not beetween keyboard and chair. So, as said before, I'll test again FS head asap and will keep you informed. Kind Regards Éric Masson
participants (4)
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Anthony Minessale
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Eric Masson
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Michael Collins
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omortimer＠gmail.com