July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to know my gateway registering is successed??
by Brad Tuan 02 Jul '09

02 Jul '09

Another question, Where does FS keep these information?? In *.db or somewhere??

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PCMU fallback for T.38
by Gabriel Kuri 01 Jul '09

01 Jul '09

hey folks, I'm trying to configure PCMU fallback for T.38. The originating endpoint (Linksys SPA-2102) sends an INVITE to FS with G729 and PCMU in the sdp. the INVITE to the provider includes G729 and PCMU as part of the sdp as well (absolute_codec_string=G729,PCMU) ... m=audio 16458 RTP/AVP 18 0 100 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 once the FAX tone is detected on the PSTN side, FS receives a T.38 re-INVITE from the provider and FS sends back a 488/Not Acceptable (proxy_media=false). at that point the provider than attempts fallback to PCMU with another reINVITE ... m=audio 16816 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 immediately after the PCMU reINVITE, FS closes the channel and the text below is in the FS logs. given the SPA-2102 included PCMU in the original INVITE, even though it was the second preferred codec, shouldn't FS fallback to using PCMU if it was re-INVITEd with PCMU by the provider? It seems like it's not passing the PCMU Re-INVITE back to the endpoint (SPA-2102), since it originally negotiated G729 with the SPA2102 as that was the 1st codec in the sdp, but trying to transcode between the two (G729 and PCMU)? 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2371 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [telephone-event:101:8000]/[G729:18:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:2407 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMU:0:8000]/[PCMU:0:8000] 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1550 sofia_glue_tech_set_codec() Changing Codec from G729 to PCMU 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1601 sofia_glue_tech_set_codec() Set Codec sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 PCMU/8000 20 ms 160 samples 2009-03-20 01:19:58 [DEBUG] sofia_glue.c:1811 sofia_glue_activate_rtp() Audio params are unchanged for sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1. 2009-03-20 01:19:58 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing Reinvite 2009-03-20 01:19:58 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 entering state [completed] 2009-03-20 01:19:58 [DEBUG] switch_core_io.c:655 switch_core_session_write_frame() sofia/cedarwireless.net/1XXXXXXXXXX@1.1.1.1 receive message [SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY] 2009-03-20 01:19:58 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-03-20 01:19:58 [ERR] switch_core_io.c:723 switch_core_session_write_frame() Codec G.729 decoder error! Thanks, Gabe

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FreeSWITCH with DimDim Integration
by Ing. Edwin Villarreal 01 Jul '09

01 Jul '09

Hello ! I’ve been trying to integrate DimDim with FreeSWITCH. What I’d like is to create a SIP leg into a FreeSWITCH audio conference room, so meetings running can be listened in conference room as well. After reading their API and “understanding” how dimdim Works (I think), this configuration may fit for FreeSWITCH. Please see img attached. With this scenario I can add DimDim to a VoIP conference room, which can host hundreads or users. Have any of you tried this before? Do you have a better idea? Thanks Edwin

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Re: [Freeswitch-users] sound_prefix is changed
by Diego Toro 01 Jul '09

01 Jul '09

The audio files are for instance on ...us/callie/digits/8000, but the last 1.0.4 pre9 version has changed, it ignores the var sound_prefix setting to SWITCH_GLOBAL_dirs.base_dir/sounds/en the path to audio files. That change is on the switch_ivr_say function Diego. --- On Wed, 7/1/09, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: From: Anthony Minessale <anthony.minessale(a)gmail.com> Subject: Re: [Freeswitch-users] sound_prefix is changed To: freeswitch-users(a)lists.freeswitch.org Date: Wednesday, July 1, 2009, 4:31 PM did you put them in that spot? the say app needs them to be in that spot. On Wed, Jul 1, 2009 at 4:10 PM, Diego Toro <dftoro(a)yahoo.com> wrote: Hello, the problem is that sound_prefix value is ignored and it's changed to SWITCH_GLOBAL_dirs.base_dir/sounds/language (with language=en), so audio files are not found . Diego --- On Wed, 7/1/09, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: From: Anthony Minessale <anthony.minessale(a)gmail.com> Subject: Re: [Freeswitch-users] sound_prefix is changed To: freeswitch-users(a)lists.freeswitch.org Date: Wednesday, July 1, 2009, 3:36 PM what is the problem exactly? what do you want it to change to? On Wed, Jul 1, 2009 at 2:41 PM, Diego Toro <dftoro(a)yahoo.com> wrote: hi all, I am working with release Pre9, I have a problem now with say module, the sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for default_language to 'es'. I checked c code on switch_ivr_say the value of sound_prefix is changed always Diego _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Re: [Freeswitch-users] sound_prefix is changed
by Diego Toro 01 Jul '09

01 Jul '09

Hello, the problem is that sound_prefix value is ignored and it's changed to SWITCH_GLOBAL_dirs.base_dir/sounds/language (with language=en), so audio files are not found . Diego --- On Wed, 7/1/09, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: From: Anthony Minessale <anthony.minessale(a)gmail.com> Subject: Re: [Freeswitch-users] sound_prefix is changed To: freeswitch-users(a)lists.freeswitch.org Date: Wednesday, July 1, 2009, 3:36 PM what is the problem exactly? what do you want it to change to? On Wed, Jul 1, 2009 at 2:41 PM, Diego Toro <dftoro(a)yahoo.com> wrote: hi all, I am working with release Pre9, I have a problem now with say module, the sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for default_language to 'es'. I checked c code on switch_ivr_say the value of sound_prefix is changed always Diego _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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G723 timer problem
by Muhammad Shahzad 01 Jul '09

01 Jul '09

Hi, I am using FS svn revision 14046 and trying to send call from SIP Dialer to a SIP gateway using G723 in passthrough mode. Everything works perfect and destination rings but then call drops with following error on FS CLI, 2009-07-02 02:39:28.790508 [WARNING] mod_sofia.c:807 We were told to use ptime 30 but what they meant to say was 60 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2009-07-02 02:39:28.790508 [WARNING] switch_core_codec.c:499 Codec G723 Exists but not at the desired implementation. 8000hz 60ms Is there any work around for this or i have downgrade my server back to Asterisk. :'-( Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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sound_prefix is changed
by Diego Toro 01 Jul '09

01 Jul '09

hi all, I am working with release Pre9, I have a problem now with say module, the sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for default_language to 'es'. I checked c code on switch_ivr_say the value of sound_prefix is changed always Diego

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freeswitch support PCMU only?
by qian ma 01 Jul '09

01 Jul '09

hi all freeswitch support PCMU only? i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, but freeswitch still support PCMU only, below is the trace: 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload to 101 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup sofia/maq/9876(a)58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal sofia/maq/9876(a)58.212.219.104 [KILL] 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal sofia/maq/9876(a)58.212.219.104 [BREAK] 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( sofia/maq/9876(a)58.212.219.104) Running State Change CS_HANGUP 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( sofia/maq/9876(a)58.212.219.104) State HANGUP how to configure the freeswitch?? support more codecs??? thx! m.q

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Freeswitch memory usage is too high
by Muhammad Danish Moosa 01 Jul '09

01 Jul '09

Hi Brian My customer is now using lua odbc ( not lua mysql anymore) and problem mentioned in jira is resolved now. * http://www.mail-archive.com/freeswitch-dev@lists.freeswitch.org/msg01352.ht… * * *This seems to answer my question, rite? BTW , starting FS with following ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 244 ulimit -l unlimited ulimit -a Date: Wed, 1 Jul 2009 09:18:57 -0500 From: Brian West <brian(a)freeswitch.org> Subject: Re: [Freeswitch-users] Freeswitch memory usage is too high To: freeswitch-users(a)lists.freeswitch.org Message-ID: <C828137D-CA89-48C3-B970-1064933D0004(a)freeswitch.org> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes You also have a jira http://jira.freeswitch.org/browse/MODAPP-298, It looks like you're using lua sql and the backtrace you attached to the jira was cut off right before the data I needed to see... can you follow up on that ASAP? It looks like a crash in libmysql from the last line but again I can't see the rest of it. /b On Jul 1, 2009, at 7:29 AM, Muhammad Danish Moosa wrote: > Hi > > Freeswitch is being used in a scenario where two endpoints are > running traffic with bypass media mode. Performance is good and all > things are smooth. > > But as the time goes after starting freeswitch, it starts consuming > almost whole of memory. Note , freeswitch is being started with - > core option, is it related? > > If this 99% memory consumption is any red alert, as we can see calls > are still connecting fine and all is going as usual. > > > -- > Muhammad Danish Moosa > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Muhammad Danish Moosa

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Could this be a bug in the SIP registry?
by Mitchel Constantin 01 Jul '09

01 Jul '09

Hello, I'm experiencing a bug that I've been working on most of today. I can not call between two SIP phones that register successfully. In order to diagnose it, I have removed my FreeSWITCH server out of the NAT/firewall to try and eliminate any such issues with these things. Here is how I ran into the issue: 1. Started from sample configs 2. Enabled xml_curl and wrote the associated script to generate XML and had the phones authenticating but was forced to use the IP address of the server as the domain in my directory XML. 3. I tested calls and they worked using this syntax: originate sofia/internal/mythicalbox%205.134.225.20 (the server's ip) sofia/internal/johndoe%205.134.225.20 4. Next to remove the "limit" on using only the IP as the domain for users, I commented out force-register-domain and force-register-db-domain in internal.xml. 5. My phones now register using the correct domain name (i.e. weavver.com) instead of the IP address (205.134.225.20) as the domain. 6. Now the problem... My originate command no longer works using the new syntax: originate sofia/internal/mythicalbox%weavver.comsofia/internal/johndoe% weavver.com The phones do show up as registered when I type "sofia status profile internal": API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 205.134.225.20 SIP-IP 205.134.225.20 URL sip:mod_sofia@205.134.225.20:5060 BIND-URL sip:mod_sofia@205.134.225.20:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 9 FAILED-CALLS-IN 3 CALLS-OUT 8 FAILED-CALLS-OUT 18 Registrations: ================================================================================================= Call-ID: ZWQ3NjRhZWI3MDc4ZDhjNTdhZDU2ZGVkM2JmYjg3NTc. User: mythicalbox(a)weavver.com Contact: "mythicalbox" <sip:mythicalbox@64.183.110.250:65438 ;rinstance=e632b810dbf91486;transport=TCP;fs_nat=yes;fs_path=sip%3Amythicalbox%4064.183.110.250%3A65438%3Brinstance%3De632b810dbf91486%3Btransport%3DTCP> Agent: eyeBeam release 1102u stamp 52344 Status: Registered(TCP-NAT)(unknown) EXP(2009-06-30 22:43:47) Host: duck.weavver.com IP: 64.183.110.250 Port: 65438 Auth-User: mythicalbox Auth-Realm: weavver.com Call-ID: Y2E0NWZiZWZjOTYxYjVhNmQ5ZDkyMzJjMzUxYWM0ZGM. User: johndoe(a)weavver.com Contact: "johndoe" <sip:johndoe@64.183.110.250:65512 ;rinstance=ce0823faf2aba98a;transport=TCP;fs_nat=yes;fs_path=sip%3Ajohndoe%4064.183.110.250%3A65512%3Brinstance%3Dce0823faf2aba98a%3Btransport%3DTCP> Agent: eyeBeam release 1102u stamp 52344 Status: Registered(TCP-NAT)(unknown) EXP(2009-06-30 22:47:37) Host: duck.weavver.com IP: 64.183.110.250 Port: 65512 Auth-User: johndoe Auth-Realm: weavver.com ================================================================================================= FYI, FreeSWITCH is on a public IP address on the 'nets and the phones are behind the same firewall on a different public IP address on the internet. Thank you in advance for any help! :) -- Mitchel Constantin Weavver, Inc.

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