March 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Core Dump on receiving a call from device with 'broken' G.722 codec.
by Keith Laaks 16 Mar '09

16 Mar '09

Hi, I am on fs 1.0.trunk (12530M) testing G.722 and found that when using a 'broken' configuration from a softphone configured for G.722, I get the warning on the cli: "We were told to use ptime 20 but what they meant to say was 820 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.. " when fs gets the invite, but then does a core dump when it tries to: <anti-action application="bridge" data="sofia/${regext_sipprofile}/${dialed_extension}%${domain_name}"/> Below are some of the traces and info output from before the core dump happens. I see this when I run gdb on the dumpfile. #0 0xb7e194d4 in switch_ivr_originate (session=0xb74640a8, bleg=0xb572b0b0, cause=0xb572b0ac, bridgeto=0xb74a1b18 "sofia/voxwan/8154%172.16.1.3", timelimit_sec=30, table=0xb7efcfc0, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=<value optimized out>) at src/switch_ivr_originate.c:1609 1609 if (switch_core_codec_init(&write_codec, (gdb) frame 1 #1 0xb6df48f5 in ?? () from /usr/local/freeswitch/mod/mod_dptools.so I wonder if anybody else has seen this behavior? This happens when the destination phone is also G.722 capable (policom). If I change the "frame per packet" setting in the softphone to 2 - All works OK (but the default is 1 - so cant risk allowing G.722 if it's going to core dump fs if a user make a wrong configuration) Best Regards Keith ************************************************************************ ************************************************************* 2009-03-16 14:37:58 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/sprof1/27879998182(a)196.99.88.77 [47e0c972-1227-11de-8b8e-1789e43c417d] <.....> 2009-03-16 14:37:58 [INFO] mod_sofia.c:1310 sofia_receive_message() Asked to send early media by sofia/sprof1/27879998182(a)196.99.88.77 2009-03-16 14:37:58 [NOTICE] sofia_glue.c:2245 sofia_glue_tech_media() Pre-Answer sofia/sprof1/27879998182(a)196.99.88.77! 2009-03-16 14:37:58 [INFO] mod_sofia.c:1351 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1237190146 1237190147 IN IP4 196.99.88.77 s=FreeSWITCH c=IN IP4 196.99.88.77 t=0 0 m=audio 16932 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-03-16 14:37:58 [INFO] switch_rtp.c:1441 rtp_common_read() Auto Changing port from 172.16.0.63:29081 to 196.22.33.44:10634 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() -- checktalktime.js -- <.. In this js I do http call to collect maximum talktime allowed ..> 2009-03-16 14:37:59 [NOTICE] checktalktime.js:1 console_log() schedparms=+3600 tbhangupwarn XML hangupwarn 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182(a)196.99.88.77 to XML[27879998154@e164route] 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing MeMe->27879998154 in context e164route 2009-03-16 14:37:59 [INFO] mod_dptools.c:945 info_function() CHANNEL_DATA: Event-Name: [CHANNEL_DATA] Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] FreeSWITCH-Hostname: [myfsbox] FreeSWITCH-IPv4: [196.99.88.77] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2009-03-16 14:37:59] Event-Date-GMT: [Mon, 16 Mar 2009 12:37:59 GMT] Event-Date-Timestamp: [1237207079387869] Event-Calling-File: [mod_dptools.c] Event-Calling-Function: [info_function] Event-Calling-Line-Number: [941] Channel-State: [CS_EXECUTE] Channel-State-Number: [4] Channel-Name: [sofia/sprof1/27879998182(a)196.99.88.77] Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Answer-State: [early] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Caller-Username: [27879998182] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [MeMe] Caller-Caller-ID-Number: [27879998182] Caller-Network-Addr: [196.22.33.44] Caller-Destination-Number: [27879998154] Caller-Unique-ID: [47e0c972-1227-11de-8b8e-1789e43c417d] Caller-Source: [mod_sofia] Caller-Context: [e164route] Caller-RDNIS: [27879998154] Caller-Channel-Name: [sofia/sprof1/27879998182(a)196.99.88.77] Caller-Profile-Index: [4] Caller-Profile-Created-Time: [1237207079387869] Caller-Channel-Created-Time: [1237207078659653] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [1237207078679638] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_sip_received_ip: [196.22.33.44] variable_sip_received_port: [36745] variable_sip_via_protocol: [udp] variable_sip_authorized: [true] variable_Event-Name: [REQUEST_PARAMS] variable_Core-UUID: [e57400ea-1223-11de-8b8e-1789e43c417d] variable_FreeSWITCH-Hostname: [myfsbox] variable_FreeSWITCH-IPv4: [196.99.88.77] variable_FreeSWITCH-IPv6: [::1] variable_Event-Date-Local: [2009-03-16 14:37:58] variable_Event-Date-GMT: [Mon, 16 Mar 2009 12:37:58 GMT] variable_Event-Date-Timestamp: [1237207078659653] variable_Event-Calling-File: [sofia_reg.c] variable_Event-Calling-Function: [sofia_reg_parse_auth] variable_Event-Calling-Line-Number: [1727] variable_sip_mailbox: [879998182] variable_sip_auth_username: [27879998182] variable_sip_auth_realm: [196.99.88.77] variable_mailbox: [879998182] variable_user_name: [27879998182] variable_domain_name: [196.99.88.77] variable_record_stereo: [true] variable_default_gateway: [verso] variable_default_areacode: [87] variable_transfer_fallback_extension: [operator] variable_sip-force-expires: [180] variable_toll_allow: [domestic,international] variable_accountcode: [tbaaaa] variable_user_context: [sprof1] variable_effective_caller_id_name: [TickyBox 99999 Phone 99 Test] variable_effective_caller_id_number: [879998182] variable_outbound_caller_id_name: [879998150] variable_outbound_caller_id_number: [879998150] variable_sip_from_user: [27879998182] variable_sip_from_uri: [27879998182(a)196.99.88.77] variable_sip_from_host: [196.99.88.77] variable_sip_from_user_stripped: [27879998182] variable_sip_from_tag: [196.99.88.77] variable_sofia_profile_name: [sprof1] variable_sofia_profile_domain_name: [196.99.88.77] variable_sip_req_user: [0879998154] variable_sip_req_uri: [0879998154(a)196.99.88.77] variable_sip_req_host: [196.99.88.77] variable_sip_to_user: [0879998154] variable_sip_to_uri: [0879998154(a)196.99.88.77] variable_sip_to_host: [196.99.88.77] variable_sip_contact_user: [27879998182] variable_sip_contact_port: [22034] variable_sip_contact_uri: [27879998182@172.16.0.63:22034] variable_sip_contact_host: [172.16.0.63] variable_channel_name: [sofia/sprof1/27879998182(a)196.99.88.77] variable_sip_call_id: [xr125298731411533c30039109e1921f(a)192.168.10.1] variable_sip_user_agent: [BrokenPhone/1.4.2] variable_sip_via_host: [172.16.0.63] variable_sip_via_port: [22034] variable_sip_via_rport: [36745] variable_presence_id: [27879998182(a)196.99.88.77] variable_switch_r_sdp: [v=0 o=2787999818 2265 2267 IN IP4 172.16.0.63 s=Broken c=IN IP4 172.16.0.63 t=0 0 m=audio 29081 RTP/AVP 9 18 101 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=candidates:-1564465265,172.16.0.63:29081,192.168.10.1:29081,192.168.20 .1:29081 ] variable_outboundcontext: [setupprepaycall] variable_remote_media_ip: [172.16.0.63] variable_remote_media_port: [29081] variable_read_codec: [G722] variable_read_rate: [16000] variable_write_codec: [G722] variable_write_rate: [16000] variable_local_media_ip: [196.99.88.77] variable_local_media_port: [16932] variable_endpoint_disposition: [EARLY MEDIA] variable_sip_nat_detected: [true] variable_api_hangup_hook: [jsapi::completecall.js] variable_talktime: [6870] variable_action: [allow] variable_status: [allowed] variable_integer: [102] variable_fraction: [85] variable_saytalktime: [60:0] variable_schedparms: [+3600 tbhangupwarn XML hangupwarn] variable_bridgejscb: [{api_hangup_hook=jsapi::completecall.js}] variable_max_forwards: [67] variable_current_application: [info] 2009-03-16 14:37:59 [WARNING] mod_sofia.c:739 sofia_read_frame() We were told to use ptime 20 but what they meant to say was 820 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.. 2009-03-16 14:37:59 [WARNING] switch_core_codec.c:499 switch_core_codec_init() Codec G722 Exists but not at the desired implementation. 8000hz 820ms 2009-03-16 14:37:59 [ERR] sofia_glue.c:1700 sofia_glue_tech_set_codec() Can't load codec? 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/sprof1/27879998182(a)196.99.88.77 has no read codec. 2009-03-16 14:37:59 [ERR] switch_core_io.c:585 switch_core_session_write_frame() sofia/sprof1/27879998182(a)196.99.88.77 has no write codec. 2009-03-16 14:37:59 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/sprof1/27879998182(a)196.99.88.77 has no read codec. 2009-03-16 14:37:59 [NOTICE] switch_ivr.c:1345 switch_ivr_session_transfer() Transfer sofia/sprof1/27879998182(a)196.99.88.77 to XML[8154@toregext] 2009-03-16 14:37:59 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing MeMe->8154 in context toregext 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/27879998182.2009-03-16- 14-37-59.wav 2009-03-16 14:37:59 [INFO] switch_ivr_async.c:1760 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features <... output from info application ...> 2009-03-16 14:38:00 [NOTICE] switch_channel.c:592 switch_channel_set_name() New Channel sofia/voxwan/8154 [48c2f400-1227-11de-8b8e-1789e43c417d] Segmentation fault (core dumped)

3 2

Start FreeSWITCH without any SQL but at the same time have all info available on realtime/runtime
by Diego Viola 16 Mar '09

16 Mar '09

Hello, I was wondering if it is possible to start FreeSWITCH without any SQL at all, yeah I know there is -nosql but I still want to have all the information available at run time, ie: show channels, sofia status, sofia status profile <profile>, everything. Would that improve performance? I ask this because I heard that SQLite causes some performance issues and I heard it's being removed from the core, I also think that making that optional would be nice along with UnixODBC support for everything. I also got a few situations where I had to rm -f /usr/local/freeswitch/db/* because there were some data there that was not supposed to be, that's why I think real time data on run time would be better. Anyway, just some feedback/question I wanted to share, FS is awesome, keep up the great work. Regards, Diego

7 17

How can I change the timeout config?
by shawn lin 16 Mar '09

16 Mar '09

Hi all, I met a problem which I thought is caused by the timeout config setting. I use SIPp UAC <-> FreeSWITCH <-> SIPp UAS to do my test. Commands: FS: ulimit -s 256; sudo ./freeswitch UAS: ulimit -s 256; sudo sipp -sn uas -i 10.67.7.224 10.67.7.46 -trace_err UAC: ulimit -s 256; sudo sipp -sn uac -i 10.67.7.213 10.67.7.29 -trace_err -l 5000 -m 5000 -d 300000 -r 5 Infr: UAC(10.67.7.213) <-> FS internal(10.67.7.29) FS external(10.67.7.46) <-> UAS 10.67.7.224 ********************** *** My problem **** ********************** I delayed uac for 300 seconds to send the BYE messages to uas, but after about 40 seconds of pausing, uas received BYE messages from FreeSWITCH: 2009-03-16 14:36:16:274 1237185376.274928: Aborting call on an unexpected BYE for call: 1-16380(a)10.67.7.213. **** Help *** Can anyone tell me how to release the timeout limit? Best Regards!

1 0

Getting a free Did number for my FS
by Meftah Tayeb 15 Mar '09

15 Mar '09

hello, please ho to get a free did number ? also, is it pocible to link it to my FS ? thanks

3 2

Problems with mp3 file formats in mod_shout
by Marc Orenberg 15 Mar '09

15 Mar '09

Hi, I'm using mod_shout to play mp3 files, and I'm having some trouble. I have two sets of mp3 files, one from a professional voice-prompt service, and one set that I converted from .wav files. In both sets, all of the files play fine in Windows Media player and other players, but in both sets, there are some files which mod_shout is complaining about. It gives me errors like this: Note: Illegal Audio-MPEG-Header 0x41504554 at offset 0x40fa. Note: Trying to resync... Note: Hit end of (available) data during resync. When it gets these errors, it often stops playing the file before it is completed. I've tried checking / re-encoding these files using Audacity, vbrfix, mp3validator and mp3gain, and I continue to have the same problem, which is making me wonder if it's really a formating problem after all, or maybe some bug in mod_shout. I'm wondering if anybody has had these problems before, and if they have any insight into what could be going on. I'd really appreciate any help or advice. By the way, I'm using FreeSwitch version 1.0.3 on CentOS linux. Thanks in advance, Marc

1 0

Problem with detecting session.cause with Linksys adaptor
by mszlazak＠aol.com 14 Mar '09

14 Mar '09

Hello, I have an analogue line connected to SPA3102 which then sends calls to my FreeSwitch (FS) application. I can originate a call from FS and dial out the analogue line almost fine. However, an issue seems to be detecting if the PSTN line is busy. Another issue might be "hanging up" the PSTN side of the adaptor from FS side. Anyway here's the problem. If I then try originating an outbound call from FS through the pstn side of the adaptor and if a caller on the analogue line has not hung up then they hear FS dialing and any automated message that FS then sends. I can not seem to detect from FS if the PSTN line is busy for some reason with the following code. var s; while (tryCalling()) {} s.hangup(); exit(); function tryCalling() { ??? s = new Session("sofia/gateway/spa3102/14082031170@10.0.0.5:5061"); ??? s.waitForAnswer(30000); ??? ??? if (s.cause == "USER_BUSY") { ??? ??? return true; ??? } ??? if (s.ready()) { ??? ??? s.sleep(1000); ??? ??? s.speak("cepstral","Callie","Hello from Gino Mick Gelato"); ??? } ??? return false; } I've also tried the above with ignore_early_media=true but no luck. Thanks. Mark.

1 0

Rewriting Remote Party ID
by rod 14 Mar '09

14 Mar '09

Dear list, I'd like to rewrite the number in the Remote Party ID header and only in this header. ex: I'd like to prefix the caller ID with a prefix code (000 in this example) in the RPID header : From: Anonymous<sip:anonymous@anonymous.invalid>;tag=1208367 Remote-Party-ID: <sip:123456@10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling should become: From: Anonymous<sip:anonymous@anonymous.invalid>;tag=1208367 Remote-Party-ID: <sip:000123456@10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling But the From field should remain unchanged. And how to strip this prefix: From: Anonymous<sip:anonymous@anonymous.invalid>;tag=1208367 Remote-Party-ID: <sip:000123456@10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling should become: From: Anonymous<sip:anonymous@anonymous.invalid>;tag=1208367 Remote-Party-ID: <sip:123456@10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling regards.

13 34

openzap with A102
by Baskar 14 Mar '09

14 Mar '09

Hi, I am using sangoma A102 card with freeswitch. I have updated all the changes in the freeswitch and have loaded openzap also. But still i cant able to make an outbound call. *openzap.conf* [span wanpipe] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 [span wanpipe] name => OpenZAP number => 2 trunk_type => E1 b-channel => 2:1-15 d-channel => 2:16 b-channel => 2:17-31 *openzap.conf.xml* <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="2"/> </settings> <pri_spans> <span id="1"> <param name="mode" value="user"/> <param name="dialect" value="euro"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> </configuration> *default.xml * <extension name="Long Distance - wanpipe"> - <condition field="destination_number" expression="^9(\d{5,15})$"> <action application="set" data="dialed_ext=$1"/> <action application="bridge" data="openzap/1/A/${dialed_ext}"/> </condition> </extension> * I have attached the freeswitch log http://pastebin.freeswitch.org/7730 Can any one correct were i am wrong. * -- Warm Regards, N.Baskar

4 5

Help with detecting hangup events.
by mszlazak 13 Mar '09

13 Mar '09

I have a problem in setting up an extension that will detect different hangup events. If an outside call comes in and gets sent to "myExtension" then that call is bridged to an application. When the application hangs up then the hangup hook "notify.js" is used to originate a call to another phone which notifies that phone that the application has been used. However, the problem here is that in the initial call, the caller can't hangup because then notify.js won't execute. <extension name="myExtension"> <condition field="destination_number" expression="^2222$"> <action application="info"/> <action application="set" data="bypass_media=true"/> <action application="set" data="effective_caller_id_number=${caller_id_number}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="api_hangup_hook=jsrun notify.js"/> <action application="bridge" data="sofia/internal/${sip_to_user}@127.0.0.1:5068"/> </condition> </extension> I haven't tried this but if I changed things so "hang_after_bridge=false" then I'm guessing that notify.js will execute when the initial caller hangs up their phone. This I want but I don't want the call originated in notify.js to happen unless the bridged to application also issued a hangup previously. Otherwise, if the caller hangs up before the application does it's thing and hasn't issued a hangup then notify.js will still originate calls and these I don't want. So, I want notify.js to originate calls after the caller hangs up but these hangups have to be those that came after the bridged to application issued a hung up. I hope this makes sense. Thanks. Mark. -- View this message in context: http://www.nabble.com/Help-with-detecting-hangup-events.-tp22507114p2250711… Sent from the Freeswitch-users mailing list archive at Nabble.com.

1 0

inband dtmf detection
by Gabriel Kuri 13 Mar '09

13 Mar '09

shoot me if I'm on the wrong track, but is it possible to use the start_dtmf app to do inband dtmf detection and "convert" the inband DTMF to rfc2833, as opposed to using the dtmf detection on a Linksys or Grandstream ATA? the reason I ask is the dtmf detection on these ATAs seems to falsely catch dtmf tones during certain conversations and falsely inject a dtmf tone into the stream (I believe the appropriate term is "talk off"). So I was thinking of setting the ATAs to do inband DTMF rather than the current rfc2833 setting, and having FS do the more reliable dtmf detection instead? Thanks, Gabe

2 2