[Freeswitch-users] what does this mean?
2008-08-11 17:40:53 [DEBUG] sofia.c:194 sofia_event_callback() event [nua_r_options] status [501][Not Implemented] session: n/a What have I not setup properly that this message constantly is displayed ona loglevel 7 mode Ilan Perez ___ 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
[Freeswitch-users] Cannot create outgoing channel of type [user] cause
Below is the error, I am getting, while running my Java program through Freeswitch dial plan. 2008-08-11 16:25:33 [NOTICE] switch_channel.c:1406 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/internal/ [EMAIL PROTECTED] 2008-08-11 16:25:33 [NOTICE] mod_sofia.c:1116 sofia_receive_message() Pre-Answer sofia/internal/[EMAIL PROTECTED] # # An unexpected error has been detected by Java Runtime Environment: # # SIGSEGV (0xb) at pc=0x7f020c4281ea, pid=9129, tid=1082808656 # # Java VM: Java HotSpot(TM) 64-Bit Server VM (11.0-b12 mixed mode linux-amd64) # Problematic frame: # C [libpthread.so.0+0xa1ea] pthread_rwlock_rdlock+0xa # # An error report file with more information is saved as: # /home/adeel/programs/freeswitch/bin/hs_err_pid9129.log # # If you would like to submit a bug report, please visit: # http://java.sun.com/webapps/bugreport/crash.jsp # Aborted (core dumped) --- Any idea, if this is because of 64bit? Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari ___ 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
[Freeswitch-users] Freeswitch is crashing.
Below is the error, I am getting, while running my Java program through Freeswitch dial plan. 2008-08-11 16:25:33 [NOTICE] switch_channel.c:1406 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/internal/ [EMAIL PROTECTED] 2008-08-11 16:25:33 [NOTICE] mod_sofia.c:1116 sofia_receive_message() Pre-Answer sofia/internal/[EMAIL PROTECTED] # # An unexpected error has been detected by Java Runtime Environment: # # SIGSEGV (0xb) at pc=0x7f020c4281ea, pid=9129, tid=1082808656 # # Java VM: Java HotSpot(TM) 64-Bit Server VM (11.0-b12 mixed mode linux-amd64) # Problematic frame: # C [libpthread.so.0+0xa1ea] pthread_rwlock_rdlock+0xa # # An error report file with more information is saved as: # /home/adeel/programs/freeswitch/bin/hs_err_pid9129.log # # If you would like to submit a bug report, please visit: # http://java.sun.com/webapps/bugreport/crash.jsp # Aborted (core dumped) --- Any idea, if this is because of 64bit? Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari ___ 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
Re: [Freeswitch-users] [Disarmed] Re: Freeswitch is crashing.
X-ECN Telecoms-MailScanner-Information: Contact ECN Telecoms X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-SpamCheck: not spam, SpamAssassin (not cached, score=-100.001, required 6, autolearn=not spam, NO_RELAYS -0.00, USER_IN_WHITELIST -100.00) X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED] X-Spam-Status: No This doesn't help, the stack doesn't show enough :(. Does the Java code start running? If so can you remote debug it and tell which command crashes? Damjan Here is the content of hs_err_pid9129.log *---**---**---**---**---* --- T H R E A D --- Current thread is native thread siginfo:si_signo=SIGSEGV: si_errno=0, si_code=1 (SEGV_MAPERR), si_addr=0x0008 Registers: RAX=0x, RBX=0x408a4dd0, RCX=0x6c7070615f746e65, RDX=0x7f020c42ee58 RSP=0x408a4db8, RBP=0x0078f3a8, RSI=0x0001, RDI=0x0008 R8 =0x7f0204030b90, R9 =0x0007, R10=0x, R11=0x R12=0x7f01fe5c33e0, R13=0x7f0204037e58, R14=0x7f0204037e58, R15=0x7f020cd9d270 RIP=0x7f020c4281ea, EFL=0x00010246, CSGSFS=0x0033, ERR=0x0006 TRAPNO=0x000e Top of Stack: (sp=0x408a4db8) 0x408a4db8: 7f020ccd3a68 7f0204037e58 0x408a4dc8: 7f01fe5c33e0 0x408a4dd8: 7f0204037e58 0078f3a8 0x408a4de8: 7f0204037e50 7f01fe5c33e0 0x408a4df8: 7f020ccd3dda 7f020cd9c9c0 0x408a4e08: 7f0204037e50 7f0204037908 0x408a4e18: 7f0204037e50 7f0204037e58 0x408a4e28: 7f0204037de0 7f0204037e50 0x408a4e38: 7f020a7860e0 7f020cd9d600 0x408a4e48: 7f020ccd74f0 7f020cd9d414 0x408a4e58: 7f0204037908 7f0204037e50 0x408a4e68: 7f0204037e58 c1a34f96 0x408a4e78: 408a4ea0 408a4ea8 0x408a4e88: 0078f3a8 7f0204037dd8 0x408a4e98: 7f0204037dd8 7f0204037dd8 0x408a4ea8: 0x408a4eb8: 0x408a4ec8: 0x408a4ed8: 0x408a4ee8: 0x408a4ef8: 0x408a4f08: 0x408a4f18: 0x408a4f28: 0x408a4f38: 0x408a4f48: 0x408a4f58: 0x408a4f68: 0001 0x408a4f78: 7f020d437358 0x408a4f88: 408a5070 7f020d437000 0x408a4f98: 7f020cc9fbb9 0x408a4fa8: Instructions: (pc=0x7f020c4281ea) 0x7f020c4281da: 90 90 90 90 90 90 4d 31 d2 be 01 00 00 00 31 c0 0x7f020c4281ea: f0 0f b1 37 75 5c 8b 47 18 85 c0 75 5f 83 7f 14 Stack: [0x4086a000,0x408a6000], sp=0x408a4db8, free space=235k Native frames: (J=compiled Java code, j=interpreted, Vv=VM code, C=native code) C [libpthread.so.0+0xa1ea] pthread_rwlock_rdlock+0xa --- P R O C E S S --- VM state:not at safepoint (normal execution) VM Mutex/Monitor currently owned by a thread: None Heap PSYoungGen total 9408K, used 325K [0x7f01ef4b, 0x7f01eff2, 0x7f01f9c0) eden space 8128K, 4% used [0x7f01ef4b,0x7f01ef501590,0x7f01efca) from space 1280K, 0% used [0x7f01efde,0x7f01efde,0x7f01eff2) to space 1280K, 0% used [0x7f01efca,0x7f01efca,0x7f01efde) PSOldGentotal 21376K, used 0K [0x7f01da60, 0x7f01dbae, 0x7f01ef4b) object space 21376K, 0% used [0x7f01da60,0x7f01da60,0x7f01dbae) PSPermGen total 21248K, used 2271K [0x7f01d520, 0x7f01d66c, 0x7f01da60) object space 21248K, 10% used [0x7f01d520,0x7f01d5437ce8,0x7f01d66c) Dynamic libraries: 0040-00403000 r-xp 08:06 2687056 /home/adeel/programs/freeswitch/bin/freeswitch 00602000-00603000 r--p 2000 08:06 2687056 /home/adeel/programs/freeswitch/bin/freeswitch 00603000-00604000 rw-p 3000 08:06 2687056 /home/adeel/programs/freeswitch/bin/freeswitch 00604000-007c7000 rw-p 00604000 00:00 0 [heap] 400df000-400e ---p 400df000 00:00 0 400e-4011b000 rwxp 400e 00:00 0 402f9000-402fa000 ---p 402f9000
[Freeswitch-users] SDP issue receiving calls from SIP connection
Afternoon everyone, I have a bit of a problem with Freeswitch receiving RTP INVITEs from my SIP provider (tesco's internet phone ... I know SIP isn't supported technically, but it sort of works...) Freeswitch (sofia) seems to be whinging about the INVITE SDP format ... but I'm not sure why ... here's a dump of my log / trace, anyone got any clues, I've tried adding extra codecs to my list and setting the late negotiation but that doesn't seem to fix it ... (registration all works fine by the way ...) INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 CSeq: 100 INVITE From: ANOTHER_PHONE_NUMBER sip:[EMAIL PROTECTED][EMAIL PROTECTED] ;user=phone;tag=SDvke8201-b2b.31deb49 To: MY_PHONE_NUMBER sip:[EMAIL PROTECTED][EMAIL PROTECTED] ;user=phone Max-Forwards: 69 Content-Type: application/sdp Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Allow: INVITE,CANCEL,ACK,BYE Accept: application/sdp Content-Length: 840 v=0 o=- 3250588766 0 IN IP4 77.75.1.10 s=- c=IN IP4 77.75.1.10 t=0 0 m=audio 54424 RTP/AVP 18 99 101 102 103 15 104 4 105 106 107 108 125 109 100 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G.729a/8000 a=rtpmap:101 G.726-16/8000 a=rtpmap:102 G.726-24/8000 a=rtpmap:103 G.726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:104 G.723.1-H/8000 a=rtpmap:4 G723/8000 a=rtpmap:105 G.723.1-L/8000 a=rtpmap:106 G.729b/8000 a=rtpmap:107 G.723.1a-H/8000 a=rtpmap:108 G.723.1a-L/8000 a=rtpmap:125 G.nX64/8000 a=rtpmap:109 AMR/8000 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=maxptime:20 a=maxptime:30 a=ptime:20 a=ptime:30 a=X-cap: 1 audio RTP/AVP 100 a=X-cap: 2 image udptl t38 a=X-sqn:0 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 tport_deliver(0x80fcbf0): msg 0x8129468 (1416 bytes) from udp/ 77.75.1.10:5060/sip next=(nil) nta: received INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 (CSeq 100) nta: canonizing sip:[EMAIL PROTECTED]:5060 with contact nta: INVITE (100) going to a default leg nta: timer set to 200 ms nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x80ef908, 0x80eccb8, 0x8120fe0) called soa_set_params(static::0x8111d08, ...) called nta_leg_tcreate(0x81200c0) soa_init_offer_answer(static::0x8111d08) called soa_set_remote_sdp(static::0x8111d08, (nil), 0x810ed20, 840) called nua(0x8120fe0): INVITE server: error parsing SDP nua: nua_invite_server_respond: entering tport_tsend(0x80fcbf0) tpn = UDP/77.75.1.10:5060 tport_resolve addrinfo = 77.75.1.10:5060 tport_by_addrinfo(0x80fcbf0): not found by name UDP/77.75.1.10:5060 tport_vsend returned 661 send 661 bytes to udp/[77.75.1.10]:5060 at 13:16:28.358779: SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 From: ANOTHER_PHONE_NUMBER sip:[EMAIL PROTECTED][EMAIL PROTECTED] ;user=phone;tag=SDvke8201-b2b.31deb49 To: MY_PHONE_NUMBER sip:[EMAIL PROTECTED][EMAIL PROTECTED] ;user=phone;tag=yjay36ZrycNmD Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 CSeq: 100 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9235 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk Content-Length: 0 nta: sent 400 Bad Session Description for INVITE (100) nta_leg_destroy(0x81200c0) soa_destroy(static::0x8111d08) called tport_wakeup_pri(0x80fcbf0): events IN tport_recv_event(0x80fcbf0) tport_recv_iovec(0x80fcbf0) msg 0x8111d08 from (udp/192.168.1.10:5060) has 442 bytes, veclen = 1 recv 442 bytes from udp/[77.75.1.10]:5060 at 13:16:28.377628: ACK sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1 CSeq: 100 ACK Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022 From: ANOTHER_PHONE_NUMBER sip:[EMAIL PROTECTED][EMAIL PROTECTED] ;user=phone;tag=SDvke8201-b2b.31deb49 To: MY_PHONE_NUMBER sip:[EMAIL PROTECTED][EMAIL PROTECTED] ;user=phone;tag=yjay36ZrycNmD Max-Forwards: 69 Content-Length: 0 tport_deliver(0x80fcbf0): msg 0x8111d08 (442
Re: [Freeswitch-users] Freeswitch is crashing.
This is already fixed in the latest SVN. The mod_java was using a older method of initialization and we forgot to bring it up to date. On Mon, Aug 11, 2008 at 3:26 AM, Adeel Ansari [EMAIL PROTECTED] wrote: Below is the error, I am getting, while running my Java program through Freeswitch dial plan. 2008-08-11 16:25:33 [NOTICE] switch_channel.c:1406 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/internal/ [EMAIL PROTECTED] 2008-08-11 16:25:33 [NOTICE] mod_sofia.c:1116 sofia_receive_message() Pre-Answer sofia/internal/[EMAIL PROTECTED] # # An unexpected error has been detected by Java Runtime Environment: # # SIGSEGV (0xb) at pc=0x7f020c4281ea, pid=9129, tid=1082808656 # # Java VM: Java HotSpot(TM) 64-Bit Server VM (11.0-b12 mixed mode linux-amd64) # Problematic frame: # C [libpthread.so.0+0xa1ea] pthread_rwlock_rdlock+0xa # # An error report file with more information is saved as: # /home/adeel/programs/freeswitch/bin/hs_err_pid9129.log # # If you would like to submit a bug report, please visit: # http://java.sun.com/webapps/bugreport/crash.jsp # Aborted (core dumped) --- Any idea, if this is because of 64bit? Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ 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
Re: [Freeswitch-users] SDP issue receiving calls from SIP connection
Michael, Great :( so probably something I'd need to mod the sofia source for really then. Not much chance of me getting Tesco (technically its a rebadge of freshtel / voicedot) to change their server so its compliant :) Cheers Kirk 2008/8/11 Michael Jerris [EMAIL PROTECTED] The issue from what I can see in the trace is the start of the s and o lines. We saw this before in a slightly different variant where those lines had extra whitespace in them after the =, this is probably the same thing, illegal chars after the =. Mike On Aug 11, 2008, at 11:09 AM, Kirk Bateman wrote: ldn't see anything specific that it was complaining about ... I've looked at the source and not figured it out yet... (really must try and memorize the spec someday). I was wondering if it was something to do with the X-NSE bit (dtmf tones extension to rfc ??) but given that I've set it to late negotiation I wouldn't expect the SDP parser to complain about that. I'm hoping the sofia dev can point me in the right direction. I have since I wrote the original mail managed to test (without any real changes) that I can make outgoing calls using the console originate command, that worked (no audio but I expected that). Cheers Kirk ___ 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 ___ 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
[Freeswitch-users] Conferencing performance
Hi list, I'm impressed by the quality of FS and I'm considering to suggest our company to use it for the conference platform. So, since I don't have currently a way to stress it, the concerns are: Q1: how is the quality under load? Q2 what are the limits of a conference bridge? Usually we get from 200 to 500 people in a system. could it be done? Q3 what about echo cancellation? Is that present on FS or is not an issue? (related to Q.1) Thank you! Fernando Gregianin Testa Voice Technnology Ltda ___ 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
[Freeswitch-users] GSM gateway MV370
I have a MV370 GSM gateway from Portech using in *.
The extension number of the gateway is 2001
To dial via the gateway I use a prefix 9.
;GSM VIA MV-370 GATEWAY)
exten = _9X.,1,Dial(SIP/2001/${EXTEN:1})
exten = _9X.,n,Congestion()
This works well.
Now I'm in the process of switching that gateway to a FS server.
The gateway is also registered as number 2001 in FS.
How do I translate the * dialplan to FS? (the two lines of code above)
Which xml file ? (default.xml ?)
Thanks
Henk
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Re: [Freeswitch-users] Call broadcasting
Ruchir, I have been using the event socket with good success. It's too much to discuss right here so I will try and start a wiki page on it. -MC Sent from my iPhone On Aug 10, 2008, at 2:03 PM, Ruchir Brahmbhatt [EMAIL PROTECTED] wrote: Hi, Which is the best method of doing call broadcasting in freeswitch? In asterisk we can do this by creating call files and asterisk handles the rest. How does freeswitch offers this feature? Do we need to use mod_event_socket only? Is it faster enough for this requirement? ___ 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 ___ 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
Re: [Freeswitch-users] RIngback and sample rate
fixed in tree, try now On Sun, Aug 10, 2008 at 11:19 PM, Michael Jerris [EMAIL PROTECTED] wrote: We discussed this a while back and could not come up with any compelling use cases where this should happen with the exception of configuration error. Do you have one? Mike On Aug 10, 2008, at 9:57 PM, Dome Charoenyost [EMAIL PROTECTED] wrote: On Mon, Aug 11, 2008 at 8:27 AM, Brian West [EMAIL PROTECTED] wrote: It should still play it and resample it to 8000. Thanks. if wav file not found i need FS still make call is posible to do ? now FS hangup /b On Aug 10, 2008, at 8:25 PM, Dome Charoenyost wrote: Dear All, Is posible ro play 32000 bit wav for ringback to gsm codec channel ? I try and got message and shoppy sound 2008-08-11 08:27:54 [DEBUG] switch_ivr_originate.c:1101 switch_ivr_originate() Play Ringback File [/home/ring.wav] 2008-08-11 08:27:54 [WARNING] switch_core_file.c:111 switch_core_perform_file_open() Sample rate doesn't match Dome C. ___ 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 ___ 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 ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ 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
[Freeswitch-users] AppGen or GUI tool for FS ?
Hi FS community, I remember, that GUI issue, and actually many other FS control access issues are of high priority, and seems there is something what might/should help: http://adhearsion.com/ It is not straitforward since it was made with Asterisk in mind, but I believe it shouldn't be big issue to port it to FS too. I just trust in Anthony The Second persuasion and influence to convince Adhearsion authors/creators to make it also FS compatible - if not directly, then maybe they can open some API or other method of integration, and FS community directly, or via bounties, will bring it to reality. Its the first thought, before reading any further about Adhearsion. What are other opinions ? /\/\arekg ___ 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
Re: [Freeswitch-users] RIngback and sample rate
I remove retun error on mod_sndfile and recompile it's work for me. Now i'm testing other feature. i think FS is excellent :) I will add Thai lang support and send back to svn. but may be only me use FS in Thailand :) Dome C. On Tue, Aug 12, 2008 at 3:41 AM, Anthony Minessale [EMAIL PROTECTED] wrote: fixed in tree, try now On Sun, Aug 10, 2008 at 11:19 PM, Michael Jerris [EMAIL PROTECTED] wrote: We discussed this a while back and could not come up with any compelling use cases where this should happen with the exception of configuration error. Do you have one? Mike On Aug 10, 2008, at 9:57 PM, Dome Charoenyost [EMAIL PROTECTED] wrote: On Mon, Aug 11, 2008 at 8:27 AM, Brian West [EMAIL PROTECTED] wrote: It should still play it and resample it to 8000. Thanks. if wav file not found i need FS still make call is posible to do ? now FS hangup /b On Aug 10, 2008, at 8:25 PM, Dome Charoenyost wrote: Dear All, Is posible ro play 32000 bit wav for ringback to gsm codec channel ? I try and got message and shoppy sound 2008-08-11 08:27:54 [DEBUG] switch_ivr_originate.c:1101 switch_ivr_originate() Play Ringback File [/home/ring.wav] 2008-08-11 08:27:54 [WARNING] switch_core_file.c:111 switch_core_perform_file_open() Sample rate doesn't match Dome C. ___ 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 ___ 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 ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED] pstn:213-799-1400 ___ 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 ___ 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
Re: [Freeswitch-users] AppGen or GUI tool for FS ? - cancel question mark
Maybe I should read first ? http://adhearsion.pbwiki.com/FreeSwitchHelloWorld Anyhow - opinions ? Hi FS community, I remember, that GUI issue, and actually many other FS control access issues are of high priority, and seems there is something what might/should help: http://adhearsion.com/ It is not straitforward since it was made with Asterisk in mind, but I believe it shouldn't be big issue to port it to FS too. I just trust in Anthony The Second persuasion and influence to convince Adhearsion authors/creators to make it also FS compatible - if not directly, then maybe they can open some API or other method of integration, and FS community directly, or via bounties, will bring it to reality. Its the first thought, before reading any further about Adhearsion. What are other opinions ? /\/\arekg ___ 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 ___ 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
Re: [Freeswitch-users] DTMF recognition and nbound calls
To all those concerned…and I know there were many of you, my problem is solved. I started from scratch basically… Because I am just learnt many things…obviously while I was still learning I must have changed a setting that caused the IVR to not work properly… Having said that I don’t know what I did… So I re-installed. Re instated all my extensions and pstn code…and now I have dtmf recognition… Ilan Perez From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael S Collins Sent: 08 August 2008 15:00 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF recognition and nbound calls I came late to the party. Could you recap what you are trying to do with the digit that is received? Is it an ivr? -MC Sent from my iPhone On Aug 7, 2008, at 9:33 PM, Ilan Perez [EMAIL PROTECTED] wrote: You got my last message…about the fact that I can see in the log that the dtmf is recognized by the system…but the system takes no action when the key is hit… Ilan Perez From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: 08 August 2008 11:18 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF recognition and nbound calls What kind of device? You shouldn't need to have the detection app in that case something else must be wrong. On Aug 7, 2008, at 6:37 PM, Ilan Perez wrote: Yes, Darren it is an analog line that is connected… ___ 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 ___ 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
Re: [Freeswitch-users] Can we have a forum
I am coming in a little late on this conversation but you can have both with keeping the existing mailman system. I added freeswitch-users to Nabble groups back in March. The same thing can be done to freeswitch-dev and we might be able to import the even older freeswitch-users data. For those not familiar Nabble ( http://www.nabble.com ) allows anyone to create a web forum based on an existing mailing list. After you signup you can input your existing mailing list auth info and when you post to the freeswitch-users forum it sends it to the mailing list and keeps track of the posting status. Here is a link to the freeswitch list: http://www.nabble.com/Freeswitch-users-f32209.html This is a works for me solution that might appease everyone. Brian S/bsnipes On Sunday 10 August 2008 6:49:32 pm Ilan Perez wrote: I am in with the forum. For whatever reason people don't like them. Voip-info.org is working use it and people will start asking more questions on it and then it will be more popular etc.. My 2 cents Ilan Perez From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Holtsclaw Sent: 11 August 2008 07:36 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Can we have a forum Personally, I also prefer some sort of bulletin board / message board based discussion. Google Groups is a great idea. To me, a straight up mailman mailing list just seems old school. I agree with Darren in that something more easily searchable would be a big plus. On 8/10/2008 at 4:54 PM, Brian West [EMAIL PROTECTED] wrote: Oh btw I have talked about using google groups which offers both in one package What does everyone thing about some sort of compromise like that? /b On Aug 10, 2008, at 3:43 PM, Lee JJ wrote: Dear Sirs : Can we have a forum like phpbb or punbb . thanks ___ ___ 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
Re: [Freeswitch-users] Conferencing performance
Just testing one I had 23 active calls in a conference ( half SIP and half PSTN ). This was with real people ( desk phones and cell phones ). Brian S/bsnipes On Monday 11 August 2008 10:30:04 am Fernando Testa wrote: Hi list, I'm impressed by the quality of FS and I'm considering to suggest our company to use it for the conference platform. So, since I don't have currently a way to stress it, the concerns are: Q1: how is the quality under load? Q2 what are the limits of a conference bridge? Usually we get from 200 to 500 people in a system. could it be done? Q3 what about echo cancellation? Is that present on FS or is not an issue? (related to Q.1) Thank you! Fernando Gregianin Testa Voice Technnology Ltda ___ 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 ___ 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
Re: [Freeswitch-users] Freeswitch is crashing.
It worked, Anthony. Thank you for the help. Cheers. On Mon, Aug 11, 2008 at 10:31 PM, Anthony Minessale [EMAIL PROTECTED] wrote: This is already fixed in the latest SVN. The mod_java was using a older method of initialization and we forgot to bring it up to date. On Mon, Aug 11, 2008 at 3:26 AM, Adeel Ansari [EMAIL PROTECTED]wrote: Below is the error, I am getting, while running my Java program through Freeswitch dial plan. 2008-08-11 16:25:33 [NOTICE] switch_channel.c:1406 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/internal/ [EMAIL PROTECTED] 2008-08-11 16:25:33 [NOTICE] mod_sofia.c:1116 sofia_receive_message() Pre-Answer sofia/internal/[EMAIL PROTECTED] # # An unexpected error has been detected by Java Runtime Environment: # # SIGSEGV (0xb) at pc=0x7f020c4281ea, pid=9129, tid=1082808656 # # Java VM: Java HotSpot(TM) 64-Bit Server VM (11.0-b12 mixed mode linux-amd64) # Problematic frame: # C [libpthread.so.0+0xa1ea] pthread_rwlock_rdlock+0xa # # An error report file with more information is saved as: # /home/adeel/programs/freeswitch/bin/hs_err_pid9129.log # # If you would like to submit a bug report, please visit: # http://java.sun.com/webapps/bugreport/crash.jsp # Aborted (core dumped) --- Any idea, if this is because of 64bit? Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ 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 -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari ___ 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
