[Freeswitch-users] Sangoma A500 - dial out from specific port group?
Hi,
Is it possible to originate calls from specific A500 ports with FreeSWITCH?
I am using a A504 (8 BRI interfaces), and I want some outbound calls to be
made from specific BRI interfaces.
I tried to modify OpenZAP config as follows:
conf/openzap.conf
[span wanpipe boostbri1]
trunk_type = bri
b-channel = 1:1-2
b-channel = 2:1-2
b-channel = 3:1-2
b-channel = 4:1-2
b-channel = 5:1-2
b-channel = 6:1-2
[span wanpipe boostbri2]
trunk_type = bri
b-channel = 7:1-2
b-channel = 8:1-2
conf/autoload_configs/openzap.conf.xml:
boost_spans
span name=boostbri1
!--param name=hold-music value=$${moh_uri}/--
!--param name=enable-analog-option value=call-swap/--
!--param name=enable-analog-option value=3-way/--
param name=dialplan value=XML/
param name=context value=isdn/
/span
span name=boostbri2
param name=dialplan value=XML/
param name=context value=isdn/
/span
/boost_spans
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Re: [Freeswitch-users] Sangoma A500 - dial out from specific port group?
Sorry I hit 'send' by mistake... Hi, Is it possible to originate calls from specific A500 ports with FreeSWITCH? I am using a A504 (8 BRI interfaces), and I want some outbound calls to be made from specific BRI interfaces. I tried to modify OpenZAP config as follows: conf/openzap.conf [span wanpipe boostbri1] trunk_type = bri b-channel = 1:1-2 b-channel = 2:1-2 b-channel = 3:1-2 b-channel = 4:1-2 b-channel = 5:1-2 b-channel = 6:1-2 [span wanpipe boostbri2] trunk_type = bri b-channel = 7:1-2 b-channel = 8:1-2 conf/autoload_configs/openzap.conf.xml: boost_spans span name=boostbri1 param name=dialplan value=XML/ param name=context value=isdn/ /span span name=boostbri2 param name=dialplan value=XML/ param name=context value=isdn/ /span /boost_spans When I try to originate call I am getting errors: freeswi...@emo-voip originate openzap/2/a/123456 music API CALL [originate(openzap/2/a/123456 music)] output: -ERR NORMAL_CIRCUIT_CONGESTION 2009-09-04 09:23:34.87253 [CRIT] ozmod_ss7_boost.c:244 SPAN is not online. 2009-09-04 09:23:34.87253 [ERR] mod_openzap.c:1043 No channels available 2009-09-04 09:23:34.87253 [ERR] switch_ivr_originate.c:1510 Cannot create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION] Then I tried to modify the /etc/wanpipe/smg_bri.conf: ;Sangoma AFT-A500 port 11 [slot:8 bus:1 span:7] group=2 country=europe operator=etsi connection_type=point_to_point signalling=bri_nt spans=7 ;Sangoma AFT-A500 port 12 [slot:8 bus:1 span:8] group=2 country=europe operator=etsi connection_type=point_to_point signalling=bri_nt spans=8 i.e. changed 'group' to 2, but this doesn't help either. Marc Celsie from Sangoma's techdesk told me that I should ' dial x...@gy with X being the number and Y being the group number'. How originate command should look like in this case? originate openzap/1/a/123...@g2 someExt ? I tried this syntax but with no effect. Marc also told me that there is a bug in FS which prevents groups from working. Should I fill bug report or feature request? Best regards, Vassil Panayotov ___ 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] mod_fax not working
Hello, On Tue, Sep 1, 2009 at 1:06 PM, Mathieu Parentmath.par...@gmail.com wrote: Hi, On Fri, Aug 28, 2009 at 7:01 PM, Steve Underwoodste...@coppice.org wrote: (snip) The log shows the same thing happening every time. A bad CRC from the far end, followed by a good DCS frame followed by what seems to be rubbish. I think I'd need an audio log from one of these calls to figure out any more. I have attached a pcap file only with SIP and RTP. I still have the same problem. Anyone can analyse the traces ? Note that it works without problems when sending a fax. Mathieu Parent ___ 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] memory leak
Hello *, the latest bugfixes for luarun (pool allocation) fixed it. It's working now with luarun (friday-monday) and bgapi (monday-today). The last test with the new operator for sched_api is currently running. Thx! Beni. attachment: mem-sz_luarun_bgapi.png___ 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] No audio on caller side when both side support speex/8000 only
Hi, Owe to the network bandwidth limitations (running on cellular phones ip link) we are using speex/8000 as our voice codec. However, when both parties are using that codec the sound is not to be heard on the caller side. looking at the log dumps one can see that a) at the caller side, it supports speex/8000 in pt=102 and receives from the server speex/8000 in pt=102 b) at the callee side FreeSwitch supports support speex/8000 in pt=98 although it receives from the client speex/8000 in pt=102 When the voice starts caller sends RTP with pt=102 and expect to receive RTP with pt=102, while the callee sends RTP with pt=98 and expect to receive RTP with pt=102. The RTP packets that received in the caller side are with pt=98 instead of 102 and thusly the client drops them. ### LOG DUMPS ### ### CALLER SIDE ## start msg (TX) INVITE sip:1...@server_domain SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:64680;rport;branch=z9hG4bKPj3caad40720064a8f124c21cf99b8b1c1 Max-Forwards: 70 From: sip:1...@server_domain;tag=6d6ef114e663e48226f2b1e598313a2e To: sip:1...@server_domain Contact: sip:1...@x.x.x.x:64680 Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23264 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 264 v=0 o=- 3461521040 3461521040 IN IP4 X.X.X.X s=pjmedia c=IN IP4 X.X.X.X t=0 0 a=X-nat:8 m=audio 64976 RTP/AVP 102 101 a=rtcp:64980 IN IP4 X.X.X.X a=rtpmap:102 speex/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- ## start msg (RX) SIP/2.0 200 OK Via: SIP/2.0/UDP 10.171.9.67:5060;rport=64680;branch=z9hG4bKPje8deea3c8476457603fe0cc301731002 From: sip:1...@server_domain;tag=6d6ef114e663e48226f2b1e598313a2e To: sip:1...@server_domain;tag=06ym4113FFcHQ Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23265 INVITE Contact: sip:mod_so...@x.x.x.x:5060;transport=udp User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 1800;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 268 v=0 o=FreeSWITCH 4446028933093139022 3405868075899860026 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 X.X.X.X t=0 0 m=audio 26664 RTP/AVP 102 101 a=rtpmap:102 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 --end msg-- ### CALLEE SIDE ## start msg (RX) INVITE sip:1...@x.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;rport;branch=z9hG4bKgvD702De7e0Se Max-Forwards: 69 From: Extension 1001 sip:1...@x.x.x.x;tag=2rH67Q3aa1rpe To: sip:1...@x.x.x.x:5060 Call-ID: e56c2918-17ad-122d-de9e-40402384297d CSeq: 120120747 INVITE Contact: sip:mod_so...@x.x.x.x:5060 User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 415 Remote-Party-ID: Extension 1001 sip:1...@x.x.x.x;screen=yes;privacy=off v=0 o=FreeSWITCH 413181116427886 953315150658749217 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 X.X.X.X t=0 0 m=audio 26662 RTP/AVP 98 0 8 3 9 99 103 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 SPEEX/16000 a=rtpmap:103 SPEEX/32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 --end msg-- ## start msg (TX) SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X;rport=5060;received=X.X.X.X;branch=z9hG4bKgvD702De7e0Se Call-ID: e56c2918-17ad-122d-de9e-40402384297d From: Extension 1001 sip:1...@x.x.x.x;tag=2rH67Q3aa1rpe To: sip:1...@x.x.x.x;tag=e4f9fb648edef1c48dbc8b8b474409e6 CSeq: 120120747 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: sip:X.X.X.X:5060 Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 256 v=0 o=- 3461503025 3461503026 IN IP4 X.X.X.X s=pjmedia c=IN IP4 X.X.X.X t=0 0 a=X-nat:5 m=audio 4000 RTP/AVP 102 101 a=rtcp:4001 IN IP4 X.X.X.X a=rtpmap:102 speex/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- Attached are the 2 files recorded from a call between 2 pjsip clients that support only speex/8000 codec. un_FSCallerSide-speexClient.TXT – is the caller side SIP messages. un_FSAnswerSide-speexClient.TXT – is the answer side of SIP
[Freeswitch-users] No audio on caller side when both side support speex/8000 only
Hi, Owe to the network bandwidth limitations (running on cellular phones ip link) we are using speex/8000 as our voice codec. However, when both parties are using that codec the sound is not to be heard on the caller side. looking at the log dumps one can see that a) at the caller side, it supports speex/8000 in pt=102 and receives from the server speex/8000 in pt=102 b) at the callee side FreeSwitch supports support speex/8000 in pt=98 although it receives from the client speex/8000 in pt=102 When the voice starts caller sends RTP with pt=102 and expect to receive RTP with pt=102, while the callee sends RTP with pt=98 and expect to receive RTP with pt=102. The RTP packets that received in the caller side are with pt=98 instead of 102 and thusly the client drops them. Attached are the 2 files recorded from a call between 2 pjsip clients that support only speex/8000 codec. un_FSCallerSide-speexClient.TXT – is the caller side SIP messages. un_FSAnswerSide-speexClient.TXT – is the answer side of SIP messages. Is there anything can be done at the configuration level to avoid this? Thanks in advance for your help /tzury --start msg (RX)-- INVITE sip:1...@95.35.241.89:5060 SIP/2.0 Via: SIP/2.0/UDP 67.23.5.142;rport;branch=z9hG4bKgvD702De7e0Se Max-Forwards: 69 From: Extension 1001 sip:1...@67.23.5.142;tag=2rH67Q3aa1rpe To: sip:1...@95.35.241.89:5060 Call-ID: e56c2918-17ad-122d-de9e-40402384297d CSeq: 120120747 INVITE Contact: sip:mod_so...@67.23.5.142:5060 User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 415 Remote-Party-ID: Extension 1001 sip:1...@67.23.5.142;screen=yes;privacy=off v=0 o=FreeSWITCH 413181116427886 953315150658749217 IN IP4 67.23.5.142 s=FreeSWITCH c=IN IP4 67.23.5.142 t=0 0 m=audio 26662 RTP/AVP 98 0 8 3 9 99 103 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 SPEEX/16000 a=rtpmap:103 SPEEX/32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 --end msg-- --start msg (TX)-- SIP/2.0 100 Trying Via: SIP/2.0/UDP 67.23.5.142;rport=5060;received=67.23.5.142;branch=z9hG4bKgvD702De7e0Se Call-ID: e56c2918-17ad-122d-de9e-40402384297d From: Extension 1001 sip:1...@67.23.5.142;tag=2rH67Q3aa1rpe To: sip:1...@95.35.241.89 CSeq: 120120747 INVITE Content-Length: 0 --end msg-- --start msg (TX)-- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 67.23.5.142;rport=5060;received=67.23.5.142;branch=z9hG4bKgvD702De7e0Se Call-ID: e56c2918-17ad-122d-de9e-40402384297d From: Extension 1001 sip:1...@67.23.5.142;tag=2rH67Q3aa1rpe To: sip:1...@95.35.241.89;tag=e4f9fb648edef1c48dbc8b8b474409e6 CSeq: 120120747 INVITE Contact: sip:95.35.241.89:5060 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 --end msg-- --start msg (RX)-- INVITE sip:1...@95.35.241.89:5060 SIP/2.0 Via: SIP/2.0/UDP 67.23.5.142;rport;branch=z9hG4bKgvD702De7e0Se Max-Forwards: 69 From: Extension 1001 sip:1...@67.23.5.142;tag=2rH67Q3aa1rpe To: sip:1...@95.35.241.89:5060 Call-ID: e56c2918-17ad-122d-de9e-40402384297d CSeq: 120120747 INVITE Contact: sip:mod_so...@67.23.5.142:5060 User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 415 Remote-Party-ID: Extension 1001 sip:1...@67.23.5.142;screen=yes;privacy=off v=0 o=FreeSWITCH 413181116427886 953315150658749217 IN IP4 67.23.5.142 s=FreeSWITCH c=IN IP4 67.23.5.142 t=0 0 m=audio 26662 RTP/AVP 98 0 8 3 9 99 103 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 SPEEX/16000 a=rtpmap:103 SPEEX/32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 --end msg-- --start msg (TX)-- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 67.23.5.142;rport=5060;received=67.23.5.142;branch=z9hG4bKgvD702De7e0Se Call-ID: e56c2918-17ad-122d-de9e-40402384297d From: Extension 1001 sip:1...@67.23.5.142;tag=2rH67Q3aa1rpe To: sip:1...@95.35.241.89;tag=e4f9fb648edef1c48dbc8b8b474409e6 CSeq: 120120747 INVITE Contact: sip:95.35.241.89:5060 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 --end msg-- --start msg (TX)-- SIP/2.0 200 OK Via: SIP/2.0/UDP
Re: [Freeswitch-users] No audio on caller side when both side support speex/8000 only
This looks and sounds like a case where pjsip isn't listening to our SDP. If we 200 OK with speex on 102 and the far end starts sending it on 98 then I suspect the client is broken if I'm not mistaken. /b On Sep 9, 2009, at 6:19 AM, Tzury Bar Yochay wrote: Hi, Owe to the network bandwidth limitations (running on cellular phones ip link) we are using speex/8000 as our voice codec. However, when both parties are using that codec the sound is not to be heard on the caller side. looking at the log dumps one can see that a) at the caller side, it supports speex/8000 in pt=102 and receives from the server speex/8000 in pt=102 b) at the callee side FreeSwitch supports support speex/8000 in pt=98 although it receives from the client speex/8000 in pt=102 When the voice starts caller sends RTP with pt=102 and expect to receive RTP with pt=102, while the callee sends RTP with pt=98 and expect to receive RTP with pt=102. The RTP packets that received in the caller side are with pt=98 instead of 102 and thusly the client drops them. Attached are the 2 files recorded from a call between 2 pjsip clients that support only speex/8000 codec. un_FSCallerSide-speexClient.TXT – is the caller side SIP messages. un_FSAnswerSide-speexClient.TXT – is the answer side of SIP messages. Is there anything can be done at the configuration level to avoid this? Thanks in advance for your help /tzury un_FSAnswerSide-speexClient.TXTun_FSCallerSide- speexClient.TXT___ 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] example configs for FS outside of NAT?
Hi there, the internal.xml and external.xml examples are for situations where FS is running inside a company's private network, behind a NAT router. So internal.xml connects the clients to FS without crossing a NAT, within the same private network, while external.xml connects SIP providers through the NAT router. But what if FS is running with a public IP (and DNS entry) outside the private network, so that the clients have to pass the NAT router to connect with FS, while FS can connect to SIP providers directly? Are there any example configs for such a configuration? Thanks in advance, Cheers, JH ___ 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] auto_hunt=true vs execute_extenstion
Hi all! Is there any difference between auto_hunt=True and execute_extenstion? ___ 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] auto_hunt=true vs execute_extenstion
So if you have an extension name that is testing and the destination number is testing then if testing is at the bottom of the dialplan auto_hunt will make it warp right to it. /b On Sep 9, 2009, at 8:29 AM, Max Ivanov wrote: Hi all! Is there any difference between auto_hunt=True and execute_extenstion? ___ 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] example configs for FS outside of NAT?
Those configs will still work. /b On Sep 9, 2009, at 6:16 AM, Jörg Hartmann wrote: Hi there, the internal.xml and external.xml examples are for situations where FS is running inside a company's private network, behind a NAT router. So internal.xml connects the clients to FS without crossing a NAT, within the same private network, while external.xml connects SIP providers through the NAT router. But what if FS is running with a public IP (and DNS entry) outside the private network, so that the clients have to pass the NAT router to connect with FS, while FS can connect to SIP providers directly? Are there any example configs for such a configuration? Thanks in advance, Cheers, JH ___ 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] auto_hunt=true vs execute_extenstion
So if you have an extension name that is testing and the destination number is testing then if testing is at the bottom of the dialplan auto_hunt will make it warp right to it. Ah, I see. Would it be correct to say that auto_hunt is similar to goto and execute_extenstion behave like include ? ___ 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] stability problems
the instructions said build latest trunk. did you actually do that? because lines of code in this gcore file do not correspond to current trunk which is why I asked you to update to it first. Did you just rebuild 1.0.4 again? If you did rebuild trunk what version was it? we can't fix problems on tarball release you have to use the development version. 2009/9/9 Christian Löschenkohl christian.loeschenk...@xpirio.com hello anthony i'm sorry the cleanup didn't solve my problem i have opend a jira bug n this - key FSCORE-432 hope this is right br On 2009-09-03 17:02, Anthony Minessale wrote: Which revision are you using? If you are not running the latest trunk, please upgrade to that in case your problem requires us to change the code we need it to be up to date. 1) Remove any binary files which may get mixed in from an older build rm /usr/local/freeswitch/bin/* rm /usr/local/freeswitch/lib/* rm /usr/local/freeswitch/mod 2) Build Latest Trunk 3) Reproduce the problem. If you get the problem keep FreeSWITCH running and capture a gcore back trace. ./scripts/freeswitch-gcore gcore.txt Send us the file as an attachment or attached to a new jira issue. http://jira.freeswitch.org 2009/9/3 Christian Löschenkohl christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com on debian lenny amd64 with the build-essential package an then with ./configure --prefix=/opt/freeswitch make make install nothing else br On 2009-09-03 16:12, Brian West wrote: Sounds like you have some build skew... can you tell us how you built FreeSWITCH? /b On Sep 3, 2009, at 2:29 AM, Christian Löschenkohl wrote: hello we have regular (every 4-6 days) stability problems with freeswitch when the problme occurs - no registers are done bythe server (olny 1 ack of the initial register) - no more calls are working - the calls are all ending with a timeout (cdr caues ORIGINATOR_CANCEL) - only a restart of the whole server cures the problem the server doesn't crash or segfault my first try was to enable the crash-protection flag, but with no difference the server is restartet every night and the last stand still was after about 15h uptime the system is an sun fire 2400 with debian 64 bit system how could i offer you more information to solve this big problem br -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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 mailto: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 -- Ing. Christian Löschenkohl Technische Leitung, Forschung Entwicklung VoIP xpirio Telekommunikation Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenk...@xpirio.com mailto:christian.loeschenk...@xpirio.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.commsn%253aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
[Freeswitch-users] [ERR] mod_sofia.c:2645 Invalid Gateway
I have phones registered internally and can call among them. However, when I dial 711 from an internal phone, freeswitch replies with 484 Address Incomplete with reason INVALID_NUMBER_FORMAT. At the server console, I see the following error: [ERR] mod_sofia.c:2645 Invalid Gateway Does anyone know why I get this error? Is there something more I must do to add the gateway below? I already added the following to the usr/local/freesitch/conf/dialplan/default.xml: extension name=Testing - Mediant 1000 condition field=destination_number expression=^(711)$ action application=bridge data=sofia/gateway/mediant1000/$1/ /condition /extension I already created a usr/local/freeswitch/conf/sip_profiles/external/mediant1000.xml file: include gateway name=192.168.72.253 param name=username value=TEOGateWay/ param name=password value=ti0w...@b/ param name=register value=false/ /gateway /include Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [ERR] mod_sofia.c:2645 Invalid Gateway
Because you named your gateway 192.168.72.253, not mediant1000. You could name it mediant1000 and set param name=proxy value=192.168.72.253 /, or use sofia/gateway/192.168.72.253/... Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 9-Sep-09, at 9:30 AM, Jerry Richards wrote: I have phones registered internally and can call among them. However, when I dial 711 from an internal phone, freeswitch replies with 484 Address Incomplete with reason INVALID_NUMBER_FORMAT. At the server console, I see the following error: [ERR] mod_sofia.c:2645 Invalid Gateway Does anyone know why I get this error? Is there something more I must do to add the gateway below? I already added the following to the usr/local/freesitch/conf/dialplan/default.xml: extension name=Testing - Mediant 1000 condition field=destination_number expression=^(711)$ action application=bridge data=sofia/gateway/ mediant1000/$1/ /condition /extension I already created a usr/local/freeswitch/conf/sip_profiles/external/mediant1000.xml file: include gateway name=192.168.72.253 param name=username value=TEOGateWay/ param name=password value=ti0w...@b/ param name=register value=false/ /gateway /include Best Regards, Jerry ___ 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] filter in fs_cli
Dear All, I'm looking for document,example for /filter command. where to get it ? BG 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
Re: [Freeswitch-users] No audio on caller side when both side support speex/8000 only
This looks and sounds like a case where pjsip isn't listening to our SDP. If we 200 OK with speex on 102 and the far end starts sending it on 98 then I suspect the client is broken if I'm not mistaken. /b Could be, anyhow, note that this happens only both side using speex/8000. If one party uses a different codec the problem does not exists. Moreover, these same two clients (with speex/8000) works fine when connected to iptel.org. The most concerning fact is that a=rtpmap:98 SPEEX/8000 sent by FS to the callee even though the caller sent a=rtpmap:102 SPEEX/8000. Does this make any sense? ___ 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] Subscribing to events in managed C# / .NET
I think the problem here is that the loader only keeps this method in scope until completion then it drops the remoted connection. Therefore you should not use threads in this method. Michael please correct me if I am wrong here. As an example of the failure simply just put a Sleep(1) call in the thread and you will see the failure. As Michael said this method was only designed to allow the option to opt out of being loaded. In order to support this perhaps a configuration flag simular to the lua startup-script should be added. Here is the error I get with the loop I mentioned. -Josh [image: Capture.PNG] On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo m...@giagnocavo.netwrote: Hi, Can you please elaborate on the crash you receive when you queue a thread during load? Thanks, Michael -- View this message in context: http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html Sent from the freeswitch-users mailing list archive at Nabble.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
[Freeswitch-users] Skypiax false DTMF event
I have a problem. After 10-20 minutes of Skype talk via cordless phone connected to ATA the latter erroneously generated DTMF 'D' event. Then skypiax looses connection while the call remain active in Skype client. The only way to terminate it is to ask another party to hang up: (...) 2009-09-09 22:20:07.474051 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||CALL 307 DURATION 500||| 2009-09-09 22:20:08.473755 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||CALL 307 DURATION 501||| 2009-09-09 22:20:09.474247 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||CALL 307 DURATION 502||| 2009-09-09 22:20:10.474611 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||CALL 307 DURATION 503||| 2009-09-09 22:20:11.474456 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||CALL 307 DURATION 504||| 2009-09-09 22:20:12.411664 [DEBUG] switch_rtp.c:2239 RTP RECV DTMF D:2000 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:633 rev 14771[(nil)|37 ][DEBUG_SKYPE 633 ][interface1][-1, 5,21] interface1 CHANNEL SEND_DTMF 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:634 rev 14771[(nil)|37 ][DEBUG_SKYPE 634 ][interface1][-1, 5,21] DTMF: D 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:882 rev 14707[(nil)|37 ][DEBUG_SKYPE 882 ][interface1][-1, 5,21] DIGIT received: D 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:1352 rev 14707[(nil)|37 ][DEBUG_SKYPE 1352 ][interface1][-1, 5,21] SENDING: |||SET CALL 307 DTMF D 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:1530 rev 14707[(nil)|37 ][DEBUG_SKYPE 1530 ][interface1][-1, 5,21] Got a 'continue' XAtom without a previous 'begin'. It's value (between vertical bars) is=|||allowed call prop||| 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||ERROR 21 Unknown/dis||| 2009-09-09 22:20:12.411664 [ERR] skypiax_protocol.c:144 rev 14707[(nil)|37 ][ERRORA 144 ][interface1][-1, 5,21] Skype got ERROR: |||ERROR 21 Unknown/dis||| 2009-09-09 22:20:12.411664 [ERR] skypiax_protocol.c:146 rev 14707[(nil)|37 ][ERRORA 146 ][interface1][-1, 5,16] skype_call now is DOWN 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:1011 rev 14771[(nil)|37 ][DEBUG_SKYPE 1011 ][interface1][-1, 1,16] skype call ended 2009-09-09 22:20:12.411664 [NOTICE] mod_skypiax.c:1022 Hangup skypiax/interface1/user2 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-09-09 22:20:12.411664 [DEBUG] switch_channel.c:1715 Send signal skypiax/interface1/user2 [KILL] 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:566 rev 14771[(nil)|37 ][DEBUG_SKYPE 566 ][interface1][-1, 1,16] interface1 CHANNEL KILL_CHANNEL 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:569 rev 14771[(nil)|37 ][DEBUG_SKYPE 569 ][interface1][-1, 1,16] skypiax/interface1/user2 CHANNEL got SWITCH_SIG_KILL 2009-09-09 22:20:12.411664 [DEBUG] switch_core_session.c:932 Send signal skypiax/interface1/user2 [BREAK] 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:566 rev 14771[(nil)|37 ][DEBUG_SKYPE 566 ][interface1][-1, 1,16] interface1 CHANNEL KILL_CHANNEL 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:589 rev 14771[(nil)|37 ][DEBUG_SKYPE 589 ][interface1][-1, 1,16] skypiax/interface1/user2 CHANNEL got SWITCH_SIG_BREAK 2009-09-09 22:20:12.428590 [DEBUG] skypiax_protocol.c:670 rev 14707[(nil)|37 ][DEBUG_SKYPE 670 ][interface1][-1, 1,16] Skype incoming audio GONE 2009-09-09 22:20:12.428590 [DEBUG] mod_skypiax.c:702 rev 14771[(nil)|37 ][DEBUG_SKYPE 702 ][interface1][-1, 1,16] CHANNEL READ FALSE 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:377 skypiax/interface1/user2 ending bridge by request from read function 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:452 BRIDGE THREAD DONE [skypiax/interface1/user2] 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:454 Send signal sofia/internal/1...@192.168.121.66[BREAK] 2009-09-09 22:20:12.428590 [DEBUG] switch_core_state_machine.c:497 (skypiax/interface1/user2) State EXCHANGE_MEDIA going to sleep 2009-09-09 22:20:12.428590 [DEBUG] switch_core_state_machine.c:398 (skypiax/interface1/user2) Running State Change CS_HANGUP 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:434 (skypiax/interface1/user2) State HANGUP 2009-09-09 22:20:12.429654 [DEBUG] mod_skypiax.c:506 rev 14771[(nil)|37 ][DEBUG_SKYPE 506 ][interface1][-1, 1,16] interface1 CHANNEL HANGUP 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:46 skypiax/interface1/user2 Standard HANGUP, cause: NORMAL_CLEARING 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:434 (skypiax/interface1/user2) State HANGUP going to sleep 2009-09-09
Re: [Freeswitch-users] filter in fs_cli
On Wed, Sep 9, 2009 at 10:17 AM, Dome Charoenyost d...@tel.co.th wrote: Dear All, I'm looking for document,example for /filter command. where to get it ? This is a handy way to add filters to what you see on the fs_cli. Event sockets allow for filters and the /filter command lets you add them to your fs_cli session. Check this page for specifics: http://wiki.freeswitch.org/wiki/Mod_event_socket#filter -MC ___ 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] Subscribing to events in managed C# / .NET
kernel32.dll!77e4bef7()
Here's that call stack.
[Frames below may be incorrect and/or missing, no symbols loaded for
kernel32.dll]
kernel32.dll!77e4bef7()
msvcr80.dll!78158e89()
mscorwks.dll!79e7a17a()
mscorwks.dll!79ea0fa8()
mscorwks.dll!79ea0eff()
mscorwks.dll!79e976cc()
mscorwks.dll!79e976b3()
mscorwks.dll!79e9e3bd()
mscorwks.dll!79e970c8()
mscorwks.dll!79f782f1()
mscorwks.dll!79eaa5c5()
mscorwks.dll!79eaad29()
mscorwks.dll!79e9a15d()
mscorwks.dll!79e9a15d()
mscorwks.dll!79e7a1f1()
mscorwks.dll!79e7a1f1()
mscorwks.dll!79e7a17a()
mscorwks.dll!79e88cca()
mscorwks.dll!79e96571()
mscorwks.dll!79e965a4()
mscorwks.dll!79e965c2()
mscorwks.dll!79f59330()
mscorwks.dll!79f59492()
mscorlib.ni.dll!792d5348()
mscorlib.ni.dll!792d514f()
mscorlib.ni.dll!792d4fde()
mscorlib.ni.dll!79799714()
mscorwks.dll!79e813e4()
mscorwks.dll!79e813ec()
FreeSwitch.dll!switch_loadable_module_load_file(char * path=0x01181250,
char * filename=0x01181240, switch_bool_t global=SWITCH_FALSE,
switch_loadable_module * * new_module=0x0012d9e0) Line 846 + 0xd bytes C
FreeSwitch.dll!switch_loadable_module_load_module_ex(char *
dir=0x003994a8, char * fname=0x01081d59, switch_bool_t runtime=SWITCH_FALSE,
switch_bool_t global=SWITCH_FALSE, const char * * err=0x0012da5c) Line 942
+ 0x15 bytes C
FreeSwitch.dll!switch_loadable_module_init() Line 1174 + 0x23 bytes C
FreeSwitch.dll!switch_core_init_and_modload(unsigned int flags=129,
switch_bool_t console=SWITCH_TRUE, const char * * err=0x0012fdec) Line 1469
+ 0x5 bytes C
FreeSwitch.exe!main(int argc=1, char * * argv=0x00394f80) Line 748 + 0x23
bytes C
FreeSwitch.exe!__tmainCRTStartup() Line 586 + 0x19 bytes C
FreeSwitch.exe!mainCRTStartup() Line 403 C
kernel32.dll!77e6f23b()
The breakpoint is: status = load_func_ptr(module_interface, pool);
Line 846 in switch_loadable_module.c
--Josh
On Tue, Sep 8, 2009 at 10:50 PM, Josh Rivers j...@radianttiger.com wrote:
I'm running of the binary release, so I don't have debug symbols for the
freeswitch core. I can do a build...but does somebody else already have one
handy? -Josh
On Tue, Sep 8, 2009 at 10:33 PM, Mathieu Rene mrene_li...@avgs.ca wrote:
Click Break, then go in Window, Debug, Stack Trace (or something similar,
I don't have any VS nearby), then copy paste that.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 8-Sep-09, at 10:30 PM, Josh Rivers wrote:
Here is the error I get with the loop I mentioned. -Josh
Capture.PNG
On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo
m...@giagnocavo.netwrote:
Hi,
Can you please elaborate on the crash you receive when
you queue a thread during load?
Thanks,
Michael
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Josh
Rivers
*Sent:* Tuesday, September 08, 2009 12:22 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# /
.NET
Thanks for the response!
I have tried putting a long-running loop here, but then it blocks
anything else managed from happening:
public class TestLoop : ILoadNotificationPlugin
{
public bool Load()
{
EventConsumer con = new EventConsumer(all, );
while (true)
{
Event ev = con.pop(0);
Log.WriteLine(LogLevel.Notice, Event: +
ev.serialized_string);
freeswitch.msleep(100);
}
}
}
However, if I fork off a thread here, freeswitch crashes:
public class TestLoop : ILoadNotificationPlugin
{
public bool Load()
{
ThreadPool.QueueUserWorkItem((o) =
{
Log.WriteLine(LogLevel.Notice, Thread Starting. );
EventConsumer con = new EventConsumer(all, );
while (true)
{
Event ev = con.pop(0);
Log.WriteLine(LogLevel.Notice, Event: +
ev.serialized_string);
freeswitch.msleep(100);
}
});
return true;
}
}
It doesn't look like this is a good place to start a long-running
process?
Thanks!
Josh
On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi
raffaele.p.gu...@gmail.com wrote:
Yes!
public class LoadDemo : ILoadNotificationPlugin {
public bool Load() {
Log.WriteLine(LogLevel.Notice, LoadDemo running.);
return true;
}
}
this example is from Michael Giagnocavo's Demo.csx which you can find
into the mod_managed svn.
And let me add that works like a charm :)
Ciao,
Raffaele
On Sun, Sep 6, 2009 at 22:50, Josh Rivers j...@radianttiger.com wrote:
Is there a way to start this when
Re: [Freeswitch-users] filter in fs_cli
On Wed, Sep 9, 2009 at 1:36 PM, Michael Collins m...@freeswitch.org wrote: On Wed, Sep 9, 2009 at 10:17 AM, Dome Charoenyost d...@tel.co.th wrote: Dear All, I'm looking for document,example for /filter command. where to get it ? This is a handy way to add filters to what you see on the fs_cli. Event sockets allow for filters and the /filter command lets you add them to your fs_cli session. Check this page for specifics: http://wiki.freeswitch.org/wiki/Mod_event_socket#filter -MC Also, I forgot to mention that this is used in conjunction with the /event' command. Open fs_cli and execute these commands: /log 0 /event plain all At this point you will get no log messages and just events. Now you can filter them as needed. Example: /filter Event-Name CHANNEL_EXECUTE Have fun! -MC ___ 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] Call Forwarding Question
On Tue, Sep 8, 2009 at 1:20 PM, Nikolai Geordzhev n.geordz...@gmail.comwrote: I`ve already tried the legs variable in cdr_csv.conf.xml, I have also tried to use the loopback endpoint and to bridge the call to the internal interface(so it can go out and in again generating the 2cdr-s I need) and still haven`t achieved any success. Can anyone please share some experience in doing CallForwarding in FreeSwitch. I beleive I`m not the only guy tryiig to achieve this, what`s the Best Practices for this task? Nik, Can you pastebin your dialplan where you do this? I'd like to see what you're doing and perhaps see if I can duplicate your scenario for testing. Thanks, MC ___ 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] Subscribing to events in managed C# / .NET
A new discovery:public bool Load()
{
ThreadPool.QueueUserWorkItem((o) =
{
Log.WriteLine(LogLevel.Notice, Thread Starting. );
EventConsumer con = new EventConsumer(all, );
while (true)
{
Event ev = con.pop(0);
if (ev == null) continue;
Log.WriteLine(LogLevel.Notice, Event: +
ev.serialized_string);
}
});
return true;
}
Does not crash. (Adding the null check prevents crash.) The backgrounded
loop runs fine. Looks like the event object goes straight to pinvokes, so a
null result just crashes?
I like the idea of a 'startup-script' for mod_managed. It would also be
excellent if there was an event or message informing the background code to
terminate nicely when the module reloads.
--Josh
On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk jl...@frontiernet.net wrote:
I think the problem here is that the loader only keeps this method in scope
until completion then it drops the remoted connection. Therefore you should
not use threads in this method. Michael please correct me if I am wrong
here.
As an example of the failure simply just put a Sleep(1) call in the
thread and you will see the failure.
As Michael said this method was only designed to allow the option to opt
out
of being loaded.
In order to support this perhaps a configuration flag simular to the lua
startup-script should be added.
Here is the error I get with the loop I mentioned. -Josh
[image: Capture.PNG]
On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo
m...@giagnocavo.netwrote:
Hi,
Can you please elaborate on the crash you receive when
you
queue a thread during load?
Thanks,
Michael
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Re: [Freeswitch-users] mod_opal segmentation fault error
On Tue, Sep 8, 2009 at 8:59 PM, Rogelio Perez rogelio.pe...@gmail.comwrote: Hi guys, My FS setup was working smoothly with mod_opal enabled until I had to rebuild everything from scratch. Now I have compiled everything following the same procedure (I even have a script for that) and mod_opal stopped working. The SVN commands are: svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/ trunk ptlib svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/branches/v3_6 opal ...and the compilation commands follow the documentation isntructions and there are no output errors. I start FS with mod_opal disabled and then when I run load mod_opalI get the error: Segmentation fault (core dumped). The log output shows nothing, and I see there are core.x files on the FS directory but I dont know how to read them. Any ideas? Visit these pages for information on how to collect a backtrace and open a JIRA bug report: http://wiki.freeswitch.org/wiki/Reporting_Bugs http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Loading_FreeSWITCH_in_GDB Before you file a bug report make sure that you update FS to latest SVN and also use the buildopal.sh file to rebuild opal and ptlib. Use make current to get your FS updated cleanly to the latest SVN and then try opal. If it still segs then open the bug report in the MODOPAL section. ( http://jira.freeswitch.org/browse/MODOPAL) Join us on IRC if you have more questions on all this. -MC ___ 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] Sangoma A500 - dial out from specific port group?
What is the output of oz list and oz dump? Put them in pastebin.freeswitch.org and link here in the mailing list. -MC On Tue, Sep 8, 2009 at 11:31 PM, Vassil Panayotov panayotov...@gmail.comwrote: Sorry I hit 'send' by mistake... Hi, Is it possible to originate calls from specific A500 ports with FreeSWITCH? I am using a A504 (8 BRI interfaces), and I want some outbound calls to be made from specific BRI interfaces. I tried to modify OpenZAP config as follows: conf/openzap.conf [span wanpipe boostbri1] trunk_type = bri b-channel = 1:1-2 b-channel = 2:1-2 b-channel = 3:1-2 b-channel = 4:1-2 b-channel = 5:1-2 b-channel = 6:1-2 [span wanpipe boostbri2] trunk_type = bri b-channel = 7:1-2 b-channel = 8:1-2 conf/autoload_configs/openzap. conf.xml: boost_spans span name=boostbri1 param name=dialplan value=XML/ param name=context value=isdn/ /span span name=boostbri2 param name=dialplan value=XML/ param name=context value=isdn/ /span /boost_spans When I try to originate call I am getting errors: freeswi...@emo-voip originate openzap/2/a/123456 music API CALL [originate(openzap/2/a/123456 music)] output: -ERR NORMAL_CIRCUIT_CONGESTION 2009-09-04 09:23:34.87253 [CRIT] ozmod_ss7_boost.c:244 SPAN is not online. 2009-09-04 09:23:34.87253 [ERR] mod_openzap.c:1043 No channels available 2009-09-04 09:23:34.87253 [ERR] switch_ivr_originate.c:1510 Cannot create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION] Then I tried to modify the /etc/wanpipe/smg_bri.conf: ;Sangoma AFT-A500 port 11 [slot:8 bus:1 span:7] group=2 country=europe operator=etsi connection_type=point_to_point signalling=bri_nt spans=7 ;Sangoma AFT-A500 port 12 [slot:8 bus:1 span:8] group=2 country=europe operator=etsi connection_type=point_to_point signalling=bri_nt spans=8 i.e. changed 'group' to 2, but this doesn't help either. Marc Celsie from Sangoma's techdesk told me that I should ' dial x...@gy with X being the number and Y being the group number'. How originate command should look like in this case? originate openzap/1/a/123...@g2 someExt ? I tried this syntax but with no effect. Marc also told me that there is a bug in FS which prevents groups from working. Should I fill bug report or feature request? Best regards, Vassil Panayotov ___ 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] Subscribing to events in managed C# / .NET
Yeah I noticed that but the thread was still terminating after a few seconds
anyway for me. Does it stay running for you?
Josh Rivers-2 wrote:
A new discovery:public bool Load()
{
ThreadPool.QueueUserWorkItem((o) =
{
Log.WriteLine(LogLevel.Notice, Thread Starting. );
EventConsumer con = new EventConsumer(all, );
while (true)
{
Event ev = con.pop(0);
if (ev == null) continue;
Log.WriteLine(LogLevel.Notice, Event: +
ev.serialized_string);
}
});
return true;
}
Does not crash. (Adding the null check prevents crash.) The backgrounded
loop runs fine. Looks like the event object goes straight to pinvokes, so
a
null result just crashes?
I like the idea of a 'startup-script' for mod_managed. It would also be
excellent if there was an event or message informing the background code
to
terminate nicely when the module reloads.
--Josh
On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk jl...@frontiernet.net wrote:
I think the problem here is that the loader only keeps this method in
scope
until completion then it drops the remoted connection. Therefore you
should
not use threads in this method. Michael please correct me if I am wrong
here.
As an example of the failure simply just put a Sleep(1) call in the
thread and you will see the failure.
As Michael said this method was only designed to allow the option to opt
out
of being loaded.
In order to support this perhaps a configuration flag simular to the lua
startup-script should be added.
Here is the error I get with the loop I mentioned. -Josh
[image: Capture.PNG]
On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo
m...@giagnocavo.netwrote:
Hi,
Can you please elaborate on the crash you receive when
you
queue a thread during load?
Thanks,
Michael
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Re: [Freeswitch-users] Sangoma A500 - dial out from specific port group?
Hi, Is it possible to originate calls from specific A500 ports with FreeSWITCH? I am using a A504 (8 BRI interfaces), and I want some outbound calls to be made from specific BRI interfaces. Hello Vassil, Unless you are using openzap trunk (and that probably means FreeSWITCH trunk as well) this is unlikely to work. Revision 825 in openzap should fix the bug in trunk group selection. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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
Re: [Freeswitch-users] Subscribing to events in managed C# / .NET
It does from a fresh start of FreeSWITCH. I've noticed, although not really
confirmed, a race condition between the unload and reload of managed code.
It seems that threads started in the newly submodule are terminated along
with the threads for the old, unloading submodule. Is that what you are
seeing as well?
On Wed, Sep 9, 2009 at 2:38 PM, Jeff Lenk jl...@frontiernet.net wrote:
Yeah I noticed that but the thread was still terminating after a few
seconds
anyway for me. Does it stay running for you?
Josh Rivers-2 wrote:
A new discovery:public bool Load()
{
ThreadPool.QueueUserWorkItem((o) =
{
Log.WriteLine(LogLevel.Notice, Thread Starting. );
EventConsumer con = new EventConsumer(all, );
while (true)
{
Event ev = con.pop(0);
if (ev == null) continue;
Log.WriteLine(LogLevel.Notice, Event: +
ev.serialized_string);
}
});
return true;
}
Does not crash. (Adding the null check prevents crash.) The backgrounded
loop runs fine. Looks like the event object goes straight to pinvokes, so
a
null result just crashes?
I like the idea of a 'startup-script' for mod_managed. It would also be
excellent if there was an event or message informing the background code
to
terminate nicely when the module reloads.
--Josh
On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk jl...@frontiernet.net
wrote:
I think the problem here is that the loader only keeps this method in
scope
until completion then it drops the remoted connection. Therefore you
should
not use threads in this method. Michael please correct me if I am wrong
here.
As an example of the failure simply just put a Sleep(1) call in the
thread and you will see the failure.
As Michael said this method was only designed to allow the option to opt
out
of being loaded.
In order to support this perhaps a configuration flag simular to the lua
startup-script should be added.
Here is the error I get with the loop I mentioned. -Josh
[image: Capture.PNG]
On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo
m...@giagnocavo.netwrote:
Hi,
Can you please elaborate on the crash you receive when
you
queue a thread during load?
Thanks,
Michael
--
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Re: [Freeswitch-users] Sangoma A500 - dial out from specific port group?
On Wed, Sep 9, 2009 at 01:20, Vassil Panayotov panayotov...@gmail.com wrote:
Hi,
Is it possible to originate calls from specific A500 ports with FreeSWITCH?
I am using a A504 (8 BRI interfaces), and I want some outbound calls to be
made from specific BRI interfaces.
You can't define several spans in openzap.conf for boost, the
sangoma_brid config file is where you define groups, so your config
should look like this:
/// smg_bri.conf
..
group=1
spans=1
group=2
spans=2
group=3
spans=3
..
/// openzap.conf
[span wanpipe BoostBRI]
trunk_type = bri
b-channel = 1:1-2
b-channel = 2:1-2
b-channel = 3:1-2
b-channel = 4:1-2
b-channel = 5:1-2
b-channel = 6:1-2
b-channel = 7:1-2
b-channel = 8:1-2
/// openzap.conf.xml
boost_spans
span name=BoostBRI
param name=local-ip value=127.0.0.65/
param name=local-port value=53000/
param name=remote-ip value=127.0.0.66/
param name=remote-port value=53000/
param name=context value=default/
param name=dialplan value=XML/
param name=tonegroup value=uk/
/span
/boost_spans
Then, you can Dial to your span/group number 3 with:
freeswitchoriginate openzap/1/a/12...@g3
exten|application_name(app_args)
freeswitchoriginate openzap/1/a/12...@g3
exten|application_name(app_args)
freeswitchoriginate openzap/1/a/12...@r3
exten|application_name(app_args)
freeswitchoriginate openzap/1/a/12...@r3
exten|application_name(app_args)
If you are using FS 1.0.4, there is a bug, you can fix it with this
-already in trunk- patch.
Index: src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c
===
--- libs/openzap/src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c.orig
+++ libs/openzap/src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c
@@ -282,6 +282,8 @@
}
ss7bc_call_init(event, caller_data-cid_num.digits, ani, r);
+ //ss7_bc_call_init will clear the trunk_group val so we need to set it
again
+ event.trunk_group=tg;
if (gr *(gr+1)) {
Best regards,
--
Octavio H. Ruiz Cervera
Tel.: (+52 55) 8590-9000 Ext. 7016
Mobile: (+52 1 55) 4358-4565
Sent from Mexico City, DF, Mexico
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Re: [Freeswitch-users] Subscribing to events in managed C# / .NET
The ILoadNotifcationPlugin is run in the appdomain created for the plugin, so it should only get unloaded when the plugin gets reloaded. Spawning threads here should work, it's definitely the intention that if you need a long-running process, you can fire it up on load and have it work. As to the race condition on reload, mod_managed should do this: - Load the new plugin into a new appdomain - Remove the entry points to the old appdomain, add entries to the new one - Old appdomain now stays alive until foreground API and APP calls finish So, you can have many versions of the same plugin active in memory. I probably need to go break compatibility and make ILoadWhateverPlugin be something like IPluginController and allow it to return loading options to control the mod_managed behavior, as well as allow it to delay shutdown of the appdomain. Part of the question is: how many people out there need compatibility, or can we go breaking all of you and make you recompile? :) Although, IIRC, if you handle AppDomain.Unload (or whatever it is), it will stay alive until your event handler completes. Hope that helps a bit. -Michael -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Wednesday, September 09, 2009 1:57 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET I think the problem here is that the loader only keeps this method in scope until completion then it drops the remoted connection. Therefore you should not use threads in this method. Michael please correct me if I am wrong here. As an example of the failure simply just put a Sleep(1) call in the thread and you will see the failure. As Michael said this method was only designed to allow the option to opt out of being loaded. In order to support this perhaps a configuration flag simular to the lua startup-script should be added. Here is the error I get with the loop I mentioned. -Josh [image: Capture.PNG] On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo m...@giagnocavo.netwrote: Hi, Can you please elaborate on the crash you receive when you queue a thread during load? Thanks, Michael -- View this message in context: http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html Sent from the freeswitch-users mailing list archive at Nabble.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 ___ 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] Subscribing to events in managed C# / .NET
I have a new thought on the crashes...I'm able to crash FreeSWITCH any time
I like, just by having an exception in a thread.
public class CrashFreeSWITCH : ILoadNotificationPlugin
{
public bool Load()
{
ThreadPool.QueueUserWorkItem((o) = { throw new
NotImplementedException(); });
return true;
}
}
Perhaps Application.ThreadException or AppDomain.UnhandledException need to
be trapped?
On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo m...@giagnocavo.netwrote:
Looks like the event object goes straight to pinvokes, so a null result
just crashes?
If it’s null, you should get a NullReferenceException. The C# compiler
should callvirt the property getter and that’ll do a null check. If that
isn’t happening, that’d be an interesting optimization somewhere along the
line.
-Michael
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Josh Rivers
*Sent:* Wednesday, September 09, 2009 3:01 PM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# /
.NET
A new discovery:
public bool Load()
{
ThreadPool.QueueUserWorkItem((o) =
{
Log.WriteLine(LogLevel.Notice, Thread Starting. );
EventConsumer con = new EventConsumer(all, );
while (true)
{
Event ev = con.pop(0);
if (ev == null) continue;
Log.WriteLine(LogLevel.Notice, Event: +
ev.serialized_string);
}
});
return true;
}
Does not crash. (Adding the null check prevents crash.) The backgrounded
loop runs fine. Looks like the event object goes straight to pinvokes, so a
null result just crashes?
I like the idea of a 'startup-script' for mod_managed. It would also be
excellent if there was an event or message informing the background code to
terminate nicely when the module reloads.
--Josh
On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk jl...@frontiernet.net wrote:
I think the problem here is that the loader only keeps this method in scope
until completion then it drops the remoted connection. Therefore you should
not use threads in this method. Michael please correct me if I am wrong
here.
As an example of the failure simply just put a Sleep(1) call in the
thread and you will see the failure.
As Michael said this method was only designed to allow the option to opt
out
of being loaded.
In order to support this perhaps a configuration flag simular to the lua
startup-script should be added.
Here is the error I get with the loop I mentioned. -Josh
[image: Capture.PNG]
On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo
m...@giagnocavo.netwrote:
Hi,
Can you please elaborate on the crash you receive when
you
queue a thread during load?
Thanks,
Michael
--
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http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html
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Re: [Freeswitch-users] Subscribing to events in managed C# / .NET
That's by design. If a thread fails, and there's no handler, then the
application could be in a corrupted state, so the CLR takes down the process.
I think there is a .NET 1.0 compat switch you can enable in the config if you
like exceptions to be silently ignored :).
-Michael
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Josh Rivers
Sent: Wednesday, September 09, 2009 6:39 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET
I have a new thought on the crashes...I'm able to crash FreeSWITCH any time I
like, just by having an exception in a thread.
public class CrashFreeSWITCH : ILoadNotificationPlugin
{
public bool Load()
{
ThreadPool.QueueUserWorkItem((o) = { throw new
NotImplementedException(); });
return true;
}
}
Perhaps Application.ThreadException or AppDomain.UnhandledException need to be
trapped?
On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo
m...@giagnocavo.netmailto:m...@giagnocavo.net wrote:
Looks like the event object goes straight to pinvokes, so a null result just
crashes?
If it's null, you should get a NullReferenceException. The C# compiler should
callvirt the property getter and that'll do a null check. If that isn't
happening, that'd be an interesting optimization somewhere along the line.
-Michael
From:
freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org]
On Behalf Of Josh Rivers
Sent: Wednesday, September 09, 2009 3:01 PM
To:
freeswitch-users@lists.freeswitch.orgmailto:freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET
A new discovery:
public bool Load()
{
ThreadPool.QueueUserWorkItem((o) =
{
Log.WriteLine(LogLevel.Notice, Thread Starting. );
EventConsumer con = new EventConsumer(all, );
while (true)
{
Event ev = con.pop(0);
if (ev == null) continue;
Log.WriteLine(LogLevel.Notice, Event: +
ev.serialized_string);
}
});
return true;
}
Does not crash. (Adding the null check prevents crash.) The backgrounded loop
runs fine. Looks like the event object goes straight to pinvokes, so a null
result just crashes?
I like the idea of a 'startup-script' for mod_managed. It would also be
excellent if there was an event or message informing the background code to
terminate nicely when the module reloads.
--Josh
On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk
jl...@frontiernet.netmailto:jl...@frontiernet.net wrote:
I think the problem here is that the loader only keeps this method in scope
until completion then it drops the remoted connection. Therefore you should
not use threads in this method. Michael please correct me if I am wrong
here.
As an example of the failure simply just put a Sleep(1) call in the
thread and you will see the failure.
As Michael said this method was only designed to allow the option to opt out
of being loaded.
In order to support this perhaps a configuration flag simular to the lua
startup-script should be added.
Here is the error I get with the loop I mentioned. -Josh
[image: Capture.PNG]
On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo
m...@giagnocavo.netmailto:m...@giagnocavo.netwrote:
Hi,
Can you please elaborate on the crash you receive when you
queue a thread during load?
Thanks,
Michael
--
View this message in context:
http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html
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[Freeswitch-users] OpenZAP No Audio In Outbound FXO for 8-10 Seconds
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello all! I am having the following issue. When I dial out over a FXO port (analog) for the first 8-10 seconds I get no audio. If I wait I will eventually hear something. On inbound calls audio works great in both directions. I used ztmonitor to record from the channel and there was in fact audio there. In the recording you can clearly tell when audio starts flowing as you can hear me twice in the recording (I was calling to my cell from a phone on my desk). Any help here would be greatly appreciated! FS is current SVN OpenZAP is current SVN. Hardware: Rhino 8 port w/6 FXO and 2 FXS ports installed. Daniel Morrigan He who says it cannot be done is interrupting the one doing it. - Chinese Proverb -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkqoeHoACgkQ3JaPN6smlEXhdQCfRf5GBKwVrtZWsCS4J1fug2e7 SEEAn1FkyhBPKiXnfUiXgvd2ggqUW87w =OFH/ -END PGP SIGNATURE- ___ 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] Subscribing to events in managed C# / .NET
The question is whether the CLR should take down the whole phone server due
to an unhandled exception...definitely the CLR should terminate...but
shouldn't it just log the exception to the console, not crash the core?
On Wed, Sep 9, 2009 at 6:30 PM, Michael Giagnocavo m...@giagnocavo.netwrote:
That’s by design. If a thread fails, and there’s no handler, then the
application could be in a corrupted state, so the CLR takes down the
process.
I think there is a .NET 1.0 compat switch you can enable in the config if
you like exceptions to be silently ignored J.
-Michael
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Josh Rivers
*Sent:* Wednesday, September 09, 2009 6:39 PM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# /
.NET
I have a new thought on the crashes...I'm able to crash FreeSWITCH any time
I like, just by having an exception in a thread.
public class CrashFreeSWITCH : ILoadNotificationPlugin
{
public bool Load()
{
ThreadPool.QueueUserWorkItem((o) = { throw new
NotImplementedException(); });
return true;
}
}
Perhaps Application.ThreadException or AppDomain.UnhandledException need
to be trapped?
On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo m...@giagnocavo.net
wrote:
Looks like the event object goes straight to pinvokes, so a null result
just crashes?
If it’s null, you should get a NullReferenceException. The C# compiler
should callvirt the property getter and that’ll do a null check. If that
isn’t happening, that’d be an interesting optimization somewhere along the
line.
-Michael
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Josh Rivers
*Sent:* Wednesday, September 09, 2009 3:01 PM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# /
.NET
A new discovery:
public bool Load()
{
ThreadPool.QueueUserWorkItem((o) =
{
Log.WriteLine(LogLevel.Notice, Thread Starting. );
EventConsumer con = new EventConsumer(all, );
while (true)
{
Event ev = con.pop(0);
if (ev == null) continue;
Log.WriteLine(LogLevel.Notice, Event: +
ev.serialized_string);
}
});
return true;
}
Does not crash. (Adding the null check prevents crash.) The backgrounded
loop runs fine. Looks like the event object goes straight to pinvokes, so a
null result just crashes?
I like the idea of a 'startup-script' for mod_managed. It would also be
excellent if there was an event or message informing the background code to
terminate nicely when the module reloads.
--Josh
On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk jl...@frontiernet.net wrote:
I think the problem here is that the loader only keeps this method in scope
until completion then it drops the remoted connection. Therefore you should
not use threads in this method. Michael please correct me if I am wrong
here.
As an example of the failure simply just put a Sleep(1) call in the
thread and you will see the failure.
As Michael said this method was only designed to allow the option to opt
out
of being loaded.
In order to support this perhaps a configuration flag simular to the lua
startup-script should be added.
Here is the error I get with the loop I mentioned. -Josh
[image: Capture.PNG]
On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo
m...@giagnocavo.netwrote:
Hi,
Can you please elaborate on the crash you receive when
you
queue a thread during load?
Thanks,
Michael
--
View this message in context:
http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html
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Re: [Freeswitch-users] outbould PHP ESL
It should be the same, except using php syntax instead of perl. Mike On Sep 2, 2009, at 11:43 AM, Dome Charoenyost wrote: 2009/9/2 Michael Collins m...@freeswitch.org: Are you trying to get a channel variable or capture DTMF input from the caller? i try to make IVR by php outbound socket. in XML dialplan we can get DTMF by read application (store in channel variable) I found it's success in perl outbound (IVR.pm) but for php how do i ? Dome C. -MC On Wed, Sep 2, 2009 at 7:56 AM, Dome Charoenyost d...@tel.co.th wrote: I follow http://wiki.freeswitch.org/wiki/PHP_ESL#ivrd how to get from php ? Dome C. -- #!/usr/bin/php -q ?php // set a couple of things so we dont kill the system ob_implicit_flush(true); set_time_limit(30); // Open stdin so we can read the AGI data in $in = fopen(php://stdin, r); // Connect echo connect\n\n; // Answer echo sendmsg\n; echo call-command: execute\n; echo execute-app-name: answer\n\n; echo sendmsg\n; echo call-command: execute\n; echo execute-app-name: read\n; echo execute-app-arg: 0 20 /opt/freeswitch/sounds/th/tuxza/welcome.wav res 5000 #\n\n; // Wait sleep(5); // Hangup echo sendmsg\n; echo call-command: hangup\n\n; fclose($in); ? 2009/9/2 Brian West br...@freeswitch.org: uuid_getvar /b On Sep 2, 2009, at 8:16 AM, Tristan Mahé wrote: Hi, just a fast 2cent: get var via channel status ? ( variable_res ) ___ 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] Implementing h extension in FS
HI,
I'm newbie in FS, I want to know how to implement h extension of asterisk to
FS. As I listed down below;
h =
{
NOOP(Call Completed with Carrier ${CARRIER});
goto add_cdr|h|1;
};
My other question is, which application/function/class is use in mod_perl to
check the channel status? i.e. busy, answer,hangup,ringing,etc.
Kindly advice me soon.
--
Regards,
Ahmed Munir
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Re: [Freeswitch-users] Mod_fifo posision in queue
You can use a phrase macro but I am not sure that we set the position in a way that you can expand it for the macro. Mike On Sep 1, 2009, at 10:37 AM, Dome Charoenyost wrote: Dear sir, I want to say posision in queue to caller but fifo_chime_list can't say more than one sound file. i try fifo_chime_list = queue/say1.wav,queue/say2.wav. Best Regards. 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
Re: [Freeswitch-users] make install failure on Solaris 10
somewhere in that mess of my commands solaris is not liking something. I tested this a lot on solaris and had it working on every box i was in, so not sure what this could be. If you can get me into a box in this state via ssh I can take a look. Mike On Sep 3, 2009, at 6:42 AM, Bruce McAlister wrote: Hi, I have just managed to complete a build of FreeSWITCH 1.0.4 on Solaris 10. The problem I am now having is that it fails on the make install part of the installation. I have attached the complete output of the make install. A snippet of the failure is below: --- Installing freeswitch *** Error code 1 The following command caused the error: for htdocsfile in `find htdocs -name \* | grep -v .svn` ; do \ dir=`echo $htdocsfile | sed -e 's|/[^/]*$||'`; \ filename=`echo $htdocsfile | sed -e 's|^.*/||'`; \ test -d /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ freeswitch/$dir || /export/home/user/packages/BUILD/freeswitch-1.0.4/ build/config/install-sh -d /var/tmp/pkgbuild-user/freeswitch-1.0.4- build/opt/freeswitch/$dir ; \ test -f /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ freeswitch/$dir/$filename || /opt/jdsbld/bin/ginstall -c -m 644 $dir/ $filename /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ freeswitch/$dir 2/dev/null; \ done make: Fatal error: Command failed for target `samples-htdocs' Current working directory /export/home/user/packages/BUILD/ freeswitch-1.0.4 *** Error code 1 The following command caused the error: test -d /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/ htdocs || make samples-htdocs make: Fatal error: Command failed for target `install-data-local' Current working directory /export/home/user/packages/BUILD/ freeswitch-1.0.4 *** Error code 1 The following command caused the error: make OUR_MODULES=$(if test -z ; then tmp_mods=$(grep -v \# / export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - e s|^.*/|| | sort | uniq ); else tmp_mods= ; fi ; mods=$(for i in $tmp_mods ; do echo $i-all ; done ); echo $mods ) OUR_CLEAN_MODULES=$(if test -z ; then tmp_mods=$(grep -v \# / export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - e s|^.*/|| | sort | uniq ); else tmp_mods= ; fi ; mods=$(for i in $tmp_mods ; do echo $i-clean ; done ); echo $mods ) OUR_INSTALL_MODULES=$(if test -z ; then tmp_mods=$(grep -v \# /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e s|^.*/|| | sort | uniq ); else tmp_mods= ; fi ; mods=$ (for i in $tmp_mods ; do echo $i-install ; done); echo $mods ) OUR_UNINSTALL_MODULES=$(if test -z ; then tmp_mods=$(grep -v \# /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e s|^.*/|| | sort | uniq ); else tmp_mods= ; fi ; mods=$ (for i in $tmp_mods ; do echo $i-uninstall ; done); echo $mods ) OUR_DISABLED_MODULES=$(tmp_mods=$(grep \# /export/home/user/ packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v \#\# | sed - e s|^.*/|| | sort | uniq ); mods=$(for i in $tmp_mods ; do echo $i-all ; done ); echo $mods ) OUR_DISABLED_CLEAN_MODULES=$ (tmp_mods=$(grep \# /export/home/user/packages/BUILD/ freeswitch-1.0.4/modules.conf | grep -v \#\# | sed -e s|^.*/|| | sort | uniq ); mods=$(for i in $tmp_mods ; do echo $i-clean ; done ); echo $mods ) OUR_DISABLED_INSTALL_MODULES=$(tmp_mods=$ (grep \# /export/home/user/packages/BUILD/freeswitch-1.0.4/ modules.conf | grep -v \#\# | sed -e s|^.*/|| | sort | uniq ); mods=$(for i in $tmp_mods ; do echo $i-install ; done); echo $mods ) OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods=$(grep \# / export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v \#\# | sed -e s|^.*/|| | sort | uniq ); mods=$(for i in $tmp_mods ; do echo $i-uninstall ; done); echo $mods ) `test -n || echo -s` install-exec-am install-data-am make: Fatal error: Command failed for target `install-am' Current working directory /export/home/user/packages/BUILD/ freeswitch-1.0.4 *** Error code 1 The following command caused the error: failcom='exit 1'; \ for f in x $MAKEFLAGS; do \ case $f in \ *=* | --[!k]*);; \ *k*) failcom='fail=yes';; \ esac; \ done; \ dot_seen=no; \ target=`echo install-recursive | sed s/-recursive//`; \ list='. src build'; for subdir in $list; do \ echo Making $target in $subdir; \ if test $subdir = .; then \ dot_seen=yes; \ local_target=$target-am; \ else \ local_target=$target; \ fi; \ (cd $subdir make OUR_MODULES=$(if test -z ; then tmp_mods=$ (grep -v \# /export/home/user/packages/BUILD/freeswitch-1.0.4/ modules.conf | sed -e s|^.*/|| | sort | uniq ); else tmp_mods= ; fi ; mods=$(for i in $tmp_mods ; do echo $i-all ; done ); echo $mods ) OUR_CLEAN_MODULES=$(if test -z ; then tmp_mods=$(grep -v \# /export/home/user/packages/BUILD/ freeswitch-1.0.4/modules.conf | sed -e s|^.*/|| | sort | uniq ); else tmp_mods= ; fi ;
Re: [Freeswitch-users] Implementing h extension in FS
You should be able to handle hangups in one of two ways:1) Register a hangup
handler in your script or dialplan. This will execute a script on the hangup
of the call.
2) Use the Event Socket Layer(ESL) to listen to hangup events and then
perform your actions there.
You can find more about these options in the wiki.
On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
HI,
I'm newbie in FS, I want to know how to implement h extension of asterisk
to FS. As I listed down below;
h =
{
NOOP(Call Completed with Carrier ${CARRIER});
goto add_cdr|h|1;
};
My other question is, which application/function/class is use in mod_perl
to check the channel status? i.e. busy, answer,hangup,ringing,etc.
Kindly advice me soon.
--
Regards,
Ahmed Munir
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