Re: [Freeswitch-users] upper registration in FS?
hi, any1 have any idea how what to sue in dialplan such that calls from a single id go to a specific gateway only with blind registration enabled, this is the only major issue im having. Regards, Bipin -- View this message in context: http://www.nabble.com/Re%3A-upper-registration-in-FS--tp22782688p22862459.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
Re: [Freeswitch-users] upper registration in FS?
xbipin bi...@xbipin.com wrote: any1 have any idea how what to sue in dialplan such that calls from a single id go to a specific gateway only with blind registration enabled, this is the only major issue im having. Perhaps you could match the source address in the dial plan and then bridge or redirect the call to the desired gateway. condition field=network_addr expression=^192.168.140.5$ for example. I tested a similar example once and it did work. ___ 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] FS failover redundancy load balancing
Henry Huang pisze: How do you load balance conference calls? Doesn't all the conference members have to be on the same freeswitch server? As I wrote I do not load balance them yet. I didn't investigate that but what comes to my mind is to setup 2 FS end register agents to one of them (load balance them), sip phones through proxy server. Then one separate FS for incoming calls and in that FS place my queue system. When incoming call needs to be connected to agent then right FS machine would be choosen. This just idea I believe that in time I will need something like that FS developers will give us some modules or other options. On Thu, Apr 2, 2009 at 1:07 PM, Szymon Olko so...@gcdf.pl wrote: Brian West pisze: what kind of hardware? I made testes on Pentium-M laptop with single core 1,6Hz. I did not write those results, it was over 100 calls that was handle good, I was just curios what will happen. Tomorrow I will make real testes. My production works on 2 core P4 and I have there only 35 agents CPU load is like 7% with 15% small peeks. All phones are sip or analog via sip gateways, PRI is currently still on asterisk which is connected via sip. /b On Apr 2, 2009, at 2:29 PM, Szymon Olko wrote: I did not described it perfectly. I made agents, queues scenarios on conferences. This what I tested was for example 100 calls, so it's 200 channels, and 100 conferences, 2 channels per conference, all are unmuted. I did that just because it is my work scenario. Brian West br...@freeswitch.org mailto:br...@freeswitch.org mailto:br...@freeswitch.org mailto:br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls [SOLVED]
Hi, Updating asterisk to version 1.4.24 solved the problem. Thanks guys. Regards. 2009/4/2 Brian West br...@freeswitch.org: Follow this thread http://lists.freeswitch.org/pipermail/freeswitch-users/2009-March/012646.html /b On Apr 1, 2009, at 5:36 PM, Alfonso Pinto wrote: Hi guys, I've using asterisk as PSTN gateway. When a call arrives from PSTN, I send the call to freeswitch and this route the call to a SIP gateway. When caller cancels the call (hangups before callee answers), I get this on asterisk CLI: chan_sip.c:13056 handle_response: Remote host can't match request CANCEL to call '271c0dad41cc80456b8de2133dc80...@1.1.1.1'. Giving up. I'm using asterisk 1.4.23.1 and freeswitch 1.0.3 This is the sip call flow: u 2009/04/01 21:59:26.402934 2.2.2.2:5060 - 1.1.1.1:5060 INVITE sip:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1. Contact: sip:99...@2.2.2.2. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Wed, 01 Apr 2009 21:03:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 29347 29347 IN IP4 2.2.2.2. s=session. c=IN IP4 2.2.2.2. t=0 0. m=audio 13846 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2009/04/01 21:59:26.403717 1.1.1.1:5060 - 2.2.2.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Content-Length: 0. . U 2009/04/01 21:59:26.414810 1.1.1.1:5060 - 2.2.2.2:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport=5060. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1;tag=ceKFmNU84B90c. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Accept: application/sdp. 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. Proxy-Authenticate: Digest realm=1.1.1.1, nonce=5df21692-1f08-11de-9d06-83e4a6e70df7, algorithm=MD5, qop=auth. Content-Length: 0. . U 2009/04/01 21:59:26.415395 2.2.2.2:5060 - 1.1.1.1:5060 ACK sip:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK2707ceb1;rport. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1;tag=ceKFmNU84B90c. Contact: sip:99...@2.2.2.2. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . U 2009/04/01 21:59:26.415648 2.2.2.2:5060 - 1.1.1.1:5060 INVITE sip:66...@1.1.1.1 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1. Contact: sip:99...@2.2.2.2. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 103 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Proxy-Authorization: Digest username=asterisk02, realm=1.1.1.1, algorithm=MD5, uri=sip:66...@1.1.1.1, nonce=5df21692-1f08-11de-9d06-83e4a6e70df7, response=cb57576192b001f79bd03ebb8bb57d0a, qop=auth, cnonce=47efcad4, nc=0001. Date: Wed, 01 Apr 2009 21:03:12 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 265. . v=0. o=root 29347 29348 IN IP4 2.2.2.2. s=session. c=IN IP4 2.2.2.2. t=0 0. m=audio 13846 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2009/04/01 21:59:26.416181 1.1.1.1:5060 - 2.2.2.2:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK176746ff;rport=5060. From: 9 sip:99...@1.1.1.1;tag=as26208773. To: sip:66...@1.1.1.1. Call-ID: 271c0dad41cc80456b8de2133dc80...@1.1.1.1. CSeq: 103 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.3-hacked. Content-Length: 0. . U 2009/04/01 21:59:26.426298 1.1.1.1:5060 - 3.3.3.3:5060 INVITE sip:66...@3.3.3.3 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;rport;branch=z9hG4bKr8XpNg0cgpSSB. Max-Forwards: 69. From: 9 sip:559066...@3.3.3.3;transport=udp;tag=e050QBXFZXN6K. To: sip:66...@3.3.3.3. Call-ID:
Re: [Freeswitch-users] Asterisk and Freeswitch: Destination keeps ringing when caller cancels calls
Thank you so much, gmane gives me correct results. Instead, trying to search the thread Brian emailed to me with site:lists.freeswitch.org doesn't give the correct response, thread doesn't appears. Regards 2009/4/2 Jason White ja...@jasonjgw.net: Alfonso Pinto elhod...@gmail.com wrote: One question more, maybe a stupid one: How can I search the archives? http://www.gmane.org/ The searching tool they use, Xapian, tends to give good relevance ranking, at least in my experience. ___ 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] Slow freeswitch shutdown
In last SVN trunk version i noticed that stopping of freeswitch takes much time. I have configuration installed with freeswitch. I added sip gateway to my asterisk instance. I don't use asterisk currently and my gateway definition is like that: gateway name=429956 param name=username value=429956/ param name=password value=429956/ param name=proxy value=10.0.0.248:5059/ param name=register value=false/ /gateway Starting freeswitch and shutting it down for console with '...' brings following logs. 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() UN-Registering example.com 2009-04-03 11:58:03 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting for worker thread 2009-04-03 11:58:03 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() deleted gateway example.com 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() UN-Registering 429956 2009-04-03 11:58:33 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting for worker thread 2009-04-03 11:58:33 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() deleted gateway 429956 Asterisk was not run at all so it should not register to it, why it hangs to unregister it? ___ 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 running on OpenWrt
Carlos Talbot Is there an interest in running FreeSWITCH on OpenWRT? I recently managed to compile and run a version for a MIPs based router, the Planex MZK-W04NU. Great news :-) I'm interested in running FS on any of this type of small hardware. Ideally, it should have a USB port so I can connect Sangoma's U100 connector to handle one or two POTS lines. Would the FS port you did handle this USB VoIP gateway? 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] Calls being cut off while recording a message
Hi Brian, I've upgraded to svn trunk but am now getting errors on load which are preventing it from working: 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_stream_pagein** 2009-04-03 11:35:52 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/local/freeswitch/lib/libjs.so.1: undefined symbol: PR_LocalTimeParameters** Sorry if this is obvious but what have I done wrong? Thanks for your help Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:40 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while recording a message Please try SVN trunk. /b On Mar 31, 2009, at 8:36 AM, Andy Ayers wrote: Hi Brian, 1.03 Thanks Andy Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.com/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Use rates from lcr in nibblebill module
Thanks for variables and explanation. Work fine! Now wait for nibblebill can hangup connection when balance hits 0.00 On Thursday 02 April 2009 15:37:28 Rupa Schomaker wrote: Update the to the latest. I've added more channel vars: eg: after doing: action application=lcr data=12148267722 default2/ (not a real number) I get the following: variable_lcr_query_digits: [12148267722] variable_lcr_query_profile: [0] variable_lcr_query_expanded_digits: [12148267722, 1214826772, 121482677, 12148267, 1214826, 121482, 12148, 1214, 121, 12, 1] variable_lcr_route_1: [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gate way/teliax/12148267722] variable_lcr_rate_1: [0.01000] variable_lcr_carrier_1: [teliax] variable_lcr_codec_1: [PCMU] variable_lcr_route_2: [[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec_string=PCMU]sofia/ga teway/vitelity/12148267722] variable_lcr_rate_2: [0.01440] variable_lcr_carrier_2: [vitelity] variable_lcr_codec_2: [PCMU] variable_lcr_route_count: [2] variable_lcr_auto_route: [[lcr_carrier=teliax,lcr_rate=0.01000,absolute_codec_string=PCMU]sofia/gate way/teliax/12148267722|[lcr_carrier=vitelity,lcr_rate=0.01440,absolute_codec _string=PCMU]sofia/gateway/vitelity/12148267722] variable_import: [lcr_carrier,lcr_rate] which, I think is what you are asking for. If you know which route you are going to use (eg: 1) then you can get it's rate by using lcr_rate_1. Alternatively, you can use the lcr_auto_route and then once the b-leg connects, query the b-leg variable for lcr_carrier and lcr_rate to see which one was actually used. You really can't use lcr_auto_route and set a single rate since each leg can be rated differently (look at example above). Normally lcr is used for your own rates between you and your carrier. That is independant of the rate table used for your customers. You can use lcr to query both. First use lcr to query your own rates using a different profile. This would return a *single* route if you setup your route table right. Save the rate in a var to be used with nibble bill. Then use lcr with your external rates so you can route according to your own cost with your carrier(s). On Thu, Apr 2, 2009 at 5:22 AM, Yuriy Ivzhenko (WP) yivzhe...@mksat.netwrote: Hi, I want to use module lcr to find a best route and his rate , then make a call and bill on that rate with nibblebill module. lcr return variable lcr_auto_route that contains [lcr_rate=xxx] variable for new channel. To use nibblebill i need to set nibble_rate = lcr_rate. What is best method to do that? Is there a way to do that with standard tools, without use external scripts? Thanks, Yuriy ___ 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] Slow freeswitch shutdown
update again and see if it's better On Fri, Apr 3, 2009 at 5:05 AM, Szymon Olko so...@gcdf.pl wrote: In last SVN trunk version i noticed that stopping of freeswitch takes much time. I have configuration installed with freeswitch. I added sip gateway to my asterisk instance. I don't use asterisk currently and my gateway definition is like that: gateway name=429956 param name=username value=429956/ param name=password value=429956/ param name=proxy value=10.0.0.248:5059/ param name=register value=false/ /gateway Starting freeswitch and shutting it down for console with '...' brings following logs. 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() UN-Registering example.com 2009-04-03 11:58:03 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting for worker thread 2009-04-03 11:58:03 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() deleted gateway example.com 2009-04-03 11:58:03 [NOTICE] sofia_reg.c:75 sofia_reg_kill_reg() UN-Registering 429956 2009-04-03 11:58:33 [NOTICE] sofia.c:895 sofia_profile_thread_run() Waiting for worker thread 2009-04-03 11:58:33 [NOTICE] sofia_glue.c:3173 sofia_glue_del_profile() deleted gateway 429956 Asterisk was not run at all so it should not register to it, why it hangs to unregister it? ___ 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:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] codecs initialization flags in endpoint modules
Hello,
I've been experimenting with the use of mod_dahdi_codec and other ways
to perform external transcoding for codecs, and came up with noticing
that transcoding resources seemed to be used up twice what I expected.
That is and 2x the number of call legs, ending up to two encoder and two
decoder instances per leg.
So, I looked at the code and noticed almost every endpoint module does
something like this (excerpt from mod_sofia, sofia_glue.c:~1800):
if (switch_core_codec_init(tech_pvt-read_codec,
tech_pvt-iananame,
tech_pvt-rm_fmtp,
tech_pvt-rm_rate,
tech_pvt-codec_ms,
1,
SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE |
tech_pvt-profile-codec_flags,
NULL, switch_core_session_get_pool(tech_pvt-session)) !=
SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, Can't load
codec?\n);
switch_goto_status(SWITCH_STATUS_FALSE, end);
}
if (switch_core_codec_init(tech_pvt-write_codec,
tech_pvt-iananame,
tech_pvt-rm_fmtp,
tech_pvt-rm_rate,
tech_pvt-codec_ms,
1,
SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE |
tech_pvt-profile-codec_flags,
NULL, switch_core_session_get_pool(tech_pvt-session)) !=
SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, Can't load
codec?\n);
switch_goto_status(SWITCH_STATUS_FALSE, end);
}
The flags being SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE
seems to be causing the apparent 'double' allocation of transcoding
resources, and I fail to understand the need for both, in both cases.
Could someone please spend a minute to explain?
thanks
lele
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Re: [Freeswitch-users] Buzzing when people speak in conference
Thanks for all your help, I finally resolved the issue by setting comfort-noise to false in the conference.conf.xml. From: stormin.nor...@hotmail.co.uk To: freeswitch-users@lists.freeswitch.org Date: Wed, 1 Apr 2009 22:09:03 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. Upgrade to Internet Explorer 8 Optimised for MSN. Download Now _ Share your photos with Windows Live Photos – Free. http://clk.atdmt.com/UKM/go/134665338/direct/01/___ 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] Buzzing when people speak in conference
Did it sound more like a machine gun? /b On Apr 3, 2009, at 9:02 AM, Stromin Normin wrote: Thanks for all your help, I finally resolved the issue by setting comfort-noise to false in the conference.conf.xml. Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] How to call multi gateways for failover with early media?
Hi, I have outbound gateways returns 403 or 503 constantly. So I tried to dial out using sofia/gateways/gw1/|sofia/gateways/gw2/|sofia/gateways/gw3... for fail over. To make it work, I need to set ignore_early_media=true. However, the caller do need to hear the early media to figure out what's going on. If I set ignore_early_media=false, only the first one tried. A little more detail: The gateway is first tier, if it cannot initiate a PSTN channel returns 403/503 immediately. If it can find a PSTN channel, but the callee fails, no answer or busy or others, it plays early media and returns 503. If I want failover, and the early media, how to do that? Thanks. regards, Seven. ___ 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] codecs initialization flags in endpoint modules
FYI, these are good questions but they probably belong on the dev list since
they are so technical in nature. :)
-MC
On Fri, Apr 3, 2009 at 6:20 AM, Lele Forzani l...@windmill.it wrote:
Hello,
I've been experimenting with the use of mod_dahdi_codec and other ways
to perform external transcoding for codecs, and came up with noticing
that transcoding resources seemed to be used up twice what I expected.
That is and 2x the number of call legs, ending up to two encoder and two
decoder instances per leg.
So, I looked at the code and noticed almost every endpoint module does
something like this (excerpt from mod_sofia, sofia_glue.c:~1800):
if (switch_core_codec_init(tech_pvt-read_codec,
tech_pvt-iananame,
tech_pvt-rm_fmtp,
tech_pvt-rm_rate,
tech_pvt-codec_ms,
1,
SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE |
tech_pvt-profile-codec_flags,
NULL, switch_core_session_get_pool(tech_pvt-session)) !=
SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, Can't load
codec?\n);
switch_goto_status(SWITCH_STATUS_FALSE, end);
}
if (switch_core_codec_init(tech_pvt-write_codec,
tech_pvt-iananame,
tech_pvt-rm_fmtp,
tech_pvt-rm_rate,
tech_pvt-codec_ms,
1,
SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE |
tech_pvt-profile-codec_flags,
NULL, switch_core_session_get_pool(tech_pvt-session)) !=
SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, Can't load
codec?\n);
switch_goto_status(SWITCH_STATUS_FALSE, end);
}
The flags being SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE
seems to be causing the apparent 'double' allocation of transcoding
resources, and I fail to understand the need for both, in both cases.
Could someone please spend a minute to explain?
thanks
lele
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Re: [Freeswitch-users] Buzzing when people speak in conference
On Fri, Apr 3, 2009 at 7:11 AM, Brian West br...@freeswitch.org wrote: Did it sound more like a machine gun? /b Comfort noise for General Douglas McArthur I guess... ___ 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] meuccisoluti...@66.96.218.5
Does anyone else seem to be getting tons of calls from this evil IP? They keep ringing me via SIP over and over again. Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] meuccisoluti...@66.96.218.5
Hi Brian, looks like this Evil is calling everywhere today on port 5060, please see my asterisk log [Apr 3 11:13:42] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as05dbf888 [Apr 3 11:25:12] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as5ab1ec7b [Apr 3 11:25:44] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as5ab1ec7b [Apr 3 11:36:17] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as5c4625af [Apr 3 11:55:22] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as4d32ad06 [Apr 3 11:55:54] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as4d32ad06 [Apr 3 11:55:56] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as324c491b [Apr 3 12:00:19] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as4ab90c05 [Apr 3 12:14:43] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as3edfecbb [Apr 3 12:23:38] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as305dbb2e [Apr 3 12:32:14] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as5bf0ab42 [Apr 3 12:49:12] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as7f56ad67 [Apr 3 12:52:21] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as0d5d32e0 [Apr 3 13:10:09] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as1b806860 [Apr 3 13:17:46] NOTICE[16617] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as487f8ecb [Apr 3 13:29:56] NOTICE[16920] chan_sip.c: Failed to authenticate user MeucciSolutions sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5 ;tag=as613a9814 On Fri, Apr 3, 2009 at 1:04 PM, Brian West br...@freeswitch.org wrote: Does anyone else seem to be getting tons of calls from this evil IP? They keep ringing me via SIP over and over again. Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] meuccisoluti...@66.96.218.5
I heard about this a few days ago, they claim it's not them, but someone trying to harm their reputation ... http://www.meucci-solutions.com/complaints.asp?id=1 Gabe Brian West wrote: Does anyone else seem to be getting tons of calls from this evil IP? They keep ringing me via SIP over and over again. Brian West br...@freeswitch.org mailto:br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.com/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] meuccisoluti...@66.96.218.5
It is strange this IP is from US 66.96.218.5USUNITED STATESPENNSYLVANIASCRANTONNETWORK OPERATIONS CENTER INC On Fri, Apr 3, 2009 at 1:53 PM, Gabriel Kuri gk...@ieee.org wrote: I heard about this a few days ago, they claim it's not them, but someone trying to harm their reputation ... http://www.meucci-solutions.com/complaints.asp?id=1 Gabe Brian West wrote: Does anyone else seem to be getting tons of calls from this evil IP? They keep ringing me via SIP over and over again. Brian West br...@freeswitch.org mailto:br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.com/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] meuccisoluti...@66.96.218.5
Yes I opened a ticket with them about it... they said it would take 24 hours to figure anything out! /b On Apr 3, 2009, at 1:02 PM, Chris Chen wrote: It is strange this IP is from US 66.96.218.5 US UNITED STATES PENNSYLVANIA SCRANTON NETWORK OPERATIONS CENTER INC Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH running on OpenWrt
This would be ideal. I'm not sure though if the wanpipe kernel driver has been ported to openwrt (or non-x86 hardware for that matter). FYI, I'm slowly working on the wiki and have faced some obstacles as openwrt.org decided to upgrade their servers this past week and have been offline for a good part of that... Carlos On Fri, Apr 3, 2009 at 5:07 AM, Fred codecompl...@free.fr wrote: Carlos Talbot Is there an interest in running FreeSWITCH on OpenWRT? I recently managed to compile and run a version for a MIPs based router, the Planex MZK-W04NU. Great news :-) I'm interested in running FS on any of this type of small hardware. Ideally, it should have a USB port so I can connect Sangoma's U100 connector to handle one or two POTS lines. Would the FS port you did handle this USB VoIP gateway? 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 ___ 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] FS failover redundancy load balancing
You could try (although it's somewhat bleeding edge) to use OpenSIPS 1.5 with load_balancer (not heavily tested, btw) in front of some FreeSWITCH machines: http://www.opensips.org/html/docs/modules/devel/load_balancer.html 2009/4/2 Ashley van Gerven ashley@gmail.com: Hi, I can't find much info on setting up a redundant or heavy load FreeSwitch implementation. Are there any links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? I imagine the entry level solution is to have two FS boxes configured identitcally, with redundant SBC software (recommendations?) in front, passing the calls to the primary FS box, or the backup FS box if the primary is not responding. Is that the easiest solution? What about a situation of having a level of concurrent calls beyond what one FS box can handle? I realise that would be a very large number of concurrent calls, but we would need a good plan on how to scale the systems. Are there recommendations for load balancing solutions? Either soft or hardware? My guess would be having 3 + 1 spare FS servers would work, where calls are distributed accross 3 FS boxes by a load balancer with one spare in event of failure. Also how would a FS box at max capacity behave? Does FS monitor available resources and reject the excess calls that it can't handle? Or would the load balancer have to be configured with the maximum number of calls per box? Would love to hear some experiences of deploying FS with failover high load. Thanks Ash ___ 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 -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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] FS failover redundancy load balancing
Hi Kristian, you're right. Definitively that will be best solution as soon as it's released as stable (it's alpha now). http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing Pablo On Fri, Apr 3, 2009 at 6:42 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: You could try (although it's somewhat bleeding edge) to use OpenSIPS 1.5 with load_balancer (not heavily tested, btw) in front of some FreeSWITCH machines: http://www.opensips.org/html/docs/modules/devel/load_balancer.html 2009/4/2 Ashley van Gerven ashley@gmail.com: Hi, I can't find much info on setting up a redundant or heavy load FreeSwitch implementation. Are there any links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment? I imagine the entry level solution is to have two FS boxes configured identitcally, with redundant SBC software (recommendations?) in front, passing the calls to the primary FS box, or the backup FS box if the primary is not responding. Is that the easiest solution? What about a situation of having a level of concurrent calls beyond what one FS box can handle? I realise that would be a very large number of concurrent calls, but we would need a good plan on how to scale the systems. Are there recommendations for load balancing solutions? Either soft or hardware? My guess would be having 3 + 1 spare FS servers would work, where calls are distributed accross 3 FS boxes by a load balancer with one spare in event of failure. Also how would a FS box at max capacity behave? Does FS monitor available resources and reject the excess calls that it can't handle? Or would the load balancer have to be configured with the maximum number of calls per box? Would love to hear some experiences of deploying FS with failover high load. Thanks Ash ___ 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 -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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] FS failover redundancy load balancing
Not opensips but the module is in alpha. In the modules doc page says alpha/new. Pablo On 4/3/09, Even André Fiskvik grev...@me.com wrote: Where do you guys read that it's in alpha? On the opensips.org they proclaim OpenSips 1.5 released, with that module being one of the new features. I don't see any mention of it being alpha/beta functionality? Best regards, Even André On 4. april. 2009, at 00.30, Pablo Hernan Saro wrote: Hi Kristian, you're right. Definitively that will be best solution as soon as it's released as stable (it's alpha now). http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing Pablo On Fri, Apr 3, 2009 at 6:42 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: You could try (although it's somewhat bleeding edge) to use OpenSIPS 1.5 with load_balancer (not heavily tested, btw) in front of some FreeSWITCH machines: http://www.opensips.org/html/docs/modules/devel/load_balancer.html 2009/4/2 Ashley van Gerven ashley@gmail.com: Hi, I can't find much info on setting up a redundant or heavy load FreeSwitch implementation. Are there any links apart from: http://wiki.freeswitch.org/wiki/Enterprise_deployment ? I imagine the entry level solution is to have two FS boxes configured identitcally, with redundant SBC software (recommendations?) in front, passing the calls to the primary FS box, or the backup FS box if the primary is not responding. Is that the easiest solution? What about a situation of having a level of concurrent calls beyond what one FS box can handle? I realise that would be a very large number of concurrent calls, but we would need a good plan on how to scale the systems. Are there recommendations for load balancing solutions? Either soft or hardware? My guess would be having 3 + 1 spare FS servers would work, where calls are distributed accross 3 FS boxes by a load balancer with one spare in event of failure. Also how would a FS box at max capacity behave? Does FS monitor available resources and reject the excess calls that it can't handle? Or would the load balancer have to be configured with the maximum number of calls per box? Would love to hear some experiences of deploying FS with failover high load. Thanks Ash ___ 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 -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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 -- Sent from Gmail for mobile | mobile.google.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] How to call multi gateways for failover with early media?
dujinfang dujinf...@gmail.com wrote: However, the caller do need to hear the early media to figure out what's going on. If I set ignore_early_media=false, only the first one tried. Could you use ring_ready? that way, the calling SIP phone should generate the ringback. ___ 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] How to call multi gateways for failover with early media?
First one to give media wins unless you ignore_early_media /b On Apr 3, 2009, at 6:53 PM, Jason White wrote: Could you use ring_ready? that way, the calling SIP phone should generate the ringback. Brian West br...@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS failover redundancy load balancing
Pablo, It is very cool and a very compelling reason to upgrade/move to OpenSIPS 1.5. I'm running (mostly) OpenSIPS 1.4.4/1.4.5 now and it's rock solid (as usual). It's really an excellent complement to FreeSWITCH! I will be doing testing with 1.5 and the new load balancer module shortly. On Fri, Apr 3, 2009 at 6:30 PM, Pablo Hernan Saro pablos...@gmail.com wrote: Hi Kristian, you're right. Definitively that will be best solution as soon as it's released as stable (it's alpha now). http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing Pablo -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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] FS failover redundancy load balancing
Hi Kristian, Let us know your experience as soon as you try it. Why not write a wiki page? =) On Fri, Apr 3, 2009 at 9:45 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Pablo, It is very cool and a very compelling reason to upgrade/move to OpenSIPS 1.5. I'm running (mostly) OpenSIPS 1.4.4/1.4.5 now and it's rock solid (as usual). It's really an excellent complement to FreeSWITCH! I will be doing testing with 1.5 and the new load balancer module shortly. On Fri, Apr 3, 2009 at 6:30 PM, Pablo Hernan Saro pablos...@gmail.com wrote: Hi Kristian, you're right. Definitively that will be best solution as soon as it's released as stable (it's alpha now). http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing Pablo -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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
[Freeswitch-users] compile problem in vista.
Hi, It's first time I install FS in Vista. After having downloaded the FS sources from svn. Follow the instruction on how to build FS on Windows. I Using Visual C++ 2008 Express Open Freeswitch.sln Right click the main solution node at the top of the Solution Explorer Right click and select Build after do this I was stoped by the problem. the error is like below, what need I to do? anyone can help me? Error 6 error C2008: '#' : unexpected in macro definition c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.h 1532 Error 8 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 9 error C2065: 'defiTE_a_15' : undeclared identifier c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 10 error C2099: initializer is not a constant c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 11 error C2061: syntax error : identifier 'defiTE_a_15' c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 12 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 13 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 14 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 15 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 16 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 17 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 18 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 19 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 20 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 21 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 22 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 23 error C2121: '#' : invalid character : possibly the result of a macro expansion c:\users\lenovo\documents\freeswitch\libs\flite-1.3.99\lang\cmulex\cmu_lts_model.c 21 Error 24 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal.c': No such file or directory c1 Error 25 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_diphone.c': No such file or directory c1 Error 26 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_lpc.c': No such file or directory c1 Error 27 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_res.c': No such file or directory c1 Error 28 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\cmu_us_kal\cmu_us_kal_residx.c': No such file or directory c1 Error 30 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_aswd.c': No such file or directory c1 Error 31 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_dur_stats.c': No such file or directory c1 Error 32 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_durz_cart.c': No such file or directory c1 Error 33 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_expand.c': No such file or directory c1 Error 34 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_f0_model.c': No such file or directory c1 Error 35 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_f0lr.c': No such file or directory c1 Error 36 fatal error C1083: Cannot open source file: '..\..\flite-1.3.99\lang\usenglish\us_ffeatures.c': No such file or directory c1 Error 37 fatal error C1083: Cannot open source file:
