Re: [Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario
Thanks, done. 2009/12/7 Michael Jerris m...@jerris.com: Please report bugs to jira.freeswitch.org. Mike On Dec 6, 2009, at 11:45 PM, Seven Du wrote: Hi, I know there's some chang on att_xfer, and after upgrade(re-bootstrap) to trunk code, no sound after att_xfer. Then I rebuild FS 15807 with a fresh checkout, but still using the old conf/ settings, sound is ok, but there are other problems: A call B, and B att_xfer C 1) origination_cancel_key not working. no even no DTMF log in FS when I press # or any other key, I tried with Zoiper and Snom(on the B leg) 2) when C answers, B immediately hangup, so B has no chance talk to C Could this be a problem? I pasted logs: http://pastebin.freeswitch.org/11417 ___ 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] Need Conference design help
Have a look at mod_conference http://wiki.freeswitch.org/wiki/Mod_conference On Sat, Dec 5, 2009 at 12:47 PM, shehzad p pmh...@gmail.com wrote: Hello Every one, I have to design conference, and I need community guidance to efficiently accomplish that. I need to create Conference which will have three kind of users: 1. Moderator (may be only one per conference) 2. User who can participate in conference without moderator interaction. 3. User who can only participate when Moderator allow them to get in. Also besides above setup I have to perform other things like Record the conference, Multicast the conference to other freeswitch server. I saw the conference Record CLI command but wondering where to setup when conference starts. I am also wondering how Multicast Conference is possible in Freeswitch and how the receiver Freeswitch configuration will look like. Thanks. msp -- View this message in context: http://old.nabble.com/Need-Conference-design-help-tp26653473p26653473.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] Choppy sound with PCMU
Here is what I found... I tried high-priority scheduling as per your suggestion, reniced the program explicitly, rewrote timer thread to sleep on cond. variable and activate only when there are timers and only when the timer actually had to be clicked, turned off SQL thread and removed polling from sofia profile thread. That pretty much eliminated all idle 1ms sleepers that were there except for three in sofia itself (su_epoll_port). And when I was about to be happy, I found that two outgoing calls through my VOIP providers when bridged together showed terrible distortions. I undid all my changes, tried 1.0.4, trunk (noticed btw that when I bridge two calls via loopback in JS in the trunk I must keep JS running, or the calls get terminated - NOT the same as in 1.0.4 where exitting JS left calls running), got pretty much the same sad results. At the same time calls bridged by freeswitch between LAN and any of the VOIP providers behaved just fine. And calls bridged by Asterisk any way were fine too. So that pretty much looked like the end of the freeswitch trials for me. But then I timed your code, mine and found that all those 1ms sleeps that your timer thread was doing (and all those pollers were doing as well) were actually 4ms sleeps because you know what unless kernel is configured with HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms (HZ=100). Mine was 250. This actually meant that the original timer thread was firing once, sleeping for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 times back-to-back, etc. It was still firing 20ms timers on time, but 30ms ones of course were not, since 30ms doesn't divide by 4 evenly. Plus whoever relied on runtime.reference or switch_micro_time_now() were kind of screwed because both were running jumpy. Plus whoever assumed that apr_sleep(1000) or cond_yield() was sleeping for 1ms were also in for a surprise. It felt satisfying to find that, however it didn't explain why the same distortions were observed with rewritten timer thread and disabled RTP timers. Anyway, I sighed (pretty much like you) and recompiled the kernel with HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes south, you need to hook up serial console and see what the heck went wrong. That eliminated distortions, ha! But made freeswitch more CPU hungry. Now the remaining 1ms threads sitting in sofia epoll were really polling for 1ms, not 4, and freeswitch was consistently sitting in the first line of the top chart showing 3% CPU utilization when idle. Don't know whether it's because of the remaining epolls in sofia or whether it's because there are still some threads left in freeswitch that I neglected to change because they were sleeping with 100ms interval, so I figured, who cares. Maybe when all things come together (sofia, 100ms*N) freeswitch ends up spending 3% of CPU while doing pretty much nothing. Btw, compared with Asterisk, the latter is not even visible on the first top's screen and spends 1% CPU when bridging two G711 calls and recording them to disk. So, at this time I have both original Asterisk and FS setups running. One is seemless but clumsy in configuration, the other one is neat and stylish but too preoccupied with smth... Should I look into sofia epollers? That's kind of deep in the code. Or should I just stick with Asterisk? Anthony Minessale-2 wrote: There is another user here with a 300mhz box. I am willing to investigate this improved performance for weak devices but I need to do it in a sane cross-platform way. On Fri, Dec 4, 2009 at 1:32 PM, Yossi Neiman freeswi...@cartissolutions.com wrote: A word to the wise to the general FreeSWITCH community: If Anthony Minessale suggests that you try to do any number of things, it's a very good idea to try all those ideas before continuing on. I've known him, MikeJ, and bkw for several years, and they almost always have very good ideas as to troubleshoot a problem in FreeSWITCH. It's extremely frustrating to try to help people out who won't try the provided suggestions first. And note directly to eaf - bogomips is quite possibly the least significant bit of data about a cpu that you will get out of /proc/cpuinfo... The name itself - bogo, means bogus. http://en.wikipedia.org/wiki/Bogomips -Yossi ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org http://pastebin.freeswitch.org/ On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. 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 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 mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net/ #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%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 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] Choppy sound with PCMU
Did you do each thing alone too to tell the difference? -hp alone, disable monotonic alone (i did not see you mention the disable monotonic) as for your 4ms thing, yes we require high resolution timing, if we ask to sleep 1000 microseconds that is what we need it to sleep for or at least as close as possible, and the main reason that thread is never sleeping is because you can't actually count on it to run every 1ms but you mostly can. Hence the whole philosophy on only making 1 thread run hot all the time to ensure that the rest don't have to repeat the same algorithm. We focus on high end performance this was the point of your experimentation because we will need to use a compile time defines and other logic to make it more efficient on your platform, a platform which we are not using. I am curious what would happen if you install Kristian's astlinux on one of your devices, i think you should also compare the kernel versions. What OS are you running anyway? Here are some more things to try (running plain trunk with no mods) do these systematically each alone and all together with/without -hp or disable monotonic etc to see what different combos create comment out this line (line 10) #define DISABLE_1MS_COND rebuild, this tells it to run a conditional at 1ms in the same timer thread which will make all the switch_cond_next share a 1ms conditional instead of doing microsleeps next some kernels/devices work better using select(0) for sleep where others work better using usleep. comment out line 109 apr_sleep(t); and try usleep(t) also mac works better using nanosleep so you could try changing it so it uses the code starting at 101 instead. also your claim about JS should be investigated because I do not think it should be the case. but you may want to move this to a jira http://jira.freeswitch.org As for the asterisk comparison, not sure how to answer you, that's your decision. On Mon, Dec 7, 2009 at 9:28 AM, eaf erandr-j...@usa.net wrote: Here is what I found... I tried high-priority scheduling as per your suggestion, reniced the program explicitly, rewrote timer thread to sleep on cond. variable and activate only when there are timers and only when the timer actually had to be clicked, turned off SQL thread and removed polling from sofia profile thread. That pretty much eliminated all idle 1ms sleepers that were there except for three in sofia itself (su_epoll_port). And when I was about to be happy, I found that two outgoing calls through my VOIP providers when bridged together showed terrible distortions. I undid all my changes, tried 1.0.4, trunk (noticed btw that when I bridge two calls via loopback in JS in the trunk I must keep JS running, or the calls get terminated - NOT the same as in 1.0.4 where exitting JS left calls running), got pretty much the same sad results. At the same time calls bridged by freeswitch between LAN and any of the VOIP providers behaved just fine. And calls bridged by Asterisk any way were fine too. So that pretty much looked like the end of the freeswitch trials for me. But then I timed your code, mine and found that all those 1ms sleeps that your timer thread was doing (and all those pollers were doing as well) were actually 4ms sleeps because you know what unless kernel is configured with HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms (HZ=100). Mine was 250. This actually meant that the original timer thread was firing once, sleeping for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 times back-to-back, etc. It was still firing 20ms timers on time, but 30ms ones of course were not, since 30ms doesn't divide by 4 evenly. Plus whoever relied on runtime.reference or switch_micro_time_now() were kind of screwed because both were running jumpy. Plus whoever assumed that apr_sleep(1000) or cond_yield() was sleeping for 1ms were also in for a surprise. It felt satisfying to find that, however it didn't explain why the same distortions were observed with rewritten timer thread and disabled RTP timers. Anyway, I sighed (pretty much like you) and recompiled the kernel with HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes south, you need to hook up serial console and see what the heck went wrong. That eliminated distortions, ha! But made freeswitch more CPU hungry. Now the remaining 1ms threads sitting in sofia epoll were really polling for 1ms, not 4, and freeswitch was consistently sitting in the first line of the top chart showing 3% CPU utilization when idle. Don't know whether it's because of the remaining epolls in sofia or whether it's because there are still some threads left in freeswitch that I neglected to change because they were sleeping with 100ms interval, so I figured, who cares. Maybe when all things come together (sofia, 100ms*N) freeswitch ends up spending 3% of CPU while doing pretty much nothing. Btw, compared
Re: [Freeswitch-users] Sporadic call drops
I will certainly shchedule time for the upgrade. Thanks for the answer On Fri, Dec 4, 2009 at 1:51 PM, Anthony Minessale anthony.miness...@gmail.com wrote: we changed that message a long time ago so people would not think that anymore We are now 3000 rev beyond the version you are at, I would like it if you try the lastest trunk. On Fri, Dec 4, 2009 at 12:16 PM, Luis F Urrea lfur...@gmail.com wrote: Hi all, Guys I know the question could be too vague, but I have a customer that just reported frequent failure to place outbound calls though a PSTN gateway on the LAN. I looked at the logs and I seem to be able to confirm that FS fails to place the call through the gateway and that the issue resides on the FS side since the first channel that s killed is tht of the internal extension registered to FS and then FS send the BYE to gw and kills the channel. What are possible causes of this? I know you always like to look at complete logs but here's a snip that could shed some light on the disconnection. (I can provide full logs if required and worthed) 2009-12-04 11:21:56 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/2...@172.16.3.5 entering state [ready][200] 2009-12-04 11:21:59 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/2...@172.16.3.5 entering state [terminated][200] 2009-12-04 11:21:59 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup sofia/internal/2...@172.16.3.5 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-04 11:21:59 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/internal/2...@172.16.3.5[kill] 2009-12-04 11:21:59 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/ 2...@172.16.3.5 [BREAK] 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:371 audio_bridge_thread() sofia/internal/2...@172.16.3.5 ending bridge by request from write function 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() sofia/pstn/22909...@172.16.3.46 receive message [UNBRIDGE] Is the 6th line normal behavior for ending the channel? FreeSWITCH Version 1.0.trunk (13484M) TIA ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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 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] Question regarding running FreeSWITCH with high priority enabled.
One thing that I forgot to mention, these 2 FreeSWITCH servers are getting calls with load balancing from another switch. Thus, the traffic type are pretty much identical and both FSs have exactly the same on configuration. Any suggestion would be appreciated. Thank you. From: DJB djbin...@yahoo.com To: FREESWITCH-USERS MAILING LIST freeswitch-users@lists.freeswitch.org Sent: Sun, December 6, 2009 5:17:14 PM Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4 (exported) with only one thing difference which is the first one is running with -hp enabled; however, I have noticed that the one with -hp option consumed double in memory usage than the other one. I wonder whether anyone can explain why. Thank you. Please see below: -- top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers Swap: 10223608k total,0k used, 10223608k free, 5417524k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch 4418 session(s) 14/100 root 30750 2.1 19.9 1879252 1634300 ? SLl Oct30 1153:50 /usr/local/freeswitch/bin/freeswitch -nc -hp -- top - 01:01:58 up 53 days, 2:45, 1 user, load average: 0.43, 0.51, 0.49 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.3%sy, 0.0%ni, 96.4%id, 2.4%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 6751260k used, 1422904k free, 203948k buffers Swap: 10223608k total,0k used, 10223608k free, 5432632k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 7147 root 15 0 1961m 755m 5164 S 9.0 9.5 1452:26 freeswitch 4478 session(s) 14/100 root 7147 1.9 9.4 2009392 774848 ? Sl Oct15 1452:37 /usr/local/freeswitch/bin/freeswitch -nc ___ 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] Question regarding running FreeSWITCH with high priority enabled.
One of the properties of -hp is to enable memlockall() which means disable swapping. This causes all memory used by FS to be resident permanently and is much more costly in memory usage. -hp also uses a RR scheduler runs the process at a less nice level and increases a few other process ulimits. This mode is designed for high end usage and uses more resources when idle with a large payout when scaling to many calls. On Mon, Dec 7, 2009 at 10:42 AM, DJB djbin...@yahoo.com wrote: One thing that I forgot to mention, these 2 FreeSWITCH servers are getting calls with load balancing from another switch. Thus, the traffic type are pretty much identical and both FSs have exactly the same on configuration. Any suggestion would be appreciated. Thank you. -- *From:* DJB djbin...@yahoo.com *To:* FREESWITCH-USERS MAILING LIST freeswitch-users@lists.freeswitch.org *Sent:* Sun, December 6, 2009 5:17:14 PM *Subject:* [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4 (exported) with only one thing difference which is the first one is running with -hp enabled; however, I have noticed that the one with -hp option consumed double in memory usage than the other one. I wonder whether anyone can explain why. Thank you. Please see below: -- top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers Swap: 10223608k total,0k used, 10223608k free, 5417524k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch 4418 session(s) 14/100 root 30750 2.1 *19.9* 1879252 1634300 ? SLl Oct30 1153:50 /usr/local/freeswitch/bin/freeswitch -nc -hp -- top - 01:01:58 up 53 days, 2:45, 1 user, load average: 0.43, 0.51, 0.49 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.3%sy, 0.0%ni, 96.4%id, 2.4%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 6751260k used, 1422904k free, 203948k buffers Swap: 10223608k total,0k used, 10223608k free, 5432632k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 7147 root 15 0 1961m 755m 5164 S 9.0 9.5 1452:26 freeswitch 4478 session(s) 14/100 root 7147 1.9 *9.4* 2009392 774848 ? Sl Oct15 1452:37 /usr/local/freeswitch/bin/freeswitch -nc ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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
Re: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled.
Anthony, Thank you for your clear response. Based on your recommendation, if I want to route more calls to the first server, should I take off -hp, or it's better to run with it. We are running FS for pass-thru traffic with signaling only. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, December 7, 2009 8:56:14 AM Subject: Re: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. One of the properties of -hp is to enable memlockall() which means disable swapping. This causes all memory used by FS to be resident permanently and is much more costly in memory usage. -hp also uses a RR scheduler runs the process at a less nice level and increases a few other process ulimits. This mode is designed for high end usage and uses more resources when idle with a large payout when scaling to many calls. On Mon, Dec 7, 2009 at 10:42 AM, DJB djbin...@yahoo.com wrote: One thing that I forgot to mention, these 2 FreeSWITCH servers are getting calls with load balancing from another switch. Thus, the traffic type are pretty much identical and both FSs have exactly the same on configuration. Any suggestion would be appreciated. Thank you. From: DJB djbin...@yahoo.com To: FREESWITCH-USERS MAILING LIST freeswitch-users@lists.freeswitch.org Sent: Sun, December 6, 2009 5:17:14 PM Subject: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4 (exported) with only one thing difference which is the first one is running with -hp enabled; however, I have noticed that the one with -hp option consumed double in memory usage than the other one. I wonder whether anyone can explain why. Thank you. Please see below: -- top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers Swap: 10223608k total,0k used, 10223608k free, 5417524k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch 4418 session(s) 14/100 root 30750 2.1 19.9 1879252 1634300 ? SLl Oct30 1153:50 /usr/local/freeswitch/bin/freeswitch -nc -hp -- top - 01:01:58 up 53 days, 2:45, 1 user, load average: 0.43, 0.51, 0.49 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.3%sy, 0.0%ni, 96.4%id, 2.4%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 6751260k used, 1422904k free, 203948k buffers Swap: 10223608k total,0k used, 10223608k free, 5432632k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 7147 root 15 0 1961m 755m 5164 S 9.0 9.5 1452:26 freeswitch 4478 session(s) 14/100 root 7147 1.9 9.4 2009392 774848 ? Sl Oct15 1452:37 /usr/local/freeswitch/bin/freeswitch -nc ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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
Re: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled.
maybe you can try both ways and see if there is a significant difference? I think -hp would help more if you were doing media than if you were not but that does not mean it could not still help performance but really the extra performance would only show up once you had consumed all the resources the box had to offer without -hp enabled in most cases. On Mon, Dec 7, 2009 at 11:12 AM, DJB djbin...@yahoo.com wrote: Anthony, Thank you for your clear response. Based on your recommendation, if I want to route more calls to the first server, should I take off -hp, or it's better to run with it. We are running FS for pass-thru traffic with signaling only. -- *From:* Anthony Minessale anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Mon, December 7, 2009 8:56:14 AM *Subject:* Re: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. One of the properties of -hp is to enable memlockall() which means disable swapping. This causes all memory used by FS to be resident permanently and is much more costly in memory usage. -hp also uses a RR scheduler runs the process at a less nice level and increases a few other process ulimits. This mode is designed for high end usage and uses more resources when idle with a large payout when scaling to many calls. On Mon, Dec 7, 2009 at 10:42 AM, DJB djbin...@yahoo.com wrote: One thing that I forgot to mention, these 2 FreeSWITCH servers are getting calls with load balancing from another switch. Thus, the traffic type are pretty much identical and both FSs have exactly the same on configuration. Any suggestion would be appreciated. Thank you. -- *From:* DJB djbin...@yahoo.com *To:* FREESWITCH-USERS MAILING LIST freeswitch-users@lists.freeswitch.org *Sent:* Sun, December 6, 2009 5:17:14 PM *Subject:* [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled. I have 2 identical Dell PowerEdge 1950 servers running FreeSWITCH Version 1.0.4 (exported) with only one thing difference which is the first one is running with -hp enabled; however, I have noticed that the one with -hp option consumed double in memory usage than the other one. I wonder whether anyone can explain why. Thank you. Please see below: -- top - 01:01:42 up 53 days, 2:45, 1 user, load average: 0.22, 0.28, 0.29 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.2%sy, 0.0%ni, 96.4%id, 2.5%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 7550092k used, 624072k free, 187568k buffers Swap: 10223608k total,0k used, 10223608k free, 5417524k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 30750 root -2 -10 1823m 1.5g 20m S 8.6 19.8 1153:40 freeswitch 4418 session(s) 14/100 root 30750 2.1 *19.9* 1879252 1634300 ? SLl Oct30 1153:50 /usr/local/freeswitch/bin/freeswitch -nc -hp -- top - 01:01:58 up 53 days, 2:45, 1 user, load average: 0.43, 0.51, 0.49 Tasks: 143 total, 1 running, 142 sleeping, 0 stopped, 0 zombie Cpu(s): 0.9%us, 0.3%sy, 0.0%ni, 96.4%id, 2.4%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 8174164k total, 6751260k used, 1422904k free, 203948k buffers Swap: 10223608k total,0k used, 10223608k free, 5432632k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 7147 root 15 0 1961m 755m 5164 S 9.0 9.5 1452:26 freeswitch 4478 session(s) 14/100 root 7147 1.9 *9.4* 2009392 774848 ? Sl Oct15 1452:37 /usr/local/freeswitch/bin/freeswitch -nc ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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
Re: [Freeswitch-users] Choppy sound with PCMU
oh and also use top -H to see which threads are using specific CPU and try to cross reference them by attaching with gdb and dumping all the thread bt On Mon, Dec 7, 2009 at 10:16 AM, Michael Jerris m...@jerris.com wrote: Also I have seen some people reporting that the new tickless timers in newer kernels work better. You may want to try those. Mike On Dec 7, 2009, at 11:00 AM, Anthony Minessale wrote: Did you do each thing alone too to tell the difference? -hp alone, disable monotonic alone (i did not see you mention the disable monotonic) as for your 4ms thing, yes we require high resolution timing, if we ask to sleep 1000 microseconds that is what we need it to sleep for or at least as close as possible, and the main reason that thread is never sleeping is because you can't actually count on it to run every 1ms but you mostly can. Hence the whole philosophy on only making 1 thread run hot all the time to ensure that the rest don't have to repeat the same algorithm. We focus on high end performance this was the point of your experimentation because we will need to use a compile time defines and other logic to make it more efficient on your platform, a platform which we are not using. I am curious what would happen if you install Kristian's astlinux on one of your devices, i think you should also compare the kernel versions. What OS are you running anyway? Here are some more things to try (running plain trunk with no mods) do these systematically each alone and all together with/without -hp or disable monotonic etc to see what different combos create comment out this line (line 10) #define DISABLE_1MS_COND rebuild, this tells it to run a conditional at 1ms in the same timer thread which will make all the switch_cond_next share a 1ms conditional instead of doing microsleeps next some kernels/devices work better using select(0) for sleep where others work better using usleep. comment out line 109 apr_sleep(t); and try usleep(t) also mac works better using nanosleep so you could try changing it so it uses the code starting at 101 instead. also your claim about JS should be investigated because I do not think it should be the case. but you may want to move this to a jira http://jira.freeswitch.org As for the asterisk comparison, not sure how to answer you, that's your decision. On Mon, Dec 7, 2009 at 9:28 AM, eaf erandr-j...@usa.net wrote: Here is what I found... I tried high-priority scheduling as per your suggestion, reniced the program explicitly, rewrote timer thread to sleep on cond. variable and activate only when there are timers and only when the timer actually had to be clicked, turned off SQL thread and removed polling from sofia profile thread. That pretty much eliminated all idle 1ms sleepers that were there except for three in sofia itself (su_epoll_port). And when I was about to be happy, I found that two outgoing calls through my VOIP providers when bridged together showed terrible distortions. I undid all my changes, tried 1.0.4, trunk (noticed btw that when I bridge two calls via loopback in JS in the trunk I must keep JS running, or the calls get terminated - NOT the same as in 1.0.4 where exitting JS left calls running), got pretty much the same sad results. At the same time calls bridged by freeswitch between LAN and any of the VOIP providers behaved just fine. And calls bridged by Asterisk any way were fine too. So that pretty much looked like the end of the freeswitch trials for me. But then I timed your code, mine and found that all those 1ms sleeps that your timer thread was doing (and all those pollers were doing as well) were actually 4ms sleeps because you know what unless kernel is configured with HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms (HZ=100). Mine was 250. This actually meant that the original timer thread was firing once, sleeping for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 times back-to-back, etc. It was still firing 20ms timers on time, but 30ms ones of course were not, since 30ms doesn't divide by 4 evenly. Plus whoever relied on runtime.reference or switch_micro_time_now() were kind of screwed because both were running jumpy. Plus whoever assumed that apr_sleep(1000) or cond_yield() was sleeping for 1ms were also in for a surprise. It felt satisfying to find that, however it didn't explain why the same distortions were observed with rewritten timer thread and disabled RTP timers. Anyway, I sighed (pretty much like you) and recompiled the kernel with HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes south, you need to hook up serial console and see what the heck went wrong. That eliminated distortions, ha! But made freeswitch more CPU hungry. Now the remaining 1ms threads sitting in sofia epoll were really polling for 1ms, not 4, and freeswitch was consistently
Re: [Freeswitch-users] lua+sqlite example?
yes if you use the lua odbc sql plugin you should be able to use that for sqlite, they may also have a native one. On Sat, Dec 5, 2009 at 9:21 PM, Steve Klein skl...@singular.com wrote: Greetings. We are attempting to add sqlite access to an IVR application we are prototyping. We are using lua for the scripts. Is there an example anywhere of a lua + sqlite script? Do we need to install luasql? Any help/pointers greatly appreciated. --Steve Klein ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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
Re: [Freeswitch-users] Mutual Registration of servers
On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah ab...@greatiam.comwrote: Pardon me if this has been addressed already. How does one go about having in the simplest instance 2 servers registering with each other on startup whereby the users registering would be able to call each other. The 2 servers are in different domains. Thanks. Are the two servers in different locations? Different LANs? Is NAT involved? Just checking. Really this is just a matter of loading the default config on each machine and then making some decisions about the dialplan: do you want prefix dialing so that you can have ext 1000 at both locations or do you want to have something like 1000~1099 at location A and 1100~1199 at location B? From there it's just a matter of creating the gateways on each machine and adding a dialplan entry to handle the routing. -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
[Freeswitch-users] esl for Mac OS X 10.4
I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can't get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation - MAC os X. I have also googled this, and don't see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB=../libesl.a SOLINK=-Xlinker -x CFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable CXX_CFLAGS= -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 ___ 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] Access to users variables
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined below:
include
user id=1000 mailbox=1000
params
param name=password value=1000/
/params
variables
variable name=smsnumber value=12345/
/variables
/user
/include
How can I use variable ${smsnumber} in a dialplan to run a perl script
using action application=system data=sms.pl/ ?
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] esl for Mac OS X 10.4
The build system for libesl and everything below that won't work 100% on the mac just yet. You have to make some changes to how its linked and you'll have to compile php yourself to get everything in there properly. The perl one however is much easier to fix. -SOLINK=-shared -Xlinker -x +SOLINK=-dynamiclib -Xlinker -x Thats all you usually fix for the mac. /b On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can’t get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation - MAC os X. I have also googled this, and don’t see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB=../libesl.a SOLINK=-Xlinker -x CFLAGS=-I/usr/src/ freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../ libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable - Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=- I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g - ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable CXX_CFLAGS= -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc - lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 ___ 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] esl for Mac OS X 10.4
Forgive me if I ask the obvious questions... Did you make in src/libs/esl before doing make phpmod ? Did you install the php-devel stuff? -MC On Mon, Dec 7, 2009 at 9:27 AM, Kendall Stauffer k...@ksac.com wrote: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can’t get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation - MAC os X. I have also googled this, and don’t see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB=../libesl.a SOLINK=-Xlinker -x CFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable CXX_CFLAGS= -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 ___ 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] Access to users variables
On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined
below:
include
user id=1000 mailbox=1000
params
param name=password value=1000/
/params
variables
variable name=smsnumber value=12345/
/variables
/user
/include
How can I use variable ${smsnumber} in a dialplan to run a perl script
using action application=system data=sms.pl/ ?
Do you just want to pass the value in smsnumber to the sms.pl script? Have
you tried this?
action application=system data=sms.pl ${smsnumber}/
-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
[Freeswitch-users] continue_on_fail
I have a Problem with continue_on_fail. I have setup a hunt group action application=set data=continue_on_fail=NO_ANSWER,USER_BUSY/ action application=bridge data=sofia/external/2...@10.11.12.243,sofia/external/2...@10.11.12.234,sofia/external/2...@10.11.12.188,sofia/external/1...@10.11.12.245/ action application=bridge data= (dialstring for fallback user ) I want the fallback user to be called whenever none of the previously called 3 gateway numbers picks up or if they are all busy. Therefore continue_on_fail=NO_ANSWER,USER_BUSY The fallback user is called, however if any of the previously called gateways picks up and then hangs up, the fallback user is called afterwards. Means: The fallback user is always called. I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire the next bridge if it gets a NORMAL_CLEARING. Am I thinking wrongly about this? I have added action application=set data=hangup_after_bridge=true/ and this works, but I would like to specify more in detail the conditions when to follow the next hunt group entry. Best regards Peter ___ 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] esl for Mac OS X 10.4
Any direction on where to start would be appreciated. I am trying to get freepbx working with this, and everything works (I think) except esl From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, December 07, 2009 1:10 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] esl for Mac OS X 10.4 The build system for libesl and everything below that won't work 100% on the mac just yet. You have to make some changes to how its linked and you'll have to compile php yourself to get everything in there properly. The perl one however is much easier to fix. -SOLINK=-shared -Xlinker -x +SOLINK=-dynamiclib -Xlinker -x Thats all you usually fix for the mac. /b On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can't get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation - MAC os X. I have also googled this, and don't see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB=../libesl.a SOLINK=-Xlinker -x CFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable CXX_CFLAGS= -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto: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] esl for Mac OS X 10.4
I did make first, but did not install any extra dev stuff, thinking I already had them. Is there a way to turn on verbose and finding out exactly what it no there that is expected? Thanksmuch!! From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, December 07, 2009 1:22 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] esl for Mac OS X 10.4 Forgive me if I ask the obvious questions... Did you make in src/libs/esl before doing make phpmod ? Did you install the php-devel stuff? -MC On Mon, Dec 7, 2009 at 9:27 AM, Kendall Stauffer k...@ksac.commailto:k...@ksac.com wrote: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can't get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation - MAC os X. I have also googled this, and don't see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB=../libesl.a SOLINK=-Xlinker -x CFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable CXX_CFLAGS= -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto: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] Access to users variables
Hi!
That's exactly what I want to do and that was the first thing I tried,
but nothing is passed to the script.
In a case like this, what defines if variable smsnumber is taken from
the A path or B path? (The A path does not have smsnumber defined)
On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins m...@freeswitch.org wrote:
On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined
below:
include
user id=1000 mailbox=1000
params
param name=password value=1000/
/params
variables
variable name=smsnumber value=12345/
/variables
/user
/include
How can I use variable ${smsnumber} in a dialplan to run a perl script
using action application=system data=sms.pl/ ?
Do you just want to pass the value in smsnumber to the sms.pl script? Have
you tried this?
action application=system data=sms.pl ${smsnumber}/
-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
___
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] compilation error of skypiax_protocol.c
Maybe, just maybe isse that make target to reconf libtiff? Regards, JM On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang jingwei.y...@gmail.com wrote: I installed libjpeg-7 following this website: http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And the previous error is replaced by a new one: gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o at_interpreter.o at_interpreter.c: In function ‘command_search’: at_interpreter.c:5299: error: ‘COMMAND_TRIE_LEN’ undeclared (first use in this function) at_interpreter.c:5299: error: (Each undeclared identifier is reported only once at_interpreter.c:5299: error: for each function it appears in.) at_interpreter.c:5308: error: ‘command_trie’ undeclared (first use in this function) at_interpreter.c: In function ‘at_interpreter’: at_interpreter.c:5424: error: ‘at_commands’ undeclared (first use in this function) make[8]: *** [at_interpreter.lo] Error 1 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 However, I'm still able to start freeswitch and mod_skypiax and make skype calls with no problem. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang jingwei.y...@gmail.comwrote: No, I didn't change or update the system libs. I just wanted to double check whether my system has this libjpeg library. ./configure was definitely executed before the source codes were rebuilt. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene mrene_li...@avgs.ca wrote: Hi, That one is on your side. If you changed/updated system libs it might be worth doing another ./configure Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: Hi Mathieu, thanks for the promptly reply. The error has been fixed. However, I encounter another one. gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: cannot open shared object file: No such file or directory make[8]: *** [at_interpreter_dictionary.h] Error 127 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 Do you have idea about this one? Thanks! On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene mrene_li...@avgs.cawrote: Consider it fixed. Committed revision 15765. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: Hi Guys, I got a compilation error of skypiax_protocol.c with the latest version r15764. Compiling skypiax_protocol.c... *cc1: warnings being treated as errors* skypiax_protocol.c: In function ‘X11_errors_handler’: skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ‘skypiax_send_message’: skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ‘skypiax_do_skypeapi_thread_func’: skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and code make[5]: *** [skypiax_protocol.o] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_skypiax-install] Error 1 make[2]: *** [install-recursive] Error 1 I personally checked the file and it shouldn't be a merge problem. Does anyone encounter this as well? ___ 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
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org http://pastebin.freeswitch.org/ On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. 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 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 mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net/ #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%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 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] Mutual Registration of servers
Michael Collins wrote: On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah ab...@greatiam.com mailto:ab...@greatiam.com wrote: Pardon me if this has been addressed already. How does one go about having in the simplest instance 2 servers registering with each other on startup whereby the users registering would be able to call each other. The 2 servers are in different domains. Thanks. Are the two servers in different locations? Different LANs? Is NAT involved? Just checking. Really this is just a matter of loading the default config on each machine and then making some decisions about the dialplan: do you want prefix dialing so that you can have ext 1000 at both locations or do you want to have something like 1000~1099 at location A and 1100~1199 at location B? From there it's just a matter of creating the gateways on each machine and adding a dialplan entry to handle the routing. -MC Hello Michael Thanks Are the two servers in different locations? Yes Different LANs? Yes Is NAT involved? Yes but for my test Nat is not . The production setup I have in mind will certainly have Nat Each location will have their won set of extension but there could be some overlap. On server A a user would dial,. for example, 98 followed by the extension number of the user on server B and the call would then be routed to the extension on server B. And the same could be from Server B to a user on Server A MC 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] Access to users variables
Check out
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
you might just need to set the user so that the vars become available on the
leg you're processing.
-MC
On Mon, Dec 7, 2009 at 10:37 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
That's exactly what I want to do and that was the first thing I tried,
but nothing is passed to the script.
In a case like this, what defines if variable smsnumber is taken from
the A path or B path? (The A path does not have smsnumber defined)
On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins m...@freeswitch.org
wrote:
On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined
below:
include
user id=1000 mailbox=1000
params
param name=password value=1000/
/params
variables
variable name=smsnumber value=12345/
/variables
/user
/include
How can I use variable ${smsnumber} in a dialplan to run a perl script
using action application=system data=sms.pl/ ?
Do you just want to pass the value in smsnumber to the sms.pl script?
Have
you tried this?
action application=system data=sms.pl ${smsnumber}/
-MC
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
[Freeswitch-users] no hangup on B leg
Hi all, I'll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I'm not seeing a hangup of the b leg at all. FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it's not being fired. Does anyone have an idea what might be causing this? Regards, ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Access to users variables
Perfect...
action application=set_user data=${dialed_extensi...@${domain}/
works like a charm.
Thanks Mike.
On Tue, Dec 8, 2009 at 5:56 AM, Michael Collins m...@freeswitch.org wrote:
Check out
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user
you might just need to set the user so that the vars become available on the
leg you're processing.
-MC
On Mon, Dec 7, 2009 at 10:37 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
That's exactly what I want to do and that was the first thing I tried,
but nothing is passed to the script.
In a case like this, what defines if variable smsnumber is taken from
the A path or B path? (The A path does not have smsnumber defined)
On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins m...@freeswitch.org
wrote:
On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined
below:
include
user id=1000 mailbox=1000
params
param name=password value=1000/
/params
variables
variable name=smsnumber value=12345/
/variables
/user
/include
How can I use variable ${smsnumber} in a dialplan to run a perl script
using action application=system data=sms.pl/ ?
Do you just want to pass the value in smsnumber to the sms.pl script?
Have
you tried this?
action application=system data=sms.pl ${smsnumber}/
-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
___
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] No NOTIFY MWI when registering via proxy.
Hello,
i now changed the $${domain} name of the server to the domain name the
phones register with.
Now messaging (MWI, notify) works.
Best regards
Peter
Peter P GMX schrieb:
Hello Anthony,
I did some checks today
Here is how the phones are registered:
mysql select sip_host, presence_hosts, server_user,server_host,
hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1;
+---+---+-+-++---+--+---+
| sip_host | presence_hosts| server_user | server_host |
hostname | sip_realm | mwi_user | mwi_host |
+---+---+-+-++---+--+---+
| sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 |
sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com |
+---+---+-+-++---+--+---+
IPs are:
10.11.12.1 sip1.mydomain.com (common cluster IP)
10.11.12.2 sip11.mydomain.com
10.11.12.3 sip12.mydomain.com (not used at this point)
XML-Curl for the directory is:
document type=freeswitch/xml
section name=directory
domain name=sip1.mydomain.com
user id=100
params
param name=password value=pass/
param name=vm-password value=pass/
param name=vm-email-all-messages value=true/
param name=vm-attach-file value=true/
param name=vm-mailto value=em...@domain.net/
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_us...@${dialed_domain})}/
param name=http-allowed-api value=voicemail/
/params
variables
variable name=accountcode value=800/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 100/
variable name=effective_caller_id_number value=100/
/variables
/user
/domain
/section
/document
The internal profile has the following alias:
profile name=internal domain=$${domain}
aliases
alias name=$${domain}/
alias name=sip1.mydomain.com/
alias name=default/
/aliases
With $${domain} being sip11.mydomain.com
Phones are registering to sip1.mydomain.com, Voicemail works, but MWI
does not. Any hint what I should change to make this work?
Best regards
Peter
Anthony Minessale schrieb:
based on your example past
sip1.mydomain.com http://sip1.mydomain.com is the domain in the
packet and thus the profile should have an alias for this.
Then the user must reside in your sip db with the user 200 and domain
sip1.mydomain.com http://sip1.mydomain.com
if you dont have this consider the force-register-domain and
force-register-db-domain to normalize the host names.
On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale
anthony.miness...@gmail.com mailto:anthony.miness...@gmail.com wrote:
Did you check the 2 replies that told you you need aliases in your
sofia profile to translate the domain found in your
message_waiting to the right profile? Both Brian and Mike
answered you.
On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX
prometheus...@gmx.net mailto:prometheus...@gmx.net wrote:
I tried now with phones directly attached to the freeswitch
(without an
OpenSIPS in between). I also added the alias. But the
behaviour is as
before:
No notify message from freeswitch, neither after register nor
after a
voicemail is recorded.
Best regards
Peter
Brian West schrieb:
Yes an alias will be required for every domain you run on
the profile
so it can find it.
/b
On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
Try an alias on the sip profile.
Mike
___
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
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter:
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
try rerunning the ./bootstrap.sh On Mon, Dec 7, 2009 at 12:49 PM, Jerry Richards jerry.richa...@teotech.comwrote: When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry -- *From:* Jerry Richards [mailto:jerry.richa...@teotech.com] *Sent:* Monday, December 07, 2009 7:44 AM *To:* 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry -- *From:* Michael Jerris [mailto:m...@jerris.com] *Sent:* Saturday, December 05, 2009 7:30 PM *To:* Jerry Richards *Subject:* Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. 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 -- 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 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 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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
Re: [Freeswitch-users] Access to users variables
On Mon, Dec 7, 2009 at 11:09 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Perfect...
action application=set_user data=${dialed_extension}@
${domain}/
works like a charm.
Another satisfied customer! :P
___
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] esl for Mac OS X 10.4
also, don't use 1.0.4, please us the latest SVN or last svn snapshot at the very least. On Mon, Dec 7, 2009 at 12:34 PM, Kendall Stauffer k...@ksac.com wrote: Any direction on where to start would be appreciated. I am trying to get freepbx working with this, and everything works (I think) except esl *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* Monday, December 07, 2009 1:10 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] esl for Mac OS X 10.4 The build system for libesl and everything below that won't work 100% on the mac just yet. You have to make some changes to how its linked and you'll have to compile php yourself to get everything in there properly. The perl one however is much easier to fix. -SOLINK=-shared -Xlinker -x +SOLINK=-dynamiclib -Xlinker -x Thats all you usually fix for the mac. /b On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can’t get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation - MAC os X. I have also googled this, and don’t see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB=../libesl.a SOLINK=-Xlinker -x CFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable CXX_CFLAGS= -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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
Re: [Freeswitch-users] Database suggestions/pointers/?
Thanks for the suggestions. We'll explore. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Tim Uckun Sent: Sunday, December 06, 2009 5:00 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Database suggestions/pointers/? On Mon, Dec 7, 2009 at 9:07 AM, Steve Klein skl...@singular.com wrote: Greetings. We need to add database access to an IVR application we are prototyping. Based on FS best practice suggestions, we are using Lua for the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite would be a recommended approach. However, we cant find any examples of this combo anywhere. So, what is the best practice scripting + database recommendation for a high-volume database-driven FS app? I would suggest you take a look at freeswitcher (http://github.com/bougyman/freeswitcher). The good thing is that it's ruby and therefore you can use any database compatible with ruby (that's all of them pretty much). You can also use an ORM of your choice or if you don't want to use an ORM you can use the amazingly fantastic sequel library. Being ruby it will run outside of the freeswitch memory space and you will have to use the inbound/outbound socket API. That may be a good thing if you want to separate your database and IVR logic from the machine running your freeswitch. Ruby is pretty easy to pick up if you don't know it and there are a wealth of libraries if you want to do other things like connect to web sites, manipulate XML, etc. There is also a liverpie http://github.com/jsgoecke/liverpie which is more of a proxy thing you can interface with any language. I am sure lua is nice but it seems like people are having some problems with ODBC, memory leaks etc when it comes to databases. If you go a ruby library that all goes away. ___ 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] lua+sqlite example?
Thanks. We'll look at that. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, December 07, 2009 9:35 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] lua+sqlite example? yes if you use the lua odbc sql plugin you should be able to use that for sqlite, they may also have a native one. On Sat, Dec 5, 2009 at 9:21 PM, Steve Klein skl...@singular.com wrote: Greetings. We are attempting to add sqlite access to an IVR application we are prototyping. We are using lua for the scripts. Is there an example anywhere of a lua + sqlite script? Do we need to install luasql? Any help/pointers greatly appreciated. --Steve Klein ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.426 / Virus Database: 270.14.83/2529 - Release Date: 12/07/09 07:33:00 ___ 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] pfSense with Freeswitch - so far so good, but no calls going through
Hi All Ok, so next episode in the saga of getting this monster of the ground :-) I've gotten the FS up and running pretty much I guess, but I'm missing something. It has been set up as per the 'multi-homed' document (http://wiki.freeswitch.org/wiki/Multi_home_tutorial). I want to use the webinterface in pfSense, as it is the easiest for me to manage, and gives me a better overview. If I do a reloadxml it gives me this output on the console: freeswi...@firewall.fribert.dk reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) 2009-12-07 21:21:40.719721 [INFO] mod_enum.c:808 ENUM Reloaded 2009-12-07 21:21:40.719721 [INFO] switch_time.c:661 Timezone reloaded 530 definitions API CALL [reloadxml()] output: +OK [Success] I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Looking at the status I see: sofia status profile internal API CALL [sofia(status profile internal)] output: = Nameinternal Domain Name N/A DBName sofia_reg_internal Pres Hosts DialplanXML Context public Challenge Realm auto_from RTP-IP 10.11.12.25 Ext-RTP-IP 10.11.12.25 SIP-IP 10.11.12.25 Ext-SIP-IP 10.11.12.25 URL sip:mod_so...@10.11.12.25:5060 BIND-URLsip:mod_so...@10.11.12.25:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS g7...@32000h,g7...@16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false CALLS-IN10 FAILED-CALLS-IN 5 CALLS-OUT 0 FAILED-CALLS-OUT0 Registrations: = Call-ID:MThhODdkOWFkMGM4YTk5OWU1MTMzMjg5NmFjOGFhNWU. User: 1...@10.11.12.25 Contact:1001 sip:1...@10.11.12.145:59650;rinstance=7c2be2513804dcea;fs_nat=yes;fs_path=sip%3A1001%4010.11.12.145%3A59650%3Brinstance%3D7c2be2513804dcea Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:03:40) Host: firewall.fribert.dk IP: 10.11.12.145 Port: 59650 Auth-User: 1001 Auth-Realm: 10.11.12.25 Call-ID:OTc2NTJkMmU3MGQ0MDNkN2NiZDgzZDFjYzQ1MzYxMDY. User: 1...@10.11.12.25 Contact:1002 sip:1...@10.11.12.195:4117;rinstance=8c18309a2c957d6a;fs_nat=yes;fs_path=sip%3A1002%4010.11.12.195%3A4117%3Brinstance%3D8c18309a2c957d6a Agent: 3CXVoipPhone 3.1.6288.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-12-07 23:25:09) Host: firewall.fribert.dk IP: 10.11.12.195 Port: 4117 Auth-User: 1002 Auth-Realm: 10.11.12.25 = And... sofia status profile external API CALL [sofia(status profile external)] output: = Nameexternal Domain Name N/A DBName sofia_reg_external Pres Hosts DialplanXML Context public Challenge Realm auto_to RTP-IP 87.61.18.196 Ext-RTP-IP 87.61.18.196 SIP-IP 87.61.18.196 Ext-SIP-IP 87.61.18.196 URL sip:mod_so...@87.61.18.196:5080 BIND-URLsip:mod_so...@87.61.18.196:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false CALLS-IN0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT2 Registrations: = = In my Dialplan
Re: [Freeswitch-users] Choppy sound with PCMU
What do you want me to check while running these tests? Sound quality (it's good now even with original 1.0.4). Or CPU utilization? It's Debian 4. Anthony Minessale-2 wrote: Did you do each thing alone too to tell the difference? -hp alone, disable monotonic alone (i did not see you mention the disable monotonic) as for your 4ms thing, yes we require high resolution timing, if we ask to sleep 1000 microseconds that is what we need it to sleep for or at least as close as possible, and the main reason that thread is never sleeping is because you can't actually count on it to run every 1ms but you mostly can. Hence the whole philosophy on only making 1 thread run hot all the time to ensure that the rest don't have to repeat the same algorithm. We focus on high end performance this was the point of your experimentation because we will need to use a compile time defines and other logic to make it more efficient on your platform, a platform which we are not using. I am curious what would happen if you install Kristian's astlinux on one of your devices, i think you should also compare the kernel versions. What OS are you running anyway? Here are some more things to try (running plain trunk with no mods) do these systematically each alone and all together with/without -hp or disable monotonic etc to see what different combos create comment out this line (line 10) #define DISABLE_1MS_COND rebuild, this tells it to run a conditional at 1ms in the same timer thread which will make all the switch_cond_next share a 1ms conditional instead of doing microsleeps next some kernels/devices work better using select(0) for sleep where others work better using usleep. comment out line 109 apr_sleep(t); and try usleep(t) also mac works better using nanosleep so you could try changing it so it uses the code starting at 101 instead. also your claim about JS should be investigated because I do not think it should be the case. but you may want to move this to a jira http://jira.freeswitch.org As for the asterisk comparison, not sure how to answer you, that's your decision. On Mon, Dec 7, 2009 at 9:28 AM, eaf erandr-j...@usa.net wrote: Here is what I found... I tried high-priority scheduling as per your suggestion, reniced the program explicitly, rewrote timer thread to sleep on cond. variable and activate only when there are timers and only when the timer actually had to be clicked, turned off SQL thread and removed polling from sofia profile thread. That pretty much eliminated all idle 1ms sleepers that were there except for three in sofia itself (su_epoll_port). And when I was about to be happy, I found that two outgoing calls through my VOIP providers when bridged together showed terrible distortions. I undid all my changes, tried 1.0.4, trunk (noticed btw that when I bridge two calls via loopback in JS in the trunk I must keep JS running, or the calls get terminated - NOT the same as in 1.0.4 where exitting JS left calls running), got pretty much the same sad results. At the same time calls bridged by freeswitch between LAN and any of the VOIP providers behaved just fine. And calls bridged by Asterisk any way were fine too. So that pretty much looked like the end of the freeswitch trials for me. But then I timed your code, mine and found that all those 1ms sleeps that your timer thread was doing (and all those pollers were doing as well) were actually 4ms sleeps because you know what unless kernel is configured with HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms (HZ=100). Mine was 250. This actually meant that the original timer thread was firing once, sleeping for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 times back-to-back, etc. It was still firing 20ms timers on time, but 30ms ones of course were not, since 30ms doesn't divide by 4 evenly. Plus whoever relied on runtime.reference or switch_micro_time_now() were kind of screwed because both were running jumpy. Plus whoever assumed that apr_sleep(1000) or cond_yield() was sleeping for 1ms were also in for a surprise. It felt satisfying to find that, however it didn't explain why the same distortions were observed with rewritten timer thread and disabled RTP timers. Anyway, I sighed (pretty much like you) and recompiled the kernel with HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes south, you need to hook up serial console and see what the heck went wrong. That eliminated distortions, ha! But made freeswitch more CPU hungry. Now the remaining 1ms threads sitting in sofia epoll were really polling for 1ms, not 4, and freeswitch was consistently sitting in the first line of the top chart showing 3% CPU utilization when idle. Don't know whether it's because of the remaining epolls in sofia or whether it's because there are still some threads left in
Re: [Freeswitch-users] no hangup on B leg
Sorry no, apart from the fact that I was seeing the hangup. I'm wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for '*' and force a hangup? I don't seem to able to see this tone on the B leg though. Regards, From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 19:12 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi all, I'll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I'm not seeing a hangup of the b leg at all. FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it's not being fired. Does anyone have an idea what might be causing this? Regards, Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? -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] Choppy sound with PCMU
Both, if it always sounds ok then I guess CPU usage. On Mon, Dec 7, 2009 at 2:58 PM, eaf erandr-j...@usa.net wrote: What do you want me to check while running these tests? Sound quality (it's good now even with original 1.0.4). Or CPU utilization? It's Debian 4. Anthony Minessale-2 wrote: Did you do each thing alone too to tell the difference? -hp alone, disable monotonic alone (i did not see you mention the disable monotonic) as for your 4ms thing, yes we require high resolution timing, if we ask to sleep 1000 microseconds that is what we need it to sleep for or at least as close as possible, and the main reason that thread is never sleeping is because you can't actually count on it to run every 1ms but you mostly can. Hence the whole philosophy on only making 1 thread run hot all the time to ensure that the rest don't have to repeat the same algorithm. We focus on high end performance this was the point of your experimentation because we will need to use a compile time defines and other logic to make it more efficient on your platform, a platform which we are not using. I am curious what would happen if you install Kristian's astlinux on one of your devices, i think you should also compare the kernel versions. What OS are you running anyway? Here are some more things to try (running plain trunk with no mods) do these systematically each alone and all together with/without -hp or disable monotonic etc to see what different combos create comment out this line (line 10) #define DISABLE_1MS_COND rebuild, this tells it to run a conditional at 1ms in the same timer thread which will make all the switch_cond_next share a 1ms conditional instead of doing microsleeps next some kernels/devices work better using select(0) for sleep where others work better using usleep. comment out line 109 apr_sleep(t); and try usleep(t) also mac works better using nanosleep so you could try changing it so it uses the code starting at 101 instead. also your claim about JS should be investigated because I do not think it should be the case. but you may want to move this to a jira http://jira.freeswitch.org As for the asterisk comparison, not sure how to answer you, that's your decision. On Mon, Dec 7, 2009 at 9:28 AM, eaf erandr-j...@usa.net wrote: Here is what I found... I tried high-priority scheduling as per your suggestion, reniced the program explicitly, rewrote timer thread to sleep on cond. variable and activate only when there are timers and only when the timer actually had to be clicked, turned off SQL thread and removed polling from sofia profile thread. That pretty much eliminated all idle 1ms sleepers that were there except for three in sofia itself (su_epoll_port). And when I was about to be happy, I found that two outgoing calls through my VOIP providers when bridged together showed terrible distortions. I undid all my changes, tried 1.0.4, trunk (noticed btw that when I bridge two calls via loopback in JS in the trunk I must keep JS running, or the calls get terminated - NOT the same as in 1.0.4 where exitting JS left calls running), got pretty much the same sad results. At the same time calls bridged by freeswitch between LAN and any of the VOIP providers behaved just fine. And calls bridged by Asterisk any way were fine too. So that pretty much looked like the end of the freeswitch trials for me. But then I timed your code, mine and found that all those 1ms sleeps that your timer thread was doing (and all those pollers were doing as well) were actually 4ms sleeps because you know what unless kernel is configured with HZ=1000, you can't sleep for less than 4ms (HZ=250) or perhaps even 10ms (HZ=100). Mine was 250. This actually meant that the original timer thread was firing once, sleeping for 4ms, firing 3 more times back-to-back, sleeping for 4ms more, firing 4 times back-to-back, etc. It was still firing 20ms timers on time, but 30ms ones of course were not, since 30ms doesn't divide by 4 evenly. Plus whoever relied on runtime.reference or switch_micro_time_now() were kind of screwed because both were running jumpy. Plus whoever assumed that apr_sleep(1000) or cond_yield() was sleeping for 1ms were also in for a surprise. It felt satisfying to find that, however it didn't explain why the same distortions were observed with rewritten timer thread and disabled RTP timers. Anyway, I sighed (pretty much like you) and recompiled the kernel with HZ=1000. Recompiling kernel on these ALIX boards is fun. If smth goes south, you need to hook up serial console and see what the heck went wrong. That eliminated distortions, ha! But made freeswitch more CPU hungry. Now the remaining 1ms threads sitting in sofia epoll were really polling for 1ms, not 4, and
[Freeswitch-users] Trapping dtmf on bridged call
Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Regards, ___ 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] Trapping dtmf on bridged call
session:execute(start_dtmf); /b On Dec 7, 2009, at 4:02 PM, Nik Middleton wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I’ve tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can’t see how to launch this within LUA Regards, ___ 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] Trapping dtmf on bridged call
session:execute(start_dtmf); this app captures inband audio tone dtmf and interprets them aka calls your callback etc. On Mon, Dec 7, 2009 at 4:02 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I’ve tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can’t see how to launch this within LUA Regards, ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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
Re: [Freeswitch-users] Trapping dtmf on bridged call
On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I’ve tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can’t see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -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] Trapping dtmf on bridged call
Can this be done in an lua script? Regards, From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 22:18 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -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] Trapping dtmf on bridged call
Once the call is bridged, while I can see an inband DTMF event being generated, it doesn't call my hook unfortuneately function onInput(session, type, obj) if type == dtmf and obj['digit'] == '*' then session:hangup(); return true; end session:execute(start_dtmf); session:execute(bridge,bridgestring ); Am I missing something? Before the bridge, the oninput function works fine Regards, From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 07 December 2009 22:15 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call session:execute(start_dtmf); this app captures inband audio tone dtmf and interprets them aka calls your callback etc. On Mon, Dec 7, 2009 at 4:02 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Regards, ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%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
Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
Hello,
i now changed the $${domain} of the server to the domain name the
phones register with.
Now messaging (MWI, notify) works. Thanks to all for your support.
Best regards
Peter
Peter P GMX schrieb:
Hello Anthony,
I did some checks today
Here is how the phones are registered:
mysql select sip_host, presence_hosts, server_user,server_host,
hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1;
+---+---+-+-++---+--+---+
| sip_host | presence_hosts| server_user | server_host |
hostname | sip_realm | mwi_user | mwi_host |
+---+---+-+-++---+--+---+
| sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 |
sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com |
+---+---+-+-++---+--+---+
IPs are:
10.11.12.1 sip1.mydomain.com (common cluster IP)
10.11.12.2 sip11.mydomain.com
10.11.12.3 sip12.mydomain.com (not used at this point)
XML-Curl for the directory is:
document type=freeswitch/xml
section name=directory
domain name=sip1.mydomain.com
user id=100
params
param name=password value=pass/
param name=vm-password value=pass/
param name=vm-email-all-messages value=true/
param name=vm-attach-file value=true/
param name=vm-mailto value=em...@domain.net/
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_us...@${dialed_domain})}/
param name=http-allowed-api value=voicemail/
/params
variables
variable name=accountcode value=800/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 100/
variable name=effective_caller_id_number value=100/
/variables
/user
/domain
/section
/document
The internal profile has the following alias:
profile name=internal domain=$${domain}
aliases
alias name=$${domain}/
alias name=sip1.mydomain.com/
alias name=default/
/aliases
With $${domain} being sip11.mydomain.com
Phones are registering to sip1.mydomain.com, Voicemail works, but MWI
does not. Any hint what I should change to make this work?
Best regards
Peter
Anthony Minessale schrieb:
based on your example past
sip1.mydomain.com http://sip1.mydomain.com is the domain in the
packet and thus the profile should have an alias for this.
Then the user must reside in your sip db with the user 200 and domain
sip1.mydomain.com http://sip1.mydomain.com
if you dont have this consider the force-register-domain and
force-register-db-domain to normalize the host names.
On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale
anthony.miness...@gmail.com mailto:anthony.miness...@gmail.com wrote:
Did you check the 2 replies that told you you need aliases in your
sofia profile to translate the domain found in your
message_waiting to the right profile? Both Brian and Mike
answered you.
On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX
prometheus...@gmx.net mailto:prometheus...@gmx.net wrote:
I tried now with phones directly attached to the freeswitch
(without an
OpenSIPS in between). I also added the alias. But the
behaviour is as
before:
No notify message from freeswitch, neither after register nor
after a
voicemail is recorded.
Best regards
Peter
Brian West schrieb:
Yes an alias will be required for every domain you run on
the profile
so it can find it.
/b
On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
Try an alias on the sip profile.
Mike
___
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
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon
Re: [Freeswitch-users] Trapping dtmf on bridged call
It can. I use it like:
session:execute(bind_meta_app, 1 b s execute_extension::dx XML
features);
session:execute(bind_meta_app, 2 b s
record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft
ime(%Y-%m-%d-%H-%M-%S)}.wav);
session:execute(bind_meta_app, 3 b s execute_extension::cf XML
features);
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik
Middleton
Sent: Monday, December 07, 2009 2:59 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call
Can this be done in an lua script?
Regards,
_
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: 07 December 2009 22:18
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call
On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi
Is it possible to trap on DTMF on a bridged call within an LUA script? I've
tried setting the gateway to use inband, but no joy. It looks like I could
use start_dtmf, but I can't see how to launch this within LUA
Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever
you want to have happen. Check it out:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app
The Local_Extension in the default.xml dialplan file has a few examples of
using this tool.
-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] Trapping dtmf on bridged call
did you set the inputcallback too? On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Can this be done in an lua script? Regards, -- *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* 07 December 2009 22:18 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Trapping dtmf on bridged call On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I’ve tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can’t see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -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 -- 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 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
Re: [Freeswitch-users] Trapping dtmf on bridged call
Yes I did, is it possible mod_vmd is interering? It's stopped before I call the start_dtmf function session:setHangupHook(myHangupHook, blah) session:setInputCallback(onInput); session:execute(vmd,start); if (session:ready() == false) then freeswitch.consoleLog(info, : Call Failed!!!\n); end session:answer(); From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 07 December 2009 23:21 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call did you set the inputcallback too? On Mon, Dec 7, 2009 at 4:59 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Can this be done in an lua script? Regards, From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 22:18 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Trapping dtmf on bridged call On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Perhaps you could use bind-meta-app to bind a key combo like *1 to whatever you want to have happen. Check it out: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_meta_app The Local_Extension in the default.xml dialplan file has a few examples of using this tool. -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 -- 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 mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%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] Skype SIP Beta
Skype have opened their beta program up to all comers. http://www.skype.com/business/products/pbx-systems/sip/support-faqs/#paddedContent Three lines in a sip_profile make FreeSWITCH talk nicely; but using the PCMU codec. Any progress on SILK native support? Last I saw was discussion back in September with Brian lamenting that Skype was hard to work with on this. I know I could use mod_skypiax; but having a native solution would be one less IT headache. Thx, Chris. ___ 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] Skype SIP Beta
Brian West br...@freeswitch.org wrote: They have yet to type make on a 64bit box and build us a binary that is 64bit. Chances are they mucked it up like the BroadVoice codecs were and it just won't work on 64bit just yet... if they would just give us the src we could be done in under two days with it I suspect. Given that they released the codec specification, perhaps someone is writing an independent C implementation? (Not that I'm much interested, but...) ___ 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] Skype SIP Beta
We can ONLY hope someone will do this and BSD/MIT the library and NOT GPL it... if they GPL it then we'll have to have someone write it all over again... love the Open Source oil and water. /b On Dec 7, 2009, at 7:39 PM, Jason White wrote: it I suspect. Given that they released the codec specification, perhaps someone is writing an independent C implementation? (Not that I'm much interested, but...) ___ 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] pfSense with Freeswitch - so far so good, but no calls going through
Question -- I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: -- Since you moved the internal profile to the lan ip address you can go ahead and dump the lan profile. Question -- If I do a reloadxml it gives me this output on the console: freeswi...@firewall.fribert.dk reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: -- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question -- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: -- This is correct as long as you have a gateway that is registered called musimi.dk Question -- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: -- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer --- On Mon, 12/7/09, mailinglist mailingl...@fribert.dk wrote: From: mailinglist mailingl...@fribert.dk Subject: [Freeswitch-users] pfSense with Freeswitch - so far so good, but no calls going through To: freeswitch-users@lists.freeswitch.org Date: Monday, December 7, 2009, 1:50 PM Hi All Ok, so next episode in the saga of getting this monster of the ground :-) I've gotten the FS up and running pretty much I guess, but I'm missing something. It has been set up as per the 'multi-homed' document (http://wiki.freeswitch.org/wiki/Multi_home_tutorial). I want to use the webinterface in pfSense, as it is the easiest for me to manage, and gives me a better overview. If I do a reloadxml it gives me this output on the console: freeswi...@firewall.fribert.dk reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) 2009-12-07 21:21:40.719721 [INFO] mod_enum.c:808 ENUM Reloaded 2009-12-07 21:21:40.719721 [INFO] switch_time.c:661 Timezone reloaded 530 definitions API CALL [reloadxml()] output: +OK [Success] I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Looking at the status I see: sofia status profile internal API CALL [sofia(status profile internal)] output: = Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.11.12.25 Ext-RTP-IP 10.11.12.25 SIP-IP 10.11.12.25 Ext-SIP-IP 10.11.12.25 URL sip:mod_so...@10.11.12.25:5060 BIND-URL sip:mod_so...@10.11.12.25:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS g7...@32000h,g7...@16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 10 FAILED-CALLS-IN 5 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: = Call-ID: MThhODdkOWFkMGM4YTk5OWU1MTMzMjg5NmFjOGFhNWU. User: 1...@10.11.12.25 Contact: 1001 sip:1...@10.11.12.145:59650;rinstance=7c2be2513804dcea;fs_nat=yes;fs_path=sip%3A1001%4010.11.12.145%3A59650%3Brinstance%3D7c2be2513804dcea Agent: X-Lite release
[Freeswitch-users] Zombie Records in core db
We have FreeSWITCH Version 1.0.4 (exported) running at a high volume traffic. I normally check the concurrent calls by looking at the number of sessions from status command. However, the number of concurrent calls in FS is normally higher than it's supposed to be after we ran traffic for about a week. Thus, I routed the traffic away from the FS and found out from show calls that there were so many old calls from previous days. We are running a pass-thru traffic in signaling only. I wonder whether there is a way to have those zombied records clean up automatically. Also, what should I do to prevent this problem? Thank you, Dorn B. ___ 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] Zombie Records in core db
I also have this problem on a trunk version more than 1000 revisions behind, so I think the best way is to upgrade to trunk and report this again if still have problem. 2009/12/8 DJB djbin...@yahoo.com: We have FreeSWITCH Version 1.0.4 (exported) running at a high volume traffic. I normally check the concurrent calls by looking at the number of sessions from status command. However, the number of concurrent calls in FS is normally higher than it's supposed to be after we ran traffic for about a week. Thus, I routed the traffic away from the FS and found out from show calls that there were so many old calls from previous days. We are running a pass-thru traffic in signaling only. I wonder whether there is a way to have those zombied records clean up automatically. Also, what should I do to prevent this problem? Thank you, Dorn B. ___ 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] Zombie Records in core db
Version 1.0.5 pre 8 is due out any minute. Definitely upgrade to trunk or at least pre8 when it's available. -MC Sent from my iPhone On Dec 7, 2009, at 6:29 PM, DJB djbin...@yahoo.com wrote: We have FreeSWITCH Version 1.0.4 (exported) running at a high volume traffic. I normally check the concurrent calls by looking at the number of sessions from status command. However, the number of concurrent calls in FS is normally higher than it's supposed to be after we ran traffic for about a week. Thus, I routed the traffic away from the FS and found out from show calls that there were so many old calls from previous days. We are running a pass-thru traffic in signaling only. I wonder whether there is a way to have those zombied records clean up automatically. Also, what should I do to prevent this problem? Thank you, Dorn B. ___ 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] Zombie Records in core db
For starters, try using the latest svn snapshot. Your version is 6 months old and several thousand revs old. On Dec 7, 2009 8:34 PM, DJB djbin...@yahoo.com wrote: We have FreeSWITCH Version 1.0.4 (exported) running at a high volume traffic. I normally check the concurrent calls by looking at the number of sessions from status command. However, the number of concurrent calls in FS is normally higher than it's supposed to be after we ran traffic for about a week. Thus, I routed the traffic away from the FS and found out from show calls that there were so many old calls from previous days. We are running a pass-thru traffic in signaling only. I wonder whether there is a way to have those zombied records clean up automatically. Also, what should I do to prevent this problem? Thank you, Dorn B. ___ 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] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through
Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0 to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001-1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1...@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1...@10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1...@10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth 08-12-2009 kl. 03:05 skrev Mark Crane mc...@yahoo.com i meddelelsen 659603.29094...@web56408.mail.re3.yahoo.com: Question -- If I do a reloadxml it gives me this output on the console: freeswi...@firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswi...@firewall.fribert.dk ) reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: -- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question -- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: -- This is correct as long as you have a gateway that is registered called musimi.dk Question -- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: -- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer ___ 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] Translating DTMF from RFC2833 to INFO
Hello all, *debug voip rtp session named-event*s shows that it receives and understands the DTMFs, but it does not send them to the PSTN (sends only those received via INFO). I haveto find some time and go to the remote site to update to the latest IOS... I will update after this has been done. Regards, __Yehavi: 2009/12/6 Anthony Minessale anthony.miness...@gmail.com Some more bad news for you, info dtmf spec has expired and has been abandoned. Wait till you see what they did accept instead.. On Dec 6, 2009 1:22 PM, Metik freeswitch-users-l...@metik.com wrote: Unless the IOS you are running is extremely buggy, debug voip ccapi commands should not provide you with that detail, what you really want to use is debug voip rtp session named-event. Normal SIP-to-PSTN calls should use both a pots and voip dial peer but DTMF relay type is determined by the voip dial peer. I haven't ran into this issue (i.e. DTMF is ignored when using RFC 2833) previously in the wild. Unlike some other SIP feature servers, I have not had issues (with RFC 2833) between FS and Cisco IOS gateways. Although unrelated to FS or any other SIP feature server, I have seen some issues when multple dtmf relay types are left enabled on a voip dial peer. Also, there are some (older) IOS versions that have issues with DTMF duration which cause digits to be misinterpreted by the far-end (PSTN/POTS) but not ignored altogether. -metik Yehavi Bourvine wrote: Hello Metik, 2009/12/6 Metik freeswitch-users-l...@metik.com mailto:freeswitch-users-l...@metik.com You previously stated that your Cisco gateway has some bug that prevents you from us... _... ___ 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
