November 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch Sonus bypass media
by Ahmed Sboor 18 Nov '12

18 Nov '12

Dear All, i've read in past or may be still Freeswitch and sonus are not good with each other . but every post shows its always rtp between FS and Sonus. In my case i am bypassing all media , so just signalling between US (using FS) and Provider (Sonus) . And ACD is extreme low as 4-5 minutes due to 100s of short calls ending with normal call clearing. Can any one Please help if in bypass still FS and Sonus can have some problem. Thanks in advance regards soFh

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FreeSWITCH High availability(Normal_temporary_failure problem)
by Rajkumar K 18 Nov '12

18 Nov '12

Hi, I am trying to achieve high availability in FreeSWITCH using heartbeat and pacemaker and I am able to switch between the server whenever one of the servers crashes. But the problem is one server is able to recover the calls when im invoking sofia recover, but another server is recovering the call one few times.(Mostly not able to recover). I have primary and secondary server installed in two different machines(centos), the FreeSWITCH instances are always running in both the PCs. I am running heartbeat and pacemaker to monitor the IP or sofia fail-over. A floating IP is configure in heartbeat to reach the active server. I succeeded in switching between the servers whenever IP or FreeSWITCH and once it reaches the another server it invokes sofia recover to recover the calls. Both freeswtich instances are using the same configuration and database is shared using ODBC connectivity. Problem is: Call is made using the primary server, and i did fsctl crash in primary server cli. Heartbeat resource switches to secondary server and it invokes "sofia profile internal restart" and "sofia recover". and it recovers the call. The call gets recovered in 4-5 seconds. At the same time i will start the freeswitch instance in the primary server. Now if i crash the secondary server using fsctl crash, the resources switches to primary server and it invokes "sofia profile internal restart" and "sofia recover". Also the server sends invite request to the clients But it ends in NORMAL_TEMPORARY_FAILURE. Wireshark log says Client is responding with "Invalid CSeq" for the Server's INVITE request. This happens always with the primary server and very few times primary server is also able to recover the calls. I have the same configurations in both the servers And also i checked by stopping the heartbeat switching, and crashed the primary server's freeswitch. Then if i start freeswitch again in the same server and invoking sofia recover will recover the calls without any problem. I have also attached the cli logs of primary and secondary servers. I am not able to identify the exact problem in this, Please help me out in this problem. Thanks Rajkumar

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uuid_media leaves behind zombie call
by Chris Martineau 18 Nov '12

18 Nov '12

Hi, I am trying to use uuid_media to switch in and out of the rtp stream and in essence it seems to work however I get some strange things happening. The first thing is that when the re-invites go out, the a and the b legs get the codec list in a different order i.e. 0 8 3 to the a leg and 8 0 3 to the b leg! If all the codecs are supported this forces a transcoding scenario as the a leg neg's pcmu and the b leg pcma! As this is done via the event socket and not the dialplan I cannot see what I am supposed to set to correct this behaviour ( the calls all start in media bypass mode so no codec neg is done at call start). Any ideas why this should do this and how I can get round it? Setting the no transcoding variable has no effect on this. The second thing is that if you go into the media path and then going out of it again and then clear down it seems to leave the a leg channel hungup in the system. Show channels lists the channel just sitting in CS_HANGUP state. In the log it says Locked, waiting on external entities? Everything externally has cleared and wireshark shows a clean dialog with no resends of the BYE messages. Cleardown is from the a leg and FS has passed the BYE on to the b leg. If you try to kill it with uuid_kill it says that the uuid does not exist? The only way I can clear it is to restart FS which on closing it hangs around with a sofia error saying waiting for 1 session. If I just go into the media path and cleardown without coming out again the bridge process seems to be broken as any BYE message does not get passed through, requiring both ends to be cleared separately. Also no UNBRIDGE event is fired in this scenario which I suppose you would expect if it wasn't bridged, however rtp is flowing through as is signalling? The same channel however is still left hanging. I may be using this incorrectly or not preparing the environment however the info around this command is pretty basic and looks simple to use. It is pretty critical for the application I am trying to build that this process works smoothly so any help you could offer would be greatly appreciated. I have attached some debug logs. Many thanks Chris

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How the IAX2 is supported in freeswitch?
by ZhouJianhua 18 Nov '12

18 Nov '12

Hi, I need to reduce the bandwidth of media stream, and RTP is too heavy for us.IAX can carry more data in one packet, but the protocol is not supported as well as SIP in FS. I just want to know that is there anyone used IAX in freeswitch ?And is it worth a try? Thanks.

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Re: [Freeswitch-users] call disconnects after 32 seconds issue - FreeSWITCH-users Digest, Vol 77, Issue 117
by kaleem rehman 17 Nov '12

17 Nov '12

Hi All, Its kaleem, i reported the issue with call disconnecting after 32 seconds, i have managed to fix the issue. please find the detailed fix below; i had to get the external Interface (internet facing) to pass 5060 TCP/UDP & 5080 TCP/UDP to freeswitch server. if you are experiencing similar issues then please follow following steps. run fswitch console using command "fs_cli" once in run command "sofia status" this should show all gateways and profiles. check if external profile has your external IP to it. (to confirm go to www.whatsmyip.org) this will confirm freeswitch is using NAT. now turn loggin on using command "sofia loglevel all 9" and "sofia profile external siptrace on" where external is name of profile which has your external IP addr. please note this will give you a lot of information and you might have digestion issues. i used putty and changed putty settings to show 3000 lines (right click on putty title bar, CHANGE SETTINGS and click on WINDOWS and change SCROLLBACK lines from default 200 to 3000) also you can enable log in putty so it saves all data in putty to a file for you. now make the call and when it disconencts go to log windows and type "/quit" to quit freeswitch terminal and no more logs are shown. now either copy and paste the data or open the log file and search for "50" and try to see what ports are being used. now log on to your router and change all port fowarding (please refer to documentation for your router) forward port you can see in log file(in my case 5060 & 5080) to freeswitch server. i restarted my server just to be on safe side and everything went fine after that. also i had a bad experience with BT SIP, British telecoms business sip, they dont respond with SRV & DNS queries so FreeSwitch doesnt register with them and gateway status goes to fail_wait. i had to play around and manage to fix it, i will do a proper guide so no one else has to suffer like me where it took me 4 days and nights to sort that out. i dont want to post messy documentation so will clean up a bit and hopefully FreeSwitch admins will import to WIKI hope this helps. regards, K On Sat, Nov 17, 2012 at 3:21 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. changing vm default announcement (andy) > 2. retreiving voicemail dropping after 30 seconds (Jason Holden) > 3. Re: call disconnects after 32 seconds (Anthony Minessale) > > > ---------- Forwarded message ---------- > From: andy <packetandy(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Fri, 16 Nov 2012 20:20:50 -0800 > Subject: [Freeswitch-users] changing vm default announcement > hi Nick, thanks for the reply. > > Greetings are in > $${base_dir}/storage/voicemail/default/$${domain}/<user>/greeting_1.wav as > specified in the wiki. > > trying to force freeswitch to use this custom greeting by using *<action application="set" data="voicemail_greeting_number=1"/> *does not work*. *Default greeting from Callie still plays*. > *If one records a greeting and then overwrites the recorded greeting with another file in the same directory - that works. > > Guys, is this a bug, or do I not understand what the 'voicemail_greeting_number variable is supposed to do? > > drew > > BTW - I am on FS 1.2.1 > > > > ---------- Forwarded message ---------- > From: "Jason Holden" <jason.holden(a)start.ca> > To: <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sat, 17 Nov 2012 01:33:04 -0500 > Subject: [Freeswitch-users] retreiving voicemail dropping after 30 seconds > > Hi.**** > > When accessing voicemail to listen to messages I am finding that it is > dropping at 30 seconds each time with a message of 100 sleep timer.**** > > Does anyone have any recommendations?**** > > I am using a Sipura 3000 connected to my freeswitch server.**** > > ** ** > > > ---------- Forwarded message ---------- > From: Anthony Minessale <anthony.minessale(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sat, 17 Nov 2012 09:21:22 -0600 > Subject: Re: [Freeswitch-users] call disconnects after 32 seconds > The only reason that there would be a failure to get the ACK would be if > the 2 endpoints have lost communication. > If you can't get an ACK to the host, you can't get a BYE to them either. > Even if you hacked it to tolerate no ACK, the session timers would fail in > another 30 seconds. > > You are just focused on your business needs and just disregarding logic. > You ask if you are missing something and several individuals are telling > you YES. This is the 4th time I am telling you that you should focus > your enthusiasm on learning rather than affirming a m00t point because we > are not going to change the behavior. > > I have not seen a seen a single case of missing ACK that did not point out > a real problem that can be simply fixed by either using some NAT related > configuration and proper formed Contact hosts. > > If this argument continues it will be ended with moderation..... > > > On Fri, Nov 16, 2012 at 6:07 PM, Yiftach Golan <yiftah(a)choochee.com>wrote: > >> Here is the reason from the RFC3261 : >> "...The reason for this separation is rooted in the importance of >> delivering all 200 (OK) responses to an INVITE to the UAC. To >> deliver them all to the UAC, the UAS alone takes responsibility >> for retransmitting them (see Section 13.3.1.4), and the UAC alone >> takes responsibility for acknowledging them with ACK (see Section >> 13.2.2.4). Since this ACK is retransmitted only by the UAC, it is >> effectively considered its own transaction..." >> >> >> On Fri, Nov 16, 2012 at 3:50 PM, Yiftach Golan <yiftah(a)choochee.com>wrote: >> >>> Yes I you are right if the RTP is not tied with SIP this theory is not >>> really valid >>> But this is again true about BYE that does not reach to the destination >>> so the risk of not closing the dialog still exist >>> In most of the voice implementation that I saw there are three options : >>> 1. Softswitch (FreeSWITCH, Asterisk, etc) >>> 2. MGW (Avaya, Cisco, etc) >>> 3. Media release but only between two phones >>> You can tie your RTP to the SIP with option 1 and option 2 >>> Option 3 is the problematic one but you will never release your media >>> for billing purpose so usually your risk will be extension to extension >>> open dialog which is less riskier >>> Again as I said it is pretty philosophical debate but from my long >>> experience in SIP I do not think that there is a practical use for it but >>> maybe I am missing something >>> >>> On Fri, Nov 16, 2012 at 2:31 PM, Sergey Okhapkin < >>> sos(a)sokhapkin.dyndns.org> wrote: >>> >>>> Don't mix signaling (SIP) and media (RTP). Signaling and media could >>>> run >>>> different ways. Why do you think FS will always be in media path? >>>> >>>> On Friday 16 November 2012 14:56:58 Yiftach Golan wrote: >>>> > This is where I am getting a bit confused, if the 200OK arrived to the >>>> > other side and we checked that the RTP exists (with mod_sofia option) >>>> we >>>> > will not get to hours of calls (unless the other side did not hanged >>>> up) >>>> > In any case there is a good chance that the BYE is getting lost, so >>>> the >>>> > danger exist even without the ACK >>>> > >>>> > I am guessing that the designers of the SIP protocol came up with the >>>> ACK >>>> > because there is a potential for open dialog that is not bound with >>>> time >>>> > and they therefore wanted to know that the other side actually ACKs >>>> the >>>> > request, but since ACK is tied to INVITE only (AFAIK) and INVITE >>>> always >>>> > tied with RTP (at least in most normal SIP implementations) I'm not >>>> sure >>>> > that this ACK is that needed >>>> > but again as I said it is more of philosophical debate, maybe a >>>> potential >>>> > request in the new RFC >>>> > >>>> > On Fri, Nov 16, 2012 at 12:58 PM, Ken Rice <krice(a)freeswitch.org> >>>> wrote: >>>> > > In this case masking the issue can lead to massive bills... Imaging >>>> > > >>>> > > paying by the minute... All of a sudden you are now leaving 2 >>>> minute calls >>>> > > up for hours on end... And continuing to get billed for them... Or >>>> you are >>>> > > not continuing to bill a customer for them... And now you have >>>> unexpected >>>> > > HUGE bills coming in... Masking it is far worse then just fixing >>>> it... >>>> > > >>>> > > If we mask this one issue, we might as well mask memory leaks, or >>>> > > passwords that don’t work, etc... Sure sometimes we might have to >>>> mask an >>>> > > issue for production to work in the short term, but that is never >>>> the >>>> > > correct answer fix a problem >>>> > > >>>> > > >>>> > > On 11/16/12 2:19 PM, "Yiftach Golan" <yiftah(a)choochee.com> wrote: >>>> > > >>>> > > While I agree on the details I disagree on the solution >>>> > > Sometimes masking the problems can be a good solution but I guess >>>> it is a >>>> > > philosophical debate >>>> > > >>>> > > >>>> > > On Fri, Nov 16, 2012 at 10:27 AM, Ken Rice <krice(a)freeswitch.org> >>>> wrote: >>>> > > >>>> > > That leaves to big a risk of open sessions and only masks the true >>>> issue >>>> > > which is a problem with FS getting the ACK back... >>>> > > >>>> > > Theres a reason FS is not getting the ACK, and FS will make several >>>> > > attempts to get an ack by retransmitting the 200 OK several times >>>> before >>>> > > that timeout occurs. >>>> > > >>>> > > The real fix here is to fix the underlying cause, not masking it.... >>>> > > >>>> > > >>>> > > On 11/16/12 11:45 AM, "Yiftach Golan" <yiftah(a)choochee.com < >>>> > > http://yiftah@choochee.com> > wrote: >>>> > > >>>> > > I know that it is kind out of the what RFC3261 instructs, but did >>>> anyone >>>> > > think on giving the option in configuration not to hang up calls in >>>> case >>>> > > of >>>> > > an ACK does not arrive? >>>> > > I know that it has the risk of open sessions but there some other >>>> ways to >>>> > > handle those cases >>>> > > >>>> > > On Thu, Nov 15, 2012 at 6:35 PM, Ken Rice <krice(a)freeswitch.org < >>>> > > http://krice@freeswitch.org> > wrote: >>>> > > >>>> > > This is probably the same scenario as this is exactly what to >>>> expect... >>>> > > Call gets answered far end doesn’t ACK FS sending them a 200OK , fs >>>> > > hangsup >>>> > > the call.... >>>> > > >>>> > > Quite common on networks with NAT issues or broken endpoints >>>> > > >>>> > > >>>> > > On 11/15/12 7:43 PM, "Vitalie Colosov" <vetali100(a)gmail.com < >>>> > > http://vetali100@gmail.com> <http://vetali100@gmail.com> > wrote: >>>> > > >>>> > > I saw this happened earlier when the remote party does not send SIP >>>> ACK >>>> > > after receiving SIP OK, so the call is being disconnected after >>>> exactly 32 >>>> > > seconds. >>>> > > Not sure if this is exact same scenario here, but just something to >>>> > > consider... >>>> > > >>>> > > Regards. >>>> > > Vitalie >>>> > > >>>> > > >>>> > > 2012/11/15 kaleem rehman <k4kaleem(a)gmail.com < >>>> http://k4kaleem@gmail.com> >>>> > > >>>> > > <http://k4kaleem@gmail.com> > >>>> > > >>>> > > Hi All, >>>> > > >>>> > > my inbound calls are fine with no issues, my outbound calls get >>>> > > disconnected after 32 seconds and its on all calls. i tried 2 >>>> different >>>> > > suppliers and its same result. >>>> > > please find the attached log file with sofia in debug mode. - >>>> caller was >>>> > > extension 1234 and desination was 01908321682 >>>> > > >>>> > > your help will be greately appreciated. >>>> > > >>>> > > regards, >>>> > > Kaleem >>>> > > >>>> > > >>>> _________________________________________________________________________ >>>> > > Professional FreeSWITCH Consulting Services: >>>> > > consulting(a)freeswitch.org <http://consulting@freeswitch.org> < >>>> > > http://consulting@freeswitch.org> >>>> > > http://www.freeswitchsolutions.com >>>> > > >>>> > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> > > http://www.cudatel.com >>>> > > >>>> > > Official FreeSWITCH Sites >>>> > > http://www.freeswitch.org >>>> > > http://wiki.freeswitch.org >>>> > > http://www.cluecon.com >>>> > > >>>> > > FreeSWITCH-users mailing list >>>> > > FreeSWITCH-users(a)lists.freeswitch.org < >>>> > > http://FreeSWITCH-users@lists.freeswitch.org> < >>>> > > http://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 >>>> > > >>>> > > >>>> > > >>>> > > ------------------------------ >>>> > > >>>> _________________________________________________________________________ >>>> > > Professional FreeSWITCH Consulting Services: >>>> > > consulting(a)freeswitch.org <http://consulting@freeswitch.org> < >>>> > > http://consulting@freeswitch.org> >>>> > > http://www.freeswitchsolutions.com >>>> > > >>>> > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> > > http://www.cudatel.com >>>> > > >>>> > > Official FreeSWITCH Sites >>>> > > http://www.freeswitch.org >>>> > > http://wiki.freeswitch.org >>>> > > http://www.cluecon.com >>>> > > >>>> > > FreeSWITCH-users mailing list >>>> > > FreeSWITCH-users(a)lists.freeswitch.org < >>>> > > http://FreeSWITCH-users@lists.freeswitch.org> < >>>> > > http://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 >>>> > > >>>> > > >>>> > > -- >>>> > > Ken >>>> > > *http://www.FreeSWITCH.org >>>> > > http://www.ClueCon.com >>>> > > http://www.OSTAG.org >>>> > > *irc.freenode.net #freeswitch >>>> > > >>>> > > >>>> _________________________________________________________________________ >>>> > > Professional FreeSWITCH Consulting Services: >>>> > > consulting(a)freeswitch.org >>>> > > http://www.freeswitchsolutions.com >>>> > > >>>> > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> > > http://www.cudatel.com >>>> > > >>>> > > Official FreeSWITCH Sites >>>> > > http://www.freeswitch.org >>>> > > http://wiki.freeswitch.org >>>> > > http://www.cluecon.com >>>> > > >>>> > > FreeSWITCH-users mailing list >>>> > > FreeSWITCH-users(a)lists.freeswitch.org >>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > > http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)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_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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call disconnects after 32 seconds
by kaleem rehman 17 Nov '12

17 Nov '12

Hi All, my inbound calls are fine with no issues, my outbound calls get disconnected after 32 seconds and its on all calls. i tried 2 different suppliers and its same result. please find the attached log file with sofia in debug mode. - caller was extension 1234 and desination was 01908321682 your help will be greately appreciated. regards, Kaleem

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G729 File version of $${us-ring}
by Tim Meade 17 Nov '12

17 Nov '12

I was wondering if anyone had a G729 file of the $${us-ring} We want to use it for playing the ringback with G729 pass through. <action application="set" data="ringback=SomeRingBackFile.G729"/> Thanks Tim

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[drm:drm_kms_helper_poll_enable] *ERROR* delayed enqueue failed -125
by Scott 17 Nov '12

17 Nov '12

Periodically, the following pops up on the running console. No discernible interval but at least once after a restart or start, only after a period of time and comparative inactivity (almost like it's a housekeeping thread or the like, but dunno). It pops as shown (no leading timestamp or [loglevel] that is typical for most console messages. [drm:drm_kms_helper_poll_enable] *ERROR* delayed enqueue failed -125 Not aware of any problem. Too new to FS to know what part of the code this is and if I RELY on it, or if it's a mod that's loaded that I don't really use but is in the default mix. FS 1.2.3+ 20121029... O/S centOS 6.3-X86-64 Question: 1. Should I care? 2. Should you, devel's, care? Let me know if/what more info wanted/needed. Thanks/Cheers,

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Friday Free-for-all!
by Michael Collins 16 Nov '12

16 Nov '12

Come join us! We're having a nice chat in the main conf room -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Re: [Freeswitch-users] Response time of ESL interface has increased a great deal from version 1.2 to 1.3.2
by Peter Olsson 16 Nov '12

16 Nov '12

Did you read my response on the jira.freeswitch.org issue? /Peter James Bravo <james.bravo(a)redmatter.com> skrev: Hi Just wondering if anyone else has experienced the following issue with later Freeswitch. Using a simple php script (see below) to issue 20 ESL requests and display the response times shows that the response time has increased from an average of 0.002 secs per request for version 1.2 to 0.06 for version 1.3.2 . I've tried issuing requests to localhost, 127.0.0.1 and from other hosts. The results are always much slowed for version 1.3.2 . The test php script is as follows: <?php require_once 'ESL.php'; $host = "localhost"; if ($argc >= 2) $host = $argv[1]; $sockFSServer = new ESLconnection($host, '8021', 'ClueCon'); function getTime() { list($usec, $sec) = explode(" ", microtime()); return ((float)$usec + (float)$sec); } for($n=0; $n < 20; $n++) { $startTime = getTime(); $res = $sockFSServer->sendRecv("api global_getvar local_ip_v4"); if (!$res) echo "request failed\n"; else echo "Got reply '" . $res->getBody() . "'. Time taken " . (getTime() - $startTime) . "\n"; } cheers James Bravo !DSPAM:50a6775532767648055020!

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