September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Easy Information Needed For Placement Of POP To Avoid Latency
by TamurlangClan 13 Sep '12

13 Sep '12

Hi, I am weak in network architecture and need help in placement of POP. Please see below: 1) Is it must to have a POP close to where your subsicriber base. For Example: If my user base is in Australia as well, Will they be affected if their calls are routed through POP in New York as it might cause Latency and packet delay thereby dropping call quality considerably? 2) I have seen some termination providers who have indirect route. Does this mean if i choose them my calls will be routed through multiple switches at multiple locations which will give poor quality voice and also high Latency and Packet Delay? So is it better choose direct routes? Please guide on this. Thanks in advance. -- View this message in context: http://old.nabble.com/Easy-Information-Needed-For-Placement-Of-POP-To-Avoid… Sent from the Freeswitch-users mailing list archive at Nabble.com.

3 2

Bridge after Leg B has connected
by Subhash 13 Sep '12

13 Sep '12

Hi, I have a simple dialplan that just do bridge for incoming calls. I'm trying to do bridge the Call Leg A with Call Leg B after getting confirmation of call connected from Call Leg B <action application="bridge" data="(sofia or freetdm)/internal/destination number"/> ** Is there any way to bridge the both the call legs after getting confirmation from call leg B? Thanks, Subhash.

4 7

Re: [Freeswitch-users] Variable continue on fail
by Mi Ke 13 Sep '12

13 Sep '12

Hi Kirill, We use the following construction: <extension name="ex1"> <condition field="destination_number" expression="your_dest_num_here"> <action application="set" data="bridge_answer_timeout=120"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION,CALL_REJECTED,RECOVERY_ON_TIMER_EXPIRE"/> <action application="set" data="fail_on_single_reject=USER_BUSY,NO_ANSWER,NO_USER_RESPONSE,ORIGINATOR_CANCEL"/> <---this prevents further failover in appropriate cases <action application="set" data="bridge_early_media=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="lua" data="get_dial_string.lua"/> <action application="bridge" data="${out_dial_string}"/> <anti-action application="pre_answer"/> <anti-action application="playback" data="/usr/local/freeswitch/recordings/number_incorrect.mp3"/> <anti-action application="hangup" data="INVALID_NUMBER_FORMAT"/> </condition> </extension> where get_dial_string app returns corresponding dialpeers as [dp1_params]sofia/external/dp1|[dp2_params]sofia/external/dp2|[dp3_params]sofia/external/dp3 It works without any issues. FreeSWITCH Version 1.2.0-rc2+git~20120815T215213Z~c6d7417aec, CentOS 6.2 x64 HTH, Mike ----- Original Message ----- From: Kirill Sysoev Sent: 09/13/12 02:13 PM Subject: Re: [Freeswitch-users] Variable continue on fail Hi David, Vitaly! Thank you for your answers. Will try to implement your advices. Best, Kirill 13.09.2012 14:52, Виталий Давудов пишет: > Hi, Kirill! > > I solved it by adding lua-script after "bridge" application in my > dialplan: > > endpoint_disposition = session:getVariable("endpoint_disposition") > last_bridge_hangup_cause = > session:getVariable("last_bridge_hangup_cause") > if (last_bridge_hangup_cause == "USER_BUSY" and endpoint_disposition > == "EARLY MEDIA" ) then > session:hangup(); > end > > You can see these variables by adding "info" application in the > corresponding extension. > > 12.09.2012 17:53, iam пишет: >> Hi Vitaly! >> >> Did you ever solve this problem? >> I have the same continue_on_fail behaviour. >> >> http://jira.freeswitch.org/browse/FS-4232 >> >> Vitaly added a comment - 26/Jun/12 3:14 AM >> Sorry for the long absence, but the problem is still actual for me. >> I replaced in my dialplan continue_on_fail to "true", but it's not give >> desired effect. >> May be some additional variables can help me? >> >> Regards, >> Kirill >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Variable-continue-on-fail-tp7… >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> >> 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

3 2

Variable continue on fail
by Виталий Давудов 13 Sep '12

13 Sep '12

Hi all! In my dialplan I've included variable continue on fail: <extension name="Dial Out SIP-M200 Mobile numbers"> <condition field="destination_number" expression="^(89\d{9})$"> <action application="export" data="t38_passthru=true"/> <action application="set" data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> <action application="info"/> <action application="bridge" data="sofia/gateway/SIP_M200/$1"/> <action application="limit_execute" data="hash limit SMG-1016M 60 bridge sofia/gateway/SMG-1016M/$1"/> </condition> </extension> And if FS recieve from first gateway USER_BUSY, then FS try to bridge this call to another gateway. Although in line <action application="set" data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> there is no code Q.850 = 17. How resolve this issue? -- Best regards, Vitaly Davudov "VIP-TELECOM-SERVICE" Ltd. ("ETERIA" Group of companies) http://www.vts24.ru

6 15

Static sound on outbound calls
by Phil Kim 12 Sep '12

12 Sep '12

Hello All, I have been using FreeSWITCH over the last two years on both inbound and outbound dialing. I have a problem with Outbound calling. This does not occur all the time, but often enough for us to be concerned. We have an IVR system that make an outbound call to our client and process the call flow using DTMF. Often times when user answers the call, there is nothing but static sound. It does not respond to any DTMF, and does not play any sound file either. It just hangs there with little static sound in background. I have consulted my SIP provider but they have no idea either. Any one has any idea what might be causing the issue? Thank you in advance. Best regards, Phil H. Kim Software Engineer The information contained in this e-mail message, and any attachment thereto, is the property of Valiant and is confidential and may not be disclosed without our express permission. If you are not the intended recipient or an employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that you have received this message in error and that any review, dissemination, distribution or copying of this message, or any attachment thereto, in whole or in part, is strictly prohibited. If you have received this message in error, please immediately notify us by telephone, fax or e-mail and delete the message and all of its attachments. Thank you. Every effort is made to keep our network free from viruses. You should, however, review this e-mail message, as well as any attachment thereto, for viruses. We take no responsibility and have no liability for any computer virus which may be transferred via this e-mail message.

2 1

Stuck Sessions
by Spencer Thomason 12 Sep '12

12 Sep '12

Hello, Recently upgraded a few of our boxes to git head and I'm seeing the following behavior: freeswitch@internal> status UP 0 years, 0 days, 15 hours, 20 minutes, 7 seconds, 552 milliseconds, 539 microseconds FreeSWITCH is ready 218 session(s) since startup 13 session(s) 0/200 5000 session(s) max min idle cpu 0.00/99.00 Current Stack Size/Max 240K/240K freeswitch@internal> show calls 0 total. freeswitch@internal> show channels 0 total. Note the discrepancy in the active sessions. I'm happy to file a jira but I'm not sure of the necessary debugging info needed. This particular machine in using a dialplan generated by FusionPBX and is close to stock. The only scripting is a huntgroup which checks if session:ready(). Thanks, Spencer

5 8

More information on FreeSwitch
by Stephan Swart 12 Sep '12

12 Sep '12

Hi Obviously i am new around here I am looking for more documentation on FreeSwitch I am aware of the Wiki I do have the cookbook What else is around I need to design a large voice deployment with Class 4 and Class 5 switches. I need to know more detail and what is out there that can manage these devices in an easier manner than editing files. Administrators can be dangerous as well sometimes. Need some proper stability etc. Tanx Stephan

5 5

FreeSWITCH Conference Call Today
by Michael Collins 12 Sep '12

12 Sep '12

Hello All, Our FreeSWITCH conference call will be taking place in a bit. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_12 We look forward to hearing from Ben and Ben about the Adhearsion project. We also have a few community updates. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

1 0

Polycom Extension Issue
by Carlos Flor 12 Sep '12

12 Sep '12

So here is the situation: I have a polycom phone with extension 101@pbxAand 101@pbxB. Someone else has a polycom phone with extension 102@pbxAand 102@pbxB. If I try to call from 101@Bto 102@B, the phone rings, but as soon as 102 answers, the call hangs up immediately. If I change 102@B to 103@B, so that the extensions are now 102@A and 103@Bon the second phone, then phone calls from the second phone to the first work, but not the other way around. If I repeat the change on the first phone, so that it now has 101@A and 104@B, then calls work correctly in both directions. So, it seems as though if you have more than one registration but use the same extension on each, the polycom has issues with the RTP piece of the call. The SIP piece seems to work ok because the call actually makes it to the other phone and rings, but as soon as you pickup (when RTP should start) the call ends. Has anyone run into anything close to this before? I am sure my description is confusing and it's much easier to explain on a whiteboard, but hopefully it makes sense. Just to clarify, the two PBXs are not related to each other. I'm not trying to call from pbxA to pbxB. Carlos

2 2

IP to PBX calls recovery
by Rajkumar Kanniappan 12 Sep '12

12 Sep '12

Hi, I'm trying to make connection between IP network and PBX network by using the PRI interface. Also I had succeeded in making the call between IP and PBX. Now I need to provide the high availability for end users in case of IP failure. I had seen the HA enterprise deployment in case of IP failover which talks about Sofia recovery. Using that I'm able to recover IP to IP calls. My question is how to recover the calls between IP and PBX. Please give your suggestion on this. Thanks ________________________________ SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

2 1