August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Do we have ESL-proxy?
by Nathan Neulinger 06 Aug '13

06 Aug '13

With HAProxy, you could just define a list of target servers. You'd configure your app to talk to the HA IP, and then each request would go to next available server in the pool you have defined. You could also simply do a DNS round robin if you wanted. All depends on what infrastructure/etc. you have available to you and how resilient it needs to be. -- Nathan On 08/06/2013 12:38 PM, SamyGo wrote: > Dear Nathan, > > I think I've missed something. I don't understand how HAproxy and Floating IP and KeepAlived will help me? I need to > have all Active FS Servers and some mechanism to send ESL commands to them in round-robin fashion. I've an autodialer > script so that Proxy will help me use many FS Servers at the same time ! > > Thanks, > Sammy > > > > On Tue, Aug 6, 2013 at 10:22 AM, Nathan Neulinger <nneul(a)mst.edu <mailto:nneul@mst.edu>> wrote: > > You might look at the keepalived page on wiki, that's what I use to float an IP in the case where only one of the FS > instances should be online. > > You could also use HAProxy to define a group of systems for it to talk to if you want multiple to be online at once, > but rotating between them. > > -- Nathan > > > On 08/06/2013 12:19 PM, SamyGo wrote: > > Hi Nathan, > > I can only think of Round Robin with failover. I definitely don't want to parallel fork an ESL command. > > -- > Sammy > > > > On Tue, Aug 6, 2013 at 10:16 AM, Nathan Neulinger <nneul(a)mst.edu <mailto:nneul@mst.edu> <mailto:nneul@mst.edu > <mailto:nneul@mst.edu>>> wrote: > > Are you saying "round robin" or "mirrored" for the commands? > > i.e. if you send a command over this proxy, would it be duplicated to all of the FS servers, or sent to "first > available"? > > -- Nathan > > > On 08/06/2013 12:05 PM, SamyGo wrote: > > Dear Users, > > I have come up with a requirement to have ESL connection made with multiple FS Servers at the same time > and send > them > commands(no event receiving is required for now) The connections to FS should be pooled and monitored > if any FS > goes down. > > Is there any ESLproxy available to be used readily? > > Thanks, > Sammy > > > -- > ------------------------------____----------------------------__-- > Nathan Neulinger nneul(a)mst.edu <mailto:nneul@mst.edu> <mailto:nneul@mst.edu <mailto:nneul@mst.edu>> > > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > > > -- > ------------------------------__------------------------------ > Nathan Neulinger nneul(a)mst.edu <mailto:nneul@mst.edu> > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > -- ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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A few problems with latest GIT
by Yehavi Bourvine 06 Aug '13

06 Aug '13

Hello, I am running FS version 1.0.x and trying to move to 1.2/1.4/latest-git. I encounter a few problems which prevents me from using these versions, and would like to know whether someone else is having it. It is quite hard to create debugging information, as with full logging the behaviour is slightly different than lower logging level. The problems appear only on my production system, around noon (when the number of active sessins goes beyond 50-60 sessions): - The voice path is lost in the middle of the session (Jira FS-5616). From TCPdump I see that FS receive the audio from both sides but stops sending it; there is nothing meaningfull in the logs. - FS starts to get sluggish, fs_cli is stuck, SIP messages are processed in delay (5-30 seconds). This goes away by itself after a few minutes. - sessions are left "in use" after the users hangup, and the number of active sessions starts rising. Doing uuid_kill succeeds once (next times fail with "no such uuid", but "show channels" still shows it. The most frustrating side is that I do not know what exact debugging tools to use in order to submit meaningfull data to FS developers. Thanks! __Yehavi:

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Do we have ESL-proxy?
by SamyGo 06 Aug '13

06 Aug '13

Dear Users, I have come up with a requirement to have ESL connection made with multiple FS Servers at the same time and send them commands(no event receiving is required for now) The connections to FS should be pooled and monitored if any FS goes down. Is there any ESLproxy available to be used readily? Thanks, Sammy

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Bridge outbound calls stay active even after hangup
by Georgi Stefanov 06 Aug '13

06 Aug '13

Hello All, Today I found something, which is strange or maybe it is some kind of an issue What I am talking about is visible here freeswitch@internal> uuid_exists a1524f1e-fece-4d4e-b0c6-1a81b02f7978 false freeswitch@internal> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch a1524f1e-fece-4d4e-b0c6-1a81b02f7978,outbound,2013-08-06 09:21:39,1375770099,sofia/internal/sip:LinkSysPhone@10.102.8.101:5060,CS_EXCHANGE_MEDIA,Outbound Call,LinkSysPhone,10.102.8.101,6001,LinkSysPhone@10.102.8.58,,ACTIVE,,,SEND,a1524f1e-fece-4d4e-b0c6-1a81b02f7978,mediagw2-stm1.dev.opencode.com,Outbound Call,GeorgiStefanov,,,,,,,,,,,,,,,,,,, 1 total. freeswitch@internal> uuid_ki [ uuid_kill] freeswitch@internal> uuid_kill a1524f1e-fece-4d4e-b0c6-1a81b02f7978 -ERR No such channel! freeswitch@internal> 1. I have made 2 outbound calls (ex: originate user/linksysphone 6001) 2. I have bridged them with uuid_bridge 3. Either phone hangup 4. Then I execute "show calls" "uuid_exist" and etc. Am I missing something ? Or it is an issue ?

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Mod_distributor question
by SamyGo 06 Aug '13

06 Aug '13

Dear users, I'm trying to figure out how this mod_distributor works. I've a few questions to help me get this going. I've defined two of my gateways in the distributor.conf.xml like this: <configuration name="distributor.conf" description="Distributor Configuration"> <lists> <list name="GWLB" total-weight="2"> <node name="OpenSIPS-A" weight="1"/> <node name="OpenSIPS-B" weight="1"/> </list> </lists> </configuration> and so when I reload the distributor from API I get the following warning: freeswitch@internal> distributor_ctl reload +ok reloaded. 2013-08-06 14:17:16.652087 [WARNING] mod_distributor.c:201 Total weight does not add up to total weight 10 So I've changed the total-weight to 10, and set weight of each node as 5. That gives the same warning as well. A Warning is fine, but as soon as I just use API to expand the list I get a proper gateway name for the first two times only and then -err starts showing up until I reload the distributor. freeswitch@internal> expand distributor GWLB OpenSIPS-A freeswitch@internal> expand distributor GWLB OpenSIPS-B freeswitch@internal> expand distributor GWLB -err freeswitch@internal> expand distributor GWLB -err freeswitch@internal> The last question is regarding use of this distributor from LUA script. I call in the distributor like this from my LUA. session = freeswitch.Session("{ignore_early_media=true,origination_caller_id_number="..outbound_cli.."}sofia/gateway/${distributor(GWLB)}/".. phone_number); and that gives a straight error: [ERR] mod_sofia.c:4668 Invalid Gateway '%{distributor(GWLB)}' I hope somebody can share their knowledge on this module and help me move forward with my project. Best Regards, Sammy

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IVR - Dial by extension
by Brian Foster 06 Aug '13

06 Aug '13

In the dialplan, I use something like this as a condition for the Local Extension "extension": <condition field="${user_exists id ${destination_number} $${domain}}" expression="^true$"> I was wondering if there was something you could do to have a similar behavior when creating IVR menus. Has anyone tried this? Our extensions don't fit a particular format, other than 4 digits, so checking beforehand whether the user id exists would be ideal before transferring. Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana

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How to attche custom "content-type" in Freeswitch SIP message
by xiaofengcanyuexp＠163.com 05 Aug '13

05 Aug '13

Dear support, I''m trying to encapsulate my private "application/isup" in the SIP Msg. Normally, it should like below example. It firstly addresses "Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR". And then can write private "content-type" like "application/sdp". Now I can see the application/sdp is encapsulated via variable "switch_r_sdp". Is there anyway to encapsulated other customized "content-type"? Appreciate to get your reply. --------------------------------------------------------------------------------------------------------------- Here is an example of SIP message which encapsulated "applicaiton/sdp" and "application/isup". INVITE sip: 87896677(a)dance.com;user=phone;SIP/2.0 From: "Caller"<sip:678923001@skynetwork.com> To: <sip:87896677@dance.com>;user=phone Call-ID: QRLVLNKeKx-WDHAuwlEkR-EwhPPcTHOP(a)skynetwork.com Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR MIME-Version: 1.0 Content-Length: 433 --QRLVLNKeKxWDHAuwlEkR Content-Type: application/sdp User-Agent: ENSR2.5.46.6-IS2-RMRG36-RG20-CPO487 Content-Length: 142 v=0 o=- 1706944438 1706944438 IN IP4 192.168.1.105 s=ENSResip t=0 0 m=audio 6793 RTP/AVP 0 a=rtpmap:0 PCMU/8000 --QRLVLNKeKxWDHAuwlEkR Content-Type: application/isup; version=ansi;base=ansi00 Content-Disposition: signal; handling=optional 01 00 60 01 0a 03 05 0b 02 c0 90 06 03 10 78 98 66 77 0a 07 83 13 76 98 32 00 0f 00 --QRLVLNKeKxWDHAuwlEkR-- Thanks ------------------- 2013-05-25

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Softphone
by Ashish Mishra 05 Aug '13

05 Aug '13

Sir i have installed xlite soft phone on my windows 8 pc and i am trying to connect to the freeswitch server running on my ubuntu machine but when i am trying to enable the account it displays an error that "account not enabled contact ur administrator" also i receive the same errror when i connect my windows 8 pc running xlite softphone with a windows 7 machine acting as server through a local LAN. Thank you Ashish Mishra

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DTMF issues
by baskar 05 Aug '13

05 Aug '13

Hi All, Our current existing set up has three different SIP service providers and all three were working fine, but all of a sudden on one of our SIP service provider we started to have DTMF issues. We followed up with the problematic service provider and they kept saying that there is no issue at their end, now we have connected the non working SIP link to our test server and we are not having any DTMF issues while using it on the test server. I have attached log on pastebin. Freeswitch Live server log http://pastebin.com/mQ27rDts Freeswitch test server log http://pastebin.com/4BEeUcSu Please some one help me to resolve this issues. Thanks, N.Baskar -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/DTMF-issues-tp7593499.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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Skype "Problem witch Audio Playback" - with skypopen-oss module.
by Jacek 05 Aug '13

05 Aug '13

Hi How should it look like fs_cli command that automatically connects the call using the following method: A call to the user, when he receives a call (pick up the phone), hears the message about the implementation of the merger, and then will be automatically connected with the number of user B. I mean the shape of the script command. Is it possible to do this with *fs_cli** -**x*, or do I have to combine the script, which will implement a similar action. Cheers Sorry for my English, my native language is Polish.

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