November 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Bad connection diagnostics? -BUMP
by Sean Devoy 10 Nov '12

10 Nov '12

BUMP. Any other thoughts anyone? What can I monitor/track/store/ect on the Server? It is Centos 5.x and I am not a unix guy. Can I store the CPU usage level stats? Sean From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sean Devoy Sent: Tuesday, November 06, 2012 5:09 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] Bad connection diagnostics? Thanks for the response AVI. I should have been more clear. This is a multi-tenant box. We use it for my 3 offices too. That is sort of why I mentioned ping and tracert results were the same, so I should have filled in the rest. The other tenants have no reported problems, and none of our 3 sites (7 phones) seem to have these issues currently. I have had all of these issues at one time or another, but write it off to temporary internet throughput variability issues/problems. I hate to use that excuse even when I truly think it is the problem. But when the problems are persistent (not constant) I am at a loss what to capture/track/test. They are looking to me to at least identify the problem and hopefully get it resolved. I do have the FS logs from at least one of these calls that had one way audio, but I am not proficient enough to spot any audio channel problems. I can hang in there through dialplan issues and sofia connections/errors and maybe even codec mismatches, but the audio connection information is still one step too far for me. Should I post those logs or can you offer another plan? Sean From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Tuesday, November 06, 2012 12:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Bad connection diagnostics? I would like an answer to this, too. However, here's a place to start: you have 2 legs - you to their VPS, and their VPS to their office phones. Perhaps there's an issue with the VPS. Can you get their VPS to playback a file and do an echo test? E.g. MOH or similar and call it directly and see if any of the same issues persist. Either way, you've narrowed down where the issue is... -Avi On Tue, Nov 6, 2012 at 6:41 PM, Sean Devoy <sdevoy(a)bizfocused.com> wrote: Hi, I have a client who has been working fine for months, but as of late they are reporting connections where the audio only works in one direction or neither and "terrible echo". I know they had some issues with their local Cable Connection, but they are supposedly resolved. Results from "ping -n 50 -l 256 " and "tracert" are virtually identical to mine from here and I have no issues. Are there some diagnostics I could run to try and pin this down. The phones are all cisco 504Gs, the server is a VPS from synapseglobal. Thanks for any ideas you may have. Sean _________________________________________________________________________ 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

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Friday FreeForAll
by Ken Rice 09 Nov '12

09 Nov '12

Hey Guys The FreeForAll is running Join us at sip:888@conference.freeswitch.org K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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conference_set_auto_outcall with Grandstream phones
by Frederick Pruneau 09 Nov '12

09 Nov '12

Hi all, I have setup a paging like solution with freeswitch. I am working with Grandstream GXP2100 phones. I have enabled "allow auto answer by Call-Info" option on the phones. Upgraded phones to the latest firmware. When i dial the paging extension, the phones freeze and I cannot do anything with the phones. I have to wait 2 or 3 minutes and phones hangup. Here is my paging config in my dialplan: <extension name="paging"> <condition field="destination_number" expression="^(12345)$"> <action application="answer"/> <action application="set" data="conference_auto_outcall_timeout=5"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/> <action application="set" data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/> <action application="set" data="conference_auto_outcall_profile=default"/> <action application="conference_set_auto_outcall" data="[sip_auto_answer=true]user/300@$${domain}"/> <action application="conference_set_auto_outcall" data="[sip_auto_answer=true]user/301@$${domain}"/> <action application="conference" data="$1@default"/> <action application="conference" data="$1 kick all"/> </condition> </extension> Is this a bug between freeswitch and Grandstream phones? Thanks! Fred

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User busy detect
by Miha 09 Nov '12

09 Nov '12

Hi, I would like to implement in my biling function that the FS will do call fwd if user is busy. How can I detect user busy? I am doing dialplan with curl. br, Miha

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New node-esl Version
by Chad Engler 09 Nov '12

09 Nov '12

node-esl just got a version bump to v0.0.7 with some great bug fixes, added tests, and more examples. As always you can find the repository at: https://github.com/englercj/node-esl Thanks to lxfontes <https://github.com/lxfontes> for his great pull requests, keep the bug reports and pull requests coming guys; need to get test coverage to 100%! Here is the full changelog: Version v0.0.7 Changelog ================================== * Added tests for esl, esl.Event, esl.Parser, esl.Server, esl.Connection * Fixed esl.setLogLevel to work correctly * Fixed Parser to correctly identify Content-Length * Made Connection callback truly asynchronous all the time * Updated Server ID generator * Connection now only automatically subscribes to events required for it to work properly * Added some "Modesl" headers to quickly check if a reply is OK or ERR * Fixed esl.Server not being able to handle empty options * Fixed an issue getting unique ID of outbound connection, issue #4 (thanks lxfontes) * Fixed an issue with execute creating invalid events, issue #5 (thanks lxfontes) * Added "myevents" option to esl.Server (thanks lxfontes) * Added more examples, including an Outbound socket example (thanks lxfontes) * Only supporting v0.8.x of node until v0.1.0 Test Coverage Report Summary (as of v0.0.7) ================================== Overall -------- 516 SLOC, 66% coverage By File -------- /esl.js: 20 SLOC, 100% coverage /esl/event.js: 83 SLOC, 100% coverage /esl/connection.js: 283 SLOC, 40% coverage /esl/server.js: 45 SLOC, 91% coverage /esl/parser.js: 85 SLOC, 96% coverage Thanks everyone for their continued support, and I look forward to hearing how people fair with the new version! Thanks, Chad Engler Web Programmer PATLive 1.800.775.7790 x746 Chad.Engler(a)patlive.com <mailto:Chad.Engler@patlive.com> Hosted Communications | Friendly Service www.patlive.com <http://www.patlive.com/>

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only port 8040 works for outbound esl connection?
by Mumuney Abdlquadri 09 Nov '12

09 Nov '12

I have posted a challenge I was having with outbound esl connection. I was using port 8022 because that was what is in the source of node-esl. Adam Kelloway suggested freeswitch is listening on that port but freeswitch listens on 8021. But I decided to change the port anyway it did not work. I then decided to change it to port 8040 like in the book and it worked. I tried other ports but they did not work. My question, is there anything that says freeswitch must only send outbound esl connection to only port 8040 and nothing else? Regards

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[Confidential] - uuid_broadcast
by Nuwan Wijerathne 09 Nov '12

09 Nov '12

Hello all, I'm resending this request as I haven't had any reply yet. Any help would be greatly appreciated. I'm trying to broadcast audio on a bridged call. My requirement is to play audio on both legs at the same time. I used uuid_broadcast in following order, Uuid_broadcast uuid 'path' both Please note that I'm sending uuid_broadcast through an esl connection. So my actual request to freeswitch is as follows, eslWriteConnection.Send("bgapi uuid_broadcast uuid 'path to audio file' both"); (eslWriteConnection is an object of .Net ESLConnection) The issue I'm having is, freeswitch not playing the audio on both channels at the same time. FreeSwitch plays the audio on one leg first, then plays on the second leg (After it finished playing on first leg). I don't have this issue with FreeSwitch 1.0.6, where it plays audio on both legs at the same time. I'm having this issue with FreeSwtich 1.2.3. Could anyone please suggest any solution. Thank you, This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.

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Continue dialplan execution on 503, NOT on early media.
by peely 09 Nov '12

09 Nov '12

Hi, I have a problem with a specific carrier which I have three interconnects with where they are rejecting with 503 on a CAPs limit. To work around this after all interconnects have been attempted I am trying to pause for 1s then attempt the three interconnects out to this carrier again: <extension name="to carrier"> <condition field="${sip_to_params}" expression="tgrp=carrier"/> <condition field="destination_number" expression="^(.+)"> <action application="set" data="bypass_media=true"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE"/> <action application="set" data="fail_on_single_reject=!NORMAL_TEMPORARY_FAILURE"/> <action application="bridge" data="sofia/gateway/carrier1/+$1|sofia/gateway/carrier2/+$1|sofia/gateway/carrier3/+$1"/> <action application="sleep" data="2000"/> <action application="bridge" data="sofia/gateway/carrier1/+$1|sofia/gateway/carrier2/+$1|sofia/gateway/carrier3/+$1"/> </condition> </extension> This all works well, except I get a large number of call attempts which result in a 183 (+ SDP) followed by a 480 and these cause the dialplan to continue with the failure reason NO_USER_RESPONSE which is not listed in the continue_on_fail reasons. I want to give up immediately on these calls. I tried the ignore_early_media=true, but the problem with this approach is that FreeSWITCH considers calls which ring for longer than 60 seconds as a timeout and clears back, which is not desirable. Is there any other approach to reroute to alternate carriers, and sleep if necessary only on 503 and not retry on all other causes, including 18x? Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Continue-dialplan-execution-o… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Connecting RTP between leg A and B directly
by Regis M 09 Nov '12

09 Nov '12

Hi, You must try to execute the api command uuid_media by setting a variable that coupled with an event. Try looking at the execute_on family : http://wiki.freeswitch.org/wiki/Channel_Variables#The_execute_on_family In your case, I think http://wiki.freeswitch.org/wiki/Variable_api_on_answer or api_after_bridge... maybe more but not documented :). or bridge_pre_execute_bleg_app All this variable/event handler can help you to run command outside script. but, i think the uuid_media off must be called after the media up this FS. Hope it helps. 2012/11/9 Víctor Vladimir Chávez Gallardo <vivlachaga(a)gmail.com> > Hi Regis, how you doing? > > see, i tried your suggestion, setting the uuid_media on both sessions > AFTER the bridge, also seting bypass_media_after_bridge = true in the call > parameters, if i put the apiExecute, the script is stalled on bridge action > until one leg trigger up an release code.. in that moment the instructions > apiExecute("uuid_media","off") are executed (the script flow continue), but > if i put those instructions BEFORE, i have an a forced release on leg A > "inline:1 Session is not active!" > > you know something about this issue? there's something wrong with my app > logic? will be very useffuly your help, thanks a lot > > originate_options_lega = > "bypass_media_after_bridge=true,ignore_early_media=true,origination_caller_id_number=1004,originate_timeout=60,leg=1"; > originate_options_legb = > "bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=60,leg=2"; > > bridge(lega_session,legb_session); > apiExecute("uuid_media", "off " + > lega_uuid); > apiExecute("uuid_media", "off " + > legb_uuid); > > > Hi, > IMHO, variables are used only in bridge in dialplan. > When doing bridge manualy via JS, you must call api uuid_media off > <uuid> after the bridge of the call, maybe with an execute_on_brigde > variable. > Take care that your A and B legs can "see" each other without funky nat, > uuid_media not renegociate RTP port. > Hope i'm right and it helps you ;) > regards > > > 2012/11/7 Víctor Vladimir Chávez Gallardo <vivlachaga(a)gmail.com> > > Hi, i have an spidermonkey script, the script place two outbound calls, > but i need to set the RTP directly between the legs (a-b) but i dont know > hoy to set the parameter bypass_media > > i tried: > > lega_session.setVariable('bypass_media', 'true'); > > bridge(lega_session,legb_session); > > also > > lega_session.setVariable('bypass_media', 'true'); > > legb_session.setVariable('bypass_media', 'true'); > > bridge(lega_session,legb_session); > > and also setting another variable bypass_media_after_bridge > > lega_session.setVariable('bypass_media_after_bridge', 'true'); > > legb_session.setVariable('bypass_media_after_bridge', 'true'); > > lega_session.setVariable('bypass_media', 'true'); > > legb_session.setVariable('bypass_media', 'true'); > > bridge(lega_session,legb_session); > > > but i dont have the RTP in the correct way, instead my freeswitch it's > behind, the rtp flow: LEG-A -----> FREESWITCH -------> LEG-B > > any idea? > > > >

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Error 606 (USER_NOT_REGISTERED) but in fact he is...
by Mi Ke 09 Nov '12

09 Nov '12

Hi, I have an IP phone behind NAT registered to also NATed FreeSWITCH (EC2): 10.10.10.107 (phone) -> 111.111.111.111 (NAT) -> internet -> 222.222.222.222 (NAT) -> 10.210.250.135 (FS) Registration part of sip trace looks OK (?): tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fdcf8004430): events IN tport.c:2845 tport_recv_event() tport_recv_event(0x7fdcf8004430) tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x7fdcf8004430) msg 0x7fdcf8010980 from (udp/10.210.250.135:5060) has 702 bytes, veclen = 1 tport.c:3004 tport_deliver() tport_deliver(0x7fdcf8004430): msg 0x7fdcf8010980 (702 bytes) from udp/111.111.111.111:5060/sip next=(nil) nta.c:2803 agent_recv_request() nta: received REGISTER sip:222.222.222.222 SIP/2.0 (CSeq 49512) nta.c:3090 agent_check_request_via() nta: Via check: received=111.111.111.111 nta.c:3161 agent_aliases() nta: canonizing sip:222.222.222.222 with contact nta.c:3002 agent_recv_request() nta: REGISTER (49512) going to a default leg 2012-11-09 01:14:34.833161 [ERR] sofia_reg.c:1613 DELETE PRESENCE SQL: delete from sip_presence where sip_user='1000' and sip_host='222.222.222.222' and profile_name='external' and open_closed='closed' nua_stack.c:529 nua_signal() nua(0x7fdcf8013190): sent signal r_respond nua_stack.c:529 nua_signal() nua(0x7fdcf8013190): sent signal r_destroy tport.c:3238 tport_tsend() tport_tsend(0x7fdcf8004430) tpn = UDP/111.111.111.111:5060 tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x7fdcf8004430): not found by name UDP/111.111.111.111:5060 nta.c:6678 incoming_reply() nta: sent 200 OK for REGISTER (49512) Here registration LED on the phone goes green, phone gives me a dialing tone i.e. behaves like registered. show registrations looks OK: reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata 1000,222.222.222.222,6a7735fa-6f81e64c@10.10.10.107,sofia/external/sip:1000@10.10.10.107:5060;fs_nat=yes;fs_path=sip%3A1000%40111.111.111.111%3A5060,1352420589,111.111.111.111,5060,udp,domU-12-31-39-09-F5-79, sofia status profile external ... REGISTRATIONS 1 Meanwhile when incoming call hits my dialplan (simple bridge to user/1000@domain) I'm getting ERR 606 (User not registered) and here is what list_users gives to me: userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number 1000|default|10.210.250.135|default|error/user_not_registered||Mike|1000 The question is - what am I doing wrong? Thanks in advance for all yours hints Mike

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