December 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Need a help with topology
by Miha 29 Dec '12

29 Dec '12

Hi, Now I have only on FS server which works very very well. As I would try to migrate more user to FS server I am implementing opensips before FS for load_balancing and also registration. I will use opensips also for cdr (if it will work ok, like fs) and for enum. On FS I have now few groups which are using group pickup in that kind of things. Is it possible to connect FS boxes together so that it will be possible to do group call pickup and still call from other FS box or would would you suggest or how to deal with this issue? The same with CFWD and etc. thanks! Miha

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Voicemail in a Cluster
by Avi Marcus 28 Dec '12

28 Dec '12

Instead of further hijacking a specific question about call recording, I'll start a new thread... It's common to have more than one FS media server to handle calls. So then, you'll have them handling voicemail too. How do you ensure that the voicemail will be available for the user? Surely this is a common thing! Some options present themselves: 1) Ensure that each user always connects to the same server (this make maintenance impossible if the VM is only on one machine!) 2) Mount the VM directory to point to Amazon S3 (this doesn't sound ideal. Perhaps for long-term archival -- if you allow such a thing -- and seemingly the network connection could be a problem) 3) Some sort of NFS - but then this become a single point of failure? 4) Some sort of network drive that has multiple copies of the file, and it could even be local -- suggestions? Gluster? I've never used any. This would be transparent to FS. 5) couchdb, riak, (rethinkdb) (effectively) have multi-master replication/multiple sharding, but FS doesn't know how to access that. (In couchdb it's not a shard, it would be a full copy. The others configure a multiple of copies that are available) 5) Abaci suggested possibly record_post_process_exec_api or if it's not in VM that you can probably add a var in mod_voicemail.c to the file_path and use that var in the post_process api/app. Then, you trigger whatever "Copy" script to the NFS. 6) Jay Binks suggest http://code.google.com/p/lsyncd/ or using inotify directly, but it sounds like you need to establish a watch per folder, and simply - it's a hack. This should be something that can be triggered (if not fully handled) in FS itself. What are people currently doing to "share" voicemail? Can we work out some sort of recipe to easily implement #4 (or #5)? -Avi Marcus BestFone

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Newbie question about Cisco phones
by Steven Schoch 28 Dec '12

28 Dec '12

I have FreeSwitch installed, and now I'm trying to test it. I'm attempting to use a Cisco UC phone, model CP-3805. I have it talking to the TFTP server and getting a configuration file that I got from the FreeSwitch wiki. It contains the IP address of the switch, and a couple of login/password lines: <authName>1001</authName> <authPassword>654987</authPassword> I have this same login/password in conf/directory/1001.xml: <user id="1001"> <params> <param name="password" value="654987"/> Using tcpwatch, I see the switch keeps sending "SIP/2.0 401 Unauthorized" and "SIP/2.0 403 Forbidden" messages to the phone, so I obviously have something wrong. I'm thinking I have to put something in conf/sip_profiles, but that's just a guess at this point. I'm assuming this is a FAQ, but I found very little in the wiki about configuring phones. -- Steve

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Re: [Freeswitch-users] repeated registration messages in logs
by Archana Venugopan 28 Dec '12

28 Dec '12

Hi, Thanks. If there is no activity for a while then calls drops out and I need to re-start the FS. I also see these messages frequently after each call and after registration and while FS is idle. Am not sure why is this happening. Regards, Archana From: Mick Stevens [mailto:mick.stevens@smartipx.com] Sent: 27 December 2012 12:49 To: Archana Venugopan Subject: Re: [Freeswitch-users] repeated registration messages in logs Hi Archana, This is just your SIP client (softphone etc) sending keep alive/SIP re-register messages. You can normally adjust this in the settings of your client. I find setting "reregister every 300 seconds" normally works ok for me. Rgds, Mick On 27 December 2012 12:40, Archana Venugopan <a.venugopan(a)mundio.com<mailto:a.venugopan@mundio.com>> wrote: Hi, After registering the phones successfully I am getting the below message continuously every few seconds in my logs and this is filling up the logs and freeswitch goes down sometimes. Can someone please tell me why I keep getting this message every few seconds? I have another server and there I don’t see this issue. 2012-12-27 12:37:44.714558 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [Abarajit(a)vectonecloud.com<mailto:Abarajit@vectonecloud.com>] from ip 192.168.2.205 2012-12-27 12:41:58.744557 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [sana(a)vectonecloud.com<mailto:sana@vectonecloud.com>] from ip 192.168.2.48 Regards, Archana _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto: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<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 -- Regards Mick Stevens Communications Technologist Smart IPX Ltd Tel/Fax. +44(0)20 7001 6869 Email. mick.stevens(a)smartipx.com<mailto:mick.stevens@smartipx.com> Skype. mick.smartipx.com<http://mick.gno247.com/> www.smartipx.com<http://www.smartipx.com/>

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Java esl client
by ashutosh bhomia 28 Dec '12

28 Dec '12

Hi, we had developed a java application. We are using jar eslclient0.9.3 to connect with freeswitch.Parser we had used is netty. Now when we start the application the thread count continually keep on increasing till they reach 32000. when we use jstack to analyse the behavior we are getting following output. Threads in state Blocked by Locks 3054 threads running in org.freeswitch.esl.client.internal.AbstractEslClientHandler.messageReceived(AbstractEslClientHandler.java:81) 515 threads running in org.freeswitch.esl.client.internal.AbstractEslClientHandler$SyncCallback.get(AbstractEslClientHandler.java:234) 515 threads running in org.jboss.netty.channel.SimpleChannelUpstreamHandler.handleUpstream(SimpleChannelUpstreamHandler.java:112) 268 threads running in System Thread 8 threads running in org.apache.tomcat.util.threads.TaskQueue.poll(TaskQueue.java:86) Total Blocked by Locks Threads: 4360 Any help on this is greatly appreciated. -- Regards Ashutosh Bhomia

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Speex@32000h@20i = no audio
by Emrah 28 Dec '12

28 Dec '12

Hi all, I am not getting any audio (in or out) when using Speex@32000h@20i. I tried multiple softphones with same results. A show channels gives me the right codecs. SPEEX,32000,44000,SPEEX,32000,44000 Using Speex@16000h@20i works fine. Any idea? Is this broken? E Version: FreeSWITCH Version 1.2.5.3+git~20121206T171012Z~d74bef3f2f (git d74bef3 2012-12-06 17:10:12Z)

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Conference Multi Channel Recording
by Bryan Vyhmeister 28 Dec '12

28 Dec '12

I'm toying with the idea of having two or more callers in a conference and having the conference session recorded with a separate channel for each caller. This would likely only potentially work with wav and similar formats. My goal is to use SILK or another wideband codec and have a great multi channel recording. I expect this isn't currently possible. Is that true? I found RECORD_STEREO but I think that only applies to recording a regular call with uuid_record and not to a conference? Perhaps I could record each incoming leg to the conference and end up with one channel as that caller and the second channel as everyone else in the conference? Any ideas? Thank you. Bryan --- Bryan Vyhmeister Sent from my iPhone

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mod limit on inbound calls
by Erik Dekkers 28 Dec '12

28 Dec '12

Hi List, I do have this setup: User has one deskphone and one portable (IP DECT) phone. Both phones have the same extention number and multiple registrations is set in the sip config This is the situation: If a user is on one of the phones and a second call comes in both phones will ring. This is the wanted situation: If a user is on one of the phones and a second call comes in a USER BUSY signal has to be send back I know you can use limit for this, but im not sure how to configure it on incoming calls when a user has 2 phones registered on the same extention number. Thank you Regards, Erik Dekkers (wvds-nl on irc) [cid:image7432f4.PNG@de4c91c0.4b889d03] ABC Westland 555 Tel: +31 174 22 50 80 P.O. Box 90 Fax: +31 174 22 50 81 Mob: +31 624 423 009 2685 ZH Poeldijk erik.dekkers(a)certhon.com<mailto:erik.dekkers@certhon.com> The Netherlands www.certhon.com<http://www.certhon.com> DISCLAIMER All our quotations, all orders and all contracts are subject to the AVAG-CONDITIONS. Op alle offertes, opdrachten en overeenkomsten zijn de AVAG-verkoopvoorwaarden van toepassing. ISO9001 CERTIFIED

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Need community input: hangup_hook recipes
by Michael Collins 28 Dec '12

28 Dec '12

Hey all, I was working on a feature for a friend of mine and it turns out that it might be useful to put on the wiki. I noticed that we don't have a place explicitly for hangup hook recipes. I was thinking about making a page just for "interesting things you can do when the call ends" and then linking to it from the dialplan recipes page. Two questions: does that sound like a good place to put these recipes? does anyone have any recipes they'd like to contribute? Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Re: [Freeswitch-users] CLIR gsmopen
by dchs dchs 27 Dec '12

27 Dec '12

To answer my own question-CLIR can be set by AT+CLIR=1.It does work with mobile partner software.However,CLIR is disabled as soon as gsmopen is loaded. I can't figure out how to configure gsmopen to invoke CLIR. -------- Оригинално писмо -------- От: dchs dchs dchs(a)abv.bg Относно: [Freeswitch-users] CLIR gsmopen До: FreeSWITCH-users(a)lists.freeswitch.org Изпратено на: Четвъртък, 2012, Декември 27 23:25:08 EET Hello, Is there any way to set up CLIR on outgoing calls placed over gsmopen?Do Huawei modems even support such an AT command? Any pointers towards any info on how to accomplish this would be much appreciated!

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