October 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Anything which encrypt SIP/SDP messages ?
by Herold Park 15 Oct '14

15 Oct '14

Is there something which encrypt SIP and SDP in the FreeSWITCH ? The SIP messages seem to be encryption with AES in FS. But the SDP doesn't. does it OK? If it is available, How can i configuration in the FreeSWITCH? * Any help with this would be greatly appreciated.*

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Cannot compile mod_cepstral
by Paul Seethaler 15 Oct '14

15 Oct '14

Hello All, I am trying to compile mod_ceptral following the instructions from https://wiki.freeswitch.org/wiki/Mod_cepstral it recommends to use 4.x instead of 5 of cesptral, but I can only download version 6. As there are some notes further down for version 6 under ubuntu, I thought I give that a try, After I disabled flit and enabled cepstral in the modules.conf , I tried to 'make' freeswich. But the proccess stumbles over the cepstral compilation with the following error: making all mod_cepstral make[5]: Entering directory `/usr/src/freeswitch.git/src/mod/asr_tts/mod_cepstral' CC mod_cepstral_la-mod_cepstral.lo CCLD mod_cepstral.la /usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when searching for -lswift /usr/bin/ld: cannot find -lswift collect2: error: ld returned 1 exit status make[5]: *** [mod_cepstral.la] Error 1 make[5]: Leaving directory `/usr/src/freeswitch.git/src/mod/asr_tts/mod_cepstral' I have tried various things now. I have reinstalled freeswtich completely following the instructions from https://confluence.freeswitch.org/display/FREESWITCH/Debian#Debian-Building… building the "stable" 1.4 version of freeswitch. But it is always the same error. I cannot find a place in the sources where the condition resides that makes it skip the /opt/swift/lib/libswift.so (which does exist, and as far as I can see is setup correclty via ldconfig) Anybody have an Ideas? Kind regards I am using a virtual box running Ubuntu 14.04 64bit Freeswitch 1.4stable

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NATIVE SQL ERR [unable to open database file]
by Linux Vince 15 Oct '14

15 Oct '14

Hi, I am trying to confure FreeSwitch for my VOIP gateway. It works really good in testing in low load. However, whenever i try to increase load it takes only around 240 concurrent calls and any call beyond that it starts giving 503, Bad Gateway errors. On console i start getting below error, which led me to believe that database issue and it seems it cannot take more connections. I tried setting up MySQL as database as well with same results. 2014-10-13 20:00:50.680811 [ERR] switch_core_sqldb.c:1196 SQL ERR: [select state,status,rpid,presence_id,uuid from sip_dialogs where uuid != '35579a26-4bc4-40ab-ae5c-a2b6f460c801' and call_info_state != 'seized' and hostname='BOX02' and profile_name='internal' and ((sip_from_user='sipp' and sip_from_host='10.0.0.14') or presence_id='sipp(a)10.0.0.14') order by rcd desc] unable to open database file 2014-10-13 20:00:50.680811 [ERR] switch_core_sqldb.c:586 NATIVE SQL ERR [unable to open database file] update sip_subscriptions set version=version+1 where hostname='BOX02' and profile_name='internal' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='sipp' and (sub_to_host='10.0.0.14' or sub_to_host='10.0.0.15' or sub_to_host='N/A' or presence_hosts like '%10.0.0.14%') and (sip_subscriptions.profile_name = 'internal' or presence_hosts like '%10.0.0.14%') 2014-10-13 20:00:50.680811 [ERR] switch_core_sqldb.c:1196 SQL ERR: [select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','10.0.0.14',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '',sip_subscriptions.orig_proto,sip_subscriptions.full_to,sip_subscriptions.network_ip, sip_subscriptions.network_port from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name and sip_subscriptions.hostname=sip_presence.hostname) where sip_subscriptions.hostname='BOX02' and sip_subscriptions.profile_name='internal' and sip_subscriptions.event != 'line-seize' and sip_subscriptions.proto='sip' and (event='presence' or event='dialog') and sub_to_user='sipp' and (sub_to_host='10.0.0.14' or sub_to_host='10.0.0.15' or sub_to_host='N/A' or presence_hosts like '%10.0.0.14%') ] unable to open database file Please help. Thanks, Vishal

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Free switch to ONLY handle RTP?
by Malcolm Lockyer 15 Oct '14

15 Oct '14

Hi, Just thought I'd throw this out to the list just in case somebody can get my started somewhere. I'm hoping to get FreeSwitch to ONLY handle the RTP/audio/media of a call while handling the SIP myself/elsewhere. I'm trying to pull apart and open up a really old in-house system that has 2 separate components for VoIP traffic. Part A - Handles SIP/presence/other stuff. - Commands Part B. Part B - Establishes and passes RTP data/mixes/conferences/plays/records etc. based on a partial SDP (session description protocol). I'm hoping to build a new server C as a replacement for B. I intend to use the command socket interface with FreeSwitch. Since I'm building a new server, I could "fake" SIP messages or something like that with FreeSwitch (i.e. have my new service C interact with FreeSwitch SIP). But I'd rather just command it to open / bridge / etc. RTP connections. Can anybody point me to something that might get me started, other than the obvious command socket interface and basic docs? I'm going to be messing around with this anyway - just wonder if somebody can suggest a head start. Thanks! -Malcolm.

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rewriting INVITE
by Anish Patel 14 Oct '14

14 Oct '14

Hello All, this is probably a stupid question, but i haven't been able to figure out it out so far. I have an invite that i am re-writing, i was wondering if its possible to just use the gateway entry i have already configured. for example, if i have a gateway called thisgateway, how would i construct the dialplan line for it? i have currently used <action application="bridge" data="{sip_invite_req_uri=sip:'.$reqesturi['destination_number'].'@'.$carrier.',sip_cid_type=pid,ignore_display_update=true}sofia/gateway/'.$carrier.'/'.$outbound['destination_number'].'"/> this doesn't seem to work for me, the $carrier variable is the name of the provider as configured in the xml files. if i replace the $carrier variable with the addresses from the xml file it works fine as well. thanks in advance, anish

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phone ring
by Pedro Santos 14 Oct '14

14 Oct '14

i want to differentiate the ring from internal and external calls. Can anyone help me ? tx it is not the ringback, but the ringing of the phone

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Abnormal TLS handshake
by Herold Park 14 Oct '14

14 Oct '14

On the FreeSWITCH-1.4.8, i setup FreeSWITCH with TLS mode using IP Phone(Ericsson-LG IP8830E). An IP Phone starts TLS handshake with FreeSWITCH one time, And the handshake finish well. It's OK. However FreeSWITCH send TLS handshake message(Client Hello) to the IP Phone one more time. This procedure make change TLS port number(5061) of FS to the other random number. It's fault. How can i remove second TLS handshake which start from the FreeSWITCH. Please help me.

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Concurrent Channels Limit
by Linux Vince 14 Oct '14

14 Oct '14

I posted a message earlier http://lists.freeswitch.org/pipermail/freeswitch-users/2014-October/108568.… I was able to fix the database error. However FreeSwitch is still not able to take more than 240 calls at a time. Help on how to troubleshoot problem would be greatly appreciated. Thanks

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Re: [Freeswitch-users] Manual generation of CDRs
by Dan Christian Bogos 13 Oct '14

13 Oct '14

Hi Neven, Although more like a work around but practical and more secure from the point of view of really having your CDRs, you can also loop the call (looping SIP signalling) through FS back (eg using fs_path) and let FreeSWITCH generate your CDR out of that looping. I prefer this approach since CDRs are real money and the risk of loosing them involving events processing over the network is more expensive than the resources used by FS to loop the call (although we rely within CGRateS quite heavily on ESL, but when it comes to CDRs we always prefer FreeSWITCH to handle them via internal modules, not external ones). You can even use a separate SIP profile listening on another IP than the public one where the initial call comes in. Hope that helps! DanB On 13.10.2014 04:25, freeswitch-users-request(a)lists.freeswitch.org wrote: > Hi guys, > > I'm new to Freeswitch and I have one question. > > It's regarding CDRs, what I would like to do is: > > 1. Customer calls number > 2. IVR responds and prompts for PIN > 3. PIN is provided by customer > 4. NEW CDR IS GENERATED FROM DIAL PLAN > 5. Call proceeds > > Is it possible to somehow generate new CDR "manually" from DialPlan? > If not what is your proposal for solving this? > > Thank you in advance, > Neven

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Strangeness on voicemail message creation
by Alex Lake 13 Oct '14

13 Oct '14

Had a report of some accounts on our service not being able to take voicemail messages. When I investigated, it appeared that some voicemail storage directories were not accessible because an actual voicemail file (.wav) had taken the place of the directory! eg. /usr/local/freeswitch/storage/voicemail/default/1.2.3.4/0982123 is a .wav file. The version of Freeswitch is 1.2.22+git~20140223T182912Z~f009fd47a1~64bit I just wondered if this apparent bug rings any bells with anyone here? I reckon I could easily implement a crude workaround - which is to schedule a job to look for non-directory files in the /usr/local/freeswitch/storage/voicemail/default/1.2.3.4 and delete them if they exist!

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