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Uncovering spoken phrases in *encrypted* VoIP conversations
by EdPimentl 15 Mar '11

15 Mar '11

New recent study at *(Johns Hopkins / UNC)* says it can *Uncover spoken phrases in *encrypted* VoIP conversations * *(Johns Hopkins / UNC) ACM paper* We evaluate our techniques on a standard speech recognition corpus containing over 2,000 phonetically rich phrases spoken by 630 distinct speakers from across the continental United States. Our results indicate that we can identify phrases within encrypted calls with an average accuracy of 50%, and with accuracy greater than 90% for some phrases. Clearly, such an attack calls into question the efficacy of current VoIP encryption standards.

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conference comfort noise: recommended setting
by David Wafula 15 Mar '11

15 Mar '11

Hi all, I wish to include background noise in a conference where users are calling in at 8000khz. What comfort-noise value would be recommended for this? Thanks. -- David Wafula

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modify ip and port of rtp server in sdp data
by Чистяков Иван 15 Mar '11

15 Mar '11

how can i modify ip and port of rtp server in sdp data on both legs before bridging for using external rtpproxy ?

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How to route ata question
by Charles Bujold 15 Mar '11

15 Mar '11

My question is I have an ATA FXO (Grandstream ht503) that currently sends calls to an extension, I want the calls to be filtered by the other Freeswitch settings and I am not certain how to proceed. I originally created a Main IVR (ext 5555) and sent the incoming FXO ATA calls to the IVR extension. Now I want to send the calls to some other extension or inbound route so that the calls are routed through all the Freeswitch settings such as holidays and time of day conditions before being sent to the IVR or some after hours message. I'm not certain how I should setup the FXO ATA to send the calls so Freeswitch processes them properly? Please advise. cjb

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How to stop SIP auth challenge (REGISTER) msgs?
by Mario G 15 Mar '11

15 Mar '11

I see this was introduced in a recent git. I know they are not an error but does anyone know if there is a way to stop these? Something I need to change on the phones? These are really a problem while trying to working on problems. The log is filled with these. Thanks. WARNING] sofia_reg.c:1246 SIP auth challenge (REGISTER) on sofia profile 'internal' for [200(a)100.200.1.7] from ip 100.200.1.12

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Question about proxy_media
by Boris Kovalenko 15 Mar '11

15 Mar '11

Hello! I have two sip profiles. One is for transit traffic with proxy_media enabled by default. Second is a SoftPBX for my clients with a features like a IVR, conference, attended call transfer and so on. So the question is - what happens when A leg is from transit profile and B leg is to SoftPBX profile. What mode of proxy_media will be set when call is connected? May my user on B leg use DTMF for call transfer for example? What mode will be used for FAX? -- С уважением, Борис Коваленко ЗАО "Тагнет" (3435) 494991

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IP Failover with Sofia Recover - not recovering?!
by Avi Marcus 15 Mar '11

15 Mar '11

I finally figured out how to get pacemaker to start profiles/trigger a sofia recover on moving the floating IP (but an FS crash it still can't handle) Anyway - I 1) set up odbc for the core, and mysql master-master replication, so I see calls & channels being logged in both 2) I set track calls in all the profiles So when I set it to transfer from one node to another, I'm kinda surprised I'm not getting a recovery of the calls. I do see that the DB has a hostname stored - "sip1" or "sip2" depending on where the call originated, but even if I change it manually, a sofia recover seems to ways say "no calls to recover" Am I missing something?! -Avi Marcus

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SIP profile and alias ip
by Boris Kovalenko 14 Mar '11

14 Mar '11

Hello! Reading documentation about sip profiles found a <param name="alias">. So I may use same profile with two IP's. Is there any disadvantages to use alias? If endpoint will be connected to alias IP what IP FS will use for outgoing RTP (alias ip or rtp-ip)? Also, if gateway (for outbound calls) is reachable (by host route table) via alias IP will it be used for RTP instead of primary rtp-ip? -- Regards, Boris

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trunk without authentication
by Borin 14 Mar '11

14 Mar '11

Hello FS users, Is it possible in FS to allow all calls from a particular IP address without authentication? like in asterisk with insecure port, invite. I tried to add a new carrier in /usr/local/freeswitch/conf/sip_profiles/external/new-carrier.xml <include> <gateway name="new-carrier"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="realm" value="carrier.ip.address"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> </gateway> </include> But after restart it stil sends 407 Proxy Authentication Required to the carrier. Is there any way to trust the carrier IP?

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Where to download zrtp/zfone - server offline?
by Peter Steinbach 14 Mar '11

14 Mar '11

I tried to install zrtp and to download Zfone, but I get http://www.zfone.com/server_problems.html "Sorry, but our download server is offline." (since 29-Jan-2011) Does anybody know of another download source? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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