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FS - DTMF and Cisco CME/UCE
by Mike Taylor 24 May '11

24 May '11

Hi, This is driving me nuts. I have a Cisco CME (a few actually) where I cannot get DTMF to work. Does someone happen to have the magic recipe of SIP/Dialplan profiles and Cisco config to make DTMF <relay> work? I spent several very very painful days trying to get RFC2833 to work, using FS to try and generate the tones itself. I ended up with all sorts of codec mismatches, delay,distortion and some really mangled DTMF tones So, now I'm back to; PSTN----CiscoGW----<external>----FS----<internal>----<CUST PABX>---Phone Where <CUST PABX> is 3 CMEs that won't talk DTMF 30+ other vendor's gateway's that WILL do DTMF Context External and internal are both set to dtmf-type info (or dtmf_type info) DTMF works FROM the Cisco GW towards our network, doesn't So, does anyone have a magic solution? Because Cisco TAC say they see DTMF being sent (as rtp-nte), when I need INFO or in-band. Thanks, Mike

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Removing SIP Auth
by Kurtis Heimerl 24 May '11

24 May '11

Hello! I've been trying to get my SIP clients to register and authenticate with freeswitch without passwords. I've been following a few wiki pages, primarily: http://wiki.freeswitch.org/wiki/Acl and http://gnuradio.org/redmine/wiki/gnuradio/OpenBTSSettingUpFreeSWITCH Most of the configs are stock, except my directory which has one new user in the "default" folder: <include> <user id="IMSI641104378340883" cidr="192.168.1.0/24,127.0.0.1/32" number-alias="1301"> <params> <param name="vm-password" value="1000"/> <param name="accept-blind-reg" value="true"/> <param name="accept-blind-auth" value="true"/> </params> ... This config is working in one one way, I do not need to register to make a call. The phone is currently able to call into FS. However, any attempts to register ()REGISTER messages) are met with a 401 authorization required response (as shown by wireshark). This means FS is unable to route calls to my handsets. My understanding was that the line, <param name="accept-blind-auth" value="true"/> when coupled with the cidr value, would remove that need. It's clearly not working. I've tried a variety of other config options (e.g., <param name="apply-register-acl" value="<acl_list|cidr>"/>), and found no joy. ACLs are complicated business, and I could take a few more stabs in the dark (using localnet.auto?). However, I thought I'd ask this list. What else do I need to do to let my users connect without a password? Thanks!

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Getting the pizza demo to work
by Yungwei Chen 24 May '11

24 May '11

Hi, I just installed the latest snapshot on CentOS 5. I want to try the pizza demo associated with 00_pizza_demo.xml, but I'm having trouble getting the pizza demo to work. According to freeswitch.log, it looks like mod_pocketsphinx doesn't exist. ...skipped... Dialplan: sofia/internal/1000(a)192.168.16.3 Regex (PASS) [pizza_demo] destination_number(74992) =~ /^(pizza|74992)$/ break=on-false Dialplan: sofia/internal/1000(a)192.168.16.3 Regex (PASS) [pizza_demo] ${module_exists(mod_spidermonkey)}(true) =~ /true/ break=on-false Dialplan: sofia/internal/1000(a)192.168.16.3 Regex (FAIL) [pizza_demo] ${module_exists(mod_pocketsphinx)}(false) =~ /true/ break=on-false Dialplan: sofia/internal/1000(a)192.168.16.3 parsing [default->Talking Clock Time] continue=false Dialplan: sofia/internal/1000(a)192.168.16.3 Regex (FAIL) [Talking Clock Time] destination_number(74992) =~ /^9170$/ break=on-false ...skipped... Dialplan: sofia/internal/1000(a)192.168.16.3 Action transfer(74992 enum) But I do see mod_pocketsphinx.so in /usr/local/freeswitch/mod folder. [root@templ0 mod]# ls /usr/local/freeswitch/mod/mod_pocketsphinx.so -l -rwxr-xr-x 1 root root 1353092 May 23 13:23 /usr/local/freeswitch/mod/mod_pocketsphinx.so I also went through http://wiki.freeswitch.org/wiki/Mod_pocketsphinx, but the problem is still there. How am I missing here? Thanks.

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Voicemail recordings being cut off on the front end.
by Jay 23 May '11

23 May '11

I need some direction on tracking down why some voicemail recordings are being cut off at the front end. There are instances where the start of the .wav file is a good 5 seconds into the intended message. what FS components could be responsible clipped voicemail recordings? is there silence detection going on? if so, are there any controls such as automatic gain control, fixed volume settings for recordings or other normalizing going on? any advice? thx.

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mod_event_zmq - sending commands?
by Peter P GMX 23 May '11

23 May '11

Hello, we have been successfully testing mod_event_zmq and we are quite pleased with this module so far. My question: In order for being able to use the same architecture for sending events: Are ther any plans for sending API commands to Freeswitch by zmq? This may be an alternate method for XML-RPC. I understand this my be a security issue, but we could manage this by some firewall rules. Best regards Peter

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mod_manage install
by budi wibowo 23 May '11

23 May '11

hi i have fs running on centos 5.5, any body knows the combination version of software need to be installed like mono etc. have tried few combination but always failed compiling thx budi

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30 Seconds - About 6 Rings
by Michael Gende 23 May '11

23 May '11

Hello, We just had an issue come up. Short version: we want to allow for about 6 rings for incoming calls. We had been using the simple logic below (from our default.xml) allowing 15 seconds (see "call_timeout" value). That gave time for about three rings. Need 6 rings? Just call the VoIP provider (Urbancom in our case) make sure they will cooperate and "up" the timeout to 30. Right? Hasn't worked out that way. Two things happen: Turns out most calls (some land line, some cellular) ring for about 20 seconds, then the caller gets an error tone and our FS returns 847, originator cancel. Strangely, when I call in from Sprint (near Chicago), the logic just goes "round and round", ringing forever. It just seems to "re-set" and rings another 20 seconds, then resets again, ringing forever. I'm thinking that Urbancom's PSTN provider may not wait for 30 seconds before canceling the call. Sprint must "come in" to Urbancom another way. But that doesn't explain not going to VM. Those are my thoughts. Anything I'm missing? Thanks, Mike  <extension name="UrbancomIn"> <condition field="destination_number" expression="^(8201)$"> <action application="set" data="*call_timeout=15*"/> <action application="set" data="domain_name=192.168.0.198"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="group/support@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default 192.168.0.198 1000"/> </condition> </extension>

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Freeswitch commands customization
by awais nazir 23 May '11

23 May '11

Hi I am using a webpanel to run freeswitch commands (for example show calls). Can we do some customization like if we want certain colums to be eliminated from "show calls" resultset? Regards

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call somebody into conference after conference established
by fieldpeak 23 May '11

23 May '11

Hi Gurus, i dial 666 and enter a conference, then i need call some body's phone number to join him into this conference... is there anyone can advise how can realize this scenario? thanks. Regards, Charles

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Question about faxing between FreeSwitch servers
by Chris Cureau 22 May '11

22 May '11

Hi, everyone! I am trying to implement a workable fax solution between two FreeSwitch servers. This is what I had in mind so far: T1 PRI <--> Sangoma A102 <--> FreeSwitch (host) <--> T.38 SIP (internet) <--> FreeSwitch (remote) <--> Sangoma B600D <--> FAX machine I am assuming here that the T.38 protocol between FreeSwitch servers will ensure that the data transfers between the two sites successfully. Would this actually be a two step process, where one server receives the fax (say, from the PRI) on one call, then sends to the second server, and the second server initiates a second call to the fax machine? The object here is to successfully pass faxes using SIP. Is there a better way to handle this? Thanks in advance, Chris

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