May 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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!

1 0

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.

2 1

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.

2 1

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

2 1

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

2 8

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>

1 0

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

3 2

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

5 11

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

5 5

Project Announcement and Request for Direction
by Kurtis Heimerl 22 May '11

22 May '11

Hello Freeswitch Users! My name is Kurtis Heimerl, and I'm a graduate researcher at the University of California, Berkeley in the Technology and Infrastructure for Emerging Regions (TIER) group under Eric Brewer. We're currently investigating the use of OpenBTS (http://openbts.sourceforge.net/) for providing cellular coverage via low-cost base stations in low-density parts of the world. This project is called The Village Base Station. In support of that project, we're also investigating the use of Freeswitch to support OpenBTS and provide a flexible, extensible, and simple platform for deploying voice/sms/data applications on the basestation itself. Towards this end, I'm asking you, the freeswitch community, for advice and direction on some of our research goals. The basic story is simple. OpenBTS uses a software radio and generic PC to provide cellular service. As a byproduct of this architecture, we also gain the ability to run freeswitch applications "locally" (concurrently with OpenBTS), and take advantage of the benefits this tight coupling provides us. These benefits are numerous; calls/SMS between BTS users can be much cheaper, as they do not use any backhaul bandwidth. Applications can query the system for more information about users, such as location or status. As an example, unlike traditional GSM telephony applications, we are able to query if users are currently available on the network. This could be used to create voice "chat lists", which tell participants which of their friends are currently within cellular range. We foresee 6 features required to support these "local" freeswitch applications on our OpenBTS system. I'm very curious how the freeswitch community feels about these possible additions, as well how they might be implemented. There are a few that are already available in freeswitch, but may be rougher than we would like. These include: 1) Identity: The ability to query for user's status, numbers, etc. This seems simple enough in the existing system. However, we'd like to provide hooks for applications to act on these sign-ons or offs. For instance, an app may hold messages until a phone logs onto the system and push them then. My understanding is that this should be simple, probably hooking onto "SIP presence" events? 2) Storage: Freeswitch currently seems to support only per-application storage, with limited support for cross-application storage (mostly user directories). This is occasionally problematic: one issue we've heard is that it is difficult to place messages into voice mailboxes from other apps. We'd like a more unified storage framework. Or... 3) "Pipes": This is the ability to pass messages between freeswitch apps. This seems pretty well supported though simple dialplan interactions, though the modules themselves may not provide enough functionality. Is there a way to do this inside of apps? I consider this an alternative to the storage framework discussed in #2 Lastly, there are three functions we don't believe are well-supported in freeswitch. These are... 1) Privacy: We expect our BTS to be used in politically sensitive areas. Given this, freeswitch could provide an anonymity layer, providing short term phone/SMS numbers, or directing communications through more secure layers (e.g., Tor). 2) Asynchrony: While freeswitch seems to support basic asynchrony though its event system, I couldn't find any way to delay events for indeterminate times. For instance, we may want to schedule a traffic warning for 1PM every Wednesday to every phone currently on the system. Is there a way to do that currently? 3) SMS: Freeswitch seems to currently support SIP chat messages (using SIMPLE?). We need to either extend OpenBTS to speak SIMPLE, or extend freeswitch to speak OpenBTS's SMS protocol. Neither seems particularly difficult. This will allow our apps to send and receive both voice and SMS messages from users. We believe these core functions will enable a wide variety of BTS applications. I have a laundry list of those, but I'll omit them for sake of space. If any members of the community (that means you!) have any directions, ideas, projects, or thoughts, please pass them on! We're just beginning this part of the project, and getting the lay of the land. Feedback is critical at this point. Thanks!

3 2