- FreeSWITCH-users - FreeSWITCH Mailing Lists

Google Voice Support?
by George F. Phelps 06 Jan '15

06 Jan '15

Now that Google has de-supported its XMPP connectivity to Google Voice, what is the Freeswitch configuration for interfacing to Google Voice? Thanks, George

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Control of IPv6 vs. IPv4
by George F. Phelps 06 Jan '15

06 Jan '15

My VoIP provider supports both IPv4 and IPV6 registrations. I can only have one registration active at a time. Freeswitch is attempting both IPv4 and IPv6 connections. Randomly, sometimes a IPv4 connection is the first (only) registration established; other times it is the IPv6 connection. How to I configure Freeswitch to deterministically only attempt one type (my choice of either IPv4 or IPv6) of connection? freeswitch(a)ip-172-31-33-109.ec2.internal> sofia status Name Type Data State ============================================================================ ===================== external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6::switch2voip.us gateway sip:XXXXXXXXXX@66.33.147.150 REGED external profile sip:mod_sofia@54.174.255.168:5080 RUNNING (0) external::switch2voip.us gateway sip:XXXXXXXXXX@66.33.147.150 TRYING (retry: 20s) 172.31.33.109 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@54.174.255.168:5060 RUNNING (0) ============================================================================ ===================== 4 profiles 1 alias freeswitch(a)ip-172-31-33-109.ec2.internal> Thanks, George

3 8

Detecting multiple tones with freeswitch
by Bunea Lucian 06 Jan '15

06 Jan '15

Hello, My SIP provider (Vodafone) decided to make me a Christmas present: they change the configuration of their SIP server. Using the old configuration they were sending SIP codes for decline, busy and unavailable. Now they are sending early media with tones (for decline and busy) and a one minute message for unavailable. Each status is followed by a SIP/480. Since I need to be able to distinguish between different statuses, I have determined the following: - if I don't receive a ring tone within 7 second, the called number is unavailable; - if I receive a busy tone within 4 seconds the called number is busy; - if I receive a busy tone after 4 second the call number has declined the call; This is how I tried to implement it: - after receiving early media (SIP/180 or SIP/183) a timer is started for call hangup (sched_hangup) - if I receive a ring tone within 7 seconds the timer is canceled (sched_cancel); - if I receive a busy tone a lua script is called; <action application="set" data="sip_ignore_remote_cause=true"/> <action application="set" data="execute_on_media=sched_hangup +7 SUBSCRIBER_ABSENT"/> <action application="tone_detect" data="busy 425 w +60000 lua 'busy_detected.lua ${uuid} ${created_time}' 2"/> <action application="tone_detect" data="ring 450 w +7000 sched_cancel"/> <action application="bridge" data="sofia/external/${destination_number}(a)XXX.XXX.XXX.XXX"/> But, for some reason, the tones are detected, but the applications are never called. If I add one of the following lines to the dialplan the corresponding application is called: <action application="set" data="execute_on_tone_detect=sched_cancel"/> or <action application="set" data="execute_on_tone_detect=lua busy_detected.lua ${uuid} ${created_time}"/> but I can't tell which tone was detected... What am I missing? PS: Tested with FreeSWITCH Version 1.4.14-1~64bit. Regards, Lucian

3 4

FreeSWITCH Week in Review (Master Branch) December 28th-January 4th
by Ken Rice 06 Jan '15

06 Jan '15

New Post on freeswitch.org from kathleen check it out at http://ift.tt/17g3uGI FreeSWITCH Week in Review (Master Branch) December 28th-January 4th Hello, again. This week in the FreeSWITCH master branch we had 10 commits. The features for this week are improvements to socket handling and the implementation of grunt for javascript management in mod_verto. Grunt is a task runner for javascript designed to automate small mundane tasks like minification, compilation, unit testing, and linting. New features that were added: bf5210b Improved mod_verto socket handling code 0db3b1e FS-7127 Add grunt to mod_verto [Jira: http://ift.tt/17g3xlC] Improvements to documentation: 7a517ee FS-7127 Update README and sync re-generate verto-min [Jira: http://ift.tt/17g3xlC] f0ec193 FS-7127 Update the README for mod_verto [Jira: http://ift.tt/17g3xlC] The following bugs were squashed: 65631ed Revert FS-7004 pending updated fixed for that issue from the original author [Jira: http://ift.tt/1pgbVcN] a067a49 FS-7046 fix warning introduced from b341ff7 properly [Jira: http://ift.tt/1yJc4rg] a961b0e Fix return size in mod_verto 1965b3b FS-7106 #resolve Fix concurrency issue in mod_httapi [Jira: http://ift.tt/1BEamXd] 0bec209 Fix fsapi in verto

1 0

Canceling sched_hangup and UUID lookup
by Kurtis Heimerl 06 Jan '15

06 Jan '15

Hello Freeswitch-users, I'm building a group billing solution for our freeswitch system. Our account management is external, so I need to make something custom. I have two questions about how to do some stuff in FS. 1) I know how to limit call length (sched_hangup). I can query an external account via REST to see how many credits they have and limit the call with that. However, when a second user on the same account calls in, I need to change that timeout. Is there a way to cancel a sched_hangup? I found sched_cancel, but that seems to be for broadcasts and transfers, neither of which I'm doing. I *guess* I could transfer to a hangup extension instead of hanging up... but that seems awful ugly. 2) Is there a way to "tag" a call (uuid) with an arbitrary text string and then look it up by that? That would allow me to tag each call with the account name and later get all calls under that account. Thanks!

4 10

How to play a stream other than mp3 with mod_shout.
by Aqs Younas 06 Jan '15

06 Jan '15

Hi, All How can i play a live stream other than mp3 with mod_shout or any module.? Is there any way to buffer the stream before playing it with mod_shout. Currently i have a list of streams and when i play them with mod_shout some work fine but others give (time out) error. How can i play mostly stream in freeswitch? Thanks

8 16

Help required
by Richard Adams 06 Jan '15

06 Jan '15

Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams(a)STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850]

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MP2
by Thomas Auge 05 Jan '15

05 Jan '15

Hello list, would it be a big task to add mp2 (transcoding) to freeswitch? What would be a realistic bounty for it? Cheers! Thomas

4 23

multi-tenant registration for Cisco spa 112
by Robert Oldham 05 Jan '15

05 Jan '15

I am a freeswitch noob, so please forgive me if my questions are not coherent with proper concepts. I have made an earnest attempt to find clues to my problem on the wiki and elsewhere, but suspect I am asking the wrong questions. Problem: I cannot get a Cisco spa 112 to register properly. Context: I am trying to put together a proof of concept multi-tenant freeswitch server. After reading the book FreeSWITCH 1.2 by Anthony Minessale, I used the FusionPBX install script to get freeswitch 1.4.14 built on a fresh Ubuntu 12.04 install. I configured a domain name to point to the new server. On the server I added the domain in the directory and 3 users/extensions along with several sip gateways. The gateways work. One of the extensions has a Grandstream GXP2160 registered successfully, another has a Zoiper soft phone registered successfully. Details: I have tried to configure and get a Cisco SPA 112 to register on the 3rd extension. However, it is failing to register. I have watched the sip trace and packet capture. Both show the same thing: the Cisco SPA sends a sip register request to which sofia responds with a 401 "unauthorized". Then, instead of sending a sip register with an authorization digest as the other phones do, the Cisco spa 112, sends another generic sip register request, which elicits another 401 "unauthorized" from sofia. This repeats a few times, and then the spa 112 gives up, waits 40 seconds and then does it all over again. I have had this device registered with an asterisk server without trouble. I have reset the device to factory defaults and updated the firmware, all without any change. The domain name is fairly long. Don't know if that might be a factor. Any clues where I might look next or tests I might try for more information? Thank you, Robert Oldham Oldham Technology Phone: (801) 877-2190 Email: robert(a)oldhamtechnology.com Website: https://www.oldhamtechnology.com

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Google Voice Support?
by George F. Phelps 05 Jan '15

05 Jan '15

Now that Google has de-supported its XMPP connectivity to Google Voice, what is the Freeswitch configuration for interfacing to Google Voice? Thanks, George

1 0