November 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch HA advices
by Federico Castro 18 Dec '14

18 Dec '14

Hi all, I'm working on a Freeswitch HA solution. Now I'm deciding what method and DB I'll use to track calls. I have installed PostgreSQL on both servers and I configured them to replicate DB asynchronously. I would like to know if someone has experience with this kind of solution and what things do I have to contemplate to deploy a solid solution. Thanks in advanced. Federico.

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Re: [Freeswitch-users] Issues with rtp audio going in slow motion
by Joel White 15 Dec '14

15 Dec '14

Is this an issue with transcoding? I am wondering how I lock it down so that transcoding only happens at a minimum Also, would it be bad to completely remove G722 from the codec list? I think the issue is that our provider is sending G711 and all of the Polycom Phones are trying to use G722 On Mon, Oct 20, 2014 at 11:43 AM, Joel White <joelewhite(a)gmail.com> wrote: > I am having an issue where some calls start out fine and then go into slow > motion (distorted slow speech) > > I noticed something in the logs stating > > Activating write resampler > > > What does this mean and how do I avoid this in the future? > > > > Thank you in advance >

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TLS versions and PFS settings
by Daniel Ivanov 12 Dec '14

12 Dec '14

What are the options to enable PFS on TLS support, i found the ability added in a changelog from feb,2014, but can't find the corresponding config file params. Also can i specify a list of allowed ciphers?

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Caller ID
by Joel White 09 Dec '14

09 Dec '14

I have several installations of FreeSWITCH. I have managed to get one to dynamically create the user directory from PostgreSQL and it properly sets the outbound caller id. I have another system, very similar in how it was configured. The second installation also dynamically generates the user directory from PostgreSQL as well. I am however having an issue with setting the caller id on this other system. I have went through with a fine tooth comb and am having a hard time locating the discrepancy between the two systems. Has anyone had an issue similar to this?

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Billing software
by Luis Daniel Lucio Quiroz 09 Dec '14

09 Dec '14

What other options for FS compatible software other than vBilling do you recommend me?

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FreeSWITCH 1.4.14 Released
by Ken Rice 08 Dec '14

08 Dec '14

New Post on freeswitch.org from krice387 check it out at http://ift.tt/1zmebiE FreeSWITCH 1.4.14 Released FreeSWITCH 1.4.14 has been released! This is routine maintenance release. Quick highlight of improvements: support for logging full timestamps with dialplan (defaults to old behavior unless requested), allow mod_http_cache to S3 services other than Amazon, and new configuration parameter add-variables-to-offer, addition of `connect-timeout` option for curl API command, allow setting CURL timeout from curl API command , respecting umask when creating new files in mod_sndfile , support for additional curl auth methods in mod_curl , fifo_position macro, improvement of accuracy in timerfd implementation , and remove support SSLv2/3 from mod_rayo due to security concerns. Tarballs and Packages available at the usual locations. New features that were added: 7c0cf50 Set managedlist to return the value to the API stream and the log in mod_managed 4037e78 Support per-module references directory in mod_managed 10ebeba Add console log level in mod_managed 33cb950 Added pure CreateStateHandlerDelegate in ManagedSession for native api usage in mod_managed 889b678 Added GetPtr to Util class for internal pointers extraction (very useful when using native api) in mod_managed 8478874 FS-6831 Addition of unset app for the chatplan and while you can already unset by calling set with var=, same with set in dialplan, this is a convenience function similar to our unset in dialplan in mod_sms [Jira: http://ift.tt/1Hwa0Xw] bc767bb Adding rfc6598.auto and adding rfc6598 space to nat.auto acl. This is the NAT444 carrier grade nat space. 1190e59 FS-6965 Set the Accept-Language into a variable when received in mod_sofia [Jira: http://ift.tt/1uLG7M3] 94278b5 FS-5159 Allow enter and exit sounds to interrupt the MOH in a wait_mod conference 1944f9a FS-6968 Changes to mod_fifo.c to add outbound_per_cycle_min in mod_fifo [Jira: http://ift.tt/1Hwa0XA] 0f2816d Add command to compile non-minified js file for testing in mod_verto f175c71 FS-6805 add support for logging full timestamps with dialplan, defaults to old behavior unless requested. Jira: http://ift.tt/11E7kpC fada4b8 FS-6977 Do not create freeswitch.serial if zrtp not enabled Jira: http://ift.tt/11E7kpG dd629c1 Add external_video_source to media handle and expose switch_core_media_start_video_thread() to start the core video thread for non-rtp based media 0eefdca FS-6947 Opus RTP payload fmtp settings ( maxaveragebitrate / maxplaybackrate ) Jira: http://ift.tt/11M58xc dd61232 FS-6979 Allow mod_http_cache to S3 services other than Amazon Jira: http://ift.tt/11E7i0R 826d428 FS-6992 Global configuration or maxplaybackrate and maxaveragebitrate from opus.conf.xml Jira: http://ift.tt/11M58xe e1c0ef5 New configuration parameter, add-variables-to-offer (default=false) (When true, all channel variables are included in the offer to rayo client in mod_rayo). 1ee325d Add `connect-timeout` option for curl API command f1df8d6 Allow setting CURL timeout from curl API command c73afe1 FS-7004 Respect unmasked when creating new files in mod_sndfile [Jira: http://ift.tt/1pgbVcN] 6f660c3 Updated mod_rayo to no longer support SSLv2/3 f198d82 FS-5666 Add ignore_connect_fail config setting to not kill the call when redis is down when using redis back-end for limit [Jira: http://ift.tt/1pgbVcP] 250234d FS-5800 Add support for additional curl auth methods in mod_curl [Jira: http://ift.tt/1pgbVcR] 1e92619 FS-6097 – added fifo_position macro [Jira: http://ift.tt/1FkkFCz] da6043f Improve timerfd implementation to be more accurate df423b8 Improve timerfd implementation to be more accurate Improvements in cross platform build supports: 0f8b993 Fix mod_say_es_ar Makefile.am 07030c6 Fix compiler warning on unmatched return type 5dee5ce FS-6953 Fix for build issues with OS X [Jira: http://ift.tt/1uLG9Um] a17be38 Add reconf target 300b8d8 FS-6973 Fix for changed Opal svn url breaking build of mod_opal [Jira: http://ift.tt/1Hwa1dY] 9673cf0 Fix for apr build issue FS-6848 freebsd arm 62a2e10 Remove hack breaking some cross compile builds. if you really need this, you should be using new enough glibc anyways. The following bugs were squashed: f87c335 Fix for issue with tracking agent availability 8f3c157 FS-6957 Fix for muting issues upon joining a conference in mod_conference [Jira: http://ift.tt/1uLG9Ue] 831832c FS-6890 Fix for FreeSWITCH forgets to send RE-INVITES [Jira: http://ift.tt/1uLG9Ug] 4eb5b38 Fix bug where re-invites needlessly re-init the codec and jb a497169 FS-6890 Additional work for FreeSWITCH forgetting to send RE-INVITES in mod_sofia [Jira: http://ift.tt/1uLG9Ug] 9c1e603 FS-6954 #comment Fix for side effect of another fix for this bug. Please test this patch. [Jira: http://ift.tt/1yzQC5V] f66f2ca FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia [Jira: http://ift.tt/1uLG9Ug] 415f82f FS-6954 Work toward fix for Freeswitch adding additional m=audio on t.38 call in mod_sofia [Jira: http://ift.tt/1yzQC5V] 5ce5199 FS-6969 Fix for unwanted profile shutdown in mod_verto [Jira: http://ift.tt/1Hwa0XE] In terms of stability these were the use cases that were fixed: f3d089a Fix a crash when trying to remove a shadow directory in mod_managed 2f1b12f OPENZAP-232 Check for digits received on sangoma isdn stack to avoid delaying moving to the ring state if all digits are received at once in overlap dialing mode. 6b8d5b2 Fix release guard timer check in freetdm 34cf3b9 FS-6980 #resolve Fix for crash when using native recording on recordstop Jira: http://ift.tt/11E7i0U ab24bde FS-5533 fix issue with busy signal being sent back to all shared lines instead of just the calling device Jira: http://ift.tt/11M58xi a279bf3 FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia Jira: http://ift.tt/1uLG9Ug 0c68bb6 FS-6957 Fix for muting issues in mod_conference Jira: http://ift.tt/1uLG9Ue 33d37ce PLIV-13 Disable SSLv3 in libcurl for mod_httapi dbc5571 FS-6983 Wrap new curl TLS macro usage with ifdefs Jira: http://ift.tt/11E7i0W 0699ea8 Fixes for various issues 87a4670 FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia Jira: http://ift.tt/1uLG9Ug 82aa331 FS-6531: #resolve set to tag on auto answer notify Jira: http://ift.tt/11M58xk 75473a7 FS-6531: #resolve set to tag on uuid_phone_event notify Jira: http://ift.tt/11M58xk 07c5cc1 FS-7003 Fix infinite loop when output sent to server without SSML configured and repeat-times=0 Jira: http://ift.tt/11E7kG0 0cf770a FS-6996: #resolve fix define change as of glibc 2.20 for _BSD_SOURCE -> _DEFAULT_SOURCE [Jira: http://ift.tt/1pgbVcT] 424df19 FS-Fix for build failure on mips[Jira: http://ift.tt/1pgbU8z] 8eaaa08 FS-6622 Set buffer size for streams based on the number of channels to avoid buffer starvation in mod_shout [Jira: http://ift.tt/1pgbVtb] 5127b64 FS-7014 Don’t touch the tech_pvt when a call has just ended, leaving us with a null tech_pvt in mod_sofia [Jira: http://ift.tt/1FkkFCF] 8330336 FS-6450 Update library [apr] Backport APR_RING_FOREACH and APR_RING_FOREACH_SAFE macros to APR for unimrcp compatibility.[Jira: http://ift.tt/1pgbVtf] c645ab3 FS-6848 #resolve #comment slight tweak to fix the logic and avoid unbalanced parens [Jira: http://ift.tt/1FkkE1B] 79de78a FS-7021 Remove extra loops when using originate_retries and the caller hangs up.[Jira: http://ift.tt/1pgbUp3]

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Recommendation for Billing for Hosted PBX Platform
by Deon Vermeulen 07 Dec '14

07 Dec '14

Hi, I've been looking at the Kazoo as a very viable option as a HostedPBX platform, but don't find any information with regards to Billing. I've also been tracking the thread about recommended GUI where 2600Hz are in the process for updated the code in git for Blue.box. This also looks like a possibility for a HostedPBX platform, but then again I have no idea about the Billing software to use for this. FusionPBX is a top contender if no the number 1 for a HostedPBX platform but again information for Billing is lacking. I would like to know what the feel out there is for Billing on a HostedPBX Platform, i.e. Cloud. If would be great if there is an opensource Billing system. I've purchased the jBilling Telecom guide, but no where can I find information about integrating with FreeSWITCH nor implementing it for a HostedPBX platform in the Cloud. Kind Regards

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Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI
by Notify Me 05 Dec '14

05 Dec '14

Hi! I am very very new to Freeswitch. I am running FreeSWITCH Version 1.5.15b+git~20141120T035109Z~79de78a0fb~64bit (git 79de78a 2014-11-20 03:51:09Z 64bit) on a CentOS 6.6 64-bit Virtual Machine. I am trying to setup freeswitch such that once it gets a call through a sip gateway, it sends the caller ID to another SIP gateway (A URI) to be processed. I am having a lot of difficulty and I am not sure if I am doing the right things correctly, and the logs give so much information it is difficult to see what is happening. Please help me verify if I am correctly configured. I've setup the gateways I expect inbound calls from and freeswitch registers correctly. example: <gateway name="sipgw081"> <param name="username" value="012345081"/> <param name="password" value="xxxxx"/> <param name="register" value="true"/> <param name="realm" value="sip.sipgwtelecoms.com"/> <param name="proxy" value="sip.sipgwtelecoms.com"/> <param name="outbound-proxy" value="sip.sipgwtelecoms.com"/> <param name="caller-id-in-from" value="true"/> </gateway> Sofia returns: external::sipgw081 gateway sip:012345081@sip.sipgwtelecoms.com REGED The gateway I expect to route caller_id_numbers to is defined, but as I was not given a password or asked to register I set it up like so (is this OK?): <gateway name="othersipgw"> <param name="username" value="user.name"/> <param name="password" value="none"/> <param name="register" value="false"/> <param name="realm" value="sip.othersipgw.in"/> <param name="proxy" value="sip.othersipgw.in"/> <param name="outbound-proxy" value="sip.othersipgw.in"/> <param name="expire-seconds" value="3600"/>  </gateway> Sofia returns: external::othersipgw gateway sip:user.name@sip.othersipgw.in NOREG >From what I can see in the documentation I've read, in order to route inbound calls anywhere in freeswitch, Calls that initially come in to the public context and are treated as untrusted—if they are not specifically routed to an extension in the default context, then they are simply disconnected. So I created a file in conf/dailplan/public with the following to dail an extension 1212 in default( I want to route caller IDs with 11 digits, all beginning with number 0): <include> <extension name="sipgw-inbound"> <condition field="caller_id_number" expression="^0(\d+)$">  <action application="transfer" data="1212 XML default"/> </condition> </extension> </include> This is where my confusion starts. I both created a file in conf/dialplan/default and edited the conf/dialplan/default.xml to route calls sent to 1212 through to the external gateway URI. I also want the caller_id prefixed with a +234 default.xml: <extension name="1212"> <condition field="destination_number" expression="^1212$"> <action application="set"data="absolute_codec_string=G729"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/othersipgw/+234$1"/> </condition> </extension> outbound.xml file in conf/dialplan/default: <include> <extension name="othersipgwoutbound"> <condition field="destination number" expression="1212"> <action application="set"data="absolute_codec_string=G729"/> <action application="bridge" data="[leg_timeout=5]sofia/gateway/othersipgw/$1"/> </condition> </extension> </include> Any assistance gratefully accepted.

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echo cancellation and mod_conference
by covici＠ccs.covici.com 01 Dec '14

01 Dec '14

Hi. I have some sip callers who call in on what are apparently bad lines and get echo when calling into conferences -- not freeswitch ones yet. So, I am wondering in mod_conference is there any attempt to do echo cancellation, if so I will try to have them call into an fs one instead. Its intermittent, so I wanted to find out first before I test. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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mod_verto behind NAT
by Ciprian Dosoftei 01 Dec '14

01 Dec '14

Hi, I am dealing with a NAT situation related to mod_verto. The switch is hosted in EC2 and the configuration works great with mod_sofia (both for receiving calls from our SIP providers and via webRTC clients). However, the counterpart configuration for mod_verto fails to expose the external IP address in the SDP. Here's the mod_verto configuration we're using: http://pastebin.com/4PDMXvW6 I can confirm the external RTP IP is set properly, however here's the returned SDP: http://pastebin.com/3vzyihRp The verto client is running behind NAT as well, but this doesn't seem to be the root of the problem as Sofia's own implementation of webRTC works fine. It is effectively sending its media to 10.178.x.x rather than the switch's public IP address. Please advise, thank you. -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender.

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