From shaun.stokes at itec-support.co.uk  Fri Dec  1 12:39:12 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Fri, 1 Dec 2017 12:39:12 +0000
Subject: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19
In-Reply-To: <CAJwe+CCcD7+sXuK8kW17Y3H1LjqNY-C83WP_z6oWac8gh7k13g@mail.gmail.com>
References: <6FD2F8B5BB72834E9939AEDF9FB802A902F78C37F9@mbx-03.sysconfig.co.uk>
 <030606f8-de4d-643b-54a6-b6af3a61d2a4@wirelessmundi.com>
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C381E@mbx-03.sysconfig.co.uk>
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C4011@mbx-03.sysconfig.co.uk>
 <CAMDdSzO5hYVXLmbKTWOvfrGotscez7=kqpfLqafRyuxLU9cAzw@mail.gmail.com>
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C599B@mbx-03.sysconfig.co.uk>,
 <CAJwe+CCcD7+sXuK8kW17Y3H1LjqNY-C83WP_z6oWac8gh7k13g@mail.gmail.com>
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A902F78C770E@mbx-03.sysconfig.co.uk>

This sounds like a great idea, however not sure how to change BLF lights on phones registered to FreeSWITCH without using PRESENCE_IN\PRESENCE_OUT events.

If the phones are registered to FreeSWITCH but Kamailio is handling presence via FS ESL events then how do we can change the BLF lights on the phones?

Using PRESENCE_IN\PRESENCE_OUT events results in FreeSWITCH checking the sip_subscriptions table which is our bottleneck.

Thanks,
Shaun

________________________________________
From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of E. Schmidbauer [eschmidbauer at gmail.com]
Sent: 30 November 2017 22:37
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19

I have found that freeswitch does not scale well with presence.
I've seen it hit a bottleneck at around that same number of subscriptions (10,000). freeswitch's presence is SQL heavy so when you are dealing with that many subscriptions and presence updates it becomes a problem.
I would recommend using something like Kamailio for presence. You can "extract" presence events from freeswitch and "publish" presence updates to kamailio using freeswitch's esl and kamailio's presence modules.
I've seen kamailio operate under extremely high volume (~100,000 subscriptions)






On Mon, Nov 27, 2017 at 4:00 PM, Shaun Stokes <shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>> wrote:
We’re still getting presence delays but it’s isolated to a single SIP profile, appears to be a bottleneck on the number of entries in the sip_subscriptions table (over 10k), we have second internal SIP profile which has less than 1000 entries and no presence issues.

I’ve looked at a few articles, Kamailio looks like a good solution for presence, would be great if there was a similar module for FreeSWITCH. Using Kamailio as a SIP proxy to improve presence is one thing, but I imagine we’ll have similar problems with presence using Verto\WebRTC via FreeSWITCH.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Colton Conor
Sent: 26 November 2017 21:47
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>

Subject: Re: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19

Unfortunately Freeswitch does not yet support RFC 4662 aka BLF Resource Lists. I would recommend you look at Kamiliio or Asterisk which both support support RFC 4662, and should significantly reduce your BLF load. Info:


Asterisk does: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158 They have a good overview page

Kamillio does: https://kamailio.org/docs/modules/3.2.x/modules_k/rls.html

Good info from Grandsteam about why to use this: http://www.grandstream.com/sites/default/files/Resources/GXP21x0_Eventlist_BLF_Guide.pdf

Same with Polycom: http://community.polycom.com/polycom/attachments/polycom/VoIP/19112/1/Technical%20Brief%20-%20Busy%20Lamp%20Field.pdf

How do we make this a feature request for Freeswitch?




On Fri, Nov 24, 2017 at 4:00 AM, Shaun Stokes <shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>> wrote:
Performance using PostgreSQL seems better but possibly too soon to tell, however we now have the ability to analyse the data and have identified that there is one domain that accounts for over 90% of the entries (over 12000, 11000 of which are presence) in our sip_subscriptions table. All domains use presence but others typically have around 100 entries in sip_subscriptions.

If the bottleneck was the DB file then PostgreSQL should have solved this, but I'm still concerned by the number of presence events in our sip_subscriptions table. How many presence events does FreeSWITCH support in the sip_subscriptions table? Are there are any other recommendations to get the best performance out of FreeSWITCH in this scenario?

We have 32 logical cores (8 physical) but CPU usage rarely goes above 10%.

Thanks,
Shaun
________________________________________
From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] on behalf of Shaun Stokes [shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>]
Sent: 23 November 2017 09:45
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19

Thanks, will give that a try and provide some feedback.
________________________________________
From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] on behalf of António Silva [asilva at wirelessmundi.com<mailto:asilva at wirelessmundi.com>]
Sent: 23 November 2017 09:32
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19

Hi,

We had the same problem using sqlite files as core db, under heavy load
sqlite backend is not that fast... switching it to postgresql solved the
issue.

Hope it helps.


On 11/23/2017 10:15 AM, Shaun Stokes wrote:
> Does anyone have experience with presence on FreeSWITCH when under load? We've considered offloading presence to a Kamailio proxy but it would great if we could understand the limitations of presence on FreeSWITCH and recommended configuration for best performance under load.
>
> We've been experiencing issues with presence (BLF) intermittently not working or being delayed by 5 minutes or more. The BLF lights may show an extension as available when they're on the phone, as ringing when they're available etc. This problem can only be re-produced on a system which has been under load but it doesn't occur immediately unless the system has been running for a couple of weeks. This effects multiple SIP profiles in a multi-tenant (multi domain) environment with roughly 800 registrations, 500 registrations on one SIP profile and 300 on another.
>
> The FreeSWITCH DB files use a 512MB RAM disk and we've balanced extensions across multiple SIP profiles using separate (not shared) DB files to distribute load.
>
> What we've noticed is the internal DB files (i.e. sofia_reg_internal.db) appear to increase exponentially over time after roughly 22MB we often start to run into problems with presence while the system is under load, restarting FreeSWITCH and flushing the internal DB files clears the problem.
>
> Interestingly we didn't have this problem on FreeSWITCH 1.4
>
> We've recently adjusted the following settings on our internal SIP profiles and are continuing to monitor:
> force-subscription-expires [900] -> [1800]
> sip-subscription-max-deviation [300] -> [600]
> max-proceeding [1000] -> [5000]
> initial-event-threads [2] -> [4]
>
> Here is a snippet of our internal SIP profile configuration:
> nonce-ttl [60]
> outbound-codec-prefs [G722,PCMA,H264]
> pass-rfc2833 [true]
> record-path [/path/freeswitch/recordings]
> record-template [${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}]
> multiple-registrations [contact]
> inbound-reg-force-matching-username [true]
> inbound-reg-in-new-thread [true]
> initial-event-threads [4]
> local-network-acl [localnet.auto]
> manage-presence [true]
> unregister-on-options-fail [true]
> tls-version [tlsv1.2]
> tls-cert-dir [/path/freeswitch/certs]
> tls-bind-params [transport=tls]
> tls [true]
> stun-enabled [false]
> stun-auto-disable [false]
> sip-trace [no]
> sip-subscription-max-deviation [600]
> sip-port [xxxx]
> sip-ip [x.x.x.x]
> sip-force-expires [900]
> sip-expires-max-deviation [300]
> sip-capture [no]
> rtp-timeout-sec [0]
> rtp-timer-name [soft]
> rtp-ip [x.x.x.x]
> rtp-hold-timeout-sec [0]
> tls-verify-in-subjects []
> tls-verify-depth [2]
> tls-verify-date [true]
> tls-sip-port [xxxx]
> tls-passphrase []
> tls-only [false]
> inbound-codec-prefs [G722,PCMA,H264]
> inbound-codec-negotiation [greedy]
> log-auth-failures [true]
> user-agent-string [FreeSWITCH]
> watchdog-enabled [no]
> watchdog-event-timeout [30000]
> watchdog-step-timeout [30000]
> ext-rtp-ip [x.x.x.x]
> accept-blind-auth [false]
> accept-blind-reg [false]
> aggressive-nat-detection [true]
> apply-inbound-acl [domains]
> apply-nat-acl [nat.auto]
> auth-all-packets [false]
> auth-calls [true]
> challenge-realm [auto_from]
> context [public]
> debug [0]
> dialplan [XML]
> dtmf-duration [2000]
> dtmf-type [rfc2833]
> enable-timer [true]
> ext-sip-ip [x.x.x.x]
> registration-thread-frequency [240]
> rfc2833-pt [101]
> force-subscription-expires [1800]
> forward-unsolicited-mwi-notify [false]
> hold-music [local_stream://default]
> nat-options-ping [true]
> NDLB-force-rport [safe]
> max-proceeding [5000]
>
> ______________________________________________________________________
> This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
> ______________________________________________________________________
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

--
Saludos / Regards / Cumprimentos
António Silva


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
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______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
______________________________________________________________________

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
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______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
______________________________________________________________________

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


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From shaun.stokes at itec-support.co.uk  Fri Dec  1 13:34:55 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Fri, 1 Dec 2017 13:34:55 +0000
Subject: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A902F78C770E@mbx-03.sysconfig.co.uk>
References: <6FD2F8B5BB72834E9939AEDF9FB802A902F78C37F9@mbx-03.sysconfig.co.uk>
 <030606f8-de4d-643b-54a6-b6af3a61d2a4@wirelessmundi.com>
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C381E@mbx-03.sysconfig.co.uk>
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C4011@mbx-03.sysconfig.co.uk>
 <CAMDdSzO5hYVXLmbKTWOvfrGotscez7=kqpfLqafRyuxLU9cAzw@mail.gmail.com>
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C599B@mbx-03.sysconfig.co.uk>,
 <CAJwe+CCcD7+sXuK8kW17Y3H1LjqNY-C83WP_z6oWac8gh7k13g@mail.gmail.com>,
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C770E@mbx-03.sysconfig.co.uk>
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A902F78C7872@mbx-03.sysconfig.co.uk>

Unless of course you are referring to using Kamailio as the registrar, we've been looking at this option but may need to bring in a Kamailio consultant.

Thanks,
Shaun
________________________________________
From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Shaun Stokes [shaun.stokes at itec-support.co.uk]
Sent: 01 December 2017 12:39
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19

This sounds like a great idea, however not sure how to change BLF lights on phones registered to FreeSWITCH without using PRESENCE_IN\PRESENCE_OUT events.

If the phones are registered to FreeSWITCH but Kamailio is handling presence via FS ESL events then how do we can change the BLF lights on the phones?

Using PRESENCE_IN\PRESENCE_OUT events results in FreeSWITCH checking the sip_subscriptions table which is our bottleneck.

Thanks,
Shaun

________________________________________
From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of E. Schmidbauer [eschmidbauer at gmail.com]
Sent: 30 November 2017 22:37
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19

I have found that freeswitch does not scale well with presence.
I've seen it hit a bottleneck at around that same number of subscriptions (10,000). freeswitch's presence is SQL heavy so when you are dealing with that many subscriptions and presence updates it becomes a problem.
I would recommend using something like Kamailio for presence. You can "extract" presence events from freeswitch and "publish" presence updates to kamailio using freeswitch's esl and kamailio's presence modules.
I've seen kamailio operate under extremely high volume (~100,000 subscriptions)






On Mon, Nov 27, 2017 at 4:00 PM, Shaun Stokes <shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>> wrote:
We’re still getting presence delays but it’s isolated to a single SIP profile, appears to be a bottleneck on the number of entries in the sip_subscriptions table (over 10k), we have second internal SIP profile which has less than 1000 entries and no presence issues.

I’ve looked at a few articles, Kamailio looks like a good solution for presence, would be great if there was a similar module for FreeSWITCH. Using Kamailio as a SIP proxy to improve presence is one thing, but I imagine we’ll have similar problems with presence using Verto\WebRTC via FreeSWITCH.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Colton Conor
Sent: 26 November 2017 21:47
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>

Subject: Re: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19

Unfortunately Freeswitch does not yet support RFC 4662 aka BLF Resource Lists. I would recommend you look at Kamiliio or Asterisk which both support support RFC 4662, and should significantly reduce your BLF load. Info:


Asterisk does: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=30278158 They have a good overview page

Kamillio does: https://kamailio.org/docs/modules/3.2.x/modules_k/rls.html

Good info from Grandsteam about why to use this: http://www.grandstream.com/sites/default/files/Resources/GXP21x0_Eventlist_BLF_Guide.pdf

Same with Polycom: http://community.polycom.com/polycom/attachments/polycom/VoIP/19112/1/Technical%20Brief%20-%20Busy%20Lamp%20Field.pdf

How do we make this a feature request for Freeswitch?




On Fri, Nov 24, 2017 at 4:00 AM, Shaun Stokes <shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>> wrote:
Performance using PostgreSQL seems better but possibly too soon to tell, however we now have the ability to analyse the data and have identified that there is one domain that accounts for over 90% of the entries (over 12000, 11000 of which are presence) in our sip_subscriptions table. All domains use presence but others typically have around 100 entries in sip_subscriptions.

If the bottleneck was the DB file then PostgreSQL should have solved this, but I'm still concerned by the number of presence events in our sip_subscriptions table. How many presence events does FreeSWITCH support in the sip_subscriptions table? Are there are any other recommendations to get the best performance out of FreeSWITCH in this scenario?

We have 32 logical cores (8 physical) but CPU usage rarely goes above 10%.

Thanks,
Shaun
________________________________________
From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] on behalf of Shaun Stokes [shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>]
Sent: 23 November 2017 09:45
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19

Thanks, will give that a try and provide some feedback.
________________________________________
From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] on behalf of António Silva [asilva at wirelessmundi.com<mailto:asilva at wirelessmundi.com>]
Sent: 23 November 2017 09:32
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19

Hi,

We had the same problem using sqlite files as core db, under heavy load
sqlite backend is not that fast... switching it to postgresql solved the
issue.

Hope it helps.


On 11/23/2017 10:15 AM, Shaun Stokes wrote:
> Does anyone have experience with presence on FreeSWITCH when under load? We've considered offloading presence to a Kamailio proxy but it would great if we could understand the limitations of presence on FreeSWITCH and recommended configuration for best performance under load.
>
> We've been experiencing issues with presence (BLF) intermittently not working or being delayed by 5 minutes or more. The BLF lights may show an extension as available when they're on the phone, as ringing when they're available etc. This problem can only be re-produced on a system which has been under load but it doesn't occur immediately unless the system has been running for a couple of weeks. This effects multiple SIP profiles in a multi-tenant (multi domain) environment with roughly 800 registrations, 500 registrations on one SIP profile and 300 on another.
>
> The FreeSWITCH DB files use a 512MB RAM disk and we've balanced extensions across multiple SIP profiles using separate (not shared) DB files to distribute load.
>
> What we've noticed is the internal DB files (i.e. sofia_reg_internal.db) appear to increase exponentially over time after roughly 22MB we often start to run into problems with presence while the system is under load, restarting FreeSWITCH and flushing the internal DB files clears the problem.
>
> Interestingly we didn't have this problem on FreeSWITCH 1.4
>
> We've recently adjusted the following settings on our internal SIP profiles and are continuing to monitor:
> force-subscription-expires [900] -> [1800]
> sip-subscription-max-deviation [300] -> [600]
> max-proceeding [1000] -> [5000]
> initial-event-threads [2] -> [4]
>
> Here is a snippet of our internal SIP profile configuration:
> nonce-ttl [60]
> outbound-codec-prefs [G722,PCMA,H264]
> pass-rfc2833 [true]
> record-path [/path/freeswitch/recordings]
> record-template [${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}]
> multiple-registrations [contact]
> inbound-reg-force-matching-username [true]
> inbound-reg-in-new-thread [true]
> initial-event-threads [4]
> local-network-acl [localnet.auto]
> manage-presence [true]
> unregister-on-options-fail [true]
> tls-version [tlsv1.2]
> tls-cert-dir [/path/freeswitch/certs]
> tls-bind-params [transport=tls]
> tls [true]
> stun-enabled [false]
> stun-auto-disable [false]
> sip-trace [no]
> sip-subscription-max-deviation [600]
> sip-port [xxxx]
> sip-ip [x.x.x.x]
> sip-force-expires [900]
> sip-expires-max-deviation [300]
> sip-capture [no]
> rtp-timeout-sec [0]
> rtp-timer-name [soft]
> rtp-ip [x.x.x.x]
> rtp-hold-timeout-sec [0]
> tls-verify-in-subjects []
> tls-verify-depth [2]
> tls-verify-date [true]
> tls-sip-port [xxxx]
> tls-passphrase []
> tls-only [false]
> inbound-codec-prefs [G722,PCMA,H264]
> inbound-codec-negotiation [greedy]
> log-auth-failures [true]
> user-agent-string [FreeSWITCH]
> watchdog-enabled [no]
> watchdog-event-timeout [30000]
> watchdog-step-timeout [30000]
> ext-rtp-ip [x.x.x.x]
> accept-blind-auth [false]
> accept-blind-reg [false]
> aggressive-nat-detection [true]
> apply-inbound-acl [domains]
> apply-nat-acl [nat.auto]
> auth-all-packets [false]
> auth-calls [true]
> challenge-realm [auto_from]
> context [public]
> debug [0]
> dialplan [XML]
> dtmf-duration [2000]
> dtmf-type [rfc2833]
> enable-timer [true]
> ext-sip-ip [x.x.x.x]
> registration-thread-frequency [240]
> rfc2833-pt [101]
> force-subscription-expires [1800]
> forward-unsolicited-mwi-notify [false]
> hold-music [local_stream://default]
> nat-options-ping [true]
> NDLB-force-rport [safe]
> max-proceeding [5000]
>
> ______________________________________________________________________
> This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
> ______________________________________________________________________
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

--
Saludos / Regards / Cumprimentos
António Silva


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________

______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
______________________________________________________________________

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________

______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
______________________________________________________________________

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________

______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
______________________________________________________________________

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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______________________________________________________________________
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______________________________________________________________________

______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
______________________________________________________________________


From eschmidbauer at gmail.com  Fri Dec  1 16:20:40 2017
From: eschmidbauer at gmail.com (E. Schmidbauer)
Date: Fri, 1 Dec 2017 11:20:40 -0500
Subject: [Freeswitch-users] Presence BLF problems on FreeSWITCH 1.6.19
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A902F78C7872@mbx-03.sysconfig.co.uk>
References: <6FD2F8B5BB72834E9939AEDF9FB802A902F78C37F9@mbx-03.sysconfig.co.uk>
 <030606f8-de4d-643b-54a6-b6af3a61d2a4@wirelessmundi.com>
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C381E@mbx-03.sysconfig.co.uk>
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C4011@mbx-03.sysconfig.co.uk>
 <CAMDdSzO5hYVXLmbKTWOvfrGotscez7=kqpfLqafRyuxLU9cAzw@mail.gmail.com>
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C599B@mbx-03.sysconfig.co.uk>
 <CAJwe+CCcD7+sXuK8kW17Y3H1LjqNY-C83WP_z6oWac8gh7k13g@mail.gmail.com>
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C770E@mbx-03.sysconfig.co.uk>
 <6FD2F8B5BB72834E9939AEDF9FB802A902F78C7872@mbx-03.sysconfig.co.uk>
Message-ID: <CAJwe+CDUKqSkutSOTvDTsQwy-ZUt4RX4iAn_pPg24ZPJndTdFw@mail.gmail.com>

You can easily make kamailio the registrar and handle presence but you can
just as easily move only presence to kamailio. regardless of that, what i
meant by using ESL was:
subscribe to channel events (i use CHANNEL_CALLSTATE), use the
`Channel-Presence-Data` event header (and some other useful ones) to
"track" calls/channels
you can even filter events to you only get "user" related events
send updates to kamailio via xhttp or nsqd or amqp (kazoo module) based on
the events from freeswitch. kamailio will update it's presence data, send
out the NOTIFYs
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From ko at sv01.de  Fri Dec  1 22:21:59 2017
From: ko at sv01.de (Kevin Olbrich)
Date: Fri, 1 Dec 2017 23:21:59 +0100
Subject: [Freeswitch-users] Issues with T.38 on outgoing fax (sofia T38
	invite failed)
Message-ID: <CA+gLzy9gigKS=g6wgPg3dkdv7HBTA_rGHV8Q7+hQ2UVPvvY8OA@mail.gmail.com>

Hi!

Our carrier supports T.38 for outgoing faxes. I am trying to send faxes
like this:

originate
> {origination_caller_id_name='kevin',origination_caller_id_number='+49123456789',ignore_early_media=true,absolute_codec_string='PCMA,PCMU',fax_use_ecm=true,fax_enable_t38=true,fax_enable_t38_request=true}sofia/gateway/carrier_1/+4912121212
> &txfax(/opt/faxout.tif)



Log:

668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.513164 [DEBUG]
> sofia_glue.c:1295 sofia/external/+4912121212 sending invite version: 1.6.19
> -36-7a77e0b 64bit
> 668fd128-f2f5-4af5-9536-6780762c2efa Local SDP:
> 668fd128-f2f5-4af5-9536-6780762c2efa v=0
> 668fd128-f2f5-4af5-9536-6780762c2efa o=FreeSWITCH 1512130268 1512130270 IN
> IP4 123.123.123.123
> 668fd128-f2f5-4af5-9536-6780762c2efa s=FreeSWITCH
> 668fd128-f2f5-4af5-9536-6780762c2efa c=IN IP4 123.123.123.123
> 668fd128-f2f5-4af5-9536-6780762c2efa t=0 0
> 668fd128-f2f5-4af5-9536-6780762c2efa m=image 30466 udptl t38
> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxVersion:0
> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38MaxBitRate:14400
> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxFillBitRemoval
> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxRateManagement:transferredTCF
> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxBuffer:2000
> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxDatagram:400
> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxUdpEC:t38UDPRedundancy
> 668fd128-f2f5-4af5-9536-6780762c2efa
> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.533158 [DEBUG]
> sofia.c:7084 Channel sofia/external/+4912121212 entering state [calling][0]
> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120 [DEBUG]
> sofia.c:6294 sofia/external/+4912121212 T38 invite failed
> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120 [DEBUG]
> sofia.c:7077 Channel sofia/external/+4912121212 skipping state [ready][488]



Carrier confirmed that T.38 is possible... What could be the cause for
this? How can I enforce T.38 instead of falling back to g711a (hangup on
failed invite)?

Thanks.

Kind regards,
Kevin
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From rbetancor at gmail.com  Sat Dec  2 08:20:05 2017
From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=)
Date: Sat, 2 Dec 2017 08:20:05 +0000
Subject: [Freeswitch-users] Issues with T.38 on outgoing fax (sofia T38
 invite failed)
In-Reply-To: <CA+gLzy9gigKS=g6wgPg3dkdv7HBTA_rGHV8Q7+hQ2UVPvvY8OA@mail.gmail.com>
References: <CA+gLzy9gigKS=g6wgPg3dkdv7HBTA_rGHV8Q7+hQ2UVPvvY8OA@mail.gmail.com>
Message-ID: <CAKsfQBijxwQ-oj89DgPiBYOChj980A3eh1KBOxMn9BPUjHwUPA@mail.gmail.com>

Some carriers gateways doesn't like a T.38 invite on advance, usually the
order that gives better results for faxing are first invite on PCMU/PCMA
(or any other supported codec) and then re-invite to T.38.

Take a look at how gonicus gofaxip do-it (https://github.com/gonicus/gofaxip
)

On Fri, Dec 1, 2017 at 10:21 PM, Kevin Olbrich <ko at sv01.de> wrote:

> Hi!
>
> Our carrier supports T.38 for outgoing faxes. I am trying to send faxes
> like this:
>
> originate {origination_caller_id_name='kevin',origination_caller_id_
>> number='+49123456789',ignore_early_media=true,absolute_
>> codec_string='PCMA,PCMU',fax_use_ecm=true,fax_enable_t38=
>> true,fax_enable_t38_request=true}sofia/gateway/carrier_1/+4912121212
>> &txfax(/opt/faxout.tif)
>
>
>
> Log:
>
> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.513164 [DEBUG]
>> sofia_glue.c:1295 sofia/external/+4912121212 sending invite version: 1.6.19
>> -36-7a77e0b 64bit
>> 668fd128-f2f5-4af5-9536-6780762c2efa Local SDP:
>> 668fd128-f2f5-4af5-9536-6780762c2efa v=0
>> 668fd128-f2f5-4af5-9536-6780762c2efa o=FreeSWITCH 1512130268 1512130270
>> IN IP4 123.123.123.123
>> 668fd128-f2f5-4af5-9536-6780762c2efa s=FreeSWITCH
>> 668fd128-f2f5-4af5-9536-6780762c2efa c=IN IP4 123.123.123.123
>> 668fd128-f2f5-4af5-9536-6780762c2efa t=0 0
>> 668fd128-f2f5-4af5-9536-6780762c2efa m=image 30466 udptl t38
>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxVersion:0
>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38MaxBitRate:14400
>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxFillBitRemoval
>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxRateManagement:
>> transferredTCF
>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxBuffer:2000
>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxDatagram:400
>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxUdpEC:t38UDPRedundancy
>> 668fd128-f2f5-4af5-9536-6780762c2efa
>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.533158 [DEBUG]
>> sofia.c:7084 Channel sofia/external/+4912121212 entering state [calling][0]
>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120 [DEBUG]
>> sofia.c:6294 sofia/external/+4912121212 T38 invite failed
>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120 [DEBUG]
>> sofia.c:7077 Channel sofia/external/+4912121212 skipping state [ready][488]
>
>
>
> Carrier confirmed that T.38 is possible... What could be the cause for
> this? How can I enforce T.38 instead of falling back to g711a (hangup on
> failed invite)?
>
> Thanks.
>
> Kind regards,
> Kevin
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmina at connectfirst.com  Sat Dec  2 22:56:24 2017
From: gmina at connectfirst.com (Geoff Mina)
Date: Sat, 2 Dec 2017 15:56:24 -0700
Subject: [Freeswitch-users] WSS Socket Disconnect
Message-ID: <CAKzROs2zFtdxSPwj6wYys1rqP1o3c+LTKBaG27AHZ3EPKjf1xQ@mail.gmail.com>

I am having an issue where the WSS connection to Freeswitch is abruptly
disconnecting under certain scenarios.  When trying to send an INVITE to
FreeSwitch, the socket disconnects.  When accepting an INVITE from
FreeSwitch everything works fine.

I am running on CentOS 7 - 3.10.0-693.el7.x86_64.  I installed Freeswitch
1.6.19 via yum.  Here is what we are seeing from the browser side.  Same
symptom in Chrome and FireFox:

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:34:52 GMT-0700 (MST) | sip.transaction.nist | Timer J
expired for non-INVITE server transaction z9hG4bKZmZS5KUSecFee

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
acquiring local media

sip-0.7.7.js:2900:3

onaddstream is deprecated! Use peerConnection.ontrack instead.

sip-0.7.7.js:11498

navigator.mozGetUserMedia has been replaced by
navigator.mediaDevices.getUserMedia

sip-0.7.7.js:10554:8

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
acquired local media streams

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
RTCIceGatheringState changed: gathering

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
ICE candidate received: candidate:0 1 UDP 2122252543 192.168.240.12 61028
typ host

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
ICE candidate received: candidate:2 1 UDP 2122187007 172.16.2.7 54757 typ
host

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
ICE candidate received: candidate:4 1 TCP 2105524479 192.168.240.12 9 typ
host tcptype active

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
ICE candidate received: candidate:5 1 TCP 2105458943 172.16.2.7 9 typ host
tcptype active

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
ICE candidate received: candidate:0 2 UDP 2122252542 192.168.240.12 61218
typ host

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
ICE candidate received: candidate:2 2 UDP 2122187006 172.16.2.7 52542 typ
host

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
ICE candidate received: candidate:4 2 TCP 2105524478 192.168.240.12 9 typ
host tcptype active

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
ICE candidate received: candidate:5 2 TCP 2105458942 172.16.2.7 9 typ host
tcptype active

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
ICE candidate received: candidate:3 1 UDP 1685987327 52.33.247.158 25669
typ srflx raddr 172.16.2.7 rport 54757

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
ICE candidate received: candidate:3 2 UDP 1685987326 52.33.247.158 16597
typ srflx raddr 172.16.2.7 rport 52542

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.invitecontext.mediahandler |
RTCIceChecking Timeout Triggered after 5000 milliseconds

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | sending WebSocket
message:


INVITE sip:*97 at c01-aorgc-reg6.somehost.biz SIP/2.0

Via: SIP/2.0/WSS hqj7ffmo2koa.invalid;branch=z9hG4bK1975235

Max-Forwards: 70

To: <sip:*97 at c01-aorgc-reg6.somehost.biz>

From: "gmina" <sip:gmina at c01-aorgc-reg6.somehost.biz>;tag=efk4hqf1dc

Call-ID: vitn3acs5ss3dikrn14a

CSeq: 7808 INVITE

Contact: <sip:7jm943k7 at hqj7ffmo2koa.invalid;transport=ws;ob>

Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

Supported: outbound

User-Agent: SIP.js/0.7.7 BB\

Content-Type: application/sdp

Content-Length: 1646


v=0

o=mozilla...THIS_IS_SDPARTA-57.0 335274293698172607 0 IN IP4 0.0.0.0

s=-

t=0 0

a=sendrecv

a=fingerprint:sha-256
BE:35:AD:35:14:DB:CB:2B:D9:A3:D5:A7:EB:7D:08:E4:29:5E:7E:25:A6:16:9F:34:EC:37:0B:7A:C7:E0:9F:5B

a=group:BUNDLE sdparta_0

a=ice-options:trickle

a=msid-semantic:WMS *

m=audio 25669 UDP/TLS/RTP/SAVPF 109 9 0 8 101

c=IN IP4 52.33.247.158

a=candidate:0 1 UDP 2122252543 192.168.240.12 61028 typ host

a=candidate:2 1 UDP 2122187007 172.16.2.7 54757 typ host

a=candidate:4 1 TCP 2105524479 192.168.240.12 9 typ host tcptype active

a=candidate:5 1 TCP 2105458943 172.16.2.7 9 typ host tcptype active

a=candidate:0 2 UDP 2122252542 192.168.240.12 61218 typ host

a=candidate:2 2 UDP 2122187006 172.16.2.7 52542 typ host

a=candidate:4 2 TCP 2105524478 192.168.240.12 9 typ host tcptype active

a=candidate:5 2 TCP 2105458942 172.16.2.7 9 typ host tcptype active

a=candidate:3 1 UDP 1685987327 52.33.247.158 25669 typ srflx raddr
172.16.2.7 rport 54757

a=candidate:3 2 UDP 1685987326 52.33.247.158 16597 typ srflx raddr
172.16.2.7 rport 52542

a=sendrecv

a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:ssrc-audio-level

a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1

a=fmtp:101 0-15

a=ice-pwd:d45de180fdae3dc61f0e26238541045c

a=ice-ufrag:50681d44

a=mid:sdparta_0

a=msid:{23ed6231-4501-1944-8577-ab85c6c721cb}
{c98743c7-76d0-5c44-89a1-f702f7d8168d}

a=rtcp:16597 IN IP4 52.33.247.158

a=rtcp-mux

a=rtpmap:109 opus/48000/2

a=rtpmap:9 G722/8000/1

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=setup:actpass

a=ssrc:3742140108 cname:{76ae6d79-b818-1e40-a9bc-4734a437b47a}



sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | WebSocket
disconnected (code: 1006)

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | WebSocket abrupt
disconnection

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.ua | connection state set to 1

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transaction.ict | transport
error occurred, deleting INVITE client transaction z9hG4bK1975235

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | Connection to
WebSocket wss://c01-aorgc-reg6.somehost.biz:8089/freeswitch severed,
attempting first reconnect

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | connecting to
WebSocket wss://c01-aorgc-reg6.somehost.biz:8089/freeswitch

sip-0.7.7.js:2900:3

Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | WebSocket wss://
c01-aorgc-reg6.somehost.biz:8089/freeswitch connected


*And here is what we are seeing from the Freeswitch side.*


tport.c:2773 tport_wakeup() tport_wakeup(0x7f4e580fc4d0): events IN

tport.c:2864 tport_recv_event() tport_recv_event(0x7f4e580fc4d0)

tport.c:2157 tport_shutdown0() tport_shutdown0(0x7f4e580fc4d0, 2)

tport.c:2090 tport_close() tport_close(0x7f4e580fc4d0): wss/
52.33.247.158:25656/sips

nua_registrar.c:200 registrar_tport_error() tport error 0: Success

tport.c:4222 tport_release() tport_release(0x7f4e580fc4d0): (nil) by
0x7f4e581361b0 with (nil)

tport.c:2263 tport_set_secondary_timer() tport(0x7f4e580fc4d0): set timer
at 0 ms because zap

*nua_stack.c:271 nua_stack_event() nua(0x7f4e581361b0): event i_media_error
500 Transport error detected*

nua_stack.c:359 nua_application_event() nua: nua_application_event: entering

tport.c:2263 tport_set_secondary_timer() tport(0x7f4e580fc4d0): set timer
at 0 ms because zap

tport_type_ws.c:521 tport_ws_deinit_secondary() 0x7f4e580fc4d0 destroy wss
transport 0x7f4e580fc6c0.

nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering

nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering

nua_stack.c:529 nua_signal() nua(0x7f4e581361b0): sent signal r_destroy

nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering

nua_stack.c:569 nua_stack_signal() nua(0x7f4e581361b0): recv signal
r_destroy

nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f4e581361b0): removing
registrar usage

nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))

soa.c:356 soa_destroy() soa_destroy(static::0x7f4e5803dea0) called

tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7f4e58004f50): events IN

tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7f4e58004f50):
new secondary tport 0x7f4e580fc4d0

tport.c:2292 tport_set_secondary_timer() tport(0x7f4e580fc4d0): set timer
at 4998 ms because keepalive

tport.c:2640 tport_accept() tport_accept(0x7f4e580fc4d0): new connection
from wss/52.33.247.158:54385/sips

tport.c:2773 tport_wakeup() tport_wakeup(0x7f4e580fc4d0): events IN

tport.c:2864 tport_recv_event() tport_recv_event(0x7f4e580fc4d0)

tport.c:2296 tport_set_secondary_timer() tport(0x7f4e580fc4d0): reset timer

tport.c:2773 tport_wakeup() tport_wakeup(0x7f4e580fc4d0): events IN


Any help in narrowing down the cause would be greatly appreciated.  I am
thinking maybe it's a max frame size on the websocket or something... but I
couldn't find anything that would indicate there was a setting to tweak.

Thanks,
Geoff
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From gregor at infomedia.si  Sun Dec  3 00:04:43 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 3 Dec 2017 01:04:43 +0100
Subject: [Freeswitch-users] Sequential enterprise originate
In-Reply-To: <010001600895b336-ee3d710f-5422-4603-a28c-bbb1a710aab7-000000@email.amazonses.com>
References: <CAC7ttFFqnSoveXR9kA_2s7jneWvD3gax8rdkMYDD5F9Qe3XRrw@mail.gmail.com>
 <B1F0DCB2-F124-45EF-9A82-5B0E4FBCC9EB@jerris.com>
 <CAC7ttFE7Qi7BmUhWh9aTaC5oEOM8Gmcw9y2rfFsgL=wKU4xz6Q@mail.gmail.com>
 <CAC7ttFH7TWKWkUuEB1Kn=U+=Wh2w3LLPXaJJubHrJB6krAFViQ@mail.gmail.com>
 <CAMtXxQkV4UWqoA+pPj3ECB4uUpy93Ab+ezTcJzX_nNMovVQQrg@mail.gmail.com>
 <CAC7ttFE1-rvhzSXmaAnEWE50LE_3Qdd+pGgVMeyaZZPRu_SvPw@mail.gmail.com>
 <0100016003992782-67e34e31-28c2-42de-9430-f983dafa3cf5-000000@email.amazonses.com>
 <CAC7ttFHvr21f+yPSaY3YJykJuZm_7=1rd+ZUmBivWbmuWsiYZA@mail.gmail.com>
 <0100016003a5789e-00b91f67-0979-4884-8d8f-f335ebd6aa8b-000000@email.amazonses.com>
 <CAC7ttFGiu7UO14brC5gTGz=ceczYgRpPRZdRZ-DdpWXNDKHVDQ@mail.gmail.com>
 <0100016003b6374b-8be01c50-aca5-4dc8-b02c-0ce2f204b898-000000@email.amazonses.com>
 <CAC7ttFFRYH0xAhiGp+5MssW-wkFWe4aOTS6aX2KQMQ9Dk18ynw@mail.gmail.com>
 <CAC7ttFE6Z1fjxGui7Q10bHCHxi6KSGhpd3FeHqNc-pDv3uL7NQ@mail.gmail.com>
 <010001600895b336-ee3d710f-5422-4603-a28c-bbb1a710aab7-000000@email.amazonses.com>
Message-ID: <CAC7ttFG=zcDLZgySpmwxqio2w56fF6kydCh6vv4J9y6TyHGqSQ@mail.gmail.com>

Avi, I read the docs several times, but it wasn't working as expected. Now
I figure it out.

Following sentences would be usefull in wiki:
"If log-b-leg is set to false, then b_only doesn't override logging of b
leg."

Now it works as exptected.

Thank you for your help.

2017-11-29 17:21 GMT+01:00 Avi Marcus <avi at avimarcus.net>:

> Please read the docs:
>
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#
> ChannelVariables-process_cdr
>
> Can be undefined or set to "false", "true", "a_only", "b_only"
>> false - indicates to not process the record.
>> true - or undefined indicates the default behavior which is to process
>> all CDR records.
>> a_only - indicates to only process CDR records on the inbound leg of a
>> call.
>> b_only - indicates to only process CDR records on the outbound leg of a
>> call.
>> Usage:
>> <action application="set" data="process_cdr=a_only"/>
>
>
> If you do a_only or false, that is supposed to override also logging the B
> leg, even if you set it with mod_xml_cdr.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From ko at sv01.de  Sat Dec  2 12:41:54 2017
From: ko at sv01.de (Kevin Olbrich)
Date: Sat, 2 Dec 2017 13:41:54 +0100
Subject: [Freeswitch-users] Issues with T.38 on outgoing fax (sofia T38
 invite failed)
In-Reply-To: <CAKsfQBijxwQ-oj89DgPiBYOChj980A3eh1KBOxMn9BPUjHwUPA@mail.gmail.com>
References: <CA+gLzy9gigKS=g6wgPg3dkdv7HBTA_rGHV8Q7+hQ2UVPvvY8OA@mail.gmail.com>
 <CAKsfQBijxwQ-oj89DgPiBYOChj980A3eh1KBOxMn9BPUjHwUPA@mail.gmail.com>
Message-ID: <CA+gLzy8ExwO8_zUewDbLCVOnKaigOHQOViMdUsM0kaGY4F5reA@mail.gmail.com>

Hi,

thanks for the link but I was unable to find information on how to
configure my installation of FreeSwitch 1.6 to make things work.

I think I need to explain a little bit more on what I am planning to do:
FreeSwitch will be used as a SBC to the carrier. It's firewall is limited
to the carriers network on the first interface.
The second interface is a local subnet where an Asterisk 13 server (PBX) is
located, connecting the phones and ATAs (last with T.38 support).
For asterisk it is only necessary so set (FAXOPT(gateway)=yes) to have it
detect CNG on audio and query both legs for T.38 (or receive a request in
advance).
It then decides if it goes to passthru mode or transcode.

As far as I understand the FreeSwitch docs, there are these options
(located in dialplan default before bridge):

> <action application="set" data="fax_enable_t38=true"/> <!-- Enable t.38
> for this call -->
> <action application="set" data="fax_enable_t38_request=true"/> <!-- Enable
> t38_gateway to send a t.38 reinvite when a fax tone is detected. If using
> t38_gateway peer then you need to export this variable instead of set -->
> <action application="set" data="execute_on_answer=t38_gateway self"/>
> <!--Execute t38_gateway on answer. self or peer. self: send a reinvite back
> to the a-leg. peer reinvite forward to the b-leg -->


<action application="export" data="fax_enable_t38=true"/> <!-- Required as
> of v1.6 -->
> <action application="export" data="t38_passthru=true"/>


*What I would like to accomplish is getting as much as T.38 I can get.* The
ATA should always be able to negotiate T.38 with Asterisk (which seems to
work atm), FreeSwitch should then negotiate T.38 if Asterisk or the carrier
asks for it (no CNG detection).
I am not sure if I can just enable t38_gateway and passthru, this case is
not listed on confluence or the old wiki. Even if I do, it does not work.

FreeSwitch might need to transcode T.38 to audio for our backup carrier who
does not support T.38.

Kind regards,
Kevin

2017-12-02 9:20 GMT+01:00 Raúl Alexis Betancor Santana <rbetancor at gmail.com>
:

> Some carriers gateways doesn't like a T.38 invite on advance, usually the
> order that gives better results for faxing are first invite on PCMU/PCMA
> (or any other supported codec) and then re-invite to T.38.
>
> Take a look at how gonicus gofaxip do-it (https://github.com/gonicus/
> gofaxip)
>
> On Fri, Dec 1, 2017 at 10:21 PM, Kevin Olbrich <ko at sv01.de> wrote:
>
>> Hi!
>>
>> Our carrier supports T.38 for outgoing faxes. I am trying to send faxes
>> like this:
>>
>> originate {origination_caller_id_name='kevin',origination_caller_id_nu
>>> mber='+49123456789',ignore_early_media=true,absolute_codec_
>>> string='PCMA,PCMU',fax_use_ecm=true,fax_enable_t38=true,
>>> fax_enable_t38_request=true}sofia/gateway/carrier_1/+4912121212
>>> &txfax(/opt/faxout.tif)
>>
>>
>>
>> Log:
>>
>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.513164 [DEBUG]
>>> sofia_glue.c:1295 sofia/external/+4912121212 sending invite version: 1.6.19
>>> -36-7a77e0b 64bit
>>> 668fd128-f2f5-4af5-9536-6780762c2efa Local SDP:
>>> 668fd128-f2f5-4af5-9536-6780762c2efa v=0
>>> 668fd128-f2f5-4af5-9536-6780762c2efa o=FreeSWITCH 1512130268 1512130270
>>> IN IP4 123.123.123.123
>>> 668fd128-f2f5-4af5-9536-6780762c2efa s=FreeSWITCH
>>> 668fd128-f2f5-4af5-9536-6780762c2efa c=IN IP4 123.123.123.123
>>> 668fd128-f2f5-4af5-9536-6780762c2efa t=0 0
>>> 668fd128-f2f5-4af5-9536-6780762c2efa m=image 30466 udptl t38
>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxVersion:0
>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38MaxBitRate:14400
>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxFillBitRemoval
>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxRateManagement:transfe
>>> rredTCF
>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxBuffer:2000
>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxDatagram:400
>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxUdpEC:t38UDPRedundancy
>>> 668fd128-f2f5-4af5-9536-6780762c2efa
>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.533158 [DEBUG]
>>> sofia.c:7084 Channel sofia/external/+4912121212 entering state [calling][0]
>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120 [DEBUG]
>>> sofia.c:6294 sofia/external/+4912121212 T38 invite failed
>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120 [DEBUG]
>>> sofia.c:7077 Channel sofia/external/+4912121212 skipping state [ready][488]
>>
>>
>>
>> Carrier confirmed that T.38 is possible... What could be the cause for
>> this? How can I enforce T.38 instead of falling back to g711a (hangup on
>> failed invite)?
>>
>> Thanks.
>>
>> Kind regards,
>> Kevin
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From abaci64 at gmail.com  Mon Dec  4 02:08:35 2017
From: abaci64 at gmail.com (Abaci B)
Date: Mon, 4 Dec 2017 02:08:35 +0000
Subject: [Freeswitch-users] mod_curl question
Message-ID: <CAFmZFiMnvUBuM4bJSxYLQrh4Jks690iuoyb+3taVH7ssSxSEww@mail.gmail.com>

I was just wondering if anyone had tried using the append_headers in
FreeSWITCH on a windows machine, I have  curl command that works on Linux
just fine but when I paste the same thing in a Windows FreeSWITCH the
serverver doesn't get the added headers. The version I test is 1.6.18 (the
last version with official binaries).
Any help or feedback would be appreciated.
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From achinthau at gmail.com  Mon Dec  4 10:01:12 2017
From: achinthau at gmail.com (Achintha)
Date: Mon, 4 Dec 2017 15:31:12 +0530
Subject: [Freeswitch-users] Call Recordings Not Working
Message-ID: <CA+JwzxtfEr7wy+NLEAYaFExCgUJ1kEaf2iq7EJNXiHZOzro4Ow@mail.gmail.com>

Dear All,

We configured freeswitch (1.6.19-36-7a77e0b~64bit) on debian 8.9. it
handled around 7000 calls per day.multiple sip trunks were terminated to
the virtual Ip-address given by keepalivd. we are using G711 and  opus
codecs.
We use the following modules.
    mod_xml_curl : Dynamic Dialplans (from rest Service)
    mod_json_cdr : for CDR  (from rest Service)
    and one module developed by our self (mod_ards) to route calls to

Server spec are 8-cors and 8GB ram.
Memory usage details:
   used : 7.7G
   cached :7.2G
   free :156M

On peak hours system handles more than 65 concurrent calls and at that time
the freeswitch service used CPU over than 110%.
Any way i reboot the server. then  calls were not recorded for more than
1.30 mins then it worked automatically.
Voice Record paths given by Dynamic dial-plan (mod_xml_curl) for Outbound
calls and mod_ards to incoming calls.

How can i fix this issue.

-- 
Best Regards..
Achintha
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From gmaruzz at gmail.com  Mon Dec  4 10:25:03 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 4 Dec 2017 11:25:03 +0100
Subject: [Freeswitch-users] Call Recordings Not Working
In-Reply-To: <CA+JwzxtfEr7wy+NLEAYaFExCgUJ1kEaf2iq7EJNXiHZOzro4Ow@mail.gmail.com>
References: <CA+JwzxtfEr7wy+NLEAYaFExCgUJ1kEaf2iq7EJNXiHZOzro4Ow@mail.gmail.com>
Message-ID: <CALXCt0rU0K_5gbvMThaWLj6B0P4=qj8WVp=Wb327e=0hgvbEGA@mail.gmail.com>

You can ask for commercial (for pay) support by writing to
consulting at freeswitch.org


Sent via mobile, please forgive typos and brevity.

cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT

On Dec 4, 2017 11:02 AM, "Achintha" <achinthau at gmail.com> wrote:

> Dear All,
>
> We configured freeswitch (1.6.19-36-7a77e0b~64bit) on debian 8.9. it
> handled around 7000 calls per day.multiple sip trunks were terminated to
> the virtual Ip-address given by keepalivd. we are using G711 and  opus
> codecs.
> We use the following modules.
>     mod_xml_curl : Dynamic Dialplans (from rest Service)
>     mod_json_cdr : for CDR  (from rest Service)
>     and one module developed by our self (mod_ards) to route calls to
>
> Server spec are 8-cors and 8GB ram.
> Memory usage details:
>    used : 7.7G
>    cached :7.2G
>    free :156M
>
> On peak hours system handles more than 65 concurrent calls and at that
> time the freeswitch service used CPU over than 110%.
> Any way i reboot the server. then  calls were not recorded for more than
> 1.30 mins then it worked automatically.
> Voice Record paths given by Dynamic dial-plan (mod_xml_curl) for Outbound
> calls and mod_ards to incoming calls.
>
> How can i fix this issue.
>
> --
> Best Regards..
> Achintha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ko at sv01.de  Sat Dec  2 15:36:01 2017
From: ko at sv01.de (Kevin Olbrich)
Date: Sat, 2 Dec 2017 16:36:01 +0100
Subject: [Freeswitch-users] Issues with T.38 on outgoing fax (sofia T38
 invite failed)
In-Reply-To: <CA+gLzy8ExwO8_zUewDbLCVOnKaigOHQOViMdUsM0kaGY4F5reA@mail.gmail.com>
References: <CA+gLzy9gigKS=g6wgPg3dkdv7HBTA_rGHV8Q7+hQ2UVPvvY8OA@mail.gmail.com>
 <CAKsfQBijxwQ-oj89DgPiBYOChj980A3eh1KBOxMn9BPUjHwUPA@mail.gmail.com>
 <CA+gLzy8ExwO8_zUewDbLCVOnKaigOHQOViMdUsM0kaGY4F5reA@mail.gmail.com>
Message-ID: <CA+gLzy8w3jNvZVP_fM6wJ0exbF3xs9fnWxfK4CsvUwNaA1wyng@mail.gmail.com>

I need to note that all components are SIP based, no other tech involved.

Kevin

2017-12-02 13:41 GMT+01:00 Kevin Olbrich <ko at sv01.de>:

> Hi,
>
> thanks for the link but I was unable to find information on how to
> configure my installation of FreeSwitch 1.6 to make things work.
>
> I think I need to explain a little bit more on what I am planning to do:
> FreeSwitch will be used as a SBC to the carrier. It's firewall is limited
> to the carriers network on the first interface.
> The second interface is a local subnet where an Asterisk 13 server (PBX)
> is located, connecting the phones and ATAs (last with T.38 support).
> For asterisk it is only necessary so set (FAXOPT(gateway)=yes) to have it
> detect CNG on audio and query both legs for T.38 (or receive a request in
> advance).
> It then decides if it goes to passthru mode or transcode.
>
> As far as I understand the FreeSwitch docs, there are these options
> (located in dialplan default before bridge):
>
>> <action application="set" data="fax_enable_t38=true"/> <!-- Enable t.38
>> for this call -->
>> <action application="set" data="fax_enable_t38_request=true"/> <!--
>> Enable t38_gateway to send a t.38 reinvite when a fax tone is detected. If
>> using t38_gateway peer then you need to export this variable instead of set
>> -->
>> <action application="set" data="execute_on_answer=t38_gateway self"/>
>> <!--Execute t38_gateway on answer. self or peer. self: send a reinvite back
>> to the a-leg. peer reinvite forward to the b-leg -->
>
>
> <action application="export" data="fax_enable_t38=true"/> <!-- Required as
>> of v1.6 -->
>> <action application="export" data="t38_passthru=true"/>
>
>
> *What I would like to accomplish is getting as much as T.38 I can get.*
> The ATA should always be able to negotiate T.38 with Asterisk (which seems
> to work atm), FreeSwitch should then negotiate T.38 if Asterisk or the
> carrier asks for it (no CNG detection).
> I am not sure if I can just enable t38_gateway and passthru, this case is
> not listed on confluence or the old wiki. Even if I do, it does not work.
>
> FreeSwitch might need to transcode T.38 to audio for our backup carrier
> who does not support T.38.
>
> Kind regards,
> Kevin
>
> 2017-12-02 9:20 GMT+01:00 Raúl Alexis Betancor Santana <
> rbetancor at gmail.com>:
>
>> Some carriers gateways doesn't like a T.38 invite on advance, usually the
>> order that gives better results for faxing are first invite on PCMU/PCMA
>> (or any other supported codec) and then re-invite to T.38.
>>
>> Take a look at how gonicus gofaxip do-it (https://github.com/gonicus/go
>> faxip)
>>
>> On Fri, Dec 1, 2017 at 10:21 PM, Kevin Olbrich <ko at sv01.de> wrote:
>>
>>> Hi!
>>>
>>> Our carrier supports T.38 for outgoing faxes. I am trying to send faxes
>>> like this:
>>>
>>> originate {origination_caller_id_name='kevin',origination_caller_id_nu
>>>> mber='+49123456789',ignore_early_media=true,absolute_codec_s
>>>> tring='PCMA,PCMU',fax_use_ecm=true,fax_enable_t38=true,fax_
>>>> enable_t38_request=true}sofia/gateway/carrier_1/+4912121212
>>>> &txfax(/opt/faxout.tif)
>>>
>>>
>>>
>>> Log:
>>>
>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.513164 [DEBUG]
>>>> sofia_glue.c:1295 sofia/external/+4912121212 sending invite version: 1.6.19
>>>> -36-7a77e0b 64bit
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa Local SDP:
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa v=0
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa o=FreeSWITCH 1512130268
>>>> 1512130270 IN IP4 123.123.123.123
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa s=FreeSWITCH
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa c=IN IP4 123.123.123.123
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa t=0 0
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa m=image 30466 udptl t38
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxVersion:0
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38MaxBitRate:14400
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxFillBitRemoval
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxRateManagement:transfe
>>>> rredTCF
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxBuffer:2000
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxDatagram:400
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxUdpEC:t38UDPRedundancy
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.533158
>>>> [DEBUG] sofia.c:7084 Channel sofia/external/+4912121212 entering state
>>>> [calling][0]
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120
>>>> [DEBUG] sofia.c:6294 sofia/external/+4912121212 T38 invite failed
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120
>>>> [DEBUG] sofia.c:7077 Channel sofia/external/+4912121212 skipping state
>>>> [ready][488]
>>>
>>>
>>>
>>> Carrier confirmed that T.38 is possible... What could be the cause for
>>> this? How can I enforce T.38 instead of falling back to g711a (hangup on
>>> failed invite)?
>>>
>>> Thanks.
>>>
>>> Kind regards,
>>> Kevin
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From ko at sv01.de  Sun Dec  3 23:00:49 2017
From: ko at sv01.de (Kevin Olbrich)
Date: Mon, 4 Dec 2017 00:00:49 +0100
Subject: [Freeswitch-users] Issues with T.38 on outgoing fax (sofia T38
 invite failed)
In-Reply-To: <CA+gLzy8w3jNvZVP_fM6wJ0exbF3xs9fnWxfK4CsvUwNaA1wyng@mail.gmail.com>
References: <CA+gLzy9gigKS=g6wgPg3dkdv7HBTA_rGHV8Q7+hQ2UVPvvY8OA@mail.gmail.com>
 <CAKsfQBijxwQ-oj89DgPiBYOChj980A3eh1KBOxMn9BPUjHwUPA@mail.gmail.com>
 <CA+gLzy8ExwO8_zUewDbLCVOnKaigOHQOViMdUsM0kaGY4F5reA@mail.gmail.com>
 <CA+gLzy8w3jNvZVP_fM6wJ0exbF3xs9fnWxfK4CsvUwNaA1wyng@mail.gmail.com>
Message-ID: <CA+gLzy8u0s32YwVhRXEe_tLHm68x7Dm88NZcnYMAncAUhiHepg@mail.gmail.com>

I just had a talk about this with our carrier and the problems seems on
their side.
This will not be fixed anytime soon, so they said it would be best, if we
use T.38 inside and t38-to-g711 when connecting to them.

I tried to set this in the incoming dialplan:


>
> <include>
>   <extension name="public_did">
>     <condition field="destination_number" expression="^\+(\d+)$|\d+$">
>       <action application="set" data="domain_name=$${domain}"/>
>       <action application="t38_gateway" data="peer"/>
>       <action application="bridge" data="sofia/internal/${
> destination_number}@192.168.31.11"/>
>     </condition>
>   </extension>
> </include>
>


IP 192.168.31.11 is an asterisk machine, connecting the T.38 ATAs at
various locations.


> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.128433 [DEBUG]
> switch_ivr_bridge.c:1614 (sofia/internal/%2B491122334455 at 192.168.31.11)
> State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.128433 [DEBUG]
> switch_core_state_machine.c:584 (sofia/internal/%
> 2B491122334455 at 192.168.31.11) Running State Change CS_EXCHANGE_MEDIA (Cur
> 2 Tot 2)
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.128433 [DEBUG]
> switch_core_state_machine.c:653 (sofia/internal/%
> 2B491122334455 at 192.168.31.11) State EXCHANGE_MEDIA
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.128433 [DEBUG]
> mod_sofia.c:631 SOFIA EXCHANGE_MEDIA
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.148489 [DEBUG]
> sofia.c:7084 Channel sofia/internal/%2B491122334455 at 192.168.31.11
> entering state [received][100]
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.148489 [DEBUG]
> sofia.c:7094 Remote SDP:
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 v=0
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 o=- 1512309774 1512309778 IN IP4
> 192.168.31.11
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 s=Asterisk
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 c=IN IP4 192.168.31.11
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 t=0 0
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 m=image 4663 udptl t38
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxVersion:1
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38MaxBitRate:14400
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxTranscodingMMR
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxTranscodingJBIG
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxRateManagement:transferredTCF
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxMaxDatagram:784
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxUdpEC:t38UDPRedundancy
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.148489 [DEBUG]
> switch_core_media.c:4028 sofia/internal/%2B491122334455 at 192.168.31.11 T38
> REFUSE on request
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.148489 [DEBUG]
> sofia.c:8007 Reinvite resulted in codec negotiation failure.
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.168441 [DEBUG]
> sofia.c:7077 Channel sofia/internal/%2B491122334455 at 192.168.31.11
> skipping state [ready][488]
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.168441 [DEBUG]
> sofia.c:7084 Channel sofia/internal/%2B491122334455 at 192.168.31.11
> entering state [received][100]
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.168441 [DEBUG]
> sofia.c:7094 Remote SDP:
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 v=0
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 o=- 1512309774 1512309779 IN IP4
> 192.168.31.11
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 s=Asterisk
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 c=IN IP4 192.168.31.11
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 t=0 0
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 m=image 4663 udptl t38
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxVersion:1
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38MaxBitRate:14400
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxTranscodingMMR
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxTranscodingJBIG
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxRateManagement:transferredTCF
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxMaxDatagram:784
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxUdpEC:t38UDPRedundancy
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.168441 [DEBUG]
> switch_core_media.c:4028 sofia/internal/%2B491122334455 at 192.168.31.11 T38
> REFUSE on request
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.168441 [DEBUG]
> sofia.c:8007 Reinvite resulted in codec negotiation failure.
> ac858869-5320-48fb-b100-1a4a9c3d0851 2017-12-03 23:43:49.188433 [DEBUG]
> switch_rtp.c:7271 Correct audio ip/port confirmed.
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.188433 [DEBUG]
> sofia.c:7077 Channel sofia/internal/%2B491122334455 at 192.168.31.11
> skipping state [ready][488]
> ac858869-5320-48fb-b100-1a4a9c3d0851 2017-12-03 23:43:49.208437 [DEBUG]
> sofia.c:7084 Channel sofia/external/anonymous at 87.234.1.183 entering state
> [ready][200]
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.208437 [DEBUG]
> switch_rtp.c:7271 Correct audio ip/port confirmed.
> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:44:33.188445 [NOTICE]
> sofia.c:1012 Hangup sofia/internal/%2B491122334455 at 192.168.31.11
> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>


What I need is:

FAX ATA (T.38) -> Asterisk (T.38) -> FreeSWITCH (T.38 to audio GW, refuse
T.38 on A-leg) -> Carrier (audio)

Carrier (audio) -> FreeSWITCH (audio to T.38, refuse T.38 on B-leg) ->
Asterisk (T.38) -> FAX ATA (T.38)

Asterisk is sending Re-INVITES as seen in the log above. Seems like I miss
some settings in my dialplan?

Kevin

2017-12-02 16:36 GMT+01:00 Kevin Olbrich <ko at sv01.de>:

> I need to note that all components are SIP based, no other tech involved.
>
> Kevin
>
> 2017-12-02 13:41 GMT+01:00 Kevin Olbrich <ko at sv01.de>:
>
>> Hi,
>>
>> thanks for the link but I was unable to find information on how to
>> configure my installation of FreeSwitch 1.6 to make things work.
>>
>> I think I need to explain a little bit more on what I am planning to do:
>> FreeSwitch will be used as a SBC to the carrier. It's firewall is limited
>> to the carriers network on the first interface.
>> The second interface is a local subnet where an Asterisk 13 server (PBX)
>> is located, connecting the phones and ATAs (last with T.38 support).
>> For asterisk it is only necessary so set (FAXOPT(gateway)=yes) to have it
>> detect CNG on audio and query both legs for T.38 (or receive a request in
>> advance).
>> It then decides if it goes to passthru mode or transcode.
>>
>> As far as I understand the FreeSwitch docs, there are these options
>> (located in dialplan default before bridge):
>>
>>> <action application="set" data="fax_enable_t38=true"/> <!-- Enable t.38
>>> for this call -->
>>> <action application="set" data="fax_enable_t38_request=true"/> <!--
>>> Enable t38_gateway to send a t.38 reinvite when a fax tone is detected. If
>>> using t38_gateway peer then you need to export this variable instead of set
>>> -->
>>> <action application="set" data="execute_on_answer=t38_gateway self"/>
>>> <!--Execute t38_gateway on answer. self or peer. self: send a reinvite back
>>> to the a-leg. peer reinvite forward to the b-leg -->
>>
>>
>> <action application="export" data="fax_enable_t38=true"/> <!-- Required
>>> as of v1.6 -->
>>> <action application="export" data="t38_passthru=true"/>
>>
>>
>> *What I would like to accomplish is getting as much as T.38 I can get.*
>> The ATA should always be able to negotiate T.38 with Asterisk (which seems
>> to work atm), FreeSwitch should then negotiate T.38 if Asterisk or the
>> carrier asks for it (no CNG detection).
>> I am not sure if I can just enable t38_gateway and passthru, this case is
>> not listed on confluence or the old wiki. Even if I do, it does not work.
>>
>> FreeSwitch might need to transcode T.38 to audio for our backup carrier
>> who does not support T.38.
>>
>> Kind regards,
>> Kevin
>>
>> 2017-12-02 9:20 GMT+01:00 Raúl Alexis Betancor Santana <
>> rbetancor at gmail.com>:
>>
>>> Some carriers gateways doesn't like a T.38 invite on advance, usually
>>> the order that gives better results for faxing are first invite on
>>> PCMU/PCMA (or any other supported codec) and then re-invite to T.38.
>>>
>>> Take a look at how gonicus gofaxip do-it (https://github.com/gonicus/go
>>> faxip)
>>>
>>> On Fri, Dec 1, 2017 at 10:21 PM, Kevin Olbrich <ko at sv01.de> wrote:
>>>
>>>> Hi!
>>>>
>>>> Our carrier supports T.38 for outgoing faxes. I am trying to send faxes
>>>> like this:
>>>>
>>>> originate {origination_caller_id_name='kevin',origination_caller_id_nu
>>>>> mber='+49123456789',ignore_early_media=true,absolute_codec_s
>>>>> tring='PCMA,PCMU',fax_use_ecm=true,fax_enable_t38=true,fax_e
>>>>> nable_t38_request=true}sofia/gateway/carrier_1/+4912121212
>>>>> &txfax(/opt/faxout.tif)
>>>>
>>>>
>>>>
>>>> Log:
>>>>
>>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.513164
>>>>> [DEBUG] sofia_glue.c:1295 sofia/external/+4912121212 sending invite
>>>>> version: 1.6.19 -36-7a77e0b 64bit
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa Local SDP:
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa v=0
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa o=FreeSWITCH 1512130268
>>>>> 1512130270 IN IP4 123.123.123.123
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa s=FreeSWITCH
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa c=IN IP4 123.123.123.123
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa t=0 0
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa m=image 30466 udptl t38
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxVersion:0
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38MaxBitRate:14400
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxFillBitRemoval
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxRateManagement:transfe
>>>>> rredTCF
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxBuffer:2000
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxDatagram:400
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxUdpEC:t38UDPRedundancy
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.533158
>>>>> [DEBUG] sofia.c:7084 Channel sofia/external/+4912121212 entering state
>>>>> [calling][0]
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120
>>>>> [DEBUG] sofia.c:6294 sofia/external/+4912121212 T38 invite failed
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120
>>>>> [DEBUG] sofia.c:7077 Channel sofia/external/+4912121212 skipping state
>>>>> [ready][488]
>>>>
>>>>
>>>>
>>>> Carrier confirmed that T.38 is possible... What could be the cause for
>>>> this? How can I enforce T.38 instead of falling back to g711a (hangup on
>>>> failed invite)?
>>>>
>>>> Thanks.
>>>>
>>>> Kind regards,
>>>> Kevin
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From gmaruzz at gmail.com  Mon Dec  4 10:35:04 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 4 Dec 2017 11:35:04 +0100
Subject: [Freeswitch-users] Issues with T.38 on outgoing fax (sofia T38
 invite failed)
In-Reply-To: <CA+gLzy8u0s32YwVhRXEe_tLHm68x7Dm88NZcnYMAncAUhiHepg@mail.gmail.com>
References: <CA+gLzy9gigKS=g6wgPg3dkdv7HBTA_rGHV8Q7+hQ2UVPvvY8OA@mail.gmail.com>
 <CAKsfQBijxwQ-oj89DgPiBYOChj980A3eh1KBOxMn9BPUjHwUPA@mail.gmail.com>
 <CA+gLzy8ExwO8_zUewDbLCVOnKaigOHQOViMdUsM0kaGY4F5reA@mail.gmail.com>
 <CA+gLzy8w3jNvZVP_fM6wJ0exbF3xs9fnWxfK4CsvUwNaA1wyng@mail.gmail.com>
 <CA+gLzy8u0s32YwVhRXEe_tLHm68x7Dm88NZcnYMAncAUhiHepg@mail.gmail.com>
Message-ID: <CALXCt0oc16knt6pm087nvqqy6en6hmMLKSKaK9QU+B-EwCtAWg@mail.gmail.com>

Try with ecm false, and early_media false

Sent via mobile, please forgive typos and brevity.

cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT

On Dec 4, 2017 11:30 AM, "Kevin Olbrich" <ko at sv01.de> wrote:

> I just had a talk about this with our carrier and the problems seems on
> their side.
> This will not be fixed anytime soon, so they said it would be best, if we
> use T.38 inside and t38-to-g711 when connecting to them.
>
> I tried to set this in the incoming dialplan:
>
>
>>
>> <include>
>>   <extension name="public_did">
>>     <condition field="destination_number" expression="^\+(\d+)$|\d+$">
>>       <action application="set" data="domain_name=$${domain}"/>
>>       <action application="t38_gateway" data="peer"/>
>>       <action application="bridge" data="sofia/internal/${destina
>> tion_number}@192.168.31.11"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>
>
> IP 192.168.31.11 is an asterisk machine, connecting the T.38 ATAs at
> various locations.
>
>
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.128433 [DEBUG]
>> switch_ivr_bridge.c:1614 (sofia/internal/%2B491122334455 at 192.168.31.11)
>> State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.128433 [DEBUG]
>> switch_core_state_machine.c:584 (sofia/internal/%2B49112233445
>> 5 at 192.168.31.11) Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 2)
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.128433 [DEBUG]
>> switch_core_state_machine.c:653 (sofia/internal/%2B49112233445
>> 5 at 192.168.31.11) State EXCHANGE_MEDIA
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.128433 [DEBUG]
>> mod_sofia.c:631 SOFIA EXCHANGE_MEDIA
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.148489 [DEBUG]
>> sofia.c:7084 Channel sofia/internal/%2B491122334455 at 192.168.31.11
>> entering state [received][100]
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.148489 [DEBUG]
>> sofia.c:7094 Remote SDP:
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 v=0
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 o=- 1512309774 1512309778 IN IP4
>> 192.168.31.11
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 s=Asterisk
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 c=IN IP4 192.168.31.11
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 t=0 0
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 m=image 4663 udptl t38
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxVersion:1
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38MaxBitRate:14400
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxTranscodingMMR
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxTranscodingJBIG
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxRateManagement:
>> transferredTCF
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxMaxDatagram:784
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxUdpEC:t38UDPRedundancy
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.148489 [DEBUG]
>> switch_core_media.c:4028 sofia/internal/%2B491122334455 at 192.168.31.11
>> T38 REFUSE on request
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.148489 [DEBUG]
>> sofia.c:8007 Reinvite resulted in codec negotiation failure.
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.168441 [DEBUG]
>> sofia.c:7077 Channel sofia/internal/%2B491122334455 at 192.168.31.11
>> skipping state [ready][488]
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.168441 [DEBUG]
>> sofia.c:7084 Channel sofia/internal/%2B491122334455 at 192.168.31.11
>> entering state [received][100]
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.168441 [DEBUG]
>> sofia.c:7094 Remote SDP:
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 v=0
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 o=- 1512309774 1512309779 IN IP4
>> 192.168.31.11
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 s=Asterisk
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 c=IN IP4 192.168.31.11
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 t=0 0
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 m=image 4663 udptl t38
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxVersion:1
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38MaxBitRate:14400
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxTranscodingMMR
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxTranscodingJBIG
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxRateManagement:
>> transferredTCF
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxMaxDatagram:784
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 a=T38FaxUdpEC:t38UDPRedundancy
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.168441 [DEBUG]
>> switch_core_media.c:4028 sofia/internal/%2B491122334455 at 192.168.31.11
>> T38 REFUSE on request
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.168441 [DEBUG]
>> sofia.c:8007 Reinvite resulted in codec negotiation failure.
>> ac858869-5320-48fb-b100-1a4a9c3d0851 2017-12-03 23:43:49.188433 [DEBUG]
>> switch_rtp.c:7271 Correct audio ip/port confirmed.
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.188433 [DEBUG]
>> sofia.c:7077 Channel sofia/internal/%2B491122334455 at 192.168.31.11
>> skipping state [ready][488]
>> ac858869-5320-48fb-b100-1a4a9c3d0851 2017-12-03 23:43:49.208437 [DEBUG]
>> sofia.c:7084 Channel sofia/external/anonymous at 87.234.1.183 entering
>> state [ready][200]
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:43:49.208437 [DEBUG]
>> switch_rtp.c:7271 Correct audio ip/port confirmed.
>> 7fb3d04b-31f4-4539-a13e-aeaea7a194e8 2017-12-03 23:44:33.188445 [NOTICE]
>> sofia.c:1012 Hangup sofia/internal/%2B491122334455 at 192.168.31.11
>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>>
>
>
> What I need is:
>
> FAX ATA (T.38) -> Asterisk (T.38) -> FreeSWITCH (T.38 to audio GW, refuse
> T.38 on A-leg) -> Carrier (audio)
>
> Carrier (audio) -> FreeSWITCH (audio to T.38, refuse T.38 on B-leg) ->
> Asterisk (T.38) -> FAX ATA (T.38)
>
> Asterisk is sending Re-INVITES as seen in the log above. Seems like I miss
> some settings in my dialplan?
>
> Kevin
>
> 2017-12-02 16:36 GMT+01:00 Kevin Olbrich <ko at sv01.de>:
>
>> I need to note that all components are SIP based, no other tech involved.
>>
>> Kevin
>>
>> 2017-12-02 13:41 GMT+01:00 Kevin Olbrich <ko at sv01.de>:
>>
>>> Hi,
>>>
>>> thanks for the link but I was unable to find information on how to
>>> configure my installation of FreeSwitch 1.6 to make things work.
>>>
>>> I think I need to explain a little bit more on what I am planning to do:
>>> FreeSwitch will be used as a SBC to the carrier. It's firewall is
>>> limited to the carriers network on the first interface.
>>> The second interface is a local subnet where an Asterisk 13 server (PBX)
>>> is located, connecting the phones and ATAs (last with T.38 support).
>>> For asterisk it is only necessary so set (FAXOPT(gateway)=yes) to have
>>> it detect CNG on audio and query both legs for T.38 (or receive a request
>>> in advance).
>>> It then decides if it goes to passthru mode or transcode.
>>>
>>> As far as I understand the FreeSwitch docs, there are these options
>>> (located in dialplan default before bridge):
>>>
>>>> <action application="set" data="fax_enable_t38=true"/> <!-- Enable t.38
>>>> for this call -->
>>>> <action application="set" data="fax_enable_t38_request=true"/> <!--
>>>> Enable t38_gateway to send a t.38 reinvite when a fax tone is detected. If
>>>> using t38_gateway peer then you need to export this variable instead of set
>>>> -->
>>>> <action application="set" data="execute_on_answer=t38_gateway self"/>
>>>> <!--Execute t38_gateway on answer. self or peer. self: send a reinvite back
>>>> to the a-leg. peer reinvite forward to the b-leg -->
>>>
>>>
>>> <action application="export" data="fax_enable_t38=true"/> <!-- Required
>>>> as of v1.6 -->
>>>> <action application="export" data="t38_passthru=true"/>
>>>
>>>
>>> *What I would like to accomplish is getting as much as T.38 I can get.*
>>> The ATA should always be able to negotiate T.38 with Asterisk (which seems
>>> to work atm), FreeSwitch should then negotiate T.38 if Asterisk or the
>>> carrier asks for it (no CNG detection).
>>> I am not sure if I can just enable t38_gateway and passthru, this case
>>> is not listed on confluence or the old wiki. Even if I do, it does not work.
>>>
>>> FreeSwitch might need to transcode T.38 to audio for our backup carrier
>>> who does not support T.38.
>>>
>>> Kind regards,
>>> Kevin
>>>
>>> 2017-12-02 9:20 GMT+01:00 Raúl Alexis Betancor Santana <
>>> rbetancor at gmail.com>:
>>>
>>>> Some carriers gateways doesn't like a T.38 invite on advance, usually
>>>> the order that gives better results for faxing are first invite on
>>>> PCMU/PCMA (or any other supported codec) and then re-invite to T.38.
>>>>
>>>> Take a look at how gonicus gofaxip do-it (https://github.com/gonicus/go
>>>> faxip)
>>>>
>>>> On Fri, Dec 1, 2017 at 10:21 PM, Kevin Olbrich <ko at sv01.de> wrote:
>>>>
>>>>> Hi!
>>>>>
>>>>> Our carrier supports T.38 for outgoing faxes. I am trying to send
>>>>> faxes like this:
>>>>>
>>>>> originate {origination_caller_id_name='kevin',origination_caller_id_nu
>>>>>> mber='+49123456789',ignore_early_media=true,absolute_codec_s
>>>>>> tring='PCMA,PCMU',fax_use_ecm=true,fax_enable_t38=true,fax_e
>>>>>> nable_t38_request=true}sofia/gateway/carrier_1/+4912121212
>>>>>> &txfax(/opt/faxout.tif)
>>>>>
>>>>>
>>>>>
>>>>> Log:
>>>>>
>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.513164
>>>>>> [DEBUG] sofia_glue.c:1295 sofia/external/+4912121212 sending invite
>>>>>> version: 1.6.19 -36-7a77e0b 64bit
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa Local SDP:
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa v=0
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa o=FreeSWITCH 1512130268
>>>>>> 1512130270 IN IP4 123.123.123.123
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa s=FreeSWITCH
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa c=IN IP4 123.123.123.123
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa t=0 0
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa m=image 30466 udptl t38
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxVersion:0
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38MaxBitRate:14400
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxFillBitRemoval
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxRateManagement:transfe
>>>>>> rredTCF
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxBuffer:2000
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxMaxDatagram:400
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa a=T38FaxUdpEC:t38UDPRedundancy
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.533158
>>>>>> [DEBUG] sofia.c:7084 Channel sofia/external/+4912121212 entering state
>>>>>> [calling][0]
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120
>>>>>> [DEBUG] sofia.c:6294 sofia/external/+4912121212 T38 invite failed
>>>>>> 668fd128-f2f5-4af5-9536-6780762c2efa 2017-12-01 21:38:57.673120
>>>>>> [DEBUG] sofia.c:7077 Channel sofia/external/+4912121212 skipping state
>>>>>> [ready][488]
>>>>>
>>>>>
>>>>>
>>>>> Carrier confirmed that T.38 is possible... What could be the cause for
>>>>> this? How can I enforce T.38 instead of falling back to g711a (hangup on
>>>>> failed invite)?
>>>>>
>>>>> Thanks.
>>>>>
>>>>> Kind regards,
>>>>> Kevin
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From asilva at wirelessmundi.com  Mon Dec  4 12:34:58 2017
From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=)
Date: Mon, 4 Dec 2017 13:34:58 +0100
Subject: [Freeswitch-users] WebSocket behind NGINX
In-Reply-To: <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997CE62@PHXEX2.vertical.com>
References: <56AD0CE7.6000607@gmail.com> <56B667DA.4010505@gmail.com>
 <1ECCD30B07AF6E4EB58BC5F1E9FFF981C997CE62@PHXEX2.vertical.com>
Message-ID: <7ada6dbc-c3c3-56c1-eec4-4ab29f5bb9ab@wirelessmundi.com>

Hi Dan,

dig you get it work? Is it possible to use:  wss -> nginx --> ws -> fs ?

I see the same behaviour, the VIA header that arrives to fs in the ws 
port contains SIP/2.0/WSS and of course, fs will reject the request 
because its a different protocol...


nta.c:3146 agent_check_request_via() nta: Via check: invalid transport 
"SIP/2.0/WSS" from 192.168.10.5:54074


I guess that what we need here is some sort of "helper" in nignx that 
replaces WSS to WS.



On 02/08/2016 05:36 PM, Dan Edwards wrote:
> Anton,
>
> I'm glad my input was useful. As for WSS vs WS, the fact you're using security bubbles up into the SIP messages themselves. I initially tried:
>
>
> Browser >> WSS >> Nginx >> WS >> FS
>
>
> FS does not like this because the protocol changes. You go from SIP/2.0/WSS to SIP/2.0/WS and FS won't allow that. Also, in some instances, you will get SIP URL changes. For example: sip:1234 at domain.com vs. sips:1234 at domain.com.
>
> The reason to go with WS to FS was to skip an encrypt/decrypt cycle on network traffic that never left the machine. I finally decided that trying to patch the SIP traffic was bound to fail at some point and we're only saving the encrypt/decrypt on the SIP traffic itself, so I went back to
>
>
> Browser >> WSS >> Nginx >>> WSS >> FS
>
>
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anton
> Sent: Saturday, February 06, 2016 4:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] WebSocket behind NGINX
>
> Hi,
>
> Sorry for not answering for a long time.
>
> Dan, thank you, your recommendation really helped me.
>
> So in order to proxy websocket request you need:
> 1. Proxy websocket requests in this way WSS -> (NGINX) -> FS WSS or WS
> -> (NGINX) -> FS WS
> 2. Modify local-network-acl
> 3. Modify apply-candidate-acl if you would like to drop more rtp candidates
>
> PS: I highly recommend to watch this video about NAT issues and ACL
> configuration:
> https://www.youtube.com/watch?v=_WSx-T6TriI
>
> BR,
> Anton Voylenko
>
> On 01/30/2016 09:20 PM, Anton wrote:
>> Hello All,
>>
>> I have to proxy all websocket requests though a nginx server. Right
>> now I am using next configuration:
>>
>> map $http_upgrade $connection_upgrade {
>>      default upgrade;
>>      ''      close;
>> }
>>
>> server {
>>      listen 443;
>>      server_name  wss.somedomain.com.ua;
>>
>>      ssl on;
>>      ssl_certificate      /etc/nginx/cert.pem;
>>      ssl_certificate_key  /etc/nginx/private.key;
>>
>>      location / {
>>          proxy_pass http://127.0.0.1:5066;
>>          proxy_http_version 1.1;
>>          proxy_set_header Upgrade $http_upgrade;
>>          proxy_set_header Connection $connection_upgrade;
>>          proxy_read_timeout 86400s;
>>     }
>>
>>      access_log /var/log/nginx/wss_access;
>>      error_log /var/log/nginx/wss_error debug; }
>>
>> I dumped traffic from nginx and found out that "switching protocol"
>> phrase was successful but INVITE message from my browser in pending
>> state.
>> Maybe FreeSWITCH wants real IP not loopback? Who have faced with
>> similar problem?
>>
>> BR,
>> Anton
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Saludos / Regards / Cumprimentos
António Silva



From prashant.lamba at gmail.com  Tue Dec  5 08:23:33 2017
From: prashant.lamba at gmail.com (Prashant Lamba)
Date: Tue, 5 Dec 2017 13:53:33 +0530
Subject: [Freeswitch-users] G729 and windows
In-Reply-To: <CAAoGpGSFuiUCS7zgyuFTF4vhk5gKsKQgJAd+pTrKxmhR3Quyvw@mail.gmail.com>
References: <CAC7ttFF-Ky_1_0_DvwfiH4bBpXi30zEubGBLqW99ZH6GvGiYSg@mail.gmail.com>
 <8BB84A89-4CE7-41AB-8669-A699506EF530@freeswitch.org>
 <CAC7ttFFCSNWf016VFU8=u0hkgj5uYf0me7xOZuGsgfNc==wtrw@mail.gmail.com>
 <15ff27235a0.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <CAAoGpGSFuiUCS7zgyuFTF4vhk5gKsKQgJAd+pTrKxmhR3Quyvw@mail.gmail.com>
Message-ID: <CAPHrvXHP2a7d6vwFYkBOFX=AF446HgO1XQ+_2psN1wLMjeLbOA@mail.gmail.com>

Although the Patent has expired, it does not imply that FS cant charge a
license / per port fee. They have developed G729 support and maintaining it.
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From kathleen at freeswitch.com  Tue Dec  5 17:58:48 2017
From: kathleen at freeswitch.com (Kathleen King)
Date: Tue, 5 Dec 2017 09:58:48 -0800
Subject: [Freeswitch-users] Turning off ring back
In-Reply-To: <f554df0bb84c4375abd2eaa23c6054f9@hil-vs-exdag01.buehner-fry.com>
References: <f554df0bb84c4375abd2eaa23c6054f9@hil-vs-exdag01.buehner-fry.com>
Message-ID: <CAOStUPYYDYsUUpE1nZwn5WtpeWY0Fb0_oysQPrxCWZu77Xuiew@mail.gmail.com>

Jerry,

We have picked your question, "FreeSwitch is presenting ring back to
calling party by default. Is there a way to shut that feature off?" to be
answered on ClueCon Weekly this week during the Community Corner segment.
You can join us live by dialing 888 at https://conference.freeswitch.org/vc/
or you can watch it live on Youtube using this link
https://youtu.be/XtMh9rAtB8w



[image: freeswitch logo giant.jpg] <https://freeswitch.com>

Kathleen King | Public Relations / Administrative Assistant

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045

Email: Kathleen at freeswitch.com

Mobile: 703-859-3757

Website: https://www.FreeSWITCH.com <https://www.freeswitch.com>

[image: color-facebook-96.png] <https://www.facebook.com/freeswitch/> [image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>

On Tue, Nov 28, 2017 at 10:08 AM, Jerry Chinn <JHChinn at thenavisway.com>
wrote:

> Good Day.
>
>
>
> FreeSwitch is presenting ring back to calling party by default.
>
> Is there a way to shut that feature off?
>
>
>
> *Jerry Chinn*
>
> *Telecom VoIP Specialist*
>
> *NAVIS *More Performance. More Profit.
>
> tel 541-330-3562 <(541)%20330-3562>
>
> www.TheNavisWay.com <http://www.thenavisway.com/>
>
> Facebook <https://www.facebook.com/theNAVISway/> | Twitter
> <https://twitter.com/NAVISway> | LinkedIn
> <https://www.linkedin.com/company/navisway> | Blog
> <https://www.thenavisway.com/blog>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmina at connectfirst.com  Wed Dec  6 02:01:02 2017
From: gmina at connectfirst.com (Geoff Mina)
Date: Wed, 06 Dec 2017 02:01:02 +0000
Subject: [Freeswitch-users] WSS Socket Disconnect
In-Reply-To: <CAKzROs2zFtdxSPwj6wYys1rqP1o3c+LTKBaG27AHZ3EPKjf1xQ@mail.gmail.com>
References: <CAKzROs2zFtdxSPwj6wYys1rqP1o3c+LTKBaG27AHZ3EPKjf1xQ@mail.gmail.com>
Message-ID: <CAKzROs1qi7HP63qFNA6TX1=G7tUV6xrh0iNxLu4JT-zE3mprYA@mail.gmail.com>

Just wondering in anyone has any thoughts on the TCP disconnect. I have
temporarily worked around the issue by putting nginx in the middle, but I
now lose the IP of the registered user - so it’s not an ideal solution long
term.

Thanks in advance.

Geoff

On Sat, Dec 2, 2017 at 3:56 PM Geoff Mina <gmina at connectfirst.com> wrote:

> I am having an issue where the WSS connection to Freeswitch is abruptly
> disconnecting under certain scenarios.  When trying to send an INVITE to
> FreeSwitch, the socket disconnects.  When accepting an INVITE from
> FreeSwitch everything works fine.
>
> I am running on CentOS 7 - 3.10.0-693.el7.x86_64.  I installed Freeswitch
> 1.6.19 via yum.  Here is what we are seeing from the browser side.  Same
> symptom in Chrome and FireFox:
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:34:52 GMT-0700 (MST) | sip.transaction.nist | Timer J
> expired for non-INVITE server transaction z9hG4bKZmZS5KUSecFee
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> acquiring local media
>
> sip-0.7.7.js:2900:3
>
> onaddstream is deprecated! Use peerConnection.ontrack instead.
>
> sip-0.7.7.js:11498
>
> navigator.mozGetUserMedia has been replaced by
> navigator.mediaDevices.getUserMedia
>
> sip-0.7.7.js:10554:8
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> acquired local media streams
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> RTCIceGatheringState changed: gathering
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> ICE candidate received: candidate:0 1 UDP 2122252543 192.168.240.12 61028
> typ host
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> ICE candidate received: candidate:2 1 UDP 2122187007 172.16.2.7 54757 typ
> host
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> ICE candidate received: candidate:4 1 TCP 2105524479 192.168.240.12 9 typ
> host tcptype active
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> ICE candidate received: candidate:5 1 TCP 2105458943 172.16.2.7 9 typ host
> tcptype active
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> ICE candidate received: candidate:0 2 UDP 2122252542 192.168.240.12 61218
> typ host
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> ICE candidate received: candidate:2 2 UDP 2122187006 172.16.2.7 52542 typ
> host
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> ICE candidate received: candidate:4 2 TCP 2105524478 192.168.240.12 9 typ
> host tcptype active
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> ICE candidate received: candidate:5 2 TCP 2105458942 172.16.2.7 9 typ host
> tcptype active
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> ICE candidate received: candidate:3 1 UDP 1685987327 52.33.247.158 25669
> typ srflx raddr 172.16.2.7 rport 54757
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:12 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> ICE candidate received: candidate:3 2 UDP 1685987326 52.33.247.158 16597
> typ srflx raddr 172.16.2.7 rport 52542
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.invitecontext.mediahandler |
> RTCIceChecking Timeout Triggered after 5000 milliseconds
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | sending
> WebSocket message:
>
>
> INVITE sip:*97 at c01-aorgc-reg6.somehost.biz SIP/2.0
>
> Via: SIP/2.0/WSS hqj7ffmo2koa.invalid;branch=z9hG4bK1975235
>
> Max-Forwards: 70
>
> To: <sip:*97 at c01-aorgc-reg6.somehost.biz>
>
> From: "gmina" <sip:gmina at c01-aorgc-reg6.somehost.biz>;tag=efk4hqf1dc
>
> Call-ID: vitn3acs5ss3dikrn14a
>
> CSeq: 7808 INVITE
>
> Contact: <sip:7jm943k7 at hqj7ffmo2koa.invalid;transport=ws;ob>
>
> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
>
> Supported: outbound
>
> User-Agent: SIP.js/0.7.7 BB\
>
> Content-Type: application/sdp
>
> Content-Length: 1646
>
>
> v=0
>
> o=mozilla...THIS_IS_SDPARTA-57.0 335274293698172607 0 IN IP4 0.0.0.0
>
> s=-
>
> t=0 0
>
> a=sendrecv
>
> a=fingerprint:sha-256
> BE:35:AD:35:14:DB:CB:2B:D9:A3:D5:A7:EB:7D:08:E4:29:5E:7E:25:A6:16:9F:34:EC:37:0B:7A:C7:E0:9F:5B
>
> a=group:BUNDLE sdparta_0
>
> a=ice-options:trickle
>
> a=msid-semantic:WMS *
>
> m=audio 25669 UDP/TLS/RTP/SAVPF 109 9 0 8 101
>
> c=IN IP4 52.33.247.158
>
> a=candidate:0 1 UDP 2122252543 192.168.240.12 61028 typ host
>
> a=candidate:2 1 UDP 2122187007 172.16.2.7 54757 typ host
>
> a=candidate:4 1 TCP 2105524479 192.168.240.12 9 typ host tcptype active
>
> a=candidate:5 1 TCP 2105458943 172.16.2.7 9 typ host tcptype active
>
> a=candidate:0 2 UDP 2122252542 192.168.240.12 61218 typ host
>
> a=candidate:2 2 UDP 2122187006 172.16.2.7 52542 typ host
>
> a=candidate:4 2 TCP 2105524478 192.168.240.12 9 typ host tcptype active
>
> a=candidate:5 2 TCP 2105458942 172.16.2.7 9 typ host tcptype active
>
> a=candidate:3 1 UDP 1685987327 52.33.247.158 25669 typ srflx raddr
> 172.16.2.7 rport 54757
>
> a=candidate:3 2 UDP 1685987326 52.33.247.158 16597 typ srflx raddr
> 172.16.2.7 rport 52542
>
> a=sendrecv
>
> a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:ssrc-audio-level
>
> a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
>
> a=fmtp:101 0-15
>
> a=ice-pwd:d45de180fdae3dc61f0e26238541045c
>
> a=ice-ufrag:50681d44
>
> a=mid:sdparta_0
>
> a=msid:{23ed6231-4501-1944-8577-ab85c6c721cb}
> {c98743c7-76d0-5c44-89a1-f702f7d8168d}
>
> a=rtcp:16597 IN IP4 52.33.247.158
>
> a=rtcp-mux
>
> a=rtpmap:109 opus/48000/2
>
> a=rtpmap:9 G722/8000/1
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=setup:actpass
>
> a=ssrc:3742140108 cname:{76ae6d79-b818-1e40-a9bc-4734a437b47a}
>
>
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | WebSocket
> disconnected (code: 1006)
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | WebSocket abrupt
> disconnection
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.ua | connection state set
> to 1
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transaction.ict | transport
> error occurred, deleting INVITE client transaction z9hG4bK1975235
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | Connection to
> WebSocket wss://c01-aorgc-reg6.somehost.biz:8089/freeswitch severed,
> attempting first reconnect
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | connecting to
> WebSocket wss://c01-aorgc-reg6.somehost.biz:8089/freeswitch
>
> sip-0.7.7.js:2900:3
>
> Sat Dec 02 2017 15:35:17 GMT-0700 (MST) | sip.transport | WebSocket wss://
> c01-aorgc-reg6.somehost.biz:8089/freeswitch connected
>
>
> *And here is what we are seeing from the Freeswitch side.*
>
>
> tport.c:2773 tport_wakeup() tport_wakeup(0x7f4e580fc4d0): events IN
>
> tport.c:2864 tport_recv_event() tport_recv_event(0x7f4e580fc4d0)
>
> tport.c:2157 tport_shutdown0() tport_shutdown0(0x7f4e580fc4d0, 2)
>
> tport.c:2090 tport_close() tport_close(0x7f4e580fc4d0): wss/
> 52.33.247.158:25656/sips
>
> nua_registrar.c:200 registrar_tport_error() tport error 0: Success
>
> tport.c:4222 tport_release() tport_release(0x7f4e580fc4d0): (nil) by
> 0x7f4e581361b0 with (nil)
>
> tport.c:2263 tport_set_secondary_timer() tport(0x7f4e580fc4d0): set timer
> at 0 ms because zap
>
> *nua_stack.c:271 nua_stack_event() nua(0x7f4e581361b0): event
> i_media_error 500 Transport error detected*
>
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
>
> tport.c:2263 tport_set_secondary_timer() tport(0x7f4e580fc4d0): set timer
> at 0 ms because zap
>
> tport_type_ws.c:521 tport_ws_deinit_secondary() 0x7f4e580fc4d0 destroy wss
> transport 0x7f4e580fc6c0.
>
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>
> nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
>
> nua_stack.c:529 nua_signal() nua(0x7f4e581361b0): sent signal r_destroy
>
> nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
>
> nua_stack.c:569 nua_stack_signal() nua(0x7f4e581361b0): recv signal
> r_destroy
>
> nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f4e581361b0):
> removing registrar usage
>
> nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
>
> soa.c:356 soa_destroy() soa_destroy(static::0x7f4e5803dea0) called
>
> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7f4e58004f50): events IN
>
> tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7f4e58004f50):
> new secondary tport 0x7f4e580fc4d0
>
> tport.c:2292 tport_set_secondary_timer() tport(0x7f4e580fc4d0): set timer
> at 4998 ms because keepalive
>
> tport.c:2640 tport_accept() tport_accept(0x7f4e580fc4d0): new connection
> from wss/52.33.247.158:54385/sips
>
> tport.c:2773 tport_wakeup() tport_wakeup(0x7f4e580fc4d0): events IN
>
> tport.c:2864 tport_recv_event() tport_recv_event(0x7f4e580fc4d0)
>
> tport.c:2296 tport_set_secondary_timer() tport(0x7f4e580fc4d0): reset timer
>
> tport.c:2773 tport_wakeup() tport_wakeup(0x7f4e580fc4d0): events IN
>
>
> Any help in narrowing down the cause would be greatly appreciated.  I am
> thinking maybe it's a max frame size on the websocket or something... but I
> couldn't find anything that would indicate there was a setting to tweak.
>
> Thanks,
> Geoff
>
-- 


*GEOFF MINA*Chief Executive Officer
Connect First / Contact Center Solutions, Built Better.

3101 Iris Ave #200, Boulder, CO 80301
720.335.5924 <javascript:void(0);>
gmina at connectfirst.com / www.connectfirst.com <http://connectfirst.com/>

 [image:
https://docs.google.com/uc?export=download&id=0B5b6KnVfm9lJTlFrQzRVUjJ2ZVE&revid=0B5b6KnVfm9lJUXpUMTFEbGJvaktwN1p5ejM3YTFkdWVWNzBzPQ]

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From royce3 at gmail.com  Wed Dec  6 03:22:25 2017
From: royce3 at gmail.com (Royce Mitchell III)
Date: Tue, 5 Dec 2017 21:22:25 -0600
Subject: [Freeswitch-users] fifo callers when > call limit until limit no
	longer exceeded
Message-ID: <CAHwzD9n29CNwS=j4iCgH1Ft-GAOsVgXmn7BtcaP7sPb2HyjwjQ@mail.gmail.com>

I am currently using hash to limit the number of calls that can get through
on a specific destination_number and giving any excess a USER_BUSY,
something like this:

<action application="limit" data="hash inbound ${destination_number}
${call_limit} !USER_BUSY"/>

However, now they're asking me to queue those calls with a recording and
let them through in order as the call_limit reduces back down and then send
the call to a different destination if it stays in that queue for a minute.

In other words, if call_limit is 5, the 6th call would get queued until one
of those first 5 calls clears. If none of the 5 calls clears before a
minute has elapsed, I want to transfer that 6th call to a different
destination. If a 7th call enters the queue, the 6th call should be the
next to get a channel because it has been in the queue longer.

I'm digging through the documentation and I haven't been able to figure out
a way to do this so far. It seems like maybe using fifo in combination with
limit would be in order, but I can't think of how to make them work
together to accomplish this.




Royce Mitchell, IT Consultant
ITAS Solutions
royce3 at itas-solutions.com
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From michael at mailworks.org  Wed Dec  6 04:54:50 2017
From: michael at mailworks.org (Michael Avers)
Date: Tue, 05 Dec 2017 21:54:50 -0700
Subject: [Freeswitch-users] Best practices for recording audio prompts
Message-ID: <1512536090.3964615.1195557968.20D962DC@webmail.messagingengine.com>

Hello,

Can anyone share some tips about recording good audio prompts? We ideally would like to have them in a variety of sample rates (as they will be used internally as well as when callers dial in from PSTN) to avoid resampling as much as possible. What should it be recorded as originally? Would we then just use sox to resample? Any good apps for Windows? (I haven't used Windows in many years but the person who will be recording would be).

Any other considerations to take into account would be greatly appreciated.. it's our first time recording our own set of prompts.

Thanks
Mike


From j.peral at airenetworks.es  Tue Dec  5 13:55:27 2017
From: j.peral at airenetworks.es (Joaquin Peral)
Date: Tue, 5 Dec 2017 14:55:27 +0100
Subject: [Freeswitch-users] Header Duplicate same name
Message-ID: <56db8c0f-1f8f-d7ef-9fa5-f8dac2f3429e@airenetworks.es>

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From nycphoneservice at gmail.com  Wed Dec  6 06:01:34 2017
From: nycphoneservice at gmail.com (NYCPhoneService)
Date: Wed, 6 Dec 2017 01:01:34 -0500
Subject: [Freeswitch-users] mod_spandsp on Windows as modem with com0com -
	port in use
In-Reply-To: <000301d369a6$0007b8b0$00172a10$@gmail.com>
References: <000301d369a6$0007b8b0$00172a10$@gmail.com>
Message-ID: <003501d36e57$ac450190$04cf04b0$@gmail.com>

Anyone please.

 

From: NYCPhoneService [mailto:nycphoneservice at gmail.com] 
Sent: Thursday, November 30, 2017 1:40 AM
To: freeswitch-users at lists.freeswitch.org
Subject: mod_spandsp on Windows as modem with com0com - port in use

 

I guess I'm missing something with this setup:

 

Freeswitch-1.6.19 built from source on Win 8.1, com0com 3.0 signed, pair on
COM6 and COM7 the and soft modem for fax server is on COM7

I had to set number of modems to 6 in config, otherwise I was just getting
no modems available.

 

Mod_spandsp sees the port:

 

2017-11-29 23:43:52.885349 [DEBUG] mod_spandsp_modem.c:371 Modem COM6 [INIT]
- Changing state to INIT

2017-11-29 23:43:52.885349 [INFO] mod_spandsp_modem.c:373 Modem
[COM6]->[(null)] Ready

 

but when I bridge to modem/6/123 I get:

 

2017-11-29 23:44:49.203916 [ERR] mod_spandsp_modem.c:1390 Modem COM6 In Use!

2017-11-29 23:44:49.203916 [ERR] mod_spandsp_modem.c:1398 No Modems
Available!

971de557-adb4-4a19-9e7d-ab4f13b954be 2017-11-29 23:44:49.203916 [ERR]
mod_spandsp_modem.c:862 Cannot find a modem.

 

 

The same com0com setup works with an old T38Modem with no issues.

 

Any suggestions are really appreciated

 

Dennis

 

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From philipp at zeitschel.net  Tue Dec  5 18:49:47 2017
From: philipp at zeitschel.net (Philipp Zeitschel)
Date: Tue, 5 Dec 2017 18:49:47 +0000
Subject: [Freeswitch-users] call interruption after 5 minutes
Message-ID: <7c443aada98b41cabf9b73e0a794e285@zeitschel.net>

Hi,

 

i'm regged to a gateway, if I initiate a call to my mobile after 5 Minutes
the call terminates but only on my mobile, my phone (connected to
freeswitch) counts the time until I hang up

 

to the time when the problem occurs I get following log entries:

 

nta.c:7159 _nta_incoming_timer() nta: timer J fired, terminate 401 response

nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout,
1/2 term, 1/2 free

nta.c:7159 _nta_incoming_timer() nta: timer J fired, terminate 200 response

nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout,
1/1 term, 1/1 free

nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate OPTIONS
(115912462)

nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout,
1/3 term, 1/3 free

nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate OPTIONS
(115912463)

nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout,
1/2 term, 1/2 free

nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate OPTIONS
(115912464)

nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout,
1/1 term, 1/1 free

auth_digest.c:226 auth_digest_a1() auth_digest_a1() has A1 = MD5(+49xxx:
mga.mnet-voip.de:xxx) = xxxx

auth_digest.c:318 auth_digest_response() A2 =
MD5(REGISTER:sip:mga.mnet-voip.de;transport=udp)

auth_digest.c:348 auth_digest_response() auth_response: xxx = MD5(xxx)
(qop=auth)

nta.c:10720 outgoing_query_aaaa() nta: for "mga.mnet-voip.de" query
"mga.mnet-voip.de" AAAA (cached)

nta.c:10774 outgoing_answer_aaaa() nta(0x7fe3740563f0): mga.mnet-voip.de. IN
AAAA 2001:a60:1:7::10

nta.c:8304 outgoing_send() nta: sent REGISTER (115912137) to
udp/2001:a60:1:7::10:5060/sip

nta.c:3299 agent_recv_response() nta: received 200 OK for REGISTER
(115912137)

nta.c:3366 agent_recv_response() nta: 200 OK is going to a transaction

nua_stack.c:271 nua_stack_event() nua(0x7fe394014e10): event r_register 200
OK

2017-12-05 19:38:12.246833 [DEBUG] switch_rtp.c:1463  [  zrtp cache]:
Storing ZRTP cache to </var/lib/freeswitch/db/zrtp.dat>...

2017-12-05 19:38:12.246833 [DEBUG] switch_rtp.c:1324 Saving ZRTP cache: OK

nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate REGISTER
(115912137)

nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout,
1/1 term, 1/1 free

nua_stack.c:569 nua_stack_signal() nua(0x7fe394083af0): recv signal
r_options

nta.c:10720 outgoing_query_aaaa() nta: for "mga.mnet-voip.de" query
"mga.mnet-voip.de" AAAA (cached)

nta.c:10774 outgoing_answer_aaaa() nta(0x7fe3740398f0): mga.mnet-voip.de. IN
AAAA 2001:a60:1:7::10

nta.c:8304 outgoing_send() nta: sent OPTIONS (115912465) to
udp/2001:a60:1:7::10:5060

nta.c:3299 agent_recv_response() nta: received 403 Forbidden for OPTIONS
(115912465)

nta.c:3366 agent_recv_response() nta: 403 Forbidden is going to a
transaction

nua_stack.c:271 nua_stack_event() nua(0x7fe394083af0): event r_options 403
Forbidden

nua_stack.c:569 nua_stack_signal() nua(0x7fe394083af0): recv signal
r_destroy

nua_stack.c:569 nua_stack_signal() nua(0x7fe394083b80): recv signal
r_options

nta.c:10720 outgoing_query_aaaa() nta: for "mga.mnet-voip.de" query
"mga.mnet-voip.de" AAAA (cached)

nta.c:10774 outgoing_answer_aaaa() nta(0x7fe37402c980): mga.mnet-voip.de. IN
AAAA 2001:a60:1:7::10

nta.c:8304 outgoing_send() nta: sent OPTIONS (115912466) to
udp/2001:a60:1:7::10:5060

nta.c:3299 agent_recv_response() nta: received 403 Forbidden for OPTIONS
(115912466)

nta.c:3366 agent_recv_response() nta: 403 Forbidden is going to a
transaction

nua_stack.c:271 nua_stack_event() nua(0x7fe394083b80): event r_options 403
Forbidden

nua_stack.c:569 nua_stack_signal() nua(0x7fe394083b80): recv signal
r_destroy

nua_stack.c:569 nua_stack_signal() nua(0x7fe39407d2d0): recv signal
r_options

nta.c:10720 outgoing_query_aaaa() nta: for "mga.mnet-voip.de" query
"mga.mnet-voip.de" AAAA (cached)

nta.c:10774 outgoing_answer_aaaa() nta(0x7fe37405b8f0): mga.mnet-voip.de. IN
AAAA 2001:a60:1:7::10

nta.c:8304 outgoing_send() nta: sent OPTIONS (115912467) to
udp/2001:a60:1:7::10:5060

nta.c:3299 agent_recv_response() nta: received 403 Forbidden for OPTIONS
(115912467)

nta.c:3366 agent_recv_response() nta: 403 Forbidden is going to a
transaction

nua_stack.c:271 nua_stack_event() nua(0x7fe39407d2d0): event r_options 403
Forbidden

nua_stack.c:569 nua_stack_signal() nua(0x7fe39407d2d0): recv signal
r_destroy

nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate OPTIONS
(115912465)

nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout,
1/3 term, 1/3 free

nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate OPTIONS
(115912466)

nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout,
1/2 term, 1/2 free

nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate OPTIONS
(115912467)

nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout,
1/1 term, 1/1 free

 

it seems like the registration is the problem but I have no glue why.

My Provider won't help cause he tries to bend me to a router he provides.

Incoming calls are no problem at all.

 

Any ideas? Anything else I can provide?

 

Thanks in advance

 

Regards

 

Philipp

 

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From lexxua at gmail.com  Wed Dec  6 08:22:11 2017
From: lexxua at gmail.com (Volodymyr Fedorov)
Date: Wed, 6 Dec 2017 09:22:11 +0100
Subject: [Freeswitch-users] mod_spandsp on Windows as modem with com0com
 - port in use
In-Reply-To: <003501d36e57$ac450190$04cf04b0$@gmail.com>
References: <000301d369a6$0007b8b0$00172a10$@gmail.com>
 <003501d36e57$ac450190$04cf04b0$@gmail.com>
Message-ID: <CAO_GGAoRsFppR72N-KWUmmjiw1m6hNEcaoTxNC2gjSjL76U0KQ@mail.gmail.com>

Hi I don't remember, but in code exist commit for Linux system which was
supposed to fix modem TTY names. But somehow it has regression on Windows
systems because modem name could not start from 0.
Br,
Volodymyr

On Dec 6, 2017 8:16 AM, "NYCPhoneService" <nycphoneservice at gmail.com> wrote:

> Anyone please…
>
>
>
> *From:* NYCPhoneService [mailto:nycphoneservice at gmail.com]
> *Sent:* Thursday, November 30, 2017 1:40 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* mod_spandsp on Windows as modem with com0com - port in use
>
>
>
> I guess I’m missing something with this setup:
>
>
>
> Freeswitch-1.6.19 built from source on Win 8.1, com0com 3.0 signed, pair
> on COM6 and COM7 the and soft modem for fax server is on COM7
>
> I had to set number of modems to 6 in config, otherwise I was just getting
> no modems available…
>
>
>
> Mod_spandsp sees the port:
>
>
>
> 2017-11-29 23:43:52.885349 [DEBUG] mod_spandsp_modem.c:371 Modem COM6
> [INIT] - Changing state to INIT
>
> 2017-11-29 23:43:52.885349 [INFO] mod_spandsp_modem.c:373 Modem
> [COM6]->[(null)] Ready
>
>
>
> but when I bridge to modem/6/123 I get:
>
>
>
> 2017-11-29 23:44:49.203916 [ERR] mod_spandsp_modem.c:1390 Modem COM6 In
> Use!
>
> 2017-11-29 23:44:49.203916 [ERR] mod_spandsp_modem.c:1398 No Modems
> Available!
>
> 971de557-adb4-4a19-9e7d-ab4f13b954be 2017-11-29 23:44:49.203916 [ERR]
> mod_spandsp_modem.c:862 Cannot find a modem.
>
>
>
>
>
> The same com0com setup works with an old T38Modem with no issues.
>
>
>
> Any suggestions are really appreciated
>
>
>
> Dennis
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From alexandr.popov at iqoption.com  Wed Dec  6 09:34:14 2017
From: alexandr.popov at iqoption.com (Alexandr Popov)
Date: Wed, 6 Dec 2017 11:34:14 +0200
Subject: [Freeswitch-users] Header Duplicate same name
In-Reply-To: <56db8c0f-1f8f-d7ef-9fa5-f8dac2f3429e@airenetworks.es>
References: <56db8c0f-1f8f-d7ef-9fa5-f8dac2f3429e@airenetworks.es>
Message-ID: <CAKeZRht9LNxmMAc7W6Tkv+vBd-qrNpK3-N4WiMDwytKx5NgWQg@mail.gmail.com>

no

2017-12-05 15:55 GMT+02:00 Joaquin Peral <j.peral at airenetworks.es>:

> Is possible to generate several headers with the same name? For example:
>
> Header-Name: XXX
> Header-Name: YYY
> Header-Name: ZZZ
>
> Using set / export is overwritten and I do not know if it can be done.
>
> <action application="set"><![CDATA[sip_h_P-Header-Name=XXX]]></action>
> <action application="set"><![CDATA[sip_h_P-Header-Name=YYY]]></action>
> <action application="set"><![CDATA[sip_h_P-Header-Name=ZZZ]]></action>
>
> Any ideas?
>
> Thanks!!
> --
>
> Joaquin Peral Cascales
>
> Dpto. de Telefonía
>
> [ j.peral at airenetworks.es ]
>
> Telf. 911090048 | 1774 | www.airenetworks.es
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vma at vallimamod.org  Wed Dec  6 11:02:21 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Wed, 6 Dec 2017 12:02:21 +0100
Subject: [Freeswitch-users] Header Duplicate same name
In-Reply-To: <56db8c0f-1f8f-d7ef-9fa5-f8dac2f3429e@airenetworks.es>
References: <56db8c0f-1f8f-d7ef-9fa5-f8dac2f3429e@airenetworks.es>
Message-ID: <86B6747A-0622-4F41-BE9D-BE8EB2B72DBF@vallimamod.org>

Hi,

The multi-header version won't work with set but you can use the equivalent compact form with comma separated list values like P-Header: XXX, YYY, ZZZ
I think I have a patch somewhere to make it work with dialplan arrays but for an older version. I need to find it and post to jira.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.

> On 5 Dec 2017, at 14:55, Joaquin Peral <j.peral at airenetworks.es> wrote:
> 
> Is possible to generate several headers with the same name? For example:
> 
> Header-Name: XXX
> Header-Name: YYY
> Header-Name: ZZZ
> 
> Using set / export is overwritten and I do not know if it can be done.
> 
> <action application="set"><![CDATA[sip_h_P-Header-Name=XXX]]></action>
> <action application="set"><![CDATA[sip_h_P-Header-Name=YYY]]></action>
> <action application="set"><![CDATA[sip_h_P-Header-Name=ZZZ]]></action>
> Any ideas?
> 
> Thanks!!
> -- 
> Joaquin Peral Cascales
> 
> Dpto. de Telefonía
> 
> [ j.peral at airenetworks.es <mailto:j.peral at airenetworks.es> ]
> 
> Telf. 911090048 | 1774 | www.airenetworks.es <http://www.airenetworks.es/><Airenetworks-Firma.jpg>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vma at vallimamod.org  Wed Dec  6 11:07:53 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Wed, 6 Dec 2017 12:07:53 +0100
Subject: [Freeswitch-users] fifo callers when > call limit until limit
 no longer exceeded
In-Reply-To: <CAHwzD9n29CNwS=j4iCgH1Ft-GAOsVgXmn7BtcaP7sPb2HyjwjQ@mail.gmail.com>
References: <CAHwzD9n29CNwS=j4iCgH1Ft-GAOsVgXmn7BtcaP7sPb2HyjwjQ@mail.gmail.com>
Message-ID: <42E7937F-B779-4048-B16F-E868C40AA50C@vallimamod.org>

Hi,

With the limit app, you can also send the call to a different extension instead of rejecting it with user_busy:

<action application="limit" data="hash inbound ${destination_number} ${call_limit} queue_ext XML queue_context"/>

The over limit calls will be transferred to the queue_context context with queue_ext extension.

But for your use case, I think the limit is no more necessary, you can directly use mod_fifo.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.


> On 6 Dec 2017, at 04:22, Royce Mitchell III <royce3 at gmail.com> wrote:
> 
> I am currently using hash to limit the number of calls that can get through on a specific destination_number and giving any excess a USER_BUSY, something like this:
> 
> <action application="limit" data="hash inbound ${destination_number} ${call_limit} !USER_BUSY"/>
> 
> However, now they're asking me to queue those calls with a recording and let them through in order as the call_limit reduces back down and then send the call to a different destination if it stays in that queue for a minute.
> 
> In other words, if call_limit is 5, the 6th call would get queued until one of those first 5 calls clears. If none of the 5 calls clears before a minute has elapsed, I want to transfer that 6th call to a different destination. If a 7th call enters the queue, the 6th call should be the next to get a channel because it has been in the queue longer.
> 
> I'm digging through the documentation and I haven't been able to figure out a way to do this so far. It seems like maybe using fifo in combination with limit would be in order, but I can't think of how to make them work together to accomplish this.
> 
> 
> 
> 
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com



From vma at vallimamod.org  Wed Dec  6 11:19:42 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Wed, 6 Dec 2017 12:19:42 +0100
Subject: [Freeswitch-users] ACL: auth_calls + apply-inbound-acl/auth-acl
In-Reply-To: <a5fcfeb9-c587-9e28-414b-41ced492651a@anatoli.ws>
References: <8fc35d90-f845-0ad2-d641-34a3b7506920@anatoli.ws>
 <d69f6fda-61c4-47bb-854f-0fb70b1a7aa4@anatoli.ws>
 <6BFB29AD-1424-4F13-A0DD-385C6C48DF6D@vallimamod.org>
 <a5fcfeb9-c587-9e28-414b-41ced492651a@anatoli.ws>
Message-ID: <AAB3D7E2-EA91-4F0C-B5BE-C92279DB6711@vallimamod.org>

Hi Anatoli,

Just saw your email.

The auth-acl is always checked first. If it passes, the call is accepted with no further check. Only if it fails:
- If auth-calls is true, digest auth is tried (that's why in logs you have: "Rejected by acl "xxx". Falling back to Digest auth.")
- else, call is rejected.

Hope this helps to make things clearer!

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.

> On 23 Nov 2017, at 22:48, Anatoli <me at anatoli.ws> wrote:
> 
> Hi Vallimamod,
> 
> Thanks a lot for your detailed explanation, sure it helps! It would be great to add these details to the documentation (not sure whom to ask about this). IMO the behavior you describe can't be inferred from the current documentation and it deals with security/authentication.
> 
> Could you please explain what would be the effect of auth-calls=true + auth-acl=<ip_range>?
> I suppose if the IP matches, it goes through the digest auth. If the IP doesn't match, sofia responds with 403 forbidden, right?
> 
> Thanks,
> Anatoli
> 
> From: Vallimamod Abdullah
> Sent: Tuesday, November 21, 2017 09:35
> To: Freeswitch Users Help
> Subject: Re: [Freeswitch-users] ACL: auth_calls + apply-inbound-acl/auth-acl
> 
> Hi,
> 
> Your mail is dense, I will try to answer at my best from my understanding of the source code:
> 
> - the default value for auth-call is false.
> 
> - When a call arrives, the apply-inbound-acl is checked first:
>   * If the IP is approved by the acl, the access is granted
>   * If the IP is rejected by the acl and auth-call is false, sofia responds with 403 forbidden (I skip the proxy-acl and X-AUTH-IP checks for simplicity)
>   * If the IP is rejected by the acl and auth-call is true, it falls back to digest auth.
> 
> - If accept-blind-auth is set with auth-call, freeswitch only checks if the From user is defined in directory. If so, user is authorized (without any password check)
> 
> - If auth-cal is set without the acl, the call go through digest authentication
> 
> - If neither is set, the call is accepted.
> 
> In your case, even if you can define directly a cidr in the apply-inbound-acl param value, it would be best to set it to a list name defined in autoload_configs/acl.conf.xml.
> 
> Hope this helps!
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From royce3 at gmail.com  Wed Dec  6 16:23:26 2017
From: royce3 at gmail.com (Royce Mitchell III)
Date: Wed, 6 Dec 2017 10:23:26 -0600
Subject: [Freeswitch-users] fifo callers when > call limit until limit
 no longer exceeded
In-Reply-To: <42E7937F-B779-4048-B16F-E868C40AA50C@vallimamod.org>
References: <CAHwzD9n29CNwS=j4iCgH1Ft-GAOsVgXmn7BtcaP7sPb2HyjwjQ@mail.gmail.com>
 <42E7937F-B779-4048-B16F-E868C40AA50C@vallimamod.org>
Message-ID: <CAHwzD9mq_+Q6Zszp6Kp5T1UDBtMSOaVONfu3vOhVGAUu5911og@mail.gmail.com>

How do I configure fifo to transfer a call out whenever a channel opens up.

I'm sending the calls through an Adtran which has a limited number of
channels available and we don't want this one DID to use all the channels
but we don't just want to USER_BUSY the calls either. We want to leave them
in a fifo listening to music until a channel becomes available to allow the
next call through. How do I configure the fifo to allow the next call
through once there are no longer 5 calls (for example) already sent to the
Adtran for that DID?

I was thinking I could execute something on orbit, but then it won't be a
fifo any more. We want them exiting the fifo in the order they enter. Also,
the orbit could mean they wait in the fifo longer than they should, so I
think that's definitely not the right approach.

I know I could write a lua script that monitors the fifo and transfers out
the next available caller, but I was hoping for a built-in way to do this.



Royce Mitchell, IT Consultant
ITAS Solutions
royce3 at itas-solutions.com

On Wed, Dec 6, 2017 at 5:07 AM, Vallimamod Abdullah <vma at vallimamod.org>
wrote:

> Hi,
>
> With the limit app, you can also send the call to a different extension
> instead of rejecting it with user_busy:
>
> <action application="limit" data="hash inbound ${destination_number}
> ${call_limit} queue_ext XML queue_context"/>
>
> The over limit calls will be transferred to the queue_context context with
> queue_ext extension.
>
> But for your use case, I think the limit is no more necessary, you can
> directly use mod_fifo.
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sipsolutions.fr
> .
>
>
> > On 6 Dec 2017, at 04:22, Royce Mitchell III <royce3 at gmail.com> wrote:
> >
> > I am currently using hash to limit the number of calls that can get
> through on a specific destination_number and giving any excess a USER_BUSY,
> something like this:
> >
> > <action application="limit" data="hash inbound ${destination_number}
> ${call_limit} !USER_BUSY"/>
> >
> > However, now they're asking me to queue those calls with a recording and
> let them through in order as the call_limit reduces back down and then send
> the call to a different destination if it stays in that queue for a minute.
> >
> > In other words, if call_limit is 5, the 6th call would get queued until
> one of those first 5 calls clears. If none of the 5 calls clears before a
> minute has elapsed, I want to transfer that 6th call to a different
> destination. If a 7th call enters the queue, the 6th call should be the
> next to get a channel because it has been in the queue longer.
> >
> > I'm digging through the documentation and I haven't been able to figure
> out a way to do this so far. It seems like maybe using fifo in combination
> with limit would be in order, but I can't think of how to make them work
> together to accomplish this.
> >
> >
> >
> >
> > Royce Mitchell, IT Consultant
> > ITAS Solutions
> > royce3 at itas-solutions.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From nneul at mst.edu  Wed Dec  6 17:39:50 2017
From: nneul at mst.edu (Nathan Neulinger)
Date: Wed, 6 Dec 2017 11:39:50 -0600
Subject: [Freeswitch-users] Any update on 1.8 production release?
Message-ID: <9b523b52-14f6-6952-9bd2-faa1b16a6b62@mst.edu>

Been in a holding pattern on deploying to our production environment until this is actually released. Any timeframe update?

-- Nathan

------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect

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From me at anatoli.ws  Wed Dec  6 18:19:28 2017
From: me at anatoli.ws (Anatoli)
Date: Wed, 6 Dec 2017 15:19:28 -0300
Subject: [Freeswitch-users] ACL: auth_calls + apply-inbound-acl/auth-acl
In-Reply-To: <AAB3D7E2-EA91-4F0C-B5BE-C92279DB6711@vallimamod.org>
References: <8fc35d90-f845-0ad2-d641-34a3b7506920@anatoli.ws>
 <d69f6fda-61c4-47bb-854f-0fb70b1a7aa4@anatoli.ws>
 <6BFB29AD-1424-4F13-A0DD-385C6C48DF6D@vallimamod.org>
 <a5fcfeb9-c587-9e28-414b-41ced492651a@anatoli.ws>
 <AAB3D7E2-EA91-4F0C-B5BE-C92279DB6711@vallimamod.org>
Message-ID: <6480d8b9-7791-7204-ed1c-c64c7ac5304f@anatoli.ws>

Hi Vallimamod,

Thanks a lot! My supposition was wrong. Quite non-trivial behavior 
again. Your explanation definitely should be added to the documentation.

Regards,
Anatoli

*From:* Vallimamod Abdullah
*Sent:* Wednesday, December 06, 2017 08:19
*To:* Freeswitch Users Help
*Subject:* Re: [Freeswitch-users] ACL: auth_calls + 
apply-inbound-acl/auth-acl

Hi Anatoli,

Just saw your email.

The auth-acl is always checked first. If it passes, the call is accepted with no further check. Only if it fails:
- If auth-calls is true, digest auth is tried (that's why in logs you have: "Rejected by acl "xxx". Falling back to Digest auth.")
- else, call is rejected.

Hope this helps to make things clearer!

Best Regards,
-- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr .

> On 23 Nov 2017, at 22:48, Anatoli<me at anatoli.ws>  wrote:
>
> Hi Vallimamod,
>
> Thanks a lot for your detailed explanation, sure it helps! It would be great to add these details to the documentation (not sure whom to ask about this). IMO the behavior you describe can't be inferred from the current documentation and it deals with security/authentication.
>
> Could you please explain what would be the effect of auth-calls=true + auth-acl=<ip_range>?
> I suppose if the IP matches, it goes through the digest auth. If the IP doesn't match, sofia responds with 403 forbidden, right?
>
> Thanks,
> Anatoli
>
> From: Vallimamod Abdullah
> Sent: Tuesday, November 21, 2017 09:35
> To: Freeswitch Users Help
> Subject: Re: [Freeswitch-users] ACL: auth_calls + apply-inbound-acl/auth-acl
>
> Hi,
>
> Your mail is dense, I will try to answer at my best from my understanding of the source code:
>
> - the default value for auth-call is false.
>
> - When a call arrives, the apply-inbound-acl is checked first:
>    * If the IP is approved by the acl, the access is granted
>    * If the IP is rejected by the acl and auth-call is false, sofia responds with 403 forbidden (I skip the proxy-acl and X-AUTH-IP checks for simplicity)
>    * If the IP is rejected by the acl and auth-call is true, it falls back to digest auth.
>
> - If accept-blind-auth is set with auth-call, freeswitch only checks if the From user is defined in directory. If so, user is authorized (without any password check)
>
> - If auth-cal is set without the acl, the call go through digest authentication
>
> - If neither is set, the call is accepted.
>
> In your case, even if you can define directly a cidr in the apply-inbound-acl param value, it would be best to set it to a list name defined in autoload_configs/acl.conf.xml.
>
> Hope this helps!

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From danielnazareth89 at gmail.com  Wed Dec  6 19:00:07 2017
From: danielnazareth89 at gmail.com (Daniel Nazareth)
Date: Wed, 6 Dec 2017 14:00:07 -0500
Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS Xenial
Message-ID: <CAM_B3yBapHfWYQRnZ81S-DUA3hJvhLOgeout63mtE-WPV5_p1g@mail.gmail.com>

Hi,

I'm having difficultly installing FS on Ubuntu 16.04 while following the
steps on the confluence page. While running apt-get update, I get *W:
http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease
<http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease>:
Signature by key 20B06EE621AB150D40F6079FD76EDC7725E010CF uses weak digest
algorithm (SHA1)*

and then

root at localhost:/usr/src# apt-get install -y freeswitch-meta-all
Reading package lists... Done
Building dependency tree
Reading state information... Done
Some packages could not be installed. This may mean that you have
requested an impossible situation or if you are using the unstable
distribution that some required packages have not yet been created
or been moved out of Incoming.
The following information may help to resolve the situation:

The following packages have unmet dependencies:
 freeswitch-meta-all : Depends: freeswitch-meta-codecs (=
1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
                       Depends: freeswitch-mod-av (=
1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
                       Depends: freeswitch-mod-soundtouch (=
1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
                       Depends: freeswitch-mod-spandsp (=
1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
                       Depends: freeswitch-mod-perl (=
1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
E: Unable to correct problems, you have held broken packages.


Anyone else having this problem or can recommend another way to install?
Thanks!

Thanks
Daniel
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From krice at freeswitch.org  Wed Dec  6 19:05:45 2017
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 6 Dec 2017 13:05:45 -0600
Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS
	Xenial
In-Reply-To: <CAM_B3yBapHfWYQRnZ81S-DUA3hJvhLOgeout63mtE-WPV5_p1g@mail.gmail.com>
References: <CAM_B3yBapHfWYQRnZ81S-DUA3hJvhLOgeout63mtE-WPV5_p1g@mail.gmail.com>
Message-ID: <2d0401d36ec5$38e571d0$aab05570$@freeswitch.org>

That’s because you are trying to install jessie packages on ubuntu. You should check out the ubuntu install pages on https://freeswitch.org/confluence

 

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Daniel Nazareth
Sent: Wednesday, December 6, 2017 1:00 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS Xenial

 

Hi,

I'm having difficultly installing FS on Ubuntu 16.04 while following the steps on the confluence page. While running apt-get update, I get W: http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease: Signature by key 20B06EE621AB150D40F6079FD76EDC7725E010CF uses weak digest algorithm (SHA1)

 

and then

 

root at localhost:/usr/src# apt-get install -y freeswitch-meta-all

Reading package lists... Done

Building dependency tree       

Reading state information... Done

Some packages could not be installed. This may mean that you have

requested an impossible situation or if you are using the unstable

distribution that some required packages have not yet been created

or been moved out of Incoming.

The following information may help to resolve the situation:

 

The following packages have unmet dependencies:

 freeswitch-meta-all : Depends: freeswitch-meta-codecs (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed

                       Depends: freeswitch-mod-av (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed

                       Depends: freeswitch-mod-soundtouch (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed

                       Depends: freeswitch-mod-spandsp (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed

                       Depends: freeswitch-mod-perl (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed

E: Unable to correct problems, you have held broken packages.

 

 

Anyone else having this problem or can recommend another way to install? Thanks!

 

Thanks

Daniel

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From danielnazareth89 at gmail.com  Wed Dec  6 19:17:35 2017
From: danielnazareth89 at gmail.com (Daniel Nazareth)
Date: Wed, 6 Dec 2017 14:17:35 -0500
Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS
	Xenial
In-Reply-To: <2d0401d36ec5$38e571d0$aab05570$@freeswitch.org>
References: <CAM_B3yBapHfWYQRnZ81S-DUA3hJvhLOgeout63mtE-WPV5_p1g@mail.gmail.com>
 <2d0401d36ec5$38e571d0$aab05570$@freeswitch.org>
Message-ID: <CAM_B3yCf8MtVsdEJzu8-2jzX8i6Nmjyy6SvgAcw6bq_kR0nDag@mail.gmail.com>

Hi Ken,

My apologies, I pasted wrong output, I tried with Debian packages as a last
resort after trying Ubuntu. When I follow specific steps for Ubuntu 16.04,
I get the below:

root at localhost:/usr/src# apt-get update && apt-get install -y
freeswitch-meta-all
Hit:1 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial InRelease
Get:2 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial-updates
InRelease [102 kB]
Get:3 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial-backports
InRelease [102 kB]
Ign:4 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie
InRelease
Ign:5 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie
Release
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main amd64 Packages
Get:7 http://security.ubuntu.com/ubuntu xenial-security InRelease [102 kB]
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main amd64 Packages
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main amd64 Packages
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en
Hit:11 http://openresty.org/package/ubuntu xenial InRelease
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main amd64 Packages
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main amd64 Packages
Hit:12 https://download.jitsi.org stable/ InRelease
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en
Err:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main amd64 Packages
  404  Not Found [IP: 209.105.235.7 80]
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
jessie/main Translation-en
Fetched 306 kB in 0s (323 kB/s)
Reading package lists... Done
*W: The repository
'http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
<http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie
Release' does not have a Release file.*
*N: Data from such a repository can't be authenticated and is therefore
potentially dangerous to use.*
*N: See apt-secure(8) manpage for repository creation and user
configuration details.*
*E: Failed to fetch
http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages
<http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages>
 404  Not Found [IP: 209.105.235.7 80]*
*E: Some index files failed to download. They have been ignored, or old
ones used instead.*

Thanks for your help.

Best
Daniel


On Wed, Dec 6, 2017 at 2:05 PM, Ken Rice <krice at freeswitch.org> wrote:

> That’s because you are trying to install jessie packages on ubuntu. You
> should check out the ubuntu install pages on https://freeswitch.org/
> confluence
>
>
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Daniel Nazareth
> *Sent:* Wednesday, December 6, 2017 1:00 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS
> Xenial
>
>
>
> Hi,
>
> I'm having difficultly installing FS on Ubuntu 16.04 while following the
> steps on the confluence page. While running apt-get update, I get *W:
> http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease
> <http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease>:
> Signature by key 20B06EE621AB150D40F6079FD76EDC7725E010CF uses weak digest
> algorithm (SHA1)*
>
>
>
> and then
>
>
>
> root at localhost:/usr/src# apt-get install -y freeswitch-meta-all
>
> Reading package lists... Done
>
> Building dependency tree
>
> Reading state information... Done
>
> Some packages could not be installed. This may mean that you have
>
> requested an impossible situation or if you are using the unstable
>
> distribution that some required packages have not yet been created
>
> or been moved out of Incoming.
>
> The following information may help to resolve the situation:
>
>
>
> The following packages have unmet dependencies:
>
>  freeswitch-meta-all : Depends: freeswitch-meta-codecs (=
> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-av (=
> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-soundtouch (=
> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-spandsp (=
> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-perl (=
> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
> E: Unable to correct problems, you have held broken packages.
>
>
>
>
>
> Anyone else having this problem or can recommend another way to install?
> Thanks!
>
>
>
> Thanks
>
> Daniel
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lexxua at gmail.com  Wed Dec  6 19:29:53 2017
From: lexxua at gmail.com (Volodymyr Fedorov)
Date: Wed, 6 Dec 2017 20:29:53 +0100
Subject: [Freeswitch-users] mod_spandsp on Windows as modem with com0com
 - port in use
In-Reply-To: <CAO_GGAoRsFppR72N-KWUmmjiw1m6hNEcaoTxNC2gjSjL76U0KQ@mail.gmail.com>
References: <000301d369a6$0007b8b0$00172a10$@gmail.com>
 <003501d36e57$ac450190$04cf04b0$@gmail.com>
 <CAO_GGAoRsFppR72N-KWUmmjiw1m6hNEcaoTxNC2gjSjL76U0KQ@mail.gmail.com>
Message-ID: <CAO_GGApvisPb9dOvK90YnfYFHr9=gt84bnUuZ_2vtLB7Jeh6YA@mail.gmail.com>

I think after resolving issue:
https://freeswitch.org/jira/browse/FS-5761

It stopped to work on Windows platforms before commits caused by that
ticket ports were used started from com5 and upwards. After commit
freeswitch wants to use com0.
You can look up as a workaround solution from Jeff Lenk:
https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_spandsp/mod_spandsp_modem.c?until=f42b17e80749250c4957601bae37b8c3ff826c39

Br,
Volodymyr

On Wed, Dec 6, 2017 at 9:22 AM, Volodymyr Fedorov <lexxua at gmail.com> wrote:

> Hi I don't remember, but in code exist commit for Linux system which was
> supposed to fix modem TTY names. But somehow it has regression on Windows
> systems because modem name could not start from 0.
> Br,
> Volodymyr
>
> On Dec 6, 2017 8:16 AM, "NYCPhoneService" <nycphoneservice at gmail.com>
> wrote:
>
>> Anyone please…
>>
>>
>>
>> *From:* NYCPhoneService [mailto:nycphoneservice at gmail.com]
>> *Sent:* Thursday, November 30, 2017 1:40 AM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* mod_spandsp on Windows as modem with com0com - port in use
>>
>>
>>
>> I guess I’m missing something with this setup:
>>
>>
>>
>> Freeswitch-1.6.19 built from source on Win 8.1, com0com 3.0 signed, pair
>> on COM6 and COM7 the and soft modem for fax server is on COM7
>>
>> I had to set number of modems to 6 in config, otherwise I was just
>> getting no modems available…
>>
>>
>>
>> Mod_spandsp sees the port:
>>
>>
>>
>> 2017-11-29 23:43:52.885349 [DEBUG] mod_spandsp_modem.c:371 Modem COM6
>> [INIT] - Changing state to INIT
>>
>> 2017-11-29 23:43:52.885349 [INFO] mod_spandsp_modem.c:373 Modem
>> [COM6]->[(null)] Ready
>>
>>
>>
>> but when I bridge to modem/6/123 I get:
>>
>>
>>
>> 2017-11-29 23:44:49.203916 [ERR] mod_spandsp_modem.c:1390 Modem COM6 In
>> Use!
>>
>> 2017-11-29 23:44:49.203916 [ERR] mod_spandsp_modem.c:1398 No Modems
>> Available!
>>
>> 971de557-adb4-4a19-9e7d-ab4f13b954be 2017-11-29 23:44:49.203916 [ERR]
>> mod_spandsp_modem.c:862 Cannot find a modem.
>>
>>
>>
>>
>>
>> The same com0com setup works with an old T38Modem with no issues.
>>
>>
>>
>> Any suggestions are really appreciated
>>
>>
>>
>> Dennis
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


-- 
Best regards,
Volodymyr
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From nycphoneservice at gmail.com  Wed Dec  6 19:51:07 2017
From: nycphoneservice at gmail.com (Nycphoneservice)
Date: Wed, 6 Dec 2017 14:51:07 -0500
Subject: [Freeswitch-users] mod_spandsp on Windows as modem with com0com
	- port in use
In-Reply-To: <CAO_GGApvisPb9dOvK90YnfYFHr9=gt84bnUuZ_2vtLB7Jeh6YA@mail.gmail.com>
References: <000301d369a6$0007b8b0$00172a10$@gmail.com>
 <003501d36e57$ac450190$04cf04b0$@gmail.com>
 <CAO_GGAoRsFppR72N-KWUmmjiw1m6hNEcaoTxNC2gjSjL76U0KQ@mail.gmail.com>
 <CAO_GGApvisPb9dOvK90YnfYFHr9=gt84bnUuZ_2vtLB7Jeh6YA@mail.gmail.com>
Message-ID: <A7D8B1C0-41EB-4CA6-A243-777DBD98663E@gmail.com>

Thank you very much - that makes sense.


Sent from my phone

> On Dec 6, 2017, at 2:29 PM, Volodymyr Fedorov <lexxua at gmail.com> wrote:
> 
> I think after resolving issue:
> https://freeswitch.org/jira/browse/FS-5761
> 
> It stopped to work on Windows platforms before commits caused by that ticket ports were used started from com5 and upwards. After commit freeswitch wants to use com0.
> You can look up as a workaround solution from Jeff Lenk:
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/diff/src/mod/applications/mod_spandsp/mod_spandsp_modem.c?until=f42b17e80749250c4957601bae37b8c3ff826c39
> 
> Br,
> Volodymyr
> 
>> On Wed, Dec 6, 2017 at 9:22 AM, Volodymyr Fedorov <lexxua at gmail.com> wrote:
>> Hi I don't remember, but in code exist commit for Linux system which was supposed to fix modem TTY names. But somehow it has regression on Windows systems because modem name could not start from 0.
>> Br,
>> Volodymyr
>> 
>>> On Dec 6, 2017 8:16 AM, "NYCPhoneService" <nycphoneservice at gmail.com> wrote:
>>> Anyone please…
>>> 
>>>  
>>> 
>>> From: NYCPhoneService [mailto:nycphoneservice at gmail.com] 
>>> Sent: Thursday, November 30, 2017 1:40 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: mod_spandsp on Windows as modem with com0com - port in use
>>> 
>>>  
>>> 
>>> I guess I’m missing something with this setup:
>>> 
>>>  
>>> 
>>> Freeswitch-1.6.19 built from source on Win 8.1, com0com 3.0 signed, pair on COM6 and COM7 the and soft modem for fax server is on COM7
>>> 
>>> I had to set number of modems to 6 in config, otherwise I was just getting no modems available…
>>> 
>>>  
>>> 
>>> Mod_spandsp sees the port:
>>> 
>>>  
>>> 
>>> 2017-11-29 23:43:52.885349 [DEBUG] mod_spandsp_modem.c:371 Modem COM6 [INIT] - Changing state to INIT
>>> 
>>> 2017-11-29 23:43:52.885349 [INFO] mod_spandsp_modem.c:373 Modem [COM6]->[(null)] Ready
>>> 
>>>  
>>> 
>>> but when I bridge to modem/6/123 I get:
>>> 
>>>  
>>> 
>>> 2017-11-29 23:44:49.203916 [ERR] mod_spandsp_modem.c:1390 Modem COM6 In Use!
>>> 
>>> 2017-11-29 23:44:49.203916 [ERR] mod_spandsp_modem.c:1398 No Modems Available!
>>> 
>>> 971de557-adb4-4a19-9e7d-ab4f13b954be 2017-11-29 23:44:49.203916 [ERR] mod_spandsp_modem.c:862 Cannot find a modem.
>>> 
>>>  
>>> 
>>>  
>>> 
>>> The same com0com setup works with an old T38Modem with no issues.
>>> 
>>>  
>>> 
>>> Any suggestions are really appreciated
>>> 
>>>  
>>> 
>>> Dennis
>>> 
>>>  
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
> 
> 
> 
> -- 
> Best regards,
> Volodymyr
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From philipp at zeitschel.net  Wed Dec  6 19:28:10 2017
From: philipp at zeitschel.net (Philipp Zeitschel)
Date: Wed, 6 Dec 2017 19:28:10 +0000
Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04
	LTS	Xenial
Message-ID: <3992600f-47a2-44ee-ba6a-5c8fdddbbe9e@zeitschel.net>

Hi,

Simply launch a Debian docker container with Host networking and Install with the Debian packages

Regards

Philipp


________________________________
Von: Daniel Nazareth <danielnazareth89 at gmail.com>
Gesendet: Mittwoch, 6. Dezember 2017 20:20
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS Xenial 

Hi Ken,

My apologies, I pasted wrong output, I tried with Debian packages as a last resort after trying Ubuntu. When I follow specific steps for Ubuntu 16.04, I get the below:

root at localhost:/usr/src# apt-get update && apt-get install -y freeswitch-meta-all
Hit:1 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial InRelease
Get:2 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial-updates InRelease [102 kB]                     
Get:3 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial-backports InRelease [102 kB]                                                    
Ign:4 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie InRelease                                                          
Ign:5 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie Release                                                     
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages          
Get:7 http://security.ubuntu.com/ubuntu xenial-security InRelease [102 kB]
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Hit:11 http://openresty.org/package/ubuntu xenial InRelease
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages
Hit:12 https://download.jitsi.org stable/ InRelease
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Err:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages
  404  Not Found [IP: 209.105.235.7 80]
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Fetched 306 kB in 0s (323 kB/s)
Reading package lists... Done
W: The repository 'http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie Release' does not have a Release file.
N: Data from such a repository can't be authenticated and is therefore potentially dangerous to use.
N: See apt-secure(8) manpage for repository creation and user configuration details.
E: Failed to fetch http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages  404  Not Found [IP: 209.105.235.7 80]
E: Some index files failed to download. They have been ignored, or old ones used instead.

Thanks for your help.

Best
Daniel


On Wed, Dec 6, 2017 at 2:05 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> That’s because you are trying to install jessie packages on ubuntu. You should check out the ubuntu install pages on https://freeswitch.org/confluence
>
>  
>
>  
>
> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Daniel Nazareth
> Sent: Wednesday, December 6, 2017 1:00 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS Xenial
>
>  
>
> Hi,
>
> I'm having difficultly installing FS on Ubuntu 16.04 while following the steps on the confluence page. While running apt-get update, I get W: http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease: Signature by key 20B06EE621AB150D40F6079FD76EDC7725E010CF uses weak digest algorithm (SHA1)
>
>  
>
> and then
>
>  
>
> root at localhost:/usr/src# apt-get install -y freeswitch-meta-all
>
> Reading package lists... Done
>
> Building dependency tree       
>
> Reading state information... Done
>
> Some packages could not be installed. This may mean that you have
>
> requested an impossible situation or if you are using the unstable
>
> distribution that some required packages have not yet been created
>
> or been moved out of Incoming.
>
> The following information may help to resolve the situation:
>
>  
>
> The following packages have unmet dependencies:
>
>  freeswitch-meta-all : Depends: freeswitch-meta-codecs (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-av (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-soundtouch (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-spandsp (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-perl (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
> E: Unable to correct problems, you have held broken packages.
>
>  
>
>  
>
> Anyone else having this problem or can recommend another way to install? Thanks!
>
>  
>
> Thanks
>
> Daniel
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From rayk at pontimax.com  Wed Dec  6 18:26:36 2017
From: rayk at pontimax.com (Ray Keating)
Date: Wed, 6 Dec 2017 13:26:36 -0500
Subject: [Freeswitch-users] Best practices for recording audio prompts
In-Reply-To: <1512536090.3964615.1195557968.20D962DC@webmail.messagingengine.com>
References: <1512536090.3964615.1195557968.20D962DC@webmail.messagingengine.com>
Message-ID: <047a01d36ebf$c244bd10$46ce3730$@pontimax.com>


Try Audacity (www.audacityteam.org).   You can set the sample rate and the various other key recording parameters.  It has a graphical
display of the recorded audio with time registration so that you can edit the prompt (to remove leading silence, for example).  Since
you can set the sample rate, you won't need to use sox to re-sample.

Use a good quality microphone and isolate it from ambient vibrations that would otherwise get recorded as background noise.

Regards,

Ray

Pontimax’s mrcpSP11-STT- the lowest cost, by far, highest recognition accuracy MRCP server available

-----Original Message-----
From: Michael Avers [mailto:michael at mailworks.org] 
Sent: Tuesday, December 5, 2017 11:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Best practices for recording audio prompts

Hello,

Can anyone share some tips about recording good audio prompts? We ideally would like to have them in a variety of sample rates (as they will be used internally as well as when callers dial in from PSTN) to avoid resampling as much as possible. What should it be recorded as originally? Would we then just use sox to resample? Any good apps for Windows? (I haven't used Windows in many years but the person who will be recording would be).

Any other considerations to take into account would be greatly appreciated.. it's our first time recording our own set of prompts.

Thanks
Mike






From mike at jerris.com  Wed Dec  6 20:48:11 2017
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Dec 2017 15:48:11 -0500
Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS
 Xenial
In-Reply-To: <CAM_B3yCf8MtVsdEJzu8-2jzX8i6Nmjyy6SvgAcw6bq_kR0nDag@mail.gmail.com>
References: <CAM_B3yBapHfWYQRnZ81S-DUA3hJvhLOgeout63mtE-WPV5_p1g@mail.gmail.com>
 <2d0401d36ec5$38e571d0$aab05570$@freeswitch.org>
 <CAM_B3yCf8MtVsdEJzu8-2jzX8i6Nmjyy6SvgAcw6bq_kR0nDag@mail.gmail.com>
Message-ID: <7B322DEC-ED2C-4134-B582-79296FFAB0E1@jerris.com>

Did you completely read this link:

https://freeswitch.org/confluence/display/FREESWITCH/Ubuntu+16.04+Xenial <https://freeswitch.org/confluence/display/FREESWITCH/Ubuntu+16.04+Xenial>


> On Dec 6, 2017, at 2:17 PM, Daniel Nazareth <danielnazareth89 at gmail.com> wrote:
> 
> Hi Ken,
> 
> My apologies, I pasted wrong output, I tried with Debian packages as a last resort after trying Ubuntu. When I follow specific steps for Ubuntu 16.04, I get the below:
> 
> root at localhost:/usr/src# apt-get update && apt-get install -y freeswitch-meta-all
> Hit:1 http://us-east-1.ec2.archive.ubuntu.com/ubuntu <http://us-east-1.ec2.archive.ubuntu.com/ubuntu> xenial InRelease
> Get:2 http://us-east-1.ec2.archive.ubuntu.com/ubuntu <http://us-east-1.ec2.archive.ubuntu.com/ubuntu> xenial-updates InRelease [102 kB]                     
> Get:3 http://us-east-1.ec2.archive.ubuntu.com/ubuntu <http://us-east-1.ec2.archive.ubuntu.com/ubuntu> xenial-backports InRelease [102 kB]                                                    
> Ign:4 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie InRelease                                                          
> Ign:5 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie Release                                                     
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages          
> Get:7 http://security.ubuntu.com/ubuntu <http://security.ubuntu.com/ubuntu> xenial-security InRelease [102 kB]
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
> Hit:11 http://openresty.org/package/ubuntu <http://openresty.org/package/ubuntu> xenial InRelease
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages
> Hit:12 https://download.jitsi.org <https://download.jitsi.org/> stable/ InRelease
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
> Err:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages
>   404  Not Found [IP: 209.105.235.7 80]
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
> Fetched 306 kB in 0s (323 kB/s)
> Reading package lists... Done
> W: The repository 'http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie Release' does not have a Release file.
> N: Data from such a repository can't be authenticated and is therefore potentially dangerous to use.
> N: See apt-secure(8) manpage for repository creation and user configuration details.
> E: Failed to fetch http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages>  404  Not Found [IP: 209.105.235.7 80]
> E: Some index files failed to download. They have been ignored, or old ones used instead.
> 
> Thanks for your help.
> 
> Best
> Daniel
> 
> 
> On Wed, Dec 6, 2017 at 2:05 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
> That’s because you are trying to install jessie packages on ubuntu. You should check out the ubuntu install pages on https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>  
> 
>  
> 
> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Daniel Nazareth
> Sent: Wednesday, December 6, 2017 1:00 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS Xenial
> 
>  
> 
> Hi,
> 
> I'm having difficultly installing FS on Ubuntu 16.04 while following the steps on the confluence page. While running apt-get update, I get W: http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease <http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease>: Signature by key 20B06EE621AB150D40F6079FD76EDC7725E010CF uses weak digest algorithm (SHA1)
> 
>  
> 
> and then
> 
>  
> 
> root at localhost:/usr/src# apt-get install -y freeswitch-meta-all
> 
> Reading package lists... Done
> 
> Building dependency tree       
> 
> Reading state information... Done
> 
> Some packages could not be installed. This may mean that you have
> 
> requested an impossible situation or if you are using the unstable
> 
> distribution that some required packages have not yet been created
> 
> or been moved out of Incoming.
> 
> The following information may help to resolve the situation:
> 
>  
> 
> The following packages have unmet dependencies:
> 
>  freeswitch-meta-all : Depends: freeswitch-meta-codecs (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
> 
>                        Depends: freeswitch-mod-av (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
> 
>                        Depends: freeswitch-mod-soundtouch (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
> 
>                        Depends: freeswitch-mod-spandsp (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
> 
>                        Depends: freeswitch-mod-perl (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
> 
> E: Unable to correct problems, you have held broken packages.
> 
>  
> 
>  
> 
> Anyone else having this problem or can recommend another way to install? Thanks!
> 
>  
> 
> Thanks
> 
> Daniel
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From danielnazareth89 at gmail.com  Wed Dec  6 21:49:30 2017
From: danielnazareth89 at gmail.com (Daniel Nazareth)
Date: Wed, 6 Dec 2017 16:49:30 -0500
Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS
	Xenial
In-Reply-To: <7B322DEC-ED2C-4134-B582-79296FFAB0E1@jerris.com>
References: <CAM_B3yBapHfWYQRnZ81S-DUA3hJvhLOgeout63mtE-WPV5_p1g@mail.gmail.com>
 <2d0401d36ec5$38e571d0$aab05570$@freeswitch.org>
 <CAM_B3yCf8MtVsdEJzu8-2jzX8i6Nmjyy6SvgAcw6bq_kR0nDag@mail.gmail.com>
 <7B322DEC-ED2C-4134-B582-79296FFAB0E1@jerris.com>
Message-ID: <CAM_B3yDOgO53RkBT2BeTH4bCHoh7GhY0W8-Su9L-pSdk+DbtEQ@mail.gmail.com>

Hi Michael,

Yes, I tried each of the three methods described on there (this is a test
server so an unstable build is ok for now)

The second method fails at apt-get update with *W:
http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/xenial/InRelease
<http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/xenial/InRelease>:
Signature by key ACAD66137D22A8A469FBB57F1FDDF413C2B201E5 uses weak digest
algorithm (SHA1)*

The third (from master branch) fails at the last step with

create mode 100644 libs/zeromq-2.1.9/tests/test_reqrep_tcp.cpp
 create mode 100644 libs/zeromq-2.1.9/tests/test_shutdown_stress.cpp
 create mode 100644 libs/zeromq-2.1.9/tests/testutil.hpp
 create mode 100755 libs/zeromq-2.1.9/version.sh
 create mode 100644 libs/zeromq-2.1.9/zeromq.spec
*xz: (stdin): Cannot allocate memory*
freeswitch-1.9.0+git~20171206T214410Z~1480362519/
freeswitch-1.9.0+git~20171206T214410Z~1480362519/.clang-format
freeswitch-1.9.0+git~20171206T214410Z~1480362519/.mailmap
freeswitch-1.9.0+git~20171206T214410Z~1480362519/.version
freeswitch-1.9.0+git~20171206T214410Z~1480362519/Freeswitch.2015.sln
*./util.sh error: unclean working tree*




cat: ../log/builds-ok.txt: No such file or directory
root at localhost:/usr/src/freeswitch-debs/freeswitch/debian#

Thanks
Daniel

On Wed, Dec 6, 2017 at 3:48 PM, Michael Jerris <mike at jerris.com> wrote:

> Did you completely read this link:
>
> https://freeswitch.org/confluence/display/FREESWITCH/Ubuntu+16.04+Xenial
>
>
> On Dec 6, 2017, at 2:17 PM, Daniel Nazareth <danielnazareth89 at gmail.com>
> wrote:
>
> Hi Ken,
>
> My apologies, I pasted wrong output, I tried with Debian packages as a
> last resort after trying Ubuntu. When I follow specific steps for Ubuntu
> 16.04, I get the below:
>
> root at localhost:/usr/src# apt-get update && apt-get install -y
> freeswitch-meta-all
> Hit:1 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial InRelease
> Get:2 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial-updates
> InRelease [102 kB]
> Get:3 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial-backports
> InRelease [102 kB]
> Ign:4 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie
> InRelease
> Ign:5 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie
> Release
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
> Get:7 http://security.ubuntu.com/ubuntu xenial-security InRelease [102 kB]
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
> Hit:11 http://openresty.org/package/ubuntu xenial InRelease
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
> Hit:12 https://download.jitsi.org stable/ InRelease
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
> Err:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
>   404  Not Found [IP: 209.105.235.7 80]
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
> Fetched 306 kB in 0s (323 kB/s)
> Reading package lists... Done
> *W: The repository
> 'http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie
> Release' does not have a Release file.*
> *N: Data from such a repository can't be authenticated and is therefore
> potentially dangerous to use.*
> *N: See apt-secure(8) manpage for repository creation and user
> configuration details.*
> *E: Failed to fetch
> http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages
> <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages>
>  404  Not Found [IP: 209.105.235.7 80]*
> *E: Some index files failed to download. They have been ignored, or old
> ones used instead.*
>
> Thanks for your help.
>
> Best
> Daniel
>
>
> On Wed, Dec 6, 2017 at 2:05 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> That’s because you are trying to install jessie packages on ubuntu. You
>> should check out the ubuntu install pages on
>> https://freeswitch.org/confluence
>>
>>
>>
>>
>>
>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
>> es at lists.freeswitch.org] *On Behalf Of *Daniel Nazareth
>> *Sent:* Wednesday, December 6, 2017 1:00 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS
>> Xenial
>>
>>
>>
>> Hi,
>>
>> I'm having difficultly installing FS on Ubuntu 16.04 while following the
>> steps on the confluence page. While running apt-get update, I get *W:
>> http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease
>> <http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease>:
>> Signature by key 20B06EE621AB150D40F6079FD76EDC7725E010CF uses weak digest
>> algorithm (SHA1)*
>>
>>
>>
>> and then
>>
>>
>>
>> root at localhost:/usr/src# apt-get install -y freeswitch-meta-all
>>
>> Reading package lists... Done
>>
>> Building dependency tree
>>
>> Reading state information... Done
>>
>> Some packages could not be installed. This may mean that you have
>>
>> requested an impossible situation or if you are using the unstable
>>
>> distribution that some required packages have not yet been created
>>
>> or been moved out of Incoming.
>>
>> The following information may help to resolve the situation:
>>
>>
>>
>> The following packages have unmet dependencies:
>>
>>  freeswitch-meta-all : Depends: freeswitch-meta-codecs (=
>> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>>
>>                        Depends: freeswitch-mod-av (=
>> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>>
>>                        Depends: freeswitch-mod-soundtouch (=
>> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>>
>>                        Depends: freeswitch-mod-spandsp (=
>> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>>
>>                        Depends: freeswitch-mod-perl (=
>> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>>
>> E: Unable to correct problems, you have held broken packages.
>>
>>
>>
>>
>>
>> Anyone else having this problem or can recommend another way to install?
>> Thanks!
>>
>>
>>
>> Thanks
>>
>> Daniel
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Dec  6 22:32:04 2017
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Dec 2017 17:32:04 -0500
Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS
 Xenial
In-Reply-To: <CAM_B3yDOgO53RkBT2BeTH4bCHoh7GhY0W8-Su9L-pSdk+DbtEQ@mail.gmail.com>
References: <CAM_B3yBapHfWYQRnZ81S-DUA3hJvhLOgeout63mtE-WPV5_p1g@mail.gmail.com>
 <2d0401d36ec5$38e571d0$aab05570$@freeswitch.org>
 <CAM_B3yCf8MtVsdEJzu8-2jzX8i6Nmjyy6SvgAcw6bq_kR0nDag@mail.gmail.com>
 <7B322DEC-ED2C-4134-B582-79296FFAB0E1@jerris.com>
 <CAM_B3yDOgO53RkBT2BeTH4bCHoh7GhY0W8-Su9L-pSdk+DbtEQ@mail.gmail.com>
Message-ID: <539904E3-AD5E-49E6-A2CA-A826FF714FA9@jerris.com>

This sounds like you are running out of ram so its failing.  

> On Dec 6, 2017, at 4:49 PM, Daniel Nazareth <danielnazareth89 at gmail.com> wrote:
> 
> Hi Michael,
> 
> Yes, I tried each of the three methods described on there (this is a test server so an unstable build is ok for now)
> 
> The second method fails at apt-get update with W: http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/xenial/InRelease <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/xenial/InRelease>: Signature by key ACAD66137D22A8A469FBB57F1FDDF413C2B201E5 uses weak digest algorithm (SHA1)
> 
> The third (from master branch) fails at the last step with 
> 
> create mode 100644 libs/zeromq-2.1.9/tests/test_reqrep_tcp.cpp
>  create mode 100644 libs/zeromq-2.1.9/tests/test_shutdown_stress.cpp
>  create mode 100644 libs/zeromq-2.1.9/tests/testutil.hpp
>  create mode 100755 libs/zeromq-2.1.9/version.sh
>  create mode 100644 libs/zeromq-2.1.9/zeromq.spec
> xz: (stdin): Cannot allocate memory
> freeswitch-1.9.0+git~20171206T214410Z~1480362519/
> freeswitch-1.9.0+git~20171206T214410Z~1480362519/.clang-format
> freeswitch-1.9.0+git~20171206T214410Z~1480362519/.mailmap
> freeswitch-1.9.0+git~20171206T214410Z~1480362519/.version
> freeswitch-1.9.0+git~20171206T214410Z~1480362519/Freeswitch.2015.sln
> ./util.sh error: unclean working tree
> 
> 
> 
> 
> cat: ../log/builds-ok.txt: No such file or directory
> root at localhost:/usr/src/freeswitch-debs/freeswitch/debian#
> 
> Thanks 
> Daniel
> 
> On Wed, Dec 6, 2017 at 3:48 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Did you completely read this link:
> 
> https://freeswitch.org/confluence/display/FREESWITCH/Ubuntu+16.04+Xenial <https://freeswitch.org/confluence/display/FREESWITCH/Ubuntu+16.04+Xenial>
> 
> 
>> On Dec 6, 2017, at 2:17 PM, Daniel Nazareth <danielnazareth89 at gmail.com <mailto:danielnazareth89 at gmail.com>> wrote:
>> 
>> Hi Ken,
>> 
>> My apologies, I pasted wrong output, I tried with Debian packages as a last resort after trying Ubuntu. When I follow specific steps for Ubuntu 16.04, I get the below:
>> 
>> root at localhost:/usr/src# apt-get update && apt-get install -y freeswitch-meta-all
>> Hit:1 http://us-east-1.ec2.archive.ubuntu.com/ubuntu <http://us-east-1.ec2.archive.ubuntu.com/ubuntu> xenial InRelease
>> Get:2 http://us-east-1.ec2.archive.ubuntu.com/ubuntu <http://us-east-1.ec2.archive.ubuntu.com/ubuntu> xenial-updates InRelease [102 kB]                     
>> Get:3 http://us-east-1.ec2.archive.ubuntu.com/ubuntu <http://us-east-1.ec2.archive.ubuntu.com/ubuntu> xenial-backports InRelease [102 kB]                                                    
>> Ign:4 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie InRelease                                                          
>> Ign:5 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie Release                                                     
>> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages          
>> Get:7 http://security.ubuntu.com/ubuntu <http://security.ubuntu.com/ubuntu> xenial-security InRelease [102 kB]
>> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
>> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
>> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
>> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages
>> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
>> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
>> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
>> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages
>> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
>> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
>> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
>> Hit:11 http://openresty.org/package/ubuntu <http://openresty.org/package/ubuntu> xenial InRelease
>> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages
>> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
>> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
>> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
>> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages
>> Hit:12 https://download.jitsi.org <https://download.jitsi.org/> stable/ InRelease
>> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
>> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
>> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
>> Err:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main amd64 Packages
>>   404  Not Found [IP: 209.105.235.7 80]
>> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main all Packages
>> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en_US
>> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie/main Translation-en
>> Fetched 306 kB in 0s (323 kB/s)
>> Reading package lists... Done
>> W: The repository 'http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie Release' does not have a Release file.
>> N: Data from such a repository can't be authenticated and is therefore potentially dangerous to use.
>> N: See apt-secure(8) manpage for repository creation and user configuration details.
>> E: Failed to fetch http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages>  404  Not Found [IP: 209.105.235.7 80]
>> E: Some index files failed to download. They have been ignored, or old ones used instead.
>> 
>> Thanks for your help.
>> 
>> Best
>> Daniel
>> 
>> 
>> On Wed, Dec 6, 2017 at 2:05 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
>> That’s because you are trying to install jessie packages on ubuntu. You should check out the ubuntu install pages on https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>>  
>> 
>>  
>> 
>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Daniel Nazareth
>> Sent: Wednesday, December 6, 2017 1:00 PM
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
>> Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS Xenial
>> 
>>  
>> 
>> Hi,
>> 
>> I'm having difficultly installing FS on Ubuntu 16.04 while following the steps on the confluence page. While running apt-get update, I get W: http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease <http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease>: Signature by key 20B06EE621AB150D40F6079FD76EDC7725E010CF uses weak digest algorithm (SHA1)
>> 
>>  
>> 
>> and then
>> 
>>  
>> 
>> root at localhost:/usr/src# apt-get install -y freeswitch-meta-all
>> 
>> Reading package lists... Done
>> 
>> Building dependency tree       
>> 
>> Reading state information... Done
>> 
>> Some packages could not be installed. This may mean that you have
>> 
>> requested an impossible situation or if you are using the unstable
>> 
>> distribution that some required packages have not yet been created
>> 
>> or been moved out of Incoming.
>> 
>> The following information may help to resolve the situation:
>> 
>>  
>> 
>> The following packages have unmet dependencies:
>> 
>>  freeswitch-meta-all : Depends: freeswitch-meta-codecs (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>> 
>>                        Depends: freeswitch-mod-av (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>> 
>>                        Depends: freeswitch-mod-soundtouch (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>> 
>>                        Depends: freeswitch-mod-spandsp (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>> 
>>                        Depends: freeswitch-mod-perl (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>> 
>> E: Unable to correct problems, you have held broken packages.
>> 
>>  
>> 
>>  
>> 
>> Anyone else having this problem or can recommend another way to install? Thanks!
>> 
>>  
>> 
>> Thanks
>> 
>> Daniel
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From marcel.haldemann at convercom.ch  Wed Dec  6 22:33:26 2017
From: marcel.haldemann at convercom.ch (Marcel Haldemann)
Date: Wed, 6 Dec 2017 22:33:26 +0000
Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04
	LTS	Xenial
In-Reply-To: <CAM_B3yDOgO53RkBT2BeTH4bCHoh7GhY0W8-Su9L-pSdk+DbtEQ@mail.gmail.com>
References: <CAM_B3yBapHfWYQRnZ81S-DUA3hJvhLOgeout63mtE-WPV5_p1g@mail.gmail.com>
 <2d0401d36ec5$38e571d0$aab05570$@freeswitch.org>
 <CAM_B3yCf8MtVsdEJzu8-2jzX8i6Nmjyy6SvgAcw6bq_kR0nDag@mail.gmail.com>
 <7B322DEC-ED2C-4134-B582-79296FFAB0E1@jerris.com>
 <CAM_B3yDOgO53RkBT2BeTH4bCHoh7GhY0W8-Su9L-pSdk+DbtEQ@mail.gmail.com>
Message-ID: <HE1PR10MB038029A91379DCF797F21881EF320@HE1PR10MB0380.EURPRD10.PROD.OUTLOOK.COM>

Hi,

Don’t know whether this is the right path for u but here is how I do it:

As FS is best working on Debian 8 I’m using Docker to run FreeSWITCH. Thus I can install it on almost any Linux Distro (any that can run Docker).
Using Network=”Host” in Docker u don’t lose any performance. And u can use a volume mapping if u want to be able to edit config files directly on the Docker Host Machine.
Meanwhile I got a yml file for Docker and can install the entire System including Freeswitch, Postgres, nginx, DotnetCore and more with one single command line (once Docker is installed)

Here is a Docker File I use to create:
I’m copying my config files into it and put it into a private registry.
FROM bitnami/minideb:jessie

# basic
RUN echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main" > /etc/apt/sources.list.d/freeswitch.list \
                  && install_packages --force-yes --no-install-recommends \
                  freeswitch \
                  freeswitch-mod-event-socket \
                  freeswitch-mod-commands \
                  freeswitch-mod-console \
                  freeswitch-mod-dialplan-xml \
                  freeswitch-mod-dptools \
                  freeswitch-mod-posix-timer \
                  freeswitch-mod-sofia \
                  libpq5 \
                  && rm -rf /var/lib/apt/lists/*

# webRTC
RUN install_packages --force-yes --no-install-recommends \
                  freeswitch-mod-av \
                  freeswitch-mod-rtc \
                  freeswitch-mod-verto \
                  && rm -rf /var/lib/apt/lists/*

# programming
RUN install_packages --force-yes --no-install-recommends \
                  freeswitch-mod-lua \
                  freeswitch-mod-curl \
                  && rm -rf /var/lib/apt/lists/*

# cdr
RUN install_packages --force-yes --no-install-recommends \
                  freeswitch-mod-cdr-pg-csv \
                  && rm -rf /var/lib/apt/lists/*

# ADD to paly files
RUN install_packages --force-yes --no-install-recommends \
                  freeswitch-mod-sndfile \
                  && rm -rf /var/lib/apt/lists/*

# copy our config dir over the existing one
WORKDIR /etc/freeswitch

ADD . /etc/freeswitch

CMD ["freeswitch", “-nonat”]
# OR:
#
#COPY ./startup.sh /
#RUN chmod +x /startup.sh
#CMD ["/startup.sh"]

I’m using yml files to deploy the entire System. So can install Freeswitch including postgres, nginx and others with one command.
PS: To run yml files u must put Docker into swarm mode, but u can do this also on a single machine.

Here some part of my yml file (using startup.sh to put env variables into the config):

….
networks:
  outside:
    external:
      name: "host"

Services:
  rtc-server:
    image: support.convercom.com:7000/csf/rtc-server
    networks:
      - outside
    secrets:
      - source: cert_wildcard_domain
        target: /etc/freeswitch/tls/dtls-srtp.pem
        mode: 0444
    environment:
      - anum=+111111111
      - carrier1_url=xxxx.com
      - carrier1_user=2222222
      - carrier1_password=3333333
      - sip_port=5060
      - pg_conn_string=host=127.0.0.1 port=5432 dbname=xxxx user=yyyyy password=zzzzzzzzzz connect_timeout=10
    deploy:
      mode: global
      restart_policy:
        condition: any
      update_config:
        parallelism: 1
        delay: 30s
…

If u want to go this path I can give u some more details on how to set it up (if aren’t already using docker it’s about time).

Just my 2 cents


Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Daniel Nazareth
Gesendet: Mittwoch, 6. Dezember 2017 22:50
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS Xenial

Hi Michael,

Yes, I tried each of the three methods described on there (this is a test server so an unstable build is ok for now)

The second method fails at apt-get update with W: http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/xenial/InRelease: Signature by key ACAD66137D22A8A469FBB57F1FDDF413C2B201E5 uses weak digest algorithm (SHA1)

The third (from master branch) fails at the last step with

create mode 100644 libs/zeromq-2.1.9/tests/test_reqrep_tcp.cpp
 create mode 100644 libs/zeromq-2.1.9/tests/test_shutdown_stress.cpp
 create mode 100644 libs/zeromq-2.1.9/tests/testutil.hpp
 create mode 100755 libs/zeromq-2.1.9/version.sh
 create mode 100644 libs/zeromq-2.1.9/zeromq.spec
xz: (stdin): Cannot allocate memory
freeswitch-1.9.0+git~20171206T214410Z~1480362519/
freeswitch-1.9.0+git~20171206T214410Z~1480362519/.clang-format
freeswitch-1.9.0+git~20171206T214410Z~1480362519/.mailmap
freeswitch-1.9.0+git~20171206T214410Z~1480362519/.version
freeswitch-1.9.0+git~20171206T214410Z~1480362519/Freeswitch.2015.sln
./util.sh error: unclean working tree




cat: ../log/builds-ok.txt: No such file or directory
root at localhost:/usr/src/freeswitch-debs/freeswitch/debian#

Thanks
Daniel

On Wed, Dec 6, 2017 at 3:48 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
Did you completely read this link:

https://freeswitch.org/confluence/display/FREESWITCH/Ubuntu+16.04+Xenial


On Dec 6, 2017, at 2:17 PM, Daniel Nazareth <danielnazareth89 at gmail.com<mailto:danielnazareth89 at gmail.com>> wrote:

Hi Ken,

My apologies, I pasted wrong output, I tried with Debian packages as a last resort after trying Ubuntu. When I follow specific steps for Ubuntu 16.04, I get the below:

root at localhost:/usr/src# apt-get update && apt-get install -y freeswitch-meta-all
Hit:1 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial InRelease
Get:2 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial-updates InRelease [102 kB]
Get:3 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial-backports InRelease [102 kB]
Ign:4 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie InRelease
Ign:5 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie Release
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages
Get:7 http://security.ubuntu.com/ubuntu xenial-security InRelease [102 kB]
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Hit:11 http://openresty.org/package/ubuntu xenial InRelease
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages
Hit:12 https://download.jitsi.org<https://download.jitsi.org/> stable/ InRelease
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Err:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main amd64 Packages
  404  Not Found [IP: 209.105.235.7 80]
Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main all Packages
Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en_US
Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie/main Translation-en
Fetched 306 kB in 0s (323 kB/s)
Reading package lists... Done
W: The repository 'http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie Release' does not have a Release file.
N: Data from such a repository can't be authenticated and is therefore potentially dangerous to use.
N: See apt-secure(8) manpage for repository creation and user configuration details.
E: Failed to fetch http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages  404  Not Found [IP: 209.105.235.7 80]
E: Some index files failed to download. They have been ignored, or old ones used instead.

Thanks for your help.

Best
Daniel


On Wed, Dec 6, 2017 at 2:05 PM, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:
That’s because you are trying to install jessie packages on ubuntu. You should check out the ubuntu install pages on https://freeswitch.org/confluence


From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Daniel Nazareth
Sent: Wednesday, December 6, 2017 1:00 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS Xenial

Hi,

I'm having difficultly installing FS on Ubuntu 16.04 while following the steps on the confluence page. While running apt-get update, I get W: http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease: Signature by key 20B06EE621AB150D40F6079FD76EDC7725E010CF uses weak digest algorithm (SHA1)

and then

root at localhost:/usr/src# apt-get install -y freeswitch-meta-all
Reading package lists... Done
Building dependency tree
Reading state information... Done
Some packages could not be installed. This may mean that you have
requested an impossible situation or if you are using the unstable
distribution that some required packages have not yet been created
or been moved out of Incoming.
The following information may help to resolve the situation:

The following packages have unmet dependencies:
 freeswitch-meta-all : Depends: freeswitch-meta-codecs (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
                       Depends: freeswitch-mod-av (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
                       Depends: freeswitch-mod-soundtouch (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
                       Depends: freeswitch-mod-spandsp (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
                       Depends: freeswitch-mod-perl (= 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
E: Unable to correct problems, you have held broken packages.


Anyone else having this problem or can recommend another way to install? Thanks!

Thanks
Daniel

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
http://confluence.freeswitch.org<http://confluence.freeswitch.org/>
http://www.cluecon.com<http://www.cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org<http://www.freeswitch.org/>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From danielnazareth89 at gmail.com  Wed Dec  6 23:03:26 2017
From: danielnazareth89 at gmail.com (Daniel Nazareth)
Date: Wed, 6 Dec 2017 18:03:26 -0500
Subject: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS
	Xenial
In-Reply-To: <HE1PR10MB038029A91379DCF797F21881EF320@HE1PR10MB0380.EURPRD10.PROD.OUTLOOK.COM>
References: <CAM_B3yBapHfWYQRnZ81S-DUA3hJvhLOgeout63mtE-WPV5_p1g@mail.gmail.com>
 <2d0401d36ec5$38e571d0$aab05570$@freeswitch.org>
 <CAM_B3yCf8MtVsdEJzu8-2jzX8i6Nmjyy6SvgAcw6bq_kR0nDag@mail.gmail.com>
 <7B322DEC-ED2C-4134-B582-79296FFAB0E1@jerris.com>
 <CAM_B3yDOgO53RkBT2BeTH4bCHoh7GhY0W8-Su9L-pSdk+DbtEQ@mail.gmail.com>
 <HE1PR10MB038029A91379DCF797F21881EF320@HE1PR10MB0380.EURPRD10.PROD.OUTLOOK.COM>
Message-ID: <CAM_B3yBDPz_mGn+mLvNJAUNFs_Sf86vGthJz5PiEqtbJe8hnJA@mail.gmail.com>

Thanks all for advice. I think i will take a couple of the suggestions here
and run it in a Docker container for my testing.

Best
Daniel

On Wed, Dec 6, 2017 at 5:33 PM, Marcel Haldemann <
marcel.haldemann at convercom.ch> wrote:

> Hi,
>
>
>
> Don’t know whether this is the right path for u but here is how I do it:
>
>
>
> As FS is best working on Debian 8 I’m using Docker to run FreeSWITCH. Thus
> I can install it on almost any Linux Distro (any that can run Docker).
>
> Using Network=”Host” in Docker u don’t lose any performance. And u can use
> a volume mapping if u want to be able to edit config files directly on the
> Docker Host Machine.
>
> Meanwhile I got a yml file for Docker and can install the entire System
> including Freeswitch, Postgres, nginx, DotnetCore and more with one single
> command line (once Docker is installed)
>
>
>
> Here is a Docker File I use to create:
>
> I’m copying my config files into it and put it into a private registry.
>
> FROM bitnami/minideb:jessie
>
>
>
> # basic
>
> RUN echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie
> main" > /etc/apt/sources.list.d/freeswitch.list \
>
>                   && install_packages --force-yes --no-install-recommends \
>
>                   freeswitch \
>
>                   freeswitch-mod-event-socket \
>
>                   freeswitch-mod-commands \
>
>                   freeswitch-mod-console \
>
>                   freeswitch-mod-dialplan-xml \
>
>                   freeswitch-mod-dptools \
>
>                   freeswitch-mod-posix-timer \
>
>                   freeswitch-mod-sofia \
>
>                   libpq5 \
>
>                   && rm -rf /var/lib/apt/lists/*
>
>
>
> # webRTC
>
> RUN install_packages --force-yes --no-install-recommends \
>
>                   freeswitch-mod-av \
>
>                   freeswitch-mod-rtc \
>
>                   freeswitch-mod-verto \
>
>                   && rm -rf /var/lib/apt/lists/*
>
>
>
> # programming
>
> RUN install_packages --force-yes --no-install-recommends \
>
>                   freeswitch-mod-lua \
>
>                   freeswitch-mod-curl \
>
>                   && rm -rf /var/lib/apt/lists/*
>
>
>
> # cdr
>
> RUN install_packages --force-yes --no-install-recommends \
>
>                   freeswitch-mod-cdr-pg-csv \
>
>                   && rm -rf /var/lib/apt/lists/*
>
>
>
> # ADD to paly files
>
> RUN install_packages --force-yes --no-install-recommends \
>
>                   freeswitch-mod-sndfile \
>
>                   && rm -rf /var/lib/apt/lists/*
>
>
>
> # copy our config dir over the existing one
>
> WORKDIR /etc/freeswitch
>
>
>
> ADD . /etc/freeswitch
>
>
>
> CMD ["freeswitch", “-nonat”]
>
> # OR:
>
> #
>
> #COPY ./startup.sh /
>
> #RUN chmod +x /startup.sh
>
> #CMD ["/startup.sh"]
>
>
>
> I’m using yml files to deploy the entire System. So can install Freeswitch
> including postgres, nginx and others with one command.
>
> PS: To run yml files u must put Docker into swarm mode, but u can do this
> also on a single machine.
>
>
>
> Here some part of my yml file (using startup.sh to put env variables into
> the config):
>
>
>
> ….
>
> networks:
>
>   outside:
>
>     external:
>
>       name: "host"
>
>
>
> Services:
>
>   rtc-server:
>
>     image: support.convercom.com:7000/csf/rtc-server
>
>     networks:
>
>       - outside
>
>     secrets:
>
>       - source: cert_wildcard_domain
>
>         target: /etc/freeswitch/tls/dtls-srtp.pem
>
>         mode: 0444
>
>     environment:
>
>       - anum=+111111111
>
>       - carrier1_url=xxxx.com
>
>       - carrier1_user=2222222
>
>       - carrier1_password=3333333
>
>       - sip_port=5060
>
>       - pg_conn_string=host=127.0.0.1 port=5432 dbname=xxxx user=yyyyy
> password=zzzzzzzzzz connect_timeout=10
>
>     deploy:
>
>       mode: global
>
>       restart_policy:
>
>         condition: any
>
>       update_config:
>
>         parallelism: 1
>
>         delay: 30s
>
> …
>
>
>
> If u want to go this path I can give u some more details on how to set it
> up (if aren’t already using docker it’s about time).
>
>
>
> Just my 2 cents
>
>
>
>
>
> *Von:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *Im Auftrag von *Daniel Nazareth
> *Gesendet:* Mittwoch, 6. Dezember 2017 22:50
> *An:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Betreff:* Re: [Freeswitch-users] Installing freeswitch on Ubuntu 16.04
> LTS Xenial
>
>
>
> Hi Michael,
>
>
>
> Yes, I tried each of the three methods described on there (this is a test
> server so an unstable build is ok for now)
>
>
>
> The second method fails at apt-get update with *W:
> http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/xenial/InRelease
> <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/xenial/InRelease>:
> Signature by key ACAD66137D22A8A469FBB57F1FDDF413C2B201E5 uses weak digest
> algorithm (SHA1)*
>
>
>
> The third (from master branch) fails at the last step with
>
>
>
> create mode 100644 libs/zeromq-2.1.9/tests/test_reqrep_tcp.cpp
>
>  create mode 100644 libs/zeromq-2.1.9/tests/test_shutdown_stress.cpp
>
>  create mode 100644 libs/zeromq-2.1.9/tests/testutil.hpp
>
>  create mode 100755 libs/zeromq-2.1.9/version.sh
>
>  create mode 100644 libs/zeromq-2.1.9/zeromq.spec
>
> *xz: (stdin): Cannot allocate memory*
>
> freeswitch-1.9.0+git~20171206T214410Z~1480362519/
>
> freeswitch-1.9.0+git~20171206T214410Z~1480362519/.clang-format
>
> freeswitch-1.9.0+git~20171206T214410Z~1480362519/.mailmap
>
> freeswitch-1.9.0+git~20171206T214410Z~1480362519/.version
>
> freeswitch-1.9.0+git~20171206T214410Z~1480362519/Freeswitch.2015.sln
>
> *./util.sh error: unclean working tree*
>
>
>
>
>
>
>
>
>
> cat: ../log/builds-ok.txt: No such file or directory
>
> root at localhost:/usr/src/freeswitch-debs/freeswitch/debian#
>
>
>
> Thanks
>
> Daniel
>
>
>
> On Wed, Dec 6, 2017 at 3:48 PM, Michael Jerris <mike at jerris.com> wrote:
>
> Did you completely read this link:
>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Ubuntu+16.04+Xenial
>
>
>
>
>
> On Dec 6, 2017, at 2:17 PM, Daniel Nazareth <danielnazareth89 at gmail.com>
> wrote:
>
>
>
> Hi Ken,
>
>
>
> My apologies, I pasted wrong output, I tried with Debian packages as a
> last resort after trying Ubuntu. When I follow specific steps for Ubuntu
> 16.04, I get the below:
>
>
>
> root at localhost:/usr/src# apt-get update && apt-get install -y
> freeswitch-meta-all
>
> Hit:1 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial InRelease
>
> Get:2 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial-updates
> InRelease [102 kB]
>
> Get:3 http://us-east-1.ec2.archive.ubuntu.com/ubuntu xenial-backports
> InRelease [102 kB]
>
> Ign:4 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie
> InRelease
>
> Ign:5 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6 jessie
> Release
>
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
>
> Get:7 http://security.ubuntu.com/ubuntu xenial-security InRelease [102 kB]
>
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
>
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
>
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
>
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
>
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
>
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
>
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
>
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
>
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
>
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
>
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
>
> Hit:11 http://openresty.org/package/ubuntu xenial InRelease
>
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
>
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
>
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
>
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
>
> Ign:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
>
> Hit:12 https://download.jitsi.org stable/ InRelease
>
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
>
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
>
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
>
> Err:6 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main amd64 Packages
>
>   404  Not Found [IP: 209.105.235.7 80]
>
> Ign:8 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main all Packages
>
> Ign:9 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en_US
>
> Ign:10 http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> jessie/main Translation-en
>
> Fetched 306 kB in 0s (323 kB/s)
>
> Reading package lists... Done
>
> *W: The repository
> 'http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6
> <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6> jessie
> Release' does not have a Release file.*
>
> *N: Data from such a repository can't be authenticated and is therefore
> potentially dangerous to use.*
>
> *N: See apt-secure(8) manpage for repository creation and user
> configuration details.*
>
> *E: Failed to fetch
> http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages
> <http://files.freeswitch.org/repo/ubuntu-1604/freeswitch-1.6/dists/jessie/main/binary-amd64/Packages>
>  404  Not Found [IP: 209.105.235.7 80]*
>
> *E: Some index files failed to download. They have been ignored, or old
> ones used instead.*
>
>
>
> Thanks for your help.
>
>
>
> Best
>
> Daniel
>
>
>
>
>
> On Wed, Dec 6, 2017 at 2:05 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> That’s because you are trying to install jessie packages on ubuntu. You
> should check out the ubuntu install pages on https://freeswitch.org/
> confluence
>
>
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Daniel Nazareth
> *Sent:* Wednesday, December 6, 2017 1:00 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Installing freeswitch on Ubuntu 16.04 LTS
> Xenial
>
>
>
> Hi,
>
> I'm having difficultly installing FS on Ubuntu 16.04 while following the
> steps on the confluence page. While running apt-get update, I get *W:
> http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease
> <http://files.freeswitch.org/repo/deb/freeswitch-1.6/dists/jessie/InRelease>:
> Signature by key 20B06EE621AB150D40F6079FD76EDC7725E010CF uses weak digest
> algorithm (SHA1)*
>
>
>
> and then
>
>
>
> root at localhost:/usr/src# apt-get install -y freeswitch-meta-all
>
> Reading package lists... Done
>
> Building dependency tree
>
> Reading state information... Done
>
> Some packages could not be installed. This may mean that you have
>
> requested an impossible situation or if you are using the unstable
>
> distribution that some required packages have not yet been created
>
> or been moved out of Incoming.
>
> The following information may help to resolve the situation:
>
>
>
> The following packages have unmet dependencies:
>
>  freeswitch-meta-all : Depends: freeswitch-meta-codecs (=
> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-av (=
> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-soundtouch (=
> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-spandsp (=
> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
>                        Depends: freeswitch-mod-perl (=
> 1.6.19~36~7a77e0b-1~jessie+1) but it is not going to be installed
>
> E: Unable to correct problems, you have held broken packages.
>
>
>
>
>
> Anyone else having this problem or can recommend another way to install?
> Thanks!
>
>
>
> Thanks
>
> Daniel
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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From michael at mailworks.org  Thu Dec  7 06:52:35 2017
From: michael at mailworks.org (Michael Avers)
Date: Wed, 06 Dec 2017 23:52:35 -0700
Subject: [Freeswitch-users] VOIP on Google Cloud
Message-ID: <1512629555.2948955.1196917528.3E95C516@webmail.messagingengine.com>

Hey, anybody running Freeswitch on Google Compute Engine? How is it working out? 

Mike


From SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com  Thu Dec  7 16:18:16 2017
From: SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com (Karanath sachidanandan, Sanooj)
Date: Thu, 7 Dec 2017 16:18:16 +0000
Subject: [Freeswitch-users] [EXTERNAL] Re: Question on integrating with
 google speech api
In-Reply-To: <CAEJMVkC0wukWQDo9BXySXiYLU=3eTJ6+FKjzxrUb0XZomfdsaw@mail.gmail.com>
References: <C02C12DA-BD47-4C9B-8DCB-8BB1453DE32D@homedepot.com>
 <CADcg8iVa-GiXucJ7E1GahiZWbcjc3Mc-+K=bfWBmhFUpnFRmfw@mail.gmail.com>
 <CBD231EA-AF90-4FBF-AE69-001554072914@homedepot.com>
 <CAEJMVkC0wukWQDo9BXySXiYLU=3eTJ6+FKjzxrUb0XZomfdsaw@mail.gmail.com>
Message-ID: <769019EA-AC3D-4BD5-A427-E04E51C3F94E@homedepot.com>

Thanks , we are able to get it working

Regards
Sanooj


From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Brian West <brian at freeswitch.com>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: Thursday, November 30, 2017 at 9:45 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [EXTERNAL] Re: [Freeswitch-users] Question on integrating with google speech api

There should be examples of interfacing with Nuance in our mrcp_profiles directory.

On Mon, Nov 20, 2017 at 8:45 AM Karanath sachidanandan, Sanooj <SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com<mailto:SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com>> wrote:
Thanks Joshua, but that is a licensed product!

Regards
Sanooj


From: Joshua Gigg <giggsey at gmail.com<mailto:giggsey at gmail.com>>
Date: Monday, November 20, 2017 at 4:51 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Question on integrating with google speech api

Not sure about the JavaScript getting the session audio, but we now use https://www.unimrcp.org/gsr<https://urldefense.proofpoint.com/v2/url?u=https-3A__www.unimrcp.org_gsr&d=DwMFaQ&c=MtgQEAMQGqekjTjiAhkudQ&r=S9Crna1EYAWVa5Sef9w2iP8TMgbiCQCxyKSTDJj2DFN4Zz0WjO7sDM6LUQr35ORe&m=JxMXx-BlEtxBaHzOrxQLA-ibLnCaZGa4jg7M7IyJr_w&s=5sxHQ36AjEinMBYfYBiGe5T-2BJTbeS7F2uy_HRO9A4&e=>

On Sun, 19 Nov 2017 at 21:20 Karanath sachidanandan, Sanooj <SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com<mailto:SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com>> wrote:
Hi Freeswitch users,

We are trying to integrate  Freeswitch  to  google speech api , is there a way to get a handle of audio stream in javascript ?

Regards
Sanooj


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--
[http://24.112.99.44/fss.png]

Brian West | General Operations Director

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3D17345-2BCivic-2BDrive-2B-25232531-2BBrookfield-2C-2BWI-2B53045-26entry-3Dgmail-26source-3Dg&d=DwMFaQ&c=MtgQEAMQGqekjTjiAhkudQ&r=S9Crna1EYAWVa5Sef9w2iP8TMgbiCQCxyKSTDJj2DFN4Zz0WjO7sDM6LUQr35ORe&m=AIyHAXtZ1Lg2WA173tuiDMKkGjGX6iSBs4LWcBDp_Fk&s=ahTvhSPZfMq9WoPLU8p2oq3qJ8wETJppTuy2RbWTsEU&e=>

Email: brian at freeswitch.com<mailto:brian at freeswitch.com>

Mobile: 918-424-9378

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________________________________

The information in this Internet Email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this Email by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients any opinions or advice contained in this Email are subject to the terms and conditions expressed in any applicable governing The Home Depot terms of business or client engagement letter. The Home Depot disclaims all responsibility and liability for the accuracy and content of this attachment and for any damages or losses arising from any inaccuracies, errors, viruses, e.g., worms, trojan horses, etc., or other items of a destructive nature, which may be contained in this attachment and shall not be liable for direct, indirect, consequential or special damages in connection with this e-mail message or its attachment.
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From callum.guy at x-on.co.uk  Thu Dec  7 17:05:41 2017
From: callum.guy at x-on.co.uk (Callum Guy)
Date: Thu, 07 Dec 2017 17:05:41 +0000
Subject: [Freeswitch-users] T38 Fax Fallback
Message-ID: <CAFjCFzn_xaFEs_=ow8tbCENBW0byUPBTC2VPVH9d0zF9Whanxw@mail.gmail.com>

Hi All,

I'll try and keep this simple.

We are experiencing difficulty transmitting fax with a certain UK provider
(BT) who have suggested the fix for the fault is that we "*should fall back
to using G711 pass through in order to negotiate/ send fax*"

The call initially connects with G711 (and the media is proxied).
Fax is detected by the originator which issues a RE-INVITE request to use
T38.
This is rejected by the gateway with a 488 at which point we would expect
the call to fall back to G711 however FreeSWITCH is reporting that this is
not possible due to a media bug:

"*Session is connected to a media bug. Re-Negotiation implicitly disabled.*"

My understanding was that a media bug would be required for the initial fax
tone detection which subsequently triggers the T38 session start?
Could anyone confirm whether we need to be looking into closing off the
media bug manually or if there is an alternative route to resolving this
issue?

We are running FreeSWITCH (Version 1.4.26  64bit).
Calls are dispatched with setting "t38_passthru=true".

Any help would be very much welcomed.

Best Regards,

Callum
-- 
Callum Guy
Head of Information Security
X-on

-- 



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From mirkobrankovic at gmail.com  Thu Dec  7 18:55:01 2017
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Thu, 7 Dec 2017 19:55:01 +0100
Subject: [Freeswitch-users] T38 Fax Fallback
In-Reply-To: <CAFjCFzn_xaFEs_=ow8tbCENBW0byUPBTC2VPVH9d0zF9Whanxw@mail.gmail.com>
References: <CAFjCFzn_xaFEs_=ow8tbCENBW0byUPBTC2VPVH9d0zF9Whanxw@mail.gmail.com>
Message-ID: <CAND18T0_nRoe4_gNYpWy57TkpKsaC8pQ59ZH=9aVMsE3y_opiA@mail.gmail.com>

I had a same question:
http://thread.gmane.org/gmane.comp.telephony.freeswitch.user/86263
Sorry google lead me only to this link...

On Dec 7, 2017 18:06, "Callum Guy" <callum.guy at x-on.co.uk> wrote:

> Hi All,
>
> I'll try and keep this simple.
>
> We are experiencing difficulty transmitting fax with a certain UK provider
> (BT) who have suggested the fix for the fault is that we "*should fall
> back to using G711 pass through in order to negotiate/ send fax*"
>
> The call initially connects with G711 (and the media is proxied).
> Fax is detected by the originator which issues a RE-INVITE request to use
> T38.
> This is rejected by the gateway with a 488 at which point we would expect
> the call to fall back to G711 however FreeSWITCH is reporting that this is
> not possible due to a media bug:
>
> "*Session is connected to a media bug. Re-Negotiation implicitly
> disabled.*"
>
> My understanding was that a media bug would be required for the initial
> fax tone detection which subsequently triggers the T38 session start?
> Could anyone confirm whether we need to be looking into closing off the
> media bug manually or if there is an alternative route to resolving this
> issue?
>
> We are running FreeSWITCH (Version 1.4.26  64bit).
> Calls are dispatched with setting "t38_passthru=true".
>
> Any help would be very much welcomed.
>
> Best Regards,
>
> Callum
> --
> Callum Guy
> Head of Information Security
> X-on
>
>
> *0333 332 0000  |  www.x-on.co.uk <http://www.x-on.co.uk>  |   **
> <https://www.linkedin.com/company/x-on>   <https://www.facebook.com/XonTel>
>   <https://twitter.com/xonuk> *
> X-on is a trading name of Storacall Technology Ltd a limited company
> registered in England and Wales.
> Registered Office : Avaland House, 110 London Road, Apsley, Hemel
> Hempstead, Herts, HP3 9SD. Company Registration No. 2578478.
> The information in this e-mail is confidential and for use by the
> addressee(s) only. If you are not the intended recipient, please notify
> X-on immediately on +44(0)333 332 0000 <+44%20333%20332%200000> and
> delete the
> message from your computer. If you are not a named addressee you must not
> use, disclose, disseminate, distribute, copy, print or reply to this email. Views
> or opinions expressed by an individual
> within this email may not necessarily reflect the views of X-on or its
> associated companies. Although X-on routinely screens for viruses,
> addressees should scan this email and any attachments
> for viruses. X-on makes no representation or warranty as to the absence of
> viruses in this email or any attachments.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From loi.dangthanh at gmail.com  Fri Dec  8 04:33:04 2017
From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=)
Date: Fri, 08 Dec 2017 04:33:04 +0000
Subject: [Freeswitch-users] T38 Fax Fallback
In-Reply-To: <CAND18T0_nRoe4_gNYpWy57TkpKsaC8pQ59ZH=9aVMsE3y_opiA@mail.gmail.com>
References: <CAFjCFzn_xaFEs_=ow8tbCENBW0byUPBTC2VPVH9d0zF9Whanxw@mail.gmail.com>
 <CAND18T0_nRoe4_gNYpWy57TkpKsaC8pQ59ZH=9aVMsE3y_opiA@mail.gmail.com>
Message-ID: <CAGsrq_umB3B1Ko=B2CjLg+AuQWqW9NaSvZv577upuyxHnKRRxA@mail.gmail.com>

Hi, media proxy mode is not willing to work fine with fax t38, since
`t38_passthru=true` dont have any effect in that mode.
Could consider getting away from media proxy mode since it's not
recommended anymore by FS's developers.
Btw, you may want to upgrade your FS to 1.6.15 or later.
http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html

rgds,
Loi Dang

On Fri, Dec 8, 2017 at 1:55 AM Mirko Brankovic <mirkobrankovic at gmail.com>
wrote:

> I had a same question:
> http://thread.gmane.org/gmane.comp.telephony.freeswitch.user/86263
> Sorry google lead me only to this link...
>
> On Dec 7, 2017 18:06, "Callum Guy" <callum.guy at x-on.co.uk> wrote:
>
>> Hi All,
>>
>> I'll try and keep this simple.
>>
>> We are experiencing difficulty transmitting fax with a certain UK
>> provider (BT) who have suggested the fix for the fault is that we "*should
>> fall back to using G711 pass through in order to negotiate/ send fax*"
>>
>> The call initially connects with G711 (and the media is proxied).
>> Fax is detected by the originator which issues a RE-INVITE request to use
>> T38.
>> This is rejected by the gateway with a 488 at which point we would expect
>> the call to fall back to G711 however FreeSWITCH is reporting that this is
>> not possible due to a media bug:
>>
>> "*Session is connected to a media bug. Re-Negotiation implicitly
>> disabled.*"
>>
>> My understanding was that a media bug would be required for the initial
>> fax tone detection which subsequently triggers the T38 session start?
>> Could anyone confirm whether we need to be looking into closing off the
>> media bug manually or if there is an alternative route to resolving this
>> issue?
>>
>> We are running FreeSWITCH (Version 1.4.26  64bit).
>> Calls are dispatched with setting "t38_passthru=true".
>>
>> Any help would be very much welcomed.
>>
>> Best Regards,
>>
>> Callum
>> --
>> Callum Guy
>> Head of Information Security
>> X-on
>>
>>
>> *0333 332 0000  |  www.x-on.co.uk <http://www.x-on.co.uk>  |   **
>> <https://www.linkedin.com/company/x-on>   <https://www.facebook.com/XonTel>
>>   <https://twitter.com/xonuk> *
>> X-on is a trading name of Storacall Technology Ltd a limited company
>> registered in England and Wales.
>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel
>> Hempstead, Herts, HP3 9SD. Company Registration No. 2578478.
>> The information in this e-mail is confidential and for use by the
>> addressee(s) only. If you are not the intended recipient, please notify
>> X-on immediately on +44(0)333 332 0000 <+44%20333%20332%200000> and
>> delete the
>> message from your computer. If you are not a named addressee you must not
>> use, disclose, disseminate, distribute, copy, print or reply to this email. Views
>> or opinions expressed by an individual
>> within this email may not necessarily reflect the views of X-on or its
>> associated companies. Although X-on routinely screens for viruses,
>> addressees should scan this email and any attachments
>> for viruses. X-on makes no representation or warranty as to the absence
>> of viruses in this email or any attachments.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From michael at mailworks.org  Fri Dec  8 07:52:45 2017
From: michael at mailworks.org (Michael Avers)
Date: Fri, 08 Dec 2017 00:52:45 -0700
Subject: [Freeswitch-users] How to bridge a call to an extension without the
	display updating?
Message-ID: <1512719565.3344027.1198209576.1A343915@webmail.messagingengine.com>

Hello,

I'm trying to create a random dialplan extension (say 5050) that when dialed into by a registered extension would bridge the call to another registered user (say 3000), but not change the display or P-Asserted-Identity.

Right now on the bridge it sends to the calling extension

*P-Asserted-Identity: "5050" <sip:3000 at sip.domain>.*

I tried setting all the possible values for sip_cid_type and setting ignore_display_updates to false, but it still sends the P-Asserted update as shown above.

The bridge itself I tried both with user/3000 at sip.domain and ${sofia_contact(566055 at rsi.solidphone.net)} - same result.

This is a problem because A. the display on the phone changes to "3000" and more importantly, B. if the calling user later hits redial it will dial 3000 instead of 5050.

Any way to achieve what I'm trying to do?

Thanks
Mike



From callum.guy at x-on.co.uk  Fri Dec  8 09:00:57 2017
From: callum.guy at x-on.co.uk (Callum Guy)
Date: Fri, 08 Dec 2017 09:00:57 +0000
Subject: [Freeswitch-users] T38 Fax Fallback
In-Reply-To: <CAGsrq_umB3B1Ko=B2CjLg+AuQWqW9NaSvZv577upuyxHnKRRxA@mail.gmail.com>
References: <CAFjCFzn_xaFEs_=ow8tbCENBW0byUPBTC2VPVH9d0zF9Whanxw@mail.gmail.com>
 <CAND18T0_nRoe4_gNYpWy57TkpKsaC8pQ59ZH=9aVMsE3y_opiA@mail.gmail.com>
 <CAGsrq_umB3B1Ko=B2CjLg+AuQWqW9NaSvZv577upuyxHnKRRxA@mail.gmail.com>
Message-ID: <CAFjCFznTx9Vxu=odDyUAOv8p1LMwC1V0ADUCqKw=SD9nJcUTZw@mail.gmail.com>

Mirko, Loi,

Thanks for your responses. I really wanted to check if there was anything
obvious which our platform was not handling properly however your responses
help to confirm that Fax/FOIP is rarely plain sailing. I had actually found
Mirko's email thread before writing this - Brian's response was helpfully
discouraging!

I'm not clear on whether it will be possible to remove the media proxy from
the call path on this occasion but it is something I'll look into.

I will also look into finding resource to test on the latest stable release
and see if the recent modifications are resolving this situation.

Once those avenues are exhausted I'll get on with the day job! If anyone
does have other suggestions then please do get in touch.

Thanks again,

Callum

*NB. When was the last time you got a fax?!?*

On Fri, Dec 8, 2017 at 4:36 AM Lợi Đặng <loi.dangthanh at gmail.com> wrote:

> Hi, media proxy mode is not willing to work fine with fax t38, since
> `t38_passthru=true` dont have any effect in that mode.
> Could consider getting away from media proxy mode since it's not
> recommended anymore by FS's developers.
> Btw, you may want to upgrade your FS to 1.6.15 or later.
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124459.html
>
> rgds,
> Loi Dang
>
> On Fri, Dec 8, 2017 at 1:55 AM Mirko Brankovic <mirkobrankovic at gmail.com>
> wrote:
>
>> I had a same question:
>> http://thread.gmane.org/gmane.comp.telephony.freeswitch.user/86263
>> Sorry google lead me only to this link...
>>
>> On Dec 7, 2017 18:06, "Callum Guy" <callum.guy at x-on.co.uk> wrote:
>>
>>> Hi All,
>>>
>>> I'll try and keep this simple.
>>>
>>> We are experiencing difficulty transmitting fax with a certain UK
>>> provider (BT) who have suggested the fix for the fault is that we "*should
>>> fall back to using G711 pass through in order to negotiate/ send fax*"
>>>
>>> The call initially connects with G711 (and the media is proxied).
>>> Fax is detected by the originator which issues a RE-INVITE request to
>>> use T38.
>>> This is rejected by the gateway with a 488 at which point we would
>>> expect the call to fall back to G711 however FreeSWITCH is reporting that
>>> this is not possible due to a media bug:
>>>
>>> "*Session is connected to a media bug. Re-Negotiation implicitly
>>> disabled.*"
>>>
>>> My understanding was that a media bug would be required for the initial
>>> fax tone detection which subsequently triggers the T38 session start?
>>> Could anyone confirm whether we need to be looking into closing off the
>>> media bug manually or if there is an alternative route to resolving this
>>> issue?
>>>
>>> We are running FreeSWITCH (Version 1.4.26  64bit).
>>> Calls are dispatched with setting "t38_passthru=true".
>>>
>>> Any help would be very much welcomed.
>>>
>>> Best Regards,
>>>
>>> Callum
>>> --
>>> Callum Guy
>>> Head of Information Security
>>> X-on
>>>
>>>
>>> *0333 332 0000  |  www.x-on.co.uk <http://www.x-on.co.uk>  |   **
>>> <https://www.linkedin.com/company/x-on>   <https://www.facebook.com/XonTel>
>>>   <https://twitter.com/xonuk> *
>>> X-on is a trading name of Storacall Technology Ltd a limited company
>>> registered in England and Wales.
>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel
>>> Hempstead, Herts, HP3 9SD. Company Registration No. 2578478.
>>> The information in this e-mail is confidential and for use by the
>>> addressee(s) only. If you are not the intended recipient, please notify
>>> X-on immediately on +44(0)333 332 0000 <+44%20333%20332%200000> and
>>> delete the
>>> message from your computer. If you are not a named addressee you must
>>> not use, disclose, disseminate, distribute, copy, print or reply to this
>>> email. Views or opinions expressed by an individual
>>> within this email may not necessarily reflect the views of X-on or its
>>> associated companies. Although X-on routinely screens for viruses,
>>> addressees should scan this email and any attachments
>>> for viruses. X-on makes no representation or warranty as to the absence
>>> of viruses in this email or any attachments.
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Callum Guy
Head of Information Security
X-on

-- 



*0333 332 0000  |  www.x-on.co.uk <http://www.x-on.co.uk>  |   ** 
<https://www.linkedin.com/company/x-on>   <https://www.facebook.com/XonTel> 
  <https://twitter.com/xonuk> * 
X-on is a trading name of Storacall Technology Ltd a limited company 
registered in England and Wales.
Registered Office : Avaland House, 110 London Road, Apsley, Hemel 
Hempstead, Herts, HP3 9SD. Company Registration No. 2578478.
The information in this e-mail is confidential and for use by the 
addressee(s) only. If you are not the intended recipient, please notify 
X-on immediately on +44(0)333 332 0000 and delete the
message from your computer. If you are not a named addressee you must not 
use, disclose, disseminate, distribute, copy, print or reply to this email. Views 
or opinions expressed by an individual
within this email may not necessarily reflect the views of X-on or its 
associated companies. Although X-on routinely screens for viruses, 
addressees should scan this email and any attachments
for viruses. X-on makes no representation or warranty as to the absence of 
viruses in this email or any attachments.

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From mirkobrankovic at gmail.com  Fri Dec  8 12:23:43 2017
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Fri, 8 Dec 2017 13:23:43 +0100
Subject: [Freeswitch-users] T38 Fax Fallback
In-Reply-To: <CAFjCFznTx9Vxu=odDyUAOv8p1LMwC1V0ADUCqKw=SD9nJcUTZw@mail.gmail.com>
References: <CAFjCFzn_xaFEs_=ow8tbCENBW0byUPBTC2VPVH9d0zF9Whanxw@mail.gmail.com>
 <CAND18T0_nRoe4_gNYpWy57TkpKsaC8pQ59ZH=9aVMsE3y_opiA@mail.gmail.com>
 <CAGsrq_umB3B1Ko=B2CjLg+AuQWqW9NaSvZv577upuyxHnKRRxA@mail.gmail.com>
 <CAFjCFznTx9Vxu=odDyUAOv8p1LMwC1V0ADUCqKw=SD9nJcUTZw@mail.gmail.com>
Message-ID: <CAND18T16oTBXMLxXqUM+Fx8=auR7Pp16a_RjXs5MDPUxp7EA5A@mail.gmail.com>

>
> *NB. When was the last time you got a fax?!?*


*Good one :D*

On Fri, Dec 8, 2017 at 10:00 AM, Callum Guy <callum.guy at x-on.co.uk> wrote:

> Mirko, Loi,
>
> Thanks for your responses. I really wanted to check if there was anything
> obvious which our platform was not handling properly however your responses
> help to confirm that Fax/FOIP is rarely plain sailing. I had actually found
> Mirko's email thread before writing this - Brian's response was helpfully
> discouraging!
>
> I'm not clear on whether it will be possible to remove the media proxy
> from the call path on this occasion but it is something I'll look into.
>
> I will also look into finding resource to test on the latest stable
> release and see if the recent modifications are resolving this situation.
>
> Once those avenues are exhausted I'll get on with the day job! If anyone
> does have other suggestions then please do get in touch.
>
> Thanks again,
>
> Callum
>
> *NB. When was the last time you got a fax?!?*
>
> On Fri, Dec 8, 2017 at 4:36 AM Lợi Đặng <loi.dangthanh at gmail.com> wrote:
>
>> Hi, media proxy mode is not willing to work fine with fax t38, since
>> `t38_passthru=true` dont have any effect in that mode.
>> Could consider getting away from media proxy mode since it's not
>> recommended anymore by FS's developers.
>> Btw, you may want to upgrade your FS to 1.6.15 or later.
>> http://lists.freeswitch.org/pipermail/freeswitch-users/
>> 2017-January/124459.html
>>
>> rgds,
>> Loi Dang
>>
>> On Fri, Dec 8, 2017 at 1:55 AM Mirko Brankovic <mirkobrankovic at gmail.com>
>> wrote:
>>
>>> I had a same question:
>>> http://thread.gmane.org/gmane.comp.telephony.freeswitch.user/86263
>>> Sorry google lead me only to this link...
>>>
>>> On Dec 7, 2017 18:06, "Callum Guy" <callum.guy at x-on.co.uk> wrote:
>>>
>>>> Hi All,
>>>>
>>>> I'll try and keep this simple.
>>>>
>>>> We are experiencing difficulty transmitting fax with a certain UK
>>>> provider (BT) who have suggested the fix for the fault is that we "*should
>>>> fall back to using G711 pass through in order to negotiate/ send fax*"
>>>>
>>>> The call initially connects with G711 (and the media is proxied).
>>>> Fax is detected by the originator which issues a RE-INVITE request to
>>>> use T38.
>>>> This is rejected by the gateway with a 488 at which point we would
>>>> expect the call to fall back to G711 however FreeSWITCH is reporting that
>>>> this is not possible due to a media bug:
>>>>
>>>> "*Session is connected to a media bug. Re-Negotiation implicitly
>>>> disabled.*"
>>>>
>>>> My understanding was that a media bug would be required for the initial
>>>> fax tone detection which subsequently triggers the T38 session start?
>>>> Could anyone confirm whether we need to be looking into closing off the
>>>> media bug manually or if there is an alternative route to resolving this
>>>> issue?
>>>>
>>>> We are running FreeSWITCH (Version 1.4.26  64bit).
>>>> Calls are dispatched with setting "t38_passthru=true".
>>>>
>>>> Any help would be very much welcomed.
>>>>
>>>> Best Regards,
>>>>
>>>> Callum
>>>> --
>>>> Callum Guy
>>>> Head of Information Security
>>>> X-on
>>>>
>>>>
>>>> *0333 332 0000  |  www.x-on.co.uk <http://www.x-on.co.uk>  |   **
>>>> <https://www.linkedin.com/company/x-on>   <https://www.facebook.com/XonTel>
>>>>   <https://twitter.com/xonuk> *
>>>> X-on is a trading name of Storacall Technology Ltd a limited company
>>>> registered in England and Wales.
>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel
>>>> Hempstead, Herts, HP3 9SD. Company Registration No. 2578478.
>>>> The information in this e-mail is confidential and for use by the
>>>> addressee(s) only. If you are not the intended recipient, please notify
>>>> X-on immediately on +44(0)333 332 0000 <+44%20333%20332%200000> and
>>>> delete the
>>>> message from your computer. If you are not a named addressee you must
>>>> not use, disclose, disseminate, distribute, copy, print or reply to this
>>>> email. Views or opinions expressed by an individual
>>>> within this email may not necessarily reflect the views of X-on or its
>>>> associated companies. Although X-on routinely screens for viruses,
>>>> addressees should scan this email and any attachments
>>>> for viruses. X-on makes no representation or warranty as to the absence
>>>> of viruses in this email or any attachments.
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Callum Guy
> Head of Information Security
> X-on
>
>
> *0333 332 0000  |  www.x-on.co.uk <http://www.x-on.co.uk>  |   **
> <https://www.linkedin.com/company/x-on>   <https://www.facebook.com/XonTel>
>   <https://twitter.com/xonuk> *
> X-on is a trading name of Storacall Technology Ltd a limited company
> registered in England and Wales.
> Registered Office : Avaland House, 110 London Road, Apsley, Hemel
> Hempstead, Herts, HP3 9SD. Company Registration No. 2578478.
> The information in this e-mail is confidential and for use by the
> addressee(s) only. If you are not the intended recipient, please notify
> X-on immediately on +44(0)333 332 0000 <+44%20333%20332%200000> and
> delete the
> message from your computer. If you are not a named addressee you must not
> use, disclose, disseminate, distribute, copy, print or reply to this email. Views
> or opinions expressed by an individual
> within this email may not necessarily reflect the views of X-on or its
> associated companies. Although X-on routinely screens for viruses,
> addressees should scan this email and any attachments
> for viruses. X-on makes no representation or warranty as to the absence of
> viruses in this email or any attachments.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Mirko
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From godson.g at gmail.com  Fri Dec  8 13:27:27 2017
From: godson.g at gmail.com (Godson Gera)
Date: Fri, 8 Dec 2017 18:57:27 +0530
Subject: [Freeswitch-users] VOIP on Google Cloud
In-Reply-To: <1512629555.2948955.1196917528.3E95C516@webmail.messagingengine.com>
References: <1512629555.2948955.1196917528.3E95C516@webmail.messagingengine.com>
Message-ID: <CAPTXd0qihga-3bkChf3cScfZ=rh7rtTorr2DaZg2m3i9ko5DZg@mail.gmail.com>

It's runs alright, no issues so far.


On Dec 7, 2017 12:26 PM, "Michael Avers" <michael at mailworks.org> wrote:

Hey, anybody running Freeswitch on Google Compute Engine? How is it working
out?

Mike

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Thanks,
Godson Gera
http://godson.in
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From euan at ensemblepourladifference.org  Fri Dec  8 10:26:45 2017
From: euan at ensemblepourladifference.org (Euan Millar)
Date: Fri, 8 Dec 2017 10:26:45 +0000
Subject: [Freeswitch-users] Can't find modules to install
Message-ID: <CAOZyRbnnOyD+ajGpFz-7rN4qBam3i2S_1K_-REEq7fWreuZb7Q@mail.gmail.com>

Hi,

I have an IVR app that I am translating into French but I get an error ...

Invalid SAY interface[fr]

I think I need to install the module "mod_say_fr" into
/usr/lib/freeswitch/mod but I cant find where online I would source this
module.

Can anyone help me with the URL for this module?  This is the first time
for me to add a module and somebody else performed the Freeswitch
installation.  Is it as simple as downloading the mod_say_fr.so file into
the directory and loading it in the config XML?

Many thanks in advance if you are able to help me.

Kind regards,

Euan
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From Martin.Gordian at c4b.de  Fri Dec  8 13:07:28 2017
From: Martin.Gordian at c4b.de (Martin Gordian)
Date: Fri, 8 Dec 2017 13:07:28 +0000
Subject: [Freeswitch-users] Directory Gateway Start On Demand
Message-ID: <80caff7b932748dc80433bf57b70a2a1@c4b.de>

Hello List,

according to https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration example 3, I have configured a gateway for a user to come up and down only when the user is registered, like this

<user id="12345">
  <params>
    <param name="password" value="12345678" />
  </params>
  <variables>
    <variable name="default_gateway" value="GW_12345"/>
  </variables>
  <gateways>
    <gateway name="GW_12345">
      <param name="username" value="54321"/>
      <param name="password" value="87654321"/>
      <param name="proxy" value="111.111.111.111:5060"/>
      <param name="register" value="false"/>
    </gateway>
  </gateways>
</user>

Now, according to the clarification on the mentioned page, I expect to see a SIP REGISTER request in Wireshark, from FreeSWITCH to 111.111.111.111 (changed the IP for the question), when the user 12345 registers. But there is none. Checking the gateway with sofia command reveals that there is no SIP registration done, the gatesway is down.

What am I missing here, so that the gateway will come up when the user 12345 registers?

Thank you.
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From paul.mateer at outlook.com  Thu Dec  7 12:13:51 2017
From: paul.mateer at outlook.com (Paul Mateer)
Date: Thu, 7 Dec 2017 12:13:51 +0000
Subject: [Freeswitch-users] Problem setting pa_ring_file with global_setvar
Message-ID: <DB5PR1001MB0776D13CECEC476C4CD9E9C5F0330@DB5PR1001MB0776.EURPRD10.PROD.OUTLOOK.COM>

I’ve run into as issue when trying to set the pa_ring_file variable to a tone stream value.
If the chosen value contains one or more elements relating to in-memory copies, volume or the number of loops then the global_setvar_function incorrectly parses the value as conditional and fails to set the value.

Anyone know of any way around this?

Paul

Sent from my Windows 10 phone

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From thilo at ginkel.com  Fri Dec  8 12:08:39 2017
From: thilo at ginkel.com (Thilo-Alexander Ginkel)
Date: Fri, 8 Dec 2017 13:08:39 +0100
Subject: [Freeswitch-users] Influencing "Update Callee ID" behavior
Message-ID: <CANvSZQ8kJ0Ki1WHY4fK=RR_xThnBq5ULuOZ5hQFBWwYDXyp2Bw@mail.gmail.com>

Hello everyone,

when placing an external outbound call via mod_sofia I am seeing the
callee's number and name being updated as soon as the called party
picks up the call:

2017-12-08 10:19:24.933767 [INFO] sofia.c:1279 sofia/external/040xxx
Update Callee ID to "Outbound Call" <40xxx>

This seems to override what has been previously set as
sip_callee_id_name (e.g., using <action application="set"
data="sip_callee_id_name=${db(select/cid/${2})}"/>).

There are two problems that I'd like to solve:

1. When sip_callee_id_name was previously set (e.g., via DB lookup)
I'd like to keep that value (or repeat the lookup) so that the
callee's name remains intact.

2. Certain destinations incorrectly strip the loeading zero from the
callee's number, which I'd like to avoid displaying as is.

Is there any way to influence what is displayed as name and number,
i.e. keep the previously set sip_callee_id_name and apply some kind of
normalization to the reported number to cope with the missing leading
zero?

Thanks,
Thilo


From konrd at yahoo.com  Thu Dec  7 13:16:32 2017
From: konrd at yahoo.com (Konrad)
Date: Thu, 7 Dec 2017 13:16:32 +0000 (UTC)
Subject: [Freeswitch-users] RTP Timestamps
References: <50456081.314082.1512652592871.ref@mail.yahoo.com>
Message-ID: <50456081.314082.1512652592871@mail.yahoo.com>

I'm running into an issue where Freeswitch is sending out RTP with old/duplicate timestamps. 
A wireshark trace will show the something like the following randomly throughout a call:
Seq=19567, Time=26560Seq=19568, Time=26880  <--Seq=19569, Time=27040Seq=19570, Time=26880  <--Seq=19571, Time=27200Seq=19572, Time=27360

Initially I thought this was related to vmware, but I am getting the same results on bare metal. Server load doesn't seem to matter. Server is a dual xeon with quad cores. Timer_test runs clean. I've tried setting <rewrite-rtp-timestamps> = true, but that didn't help. 
OS is Debian Jessie with FreeSWITCH (Version 1.6.19 -36-7a77e0b 64bit)

Any thoughts?
Thanks,Konrad

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From SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com  Fri Dec  8 22:12:39 2017
From: SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com (Karanath sachidanandan, Sanooj)
Date: Fri, 8 Dec 2017 22:12:39 +0000
Subject: [Freeswitch-users] FS connecting to Avaya
Message-ID: <E338CEDA-8B96-4201-A6A5-38B85D3C2529@homedepot.com>

Hi All,
Is there any documentation on connecting FS to AVAYA CM/AES ? or any pointers ?

Regards
Sanooj


________________________________

The information in this Internet Email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this Email by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients any opinions or advice contained in this Email are subject to the terms and conditions expressed in any applicable governing The Home Depot terms of business or client engagement letter. The Home Depot disclaims all responsibility and liability for the accuracy and content of this attachment and for any damages or losses arising from any inaccuracies, errors, viruses, e.g., worms, trojan horses, etc., or other items of a destructive nature, which may be contained in this attachment and shall not be liable for direct, indirect, consequential or special damages in connection with this e-mail message or its attachment.
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From bipin at xbipin.com  Sat Dec  9 06:01:40 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Sat, 9 Dec 2017 10:01:40 +0400
Subject: [Freeswitch-users] FS weird issue with TLS/SRTP and xml_curl on
 windows
In-Reply-To: <e0bf7374-eed8-8485-aa19-e3be0775d9d4@xbipin.com>
References: <a63ef485-e412-dba5-9961-17e2e8982ae1@xbipin.com>
 <CAPTXd0pJ1ivi64G2tN+3JzzOZsSjn8mXWcawVabxRrH63fnF8g@mail.gmail.com>
 <e0bf7374-eed8-8485-aa19-e3be0775d9d4@xbipin.com>
Message-ID: <26234fe5-bb5d-a99a-3a85-cc1cd8778b90@xbipin.com>

hi,

after setting timeout etc it is better but there is something still 
wrong because xml_curl doesnt seem to cache properly as the time now 
shows negative and it still calls script for registration details

Time Now:       -101036188
Expires:        1044723532
2017-12-09 09:57:29.764708 [DEBUG] switch_xml.c:1990 Cache expired for 
user at 1.2.3.4, doing fresh lookup
2017-12-09 09:57:29.804735 [DEBUG] switch_xml.c:2068 caching lookup for 
user user at 1.2.3.4 for 1200000 milliseconds

earlier both time now and expires was negative but after running FS with 
-monotonic-clock expires came to positive but time now still negative 
and the php script is always called rather than using cache


Regards,
Bipin


------------------------------------------------------------------------
-------- Original Message --------
Subject: Re: [Freeswitch-users] FS weird issue with TLS/SRTP and 
xml_curl on windows
From: Bipin Patel <bipin at xbipin.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 10/17/2017, 10:54:11 AM
> hi,
>
> the webserver and server where FS runs is same, for now i have added 
> the timeout options in xml_curl so lets see if it happens again and 
> ill report back, the other issue is for no reason at times FS just 
> hangs or so, running the fs_cli app shows blank, this happened today 
> morning itself, after killing the process and restarting the service 
> it went back to normal
>
>
> Regards,
> Bipin
>
>
> ------------------------------------------------------------------------
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] FS weird issue with TLS/SRTP and 
> xml_curl on windows
> From: Godson Gera <godson.g at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: 10/16/2017, 10:45:08 AM
>> Hi Bipin,
>>
>> I ran into some thing like this once. Turned out to be a the 
>> webservice which is called by xml_curl took too long to repond and FS 
>> waited there forever with out timing out. Later on timeout option is 
>> introduced in xml curl config. Try setting a reasonable timeout and 
>> see if that resolves your issue.
>>
>> On Fri, Oct 6, 2017 at 7:39 PM, Bipin Patel <bipin at xbipin.com 
>> <mailto:bipin at xbipin.com>> wrote:
>>
>>     hi,
>>
>>     i have two instances of FS running on a windows server, one is
>>     used for routing to carriers and that uses xml files and that
>>     works fine, the second instance is set up to allow clients to
>>     register to it using tls and srtp and users are authenticated
>>     using xml_curl which calls a php script which inturn sends the
>>     directory users details on a register from a client. The problem
>>     is every few hours or so FS stops accepting new clients unless i
>>     restart the service, i check the php script and the webserver and
>>     those are running just fine so no idea whats causing FS to stop
>>     calling the script or something else.
>>
>>
>>     -- 
>>     Regards,
>>     Bipin
>>
>>
>>     ------------------------------------------------------------------------
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>     <http://www.freeswitchsolutions.com>
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>     http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Thanks & Regards,
>> Godson Gera
>> FreeSWITCH Consultant <http://godson.in>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ssinyagin at gmail.com  Sun Dec 10 21:08:04 2017
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 10 Dec 2017 22:08:04 +0100
Subject: [Freeswitch-users] VOIP on Google Cloud
In-Reply-To: <CAPTXd0qihga-3bkChf3cScfZ=rh7rtTorr2DaZg2m3i9ko5DZg@mail.gmail.com>
References: <1512629555.2948955.1196917528.3E95C516@webmail.messagingengine.com>
 <CAPTXd0qihga-3bkChf3cScfZ=rh7rtTorr2DaZg2m3i9ko5DZg@mail.gmail.com>
Message-ID: <CACOYK=jVWHdVCxeJx3qgGRRKOtY5=n+oMeLQbM4=egwDvyzUgg@mail.gmail.com>

are you only bridging the calls, or playing back media files as well?

You would usually not notice much of distortion ob bridged calls,
unless the host CPU is too much overloaded. But media playback and
conferences are very sensitive to the quality of the host.

On Fri, Dec 8, 2017 at 2:27 PM, Godson Gera <godson.g at gmail.com> wrote:
> It's runs alright, no issues so far.
>
>
> On Dec 7, 2017 12:26 PM, "Michael Avers" <michael at mailworks.org> wrote:
>
> Hey, anybody running Freeswitch on Google Compute Engine? How is it working
> out?
>
> Mike
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> Thanks,
> Godson Gera
> http://godson.in
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From tom at tomlynn.com  Mon Dec 11 00:28:54 2017
From: tom at tomlynn.com (Tom Lynn)
Date: Sun, 10 Dec 2017 16:28:54 -0800
Subject: [Freeswitch-users] FS connecting to Avaya
In-Reply-To: <E338CEDA-8B96-4201-A6A5-38B85D3C2529@homedepot.com>
References: <E338CEDA-8B96-4201-A6A5-38B85D3C2529@homedepot.com>
Message-ID: <CAERdjibo-K921xiPSUPRq_rXEjiesUGr4CSmvE+YvV=+YngryA@mail.gmail.com>

Are you planning to peer directly to Communication Manager or using Avaya
Session Manager in between?  It can be done both ways.  If the requirement
is to transact only with Communication Manager a direct connection may work
out.

In that case I'd suggest you'll need to add a node-name (which is like a
host table entry) for FS and a signaling group of type SIP, along with a
trunk group whose channels are controlled by said signaling group.  Assume
TCP signaling to begin with and pay attention to your port numbers.  Some
may already be used by other signaling groups.

Codecs are assigned based on a network-region construct which is used to
model your voip network against your physical topology, supporting call
admission control and where DSP channels are sourced.  Since Avaya doesn't
use it's server host to mix audio, you'll have to know that the network
region you choose either has DSPs or a logical path to a region which does
is available.  The network region is assigned to the SIP signaling group in
a field called Far End Network region.

To point calls at the trunk group you'll have to set up a route pattern and
then place a dial string match in either an ARS analysis table (PSTN) or an
AAR analysis table (private network).

How are you thinking about using AES?  I don't see an easy interface, but
if you program TSAPI, it may be useful to get call events from the Avaya
side.

Tom Lynn


On Fri, Dec 8, 2017 at 2:12 PM, Karanath sachidanandan, Sanooj <
SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com> wrote:

> Hi All,
>
> Is there any documentation on connecting FS to AVAYA CM/AES ? or any
> pointers ?
>
>
>
> Regards
>
> Sanooj
>
>
>
> ------------------------------
>
> The information in this Internet Email is confidential and may be legally
> privileged. It is intended solely for the addressee. Access to this Email
> by anyone else is unauthorized. If you are not the intended recipient, any
> disclosure, copying, distribution or any action taken or omitted to be
> taken in reliance on it, is prohibited and may be unlawful. When addressed
> to our clients any opinions or advice contained in this Email are subject
> to the terms and conditions expressed in any applicable governing The Home
> Depot terms of business or client engagement letter. The Home Depot
> disclaims all responsibility and liability for the accuracy and content of
> this attachment and for any damages or losses arising from any
> inaccuracies, errors, viruses, e.g., worms, trojan horses, etc., or other
> items of a destructive nature, which may be contained in this attachment
> and shall not be liable for direct, indirect, consequential or special
> damages in connection with this e-mail message or its attachment.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tchen61 at gmail.com  Sun Dec 10 17:36:19 2017
From: tchen61 at gmail.com (Thomas Chen)
Date: Sun, 10 Dec 2017 12:36:19 -0500
Subject: [Freeswitch-users] SOFIA_PRESENCE check
Message-ID: <5531fc9a-fa94-5894-3ba6-fa5be2046fd9@gmail.com>

i am trying to send CHAT/SMS  to external user (outside of the FS box)

i was hoping to handle that in chatplan similar to handling external 
phone # in dialplan with ESL

however, seems like the SOFIA_PRESENCE.c is checking whether the ADDR at IP 
is present first

before even activating chatplan  (so i keep on getting error message  
"nobody to send to / internal profile)


is there anyway to bypass this check (and accept all destination "to:" 
to the chatplan ???


thanks




From lists at telefaks.de  Mon Dec 11 16:30:45 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Mon, 11 Dec 2017 17:30:45 +0100
Subject: [Freeswitch-users] Lose Race instead of missed call
Message-ID: <5A2EB2B5.9030907@telefaks.de>

Hello,

we have some Problems with the right signalling of missed calls when
calling multiple phones in parallel

Here's the scenario:
Phone no 49 is calling a group with 2170 and 3275 with the following
dialstring

<action application="bridge"
data="{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59eee5e0-c0bc-0135-7dc9-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/2170 at flex.mydomain.de:_:{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59f07af0-c0bc-0135-7dca-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/3275 at flex.mydomain.de"
/>

Destination dialstrings are seperated by ":_:" ("Enterprise
Origination"). We use curly brackets instead of "<" as we sometimes have
to insert asserted identy tags into the dialstring.

We checked 3 versions of Freeswitch for this

  * Version  Feb 2016 shows missed calls on both phones. Even if one
    phone answers, the other phone one still shows a missed call (reason
    for upgrading to newer Freeswitch)
  * Version  Aug 2017 never shows missed call, see logs and hangup
    message below
  * Version  10/Dec 2017(yesterday) never shows missed call, as above

So for the 2 never Freeswitch Versions, here are the logs at hangup
2017-12-11 17:07:46.022450 [DEBUG] sofia.c:7283 Channel
sofia/internal/49 at flex.mydomain.de:5060 entering state [terminated][487]
2017-12-11 17:07:46.022450 [NOTICE] sofia.c:8474 Hangup
sofia/internal/*49*@flex.mydomain.de:5060 [CS_EXECUTE] *[ORIGINATOR_CANCEL]*
2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup
sofia/internal/*2170*@94.xx.xxx.xx:42170 [CS_CONSUME_MEDIA] *[LOSE_RACE]*
2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot
create outgoing channel of type [user] cause: [LOSE_RACE]
2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863 Originate
Resulted in Error Cause: 502 [LOSE_RACE]
2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:584
(sofia/internal/2170 at 94.xx.xxx.xx:42170) Running State Change CS_HANGUP
(Cur 3 Tot 22)
2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:850
(sofia/internal/2170 at 94.xx.xxx.xx:42170) Callstate Change RINGING -> HANGUP
2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:852
(sofia/internal/2170 at 94.xx.xxx.xx:42170) State HANGUP
2017-12-11 17:07:46.042331 [DEBUG] mod_sofia.c:449 Channel
sofia/internal/2170 at 94.xx.xxx.xx:42170 hanging up, cause: LOSE_RACE
2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup
sofia/internal/*3275*@94.xx.xxx.xx:43275 [CS_CONSUME_MEDIA] [*LOSE_RACE*]
2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot
create outgoing channel of type [user] cause: [LOSE_RACE]
2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863 Originate
Resulted in Error Cause: 502 [LOSE_RACE]

Here is the Cancel message for one of the called phones:
U 2017/12/11 17:07:46.045120 144.xx.xxx.xx:5060 -> 94.xx.xxx.xx:42170
*CANCEL sip:2170*@94.xx.xxx.xx:42170 SIP/2.0.
Via: SIP/2.0/UDP 144.xx.xxx.xx;rport;branch=z9hG4bK07r3ce94cvjpp.
Max-Forwards: 70.
From: "Test" <sip:49 at flex.mydomain.de>;tag=mQS1eFDpp1peS.
To: <sip:2170 at 94.xx.xxx.xx:42170>.
Call-ID: 4238b5a7-5930-1236-b2ab-00505600a1a5.
CSeq: 116168167 CANCEL.
*Reason: SIP;cause=200;text="Call completed elsewhere"*.
Content-Length: 0.

Any hints why this happens, or anyone has this scenario working?

-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

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From jos at firstcom.dk  Mon Dec 11 20:11:50 2017
From: jos at firstcom.dk (=?utf-8?B?Sm9uIFNjaMO4cHppbnNreQ==?=)
Date: Mon, 11 Dec 2017 20:11:50 +0000
Subject: [Freeswitch-users] Cancel reason on loose_race
Message-ID: <1F0B51F9-4616-4396-9326-BFCA098B00C8@firstcom.dk>

Hi list,

We have the following setup, with a queue server, running our own queue software, a main switch and a SBC.

Queue FreeSWITCH -> Master FreeSWITCH -> OpenSIPS -> UA

When a queue dials multiple UA’s, an one of them answers, the rest is correctly sent a CANCEL with the LOOSE_RACE as X-Reason header.
This is sent from the Queue FreeSWITCH to the Master FreeSWITCH, but when the Master FreeSWITCH sends this on to the OpenSIPS, the X-Reason header gets changed to NORMAL_CLEARING.

Is there any way we can get this header “proxied” through? If not readily available, where should we look to implement this functionality ourselves?

We thought of using our own X- headers, to indicate the reason to the OpenSIPS, but there doesn’t seem to be a way to affect headers on CANCEL messages, only on BYE messages.

Thank you,
Jon Schøpzinsky
ThisIsUniverse
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From Rich.freeswitch at Branham.us  Tue Dec 12 00:41:36 2017
From: Rich.freeswitch at Branham.us (Richard A. Branham, Jr.)
Date: Mon, 11 Dec 2017 20:41:36 -0400
Subject: [Freeswitch-users] Subscribing to event channels with Verto
Message-ID: <84cce92b38a57848d86dec75582c00cc.squirrel@branham.us>

I have a web application in which phone calling has been implemented using
Verto.  Calls have been working for quite a while now and we are happy
with Verto and FS.  I am now attempting to create other event channels
using Verto.

Two clients subscribe to an event channel with the following:

myVertoHandle.sendMethod('verto.subscribe', { "eventChannel":
"my-channel", "subParams": { "callID": null, "someparam":"someparam value"
} });

The subscription requests are received by the server and
subscribedChannels includes "my-channel" in both cases.

However, when I use the following to broadcast a message to the channel,
it does not seem to be sent to the clients:

myVertoHandle.broadcast("my-channel", {"someval":"val1",
"anotherval":"val2"});

The server receives the event and sends a reply to the sender, but other
clients do not receive the event.  (The reply from the server includes
message = "MCAST Data Sent".)

We are using FreeSWITCH Version 1.6.19~64bit ( 64bit).


Some questions:
--Is it possible to implement non-call communication using this approach,
or am I way off track?
--If this is the correct approach, what am I missing in the implementation?

Thanks!




From rw at panorgan.ch  Tue Dec 12 15:21:04 2017
From: rw at panorgan.ch (=?UTF-8?Q?Ren=c3=a9_Weiss?=)
Date: Tue, 12 Dec 2017 16:21:04 +0100
Subject: [Freeswitch-users] esl originate + sessions-per-second
Message-ID: <83618cf3-e72f-e10d-cbe8-c500e5eb1ae9@panorgan.ch>

Hi,

Is there a way do detect that an "originate" call over ESL has failed 
because of the "sessions-per-second" setting?

I see "Throttle Error!" in the Freeswitch logfile and
"503 Maximum Calls In Progress" on the SIP level, but the originate call 
itself just returns "-ERR DESTINATION_OUT_OF_ORDER" with no indication 
that the "sessions-per-second" limit was the reason for it.

Thanks,
René


From kathleen at freeswitch.com  Tue Dec 12 18:57:07 2017
From: kathleen at freeswitch.com (Kathleen King)
Date: Tue, 12 Dec 2017 10:57:07 -0800
Subject: [Freeswitch-users] Can't find modules to install
In-Reply-To: <CAOZyRbnnOyD+ajGpFz-7rN4qBam3i2S_1K_-REEq7fWreuZb7Q@mail.gmail.com>
References: <CAOZyRbnnOyD+ajGpFz-7rN4qBam3i2S_1K_-REEq7fWreuZb7Q@mail.gmail.com>
Message-ID: <CAOStUPb4ud4Xi-x8fmZo+hD9y8WtkF5t3CydFMu-cjcOPcAFNA@mail.gmail.com>

Hello,


We are going to have one of our team members answer this on our weekly
conference call this week. You can call in to join us live by dialing 888
at https://conference.freeswitch.org/vc/ or watch it live and after the
fact on youtube at https://www.youtube.com/watch?v=W9348CoRhjI


Please let me know if you have any other questions.

[image: freeswitch logo giant.jpg] <https://freeswitch.com>

Kathleen King | Public Relations / Administrative Assistant

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045

Email: Kathleen at freeswitch.com

Mobile: 703-859-3757

Website: https://www.FreeSWITCH.com <https://www.freeswitch.com>

[image: color-facebook-96.png] <https://www.facebook.com/freeswitch/> [image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>

On Fri, Dec 8, 2017 at 2:26 AM, Euan Millar <
euan at ensemblepourladifference.org> wrote:

> Hi,
>
> I have an IVR app that I am translating into French but I get an error ...
>
> Invalid SAY interface[fr]
>
> I think I need to install the module "mod_say_fr" into
> /usr/lib/freeswitch/mod but I cant find where online I would source this
> module.
>
> Can anyone help me with the URL for this module?  This is the first time
> for me to add a module and somebody else performed the Freeswitch
> installation.  Is it as simple as downloading the mod_say_fr.so file into
> the directory and loading it in the config XML?
>
> Many thanks in advance if you are able to help me.
>
> Kind regards,
>
> Euan
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From euan at ensemblepourladifference.org  Tue Dec 12 19:06:44 2017
From: euan at ensemblepourladifference.org (Euan Millar)
Date: Tue, 12 Dec 2017 19:06:44 +0000
Subject: [Freeswitch-users] Can't find modules to install
In-Reply-To: <CAOStUPb4ud4Xi-x8fmZo+hD9y8WtkF5t3CydFMu-cjcOPcAFNA@mail.gmail.com>
References: <CAOZyRbnnOyD+ajGpFz-7rN4qBam3i2S_1K_-REEq7fWreuZb7Q@mail.gmail.com>
 <CAOStUPb4ud4Xi-x8fmZo+hD9y8WtkF5t3CydFMu-cjcOPcAFNA@mail.gmail.com>
Message-ID: <CAOZyRbn9hCsYo_HhYyOvX0Y1eOwgxJXefcy4Qn=Dkr3gJdBtSQ@mail.gmail.com>

That's great thanks. Looking forward to it.

Euan

On 12 Dec 2017 7:02 p.m., "Kathleen King" <kathleen at freeswitch.com> wrote:

> Hello,
>
>
> We are going to have one of our team members answer this on our weekly
> conference call this week. You can call in to join us live by dialing 888
> at https://conference.freeswitch.org/vc/ or watch it live and after the
> fact on youtube at https://www.youtube.com/watch?v=W9348CoRhjI
>
>
> Please let me know if you have any other questions.
>
> [image: freeswitch logo giant.jpg] <https://freeswitch.com>
>
> Kathleen King | Public Relations / Administrative Assistant
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
>
> Email: Kathleen at freeswitch.com
>
> Mobile: 703-859-3757 <(703)%20859-3757>
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com>
>
> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/> [image:
> color-twitter-96.png]
> <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
>
> On Fri, Dec 8, 2017 at 2:26 AM, Euan Millar <euan@
> ensemblepourladifference.org> wrote:
>
>> Hi,
>>
>> I have an IVR app that I am translating into French but I get an error ...
>>
>> Invalid SAY interface[fr]
>>
>> I think I need to install the module "mod_say_fr" into
>> /usr/lib/freeswitch/mod but I cant find where online I would source this
>> module.
>>
>> Can anyone help me with the URL for this module?  This is the first time
>> for me to add a module and somebody else performed the Freeswitch
>> installation.  Is it as simple as downloading the mod_say_fr.so file into
>> the directory and loading it in the config XML?
>>
>> Many thanks in advance if you are able to help me.
>>
>> Kind regards,
>>
>> Euan
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mario_fs at mgtech.com  Tue Dec 12 19:24:16 2017
From: mario_fs at mgtech.com (Mario)
Date: Tue, 12 Dec 2017 11:24:16 -0800
Subject: [Freeswitch-users] Major change to macOS Xcode/CLT requirements
Message-ID: <0F9E5D80-4892-46DC-B27D-E24B7194139A@mgtech.com>

FYI for macOS folks running FreeSWITCH:

Homebrew made major changes in the last few weeks that affect the installation instructions for FreeSwitch on macOS. It also affects Homebrew update/upgrade functions. I updated 3 pages of the macOS wiki and the MacOS FreeSwitch Installer, macFI. A brief summary:

Homebrew always required the full Xcode instead of the standalone Apple Command Line Tools. That is now the opposite…

When Apple updated Xcode it didn’t always provide the correct version of CLT for the previous macOS. For instance, 10.13 and 10.12 use Xcode 9.2, but 9.2 may install the CLT for 10.13 on 10.12 meaning the CLT was for building 10.13 apps. Homebrew would detect this as a showstopper. This has happened a couple of times, and that’s why we sometimes had to use an older Xcode for a macOS that was not the latest to build FreeSwitch. This didn’t just affect FreeSwitch but some Homebrew formulas.

So…. The Homebrew folks decided to fix this once and for all: Now, when Homebrew is installed, it looks for the correct CLT. If the latest Xcode (9.2) is already installed on the latest macOS (10.13), nothing needs to happen because the CLT that is part of Xcode is already installed and correct. If Homebrew does not detect the CLT, it will locate them online, download and install the correct version for the macOS running!

The big benefit is that if you don’t need Xcode you will save about 4GB of space, and of course it takes a lot less time installing FreeSwitch from scratch.

Again, the wiki and macFI are updated, I tested them on 10.13, 10.12, 10.11, and 10.10.
Mario G

From babak.freeswitch at gmail.com  Wed Dec 13 08:50:10 2017
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Wed, 13 Dec 2017 12:20:10 +0330
Subject: [Freeswitch-users] How should I configure xml_curl bindings for
	phrases?
Message-ID: <CAEVfcab9V_hMj3gbCjuOx3d=7yoVGWXwD66o9cX_AzbDp8U8fA@mail.gmail.com>

Hi
how can I use xml_curl to fetch phrases?
is this right?
<binding name="languages">
    <param name="gateway-url" value="http://localhost:3300/fs/phrases"
bindings="phrases" />
  </binding>
or I should use languages instead of phrases?

and what should be the format of generated xml from server?
thanks
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From shaun.stokes at itec-support.co.uk  Wed Dec 13 11:48:16 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Wed, 13 Dec 2017 11:48:16 +0000
Subject: [Freeswitch-users] FreeSWITCH crash due to Signal 6 Abort while
 executing assert in nua_stack.c
In-Reply-To: <CAEVfcab9V_hMj3gbCjuOx3d=7yoVGWXwD66o9cX_AzbDp8U8fA@mail.gmail.com>
References: <CAEVfcab9V_hMj3gbCjuOx3d=7yoVGWXwD66o9cX_AzbDp8U8fA@mail.gmail.com>
Message-ID: <1513165822345.35740@itec-support.co.uk>

Hi,


Recently experienced a crash (Signal 6 Abort) on FreeSWITCH, daemon log as follows.


systemd[1]: freeswitch.service: main process exited, code=killed, status=6/ABRT
fs_cli[5866]: [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
systemd[1]: freeswitch.service: control process exited, code=exited status=255
systemd[1]: Unit freeswitch.service entered failed state.
systemd[1]: freeswitch.service holdoff time over, scheduling restart.
systemd[1]: Stopping freeswitch...
systemd[1]: Starting freeswitch...


We don't have the FreeSWITCH logs for this particular incident, back-trace as follows.


================================================================================
# bt full
================================================================================
#0  0x00007f6b05f79067 in __GI_raise (sig=sig at entry=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:56
        resultvar = 0
        pid = 20383
        selftid = 20402
#1  0x00007f6b05f7a448 in __GI_abort () at abort.c:89
        save_stage = 2
        act = {__sigaction_handler = {sa_handler = 0x7fff30339e75, sa_sigaction = 0x7fff30339e75}, sa_mask = {__val = {140097639576617, 140097517785411, 959, 4, 140097512340528, 0, 140096967610088, 4294967296, 0, 0, 0, 21474836480, 140097639576143, 140097512340680, 140097676636160, 140097639591704}}, sa_flags = -20442716, sa_restorer = 0x7f6afec8139a <__PRETTY_FUNCTION__.16073>}
        sigs = {__val = {32, 0 <repeats 15 times>}}
#2  0x00007f6b05f72266 in __assert_fail_base (fmt=0x7f6b060aaf18 "%s%s%s:%u: %s%sAssertion `%s' failed.\n%n", assertion=assertion at entry=0x7f6afec811a4 "*nh->nh_prev == nh", file=file at entry=0x7f6afec81143 "nua_stack.c", line=line at entry=959, function=function at entry=0x7f6afec8139a <__PRETTY_FUNCTION__.16073> "nh_remove") at assert.c:92
        str = 0x7f67e44770f0 ""
        total = 4096
#3  0x00007f6b05f72312 in __GI___assert_fail (assertion=assertion at entry=0x7f6afec811a4 "*nh->nh_prev == nh", file=file at entry=0x7f6afec81143 "nua_stack.c", line=line at entry=959, function=function at entry=0x7f6afec8139a <__PRETTY_FUNCTION__.16073> "nh_remove") at assert.c:101
No locals.
#4  0x00007f6afebf3a36 in nh_remove (nua=0x7f6aec019e70, nh=0x7f6adfb244a0) at nua_stack.c:959
No locals.
#5  nh_destroy (nua=0x7f6aec019e70, nh=0x7f6adfb244a0) at nua_stack.c:998
        nh = 0x7f6adfb244a0
        nua = 0x7f6aec019e70
#6  0x00007f6afebf4ba5 in nua_stack_destroy_handle (tags=<optimized out>, nh=<optimized out>, nua=<optimized out>) at nua_stack.c:660
No locals.
#7  nua_stack_signal (nua=0x7f6aec019e70, msg=0x31, ee=0x7f6ac401fba8) at nua_stack.c:661
        nh = 0x7f6adfb244a0
        tags = 0x7f6ac401fbd0
        event = nua_r_destroy
        error = 0
        __func__ = "nua_stack_signal"
#8  0x00007f6afec276d2 in su_base_port_execute_msgs (queue=0x7f67ea5624b0) at su_base_port.c:280
        root = <optimized out>
        f = <optimized out>
        msg = 0x0
        n = 12
#9  0x00007f6afec27bcd in su_base_port_run (self=0x7f6adc0008c0) at su_base_port.c:335
        tout = 15000
        tout2 = 0
        __PRETTY_FUNCTION__ = "su_base_port_run"
#10 0x00007f6afec2c900 in su_pthread_port_clone_main (varg=0x7f6afea2e750) at su_pthread_port.c:343
        arg = 0x0
        task = {{sut_port = 0x7f6adc0008c0, sut_root = 0x7f6adc001130}}
        zap = 1
#11 0x00007f6b062f7064 in start_thread (arg=0x7f6afe750700) at pthread_create.c:309
        __res = <optimized out>
        pd = 0x7f6afe750700
        now = <optimized out>
        unwind_buf = {cancel_jmp_buf = {{jmp_buf = {140097512343296, 1338962068899872702, 0, 140097676709984, 27192672, 140097512343296, -1422840606975082562, -1422294471635779650}, mask_was_saved = 0}}, priv = {pad = {0x0, 0x0, 0x0, 0x0}, data = {prev = 0x0, cleanup = 0x0, canceltype = 0}}}
        not_first_call = <optimized out>
        pagesize_m1 = <optimized out>
        sp = <optimized out>
        freesize = <optimized out>
        __PRETTY_FUNCTION__ = "start_thread"
#12 0x00007f6b0602c62d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111
No locals.

Essentially it failed on 'assert(*nh->nh_prev == nh);?' while executing this function in 'nua_stack.c'.
###############
/** @internal Remove a handle from list of handles */
static
void nh_remove(nua_t *nua, nua_handle_t *nh)
{
  assert(nh_is_inserted(nh)); assert(*nh->nh_prev == nh);

  if (nh->nh_next)
    nh->nh_next->nh_prev = nh->nh_prev;
  else
    nua->nua_handles_tail = nh->nh_prev;

  *nh->nh_prev = nh->nh_next;

  nh->nh_prev = NULL;
  nh->nh_next = NULL;
}
###############


It's difficult for us to understand the exact cause with-out a copy of the logs,? if anyone has seem something similar or might be able to point us in the right direction we would be greatful.

Thanks,
S




______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
______________________________________________________________________
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From shaun.stokes at itec-support.co.uk  Wed Dec 13 12:11:01 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Wed, 13 Dec 2017 12:11:01 +0000
Subject: [Freeswitch-users] FreeSWITCH crash due to Signal 6 Abort while
 executing assert in nua_stack.c
In-Reply-To: <1513165822345.35740@itec-support.co.uk>
References: <CAEVfcab9V_hMj3gbCjuOx3d=7yoVGWXwD66o9cX_AzbDp8U8fA@mail.gmail.com>,
 <1513165822345.35740@itec-support.co.uk>
Message-ID: <1513167187691.96833@itec-support.co.uk>

​FYI.


FreeSWITCH version 1.6.19

Debian 8.9


________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Shaun Stokes <shaun.stokes at itec-support.co.uk>
Sent: 13 December 2017 11:48
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] FreeSWITCH crash due to Signal 6 Abort while executing assert in nua_stack.c


Hi,


Recently experienced a crash (Signal 6 Abort) on FreeSWITCH, daemon log as follows.


systemd[1]: freeswitch.service: main process exited, code=killed, status=6/ABRT
fs_cli[5866]: [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
systemd[1]: freeswitch.service: control process exited, code=exited status=255
systemd[1]: Unit freeswitch.service entered failed state.
systemd[1]: freeswitch.service holdoff time over, scheduling restart.
systemd[1]: Stopping freeswitch...
systemd[1]: Starting freeswitch...


We don't have the FreeSWITCH logs for this particular incident, back-trace as follows.


================================================================================
# bt full
================================================================================
#0  0x00007f6b05f79067 in __GI_raise (sig=sig at entry=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:56
        resultvar = 0
        pid = 20383
        selftid = 20402
#1  0x00007f6b05f7a448 in __GI_abort () at abort.c:89
        save_stage = 2
        act = {__sigaction_handler = {sa_handler = 0x7fff30339e75, sa_sigaction = 0x7fff30339e75}, sa_mask = {__val = {140097639576617, 140097517785411, 959, 4, 140097512340528, 0, 140096967610088, 4294967296, 0, 0, 0, 21474836480, 140097639576143, 140097512340680, 140097676636160, 140097639591704}}, sa_flags = -20442716, sa_restorer = 0x7f6afec8139a <__PRETTY_FUNCTION__.16073>}
        sigs = {__val = {32, 0 <repeats 15 times>}}
#2  0x00007f6b05f72266 in __assert_fail_base (fmt=0x7f6b060aaf18 "%s%s%s:%u: %s%sAssertion `%s' failed.\n%n", assertion=assertion at entry=0x7f6afec811a4 "*nh->nh_prev == nh", file=file at entry=0x7f6afec81143 "nua_stack.c", line=line at entry=959, function=function at entry=0x7f6afec8139a <__PRETTY_FUNCTION__.16073> "nh_remove") at assert.c:92
        str = 0x7f67e44770f0 ""
        total = 4096
#3  0x00007f6b05f72312 in __GI___assert_fail (assertion=assertion at entry=0x7f6afec811a4 "*nh->nh_prev == nh", file=file at entry=0x7f6afec81143 "nua_stack.c", line=line at entry=959, function=function at entry=0x7f6afec8139a <__PRETTY_FUNCTION__.16073> "nh_remove") at assert.c:101
No locals.
#4  0x00007f6afebf3a36 in nh_remove (nua=0x7f6aec019e70, nh=0x7f6adfb244a0) at nua_stack.c:959
No locals.
#5  nh_destroy (nua=0x7f6aec019e70, nh=0x7f6adfb244a0) at nua_stack.c:998
        nh = 0x7f6adfb244a0
        nua = 0x7f6aec019e70
#6  0x00007f6afebf4ba5 in nua_stack_destroy_handle (tags=<optimized out>, nh=<optimized out>, nua=<optimized out>) at nua_stack.c:660
No locals.
#7  nua_stack_signal (nua=0x7f6aec019e70, msg=0x31, ee=0x7f6ac401fba8) at nua_stack.c:661
        nh = 0x7f6adfb244a0
        tags = 0x7f6ac401fbd0
        event = nua_r_destroy
        error = 0
        __func__ = "nua_stack_signal"
#8  0x00007f6afec276d2 in su_base_port_execute_msgs (queue=0x7f67ea5624b0) at su_base_port.c:280
        root = <optimized out>
        f = <optimized out>
        msg = 0x0
        n = 12
#9  0x00007f6afec27bcd in su_base_port_run (self=0x7f6adc0008c0) at su_base_port.c:335
        tout = 15000
        tout2 = 0
        __PRETTY_FUNCTION__ = "su_base_port_run"
#10 0x00007f6afec2c900 in su_pthread_port_clone_main (varg=0x7f6afea2e750) at su_pthread_port.c:343
        arg = 0x0
        task = {{sut_port = 0x7f6adc0008c0, sut_root = 0x7f6adc001130}}
        zap = 1
#11 0x00007f6b062f7064 in start_thread (arg=0x7f6afe750700) at pthread_create.c:309
        __res = <optimized out>
        pd = 0x7f6afe750700
        now = <optimized out>
        unwind_buf = {cancel_jmp_buf = {{jmp_buf = {140097512343296, 1338962068899872702, 0, 140097676709984, 27192672, 140097512343296, -1422840606975082562, -1422294471635779650}, mask_was_saved = 0}}, priv = {pad = {0x0, 0x0, 0x0, 0x0}, data = {prev = 0x0, cleanup = 0x0, canceltype = 0}}}
        not_first_call = <optimized out>
        pagesize_m1 = <optimized out>
        sp = <optimized out>
        freesize = <optimized out>
        __PRETTY_FUNCTION__ = "start_thread"
#12 0x00007f6b0602c62d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111
No locals.

Essentially it failed on 'assert(*nh->nh_prev == nh);​' while executing this function in 'nua_stack.c'.
###############
/** @internal Remove a handle from list of handles */
static
void nh_remove(nua_t *nua, nua_handle_t *nh)
{
  assert(nh_is_inserted(nh)); assert(*nh->nh_prev == nh);

  if (nh->nh_next)
    nh->nh_next->nh_prev = nh->nh_prev;
  else
    nua->nua_handles_tail = nh->nh_prev;

  *nh->nh_prev = nh->nh_next;

  nh->nh_prev = NULL;
  nh->nh_next = NULL;
}
###############


It's difficult for us to understand the exact cause with-out a copy of the logs,​ if anyone has seem something similar or might be able to point us in the right direction we would be greatful.

Thanks,
S




______________________________________________________________________
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______________________________________________________________________

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From igorolhovskiy at gmail.com  Wed Dec 13 13:48:28 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Wed, 13 Dec 2017 15:48:28 +0200
Subject: [Freeswitch-users] FS not reacting to BYE
Message-ID: <4f36450f-7499-4c7f-994c-e3cce8eb6170@Spark>

Hi all!


Got into issue with one of providers, and can’t get some things...
For what reason FS is not reacting to BYE on external profile?

FS -> Prov
INVITE sip:3809XXXXXXXX at sip.telecomax.net SIP/2.0
Via: SIP/2.0/UDP 138.68.65.164:5080;rport;branch=z9hG4bKt7cUUFUKyea3r
Max-Forwards: 69
From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
To: <sip:3809XXXXXXXX at sip.telecomax.net>
Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
CSeq: 116249601 INVITE
Contact: <sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c>
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
X-Serialnumber: 000413719560
P-Key-Flags: resolution="31x13", keys="4"
X-accountcode: fusion.contactise.com
X-FS-Support: update_display,send_info
Remote-Party-ID: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1513140493 1513140494 IN IP4 138.68.65.164
s=FreeSWITCH
c=IN IP4 138.68.65.164
t=0 0
m=audio 30838 RTP/AVP 0 8 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

FS <- Prov
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 138.68.65.164:5080;branch=z9hG4bKUg6KXacQUQ0Nm;received=138.68.65.164;rport=5080
From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
To: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
CSeq: 116249602 INVITE
User-Agent: Free World Web Client
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:3809XXXXXXXX at 80.242.134.195>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 21659 21659 IN IP4 80.242.134.195
s=session
c=IN IP4 80.242.134.195
t=0 0
m=audio 13494 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

FS <- Prov
SIP/2.0 200 OK
Via: SIP/2.0/UDP 138.68.65.164:5080;branch=z9hG4bKUg6KXacQUQ0Nm;received=138.68.65.164;rport=5080
From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
To: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
CSeq: 116249602 INVITE
User-Agent: Free World Web Client
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:3809XXXXXXXX at 80.242.134.195>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 21659 21660 IN IP4 80.242.134.195
s=session
c=IN IP4 80.242.134.195
t=0 0
m=audio 13494 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

FS -> Prov
ACK sip:3809XXXXXXXX at 80.242.134.195 SIP/2.0
Via: SIP/2.0/UDP 138.68.65.164:5080;rport;branch=z9hG4bKvSZcZ5vtr0p8F
Max-Forwards: 70
From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
To: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
CSeq: 116249602 ACK
Contact: <sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c>
Proxy-Authorization: Digest username="456XXXXXXX", realm="Free World Web Client", nonce="6d32caff", algorithm=MD5, uri="sip:3809XXXXXXXX at sip.telecomax.net", response="c0aa24f024fc447594290b580bcedbfa"
Content-Length: 0

<- This BYE is received, but no react to it in CLI at all.

FS <- Prov
BYE sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c SIP/2.0
Via: SIP/2.0/UDP 80.242.134.195:5060;branch=z9hG4bK06dce488;rport
From: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
To: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
CSeq: 102 BYE
User-Agent: Free World Web Client
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0






I've set up sofia debug all 9 and received this
   ------------------------------------------------------------------------
BYE sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c SIP/2.0
   Via: SIP/2.0/UDP 80.242.134.195:5060;branch=z9hG4bK06dce488;rport
   From: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as77b4319c
   To: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
   Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
   CSeq: 102 BYE
   User-Agent: Free World Web Client
   Max-Forwards: 70
   X-Asterisk-HangupCause: Normal Clearing
   X-Asterisk-HangupCauseCode: 16
   Content-Length: 0

   ------------------------------------------------------------------------
tport.c:3023 tport_deliver() tport_deliver(0x7ff1e0004650): msg 0x7ff1e0025700 (536 bytes) from udp/80.242.134.195:5080/sip next=(nil)
nta.c:2880 agent_recv_request() nta: received BYE sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c SIP/2.0 (CSeq 102)
nta.c:3248 agent_aliases() nta: canonizing sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080 with contact
nta.c:3060 agent_recv_request() nta: BYE (102) going to existing leg
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
nua_stack.c:271 nua_stack_event() nua(0x7ff204000b60): event i_bye 100 Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nta.c:7134 _nta_incoming_timer() nta: timer I fired, terminate 200 response
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7ff20dc4ac60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta.c:1289 agent_timer() nta: timer not set
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7ff1e8004410): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7ff1e8004410)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7ff1e8004410) msg 0x7ff1e80c90c0 from (udp/138.68.65.164:5060) has 4 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7ff1e8004410): bad msg 0x7ff1e80c90c0 (4 bytes) from udp/176.104.56.91:5060/sip next=(nil)
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7ff1e0004650): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7ff1e0004650)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7ff1e0004650) msg 0x7ff1e00582d0 from (udp/138.68.65.164:5080) has 536 bytes, veclen = 1



And this is it….
Also there is one-way sound, but I think it’s somehow related to this.

FS 1.6.19 (FusionPBX)

Regards, Igor
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From krice at freeswitch.org  Wed Dec 13 14:31:32 2017
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 13 Dec 2017 08:31:32 -0600
Subject: [Freeswitch-users] FreeSWITCH crash due to Signal 6 Abort while
	executing assert in nua_stack.c
In-Reply-To: <1513167187691.96833@itec-support.co.uk>
References: <CAEVfcab9V_hMj3gbCjuOx3d=7yoVGWXwD66o9cX_AzbDp8U8fA@mail.gmail.com>,
 <1513165822345.35740@itec-support.co.uk>
 <1513167187691.96833@itec-support.co.uk>
Message-ID: <1a1601d3741f$1324e6b0$396eb410$@freeswitch.org>

Bug reports go to https://freeswitch.org/jira not the mailing list

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes
Sent: Wednesday, December 13, 2017 6:11 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSWITCH crash due to Signal 6 Abort while executing assert in nua_stack.c

 

​FYI.

 

FreeSWITCH version 1.6.19

Debian 8.9

 

  _____  

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > on behalf of Shaun Stokes <shaun.stokes at itec-support.co.uk <mailto:shaun.stokes at itec-support.co.uk> >
Sent: 13 December 2017 11:48
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] FreeSWITCH crash due to Signal 6 Abort while executing assert in nua_stack.c 

 

Hi,

 

Recently experienced a crash (Signal 6 Abort) on FreeSWITCH, daemon log as follows.

 

systemd[1]: freeswitch.service: main process exited, code=killed, status=6/ABRT

fs_cli[5866]: [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]

systemd[1]: freeswitch.service: control process exited, code=exited status=255

systemd[1]: Unit freeswitch.service entered failed state.

systemd[1]: freeswitch.service holdoff time over, scheduling restart.

systemd[1]: Stopping freeswitch...

systemd[1]: Starting freeswitch...

 

We don't have the FreeSWITCH logs for this particular incident, back-trace as follows.

 

================================================================================

# bt full

================================================================================

#0  0x00007f6b05f79067 in __GI_raise (sig=sig at entry=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:56

        resultvar = 0

        pid = 20383

        selftid = 20402

#1  0x00007f6b05f7a448 in __GI_abort () at abort.c:89

        save_stage = 2

        act = {__sigaction_handler = {sa_handler = 0x7fff30339e75, sa_sigaction = 0x7fff30339e75}, sa_mask = {__val = {140097639576617, 140097517785411, 959, 4, 140097512340528, 0, 140096967610088, 4294967296, 0, 0, 0, 21474836480, 140097639576143, 140097512340680, 140097676636160, 140097639591704}}, sa_flags = -20442716, sa_restorer = 0x7f6afec8139a <__PRETTY_FUNCTION__.16073>}

        sigs = {__val = {32, 0 <repeats 15 times>}}

#2  0x00007f6b05f72266 in __assert_fail_base (fmt=0x7f6b060aaf18 "%s%s%s:%u: %s%sAssertion `%s' failed.\n%n", assertion=assertion at entry=0x7f6afec811a4 "*nh->nh_prev == nh", file=file at entry=0x7f6afec81143 "nua_stack.c", line=line at entry=959, function=function at entry=0x7f6afec8139a <__PRETTY_FUNCTION__.16073> "nh_remove") at assert.c:92

        str = 0x7f67e44770f0 ""

        total = 4096

#3  0x00007f6b05f72312 in __GI___assert_fail (assertion=assertion at entry=0x7f6afec811a4 "*nh->nh_prev == nh", file=file at entry=0x7f6afec81143 "nua_stack.c", line=line at entry=959, function=function at entry=0x7f6afec8139a <__PRETTY_FUNCTION__.16073> "nh_remove") at assert.c:101

No locals.

#4  0x00007f6afebf3a36 in nh_remove (nua=0x7f6aec019e70, nh=0x7f6adfb244a0) at nua_stack.c:959

No locals.

#5  nh_destroy (nua=0x7f6aec019e70, nh=0x7f6adfb244a0) at nua_stack.c:998

        nh = 0x7f6adfb244a0

        nua = 0x7f6aec019e70

#6  0x00007f6afebf4ba5 in nua_stack_destroy_handle (tags=<optimized out>, nh=<optimized out>, nua=<optimized out>) at nua_stack.c:660

No locals.

#7  nua_stack_signal (nua=0x7f6aec019e70, msg=0x31, ee=0x7f6ac401fba8) at nua_stack.c:661

        nh = 0x7f6adfb244a0

        tags = 0x7f6ac401fbd0

        event = nua_r_destroy

        error = 0

        __func__ = "nua_stack_signal"

#8  0x00007f6afec276d2 in su_base_port_execute_msgs (queue=0x7f67ea5624b0) at su_base_port.c:280

        root = <optimized out>

        f = <optimized out>

        msg = 0x0

        n = 12

#9  0x00007f6afec27bcd in su_base_port_run (self=0x7f6adc0008c0) at su_base_port.c:335

        tout = 15000

        tout2 = 0

        __PRETTY_FUNCTION__ = "su_base_port_run"

#10 0x00007f6afec2c900 in su_pthread_port_clone_main (varg=0x7f6afea2e750) at su_pthread_port.c:343

        arg = 0x0

        task = {{sut_port = 0x7f6adc0008c0, sut_root = 0x7f6adc001130}}

        zap = 1

#11 0x00007f6b062f7064 in start_thread (arg=0x7f6afe750700) at pthread_create.c:309

        __res = <optimized out>

        pd = 0x7f6afe750700

        now = <optimized out>

        unwind_buf = {cancel_jmp_buf = {{jmp_buf = {140097512343296, 1338962068899872702, 0, 140097676709984, 27192672, 140097512343296, -1422840606975082562, -1422294471635779650}, mask_was_saved = 0}}, priv = {pad = {0x0, 0x0, 0x0, 0x0}, data = {prev = 0x0, cleanup = 0x0, canceltype = 0}}}

        not_first_call = <optimized out>

        pagesize_m1 = <optimized out>

        sp = <optimized out>

        freesize = <optimized out>

        __PRETTY_FUNCTION__ = "start_thread"

#12 0x00007f6b0602c62d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111

No locals.

 

Essentially it failed on 'assert(*nh->nh_prev == nh);​' while executing this function in 'nua_stack.c'.

###############

/** @internal Remove a handle from list of handles */

static

void nh_remove(nua_t *nua, nua_handle_t *nh)

{

  assert(nh_is_inserted(nh)); assert(*nh->nh_prev == nh);

 

  if (nh->nh_next)

    nh->nh_next->nh_prev = nh->nh_prev;

  else

    nua->nua_handles_tail = nh->nh_prev;

 

  *nh->nh_prev = nh->nh_next;

 

  nh->nh_prev = NULL;

  nh->nh_next = NULL;

}

###############

 

 

It's difficult for us to understand the exact cause with-out a copy of the logs,​ if anyone has seem something similar or might be able to point us in the right direction we would be greatful.

 

Thanks,

S

 

 

 


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This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
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From shaun.stokes at itec-support.co.uk  Wed Dec 13 15:39:22 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Wed, 13 Dec 2017 15:39:22 +0000
Subject: [Freeswitch-users] FreeSWITCH crash due to Signal 6 Abort
	while	executing assert in nua_stack.c
In-Reply-To: <1a1601d3741f$1324e6b0$396eb410$@freeswitch.org>
References: <CAEVfcab9V_hMj3gbCjuOx3d=7yoVGWXwD66o9cX_AzbDp8U8fA@mail.gmail.com>, 
 <1513165822345.35740@itec-support.co.uk>
 <1513167187691.96833@itec-support.co.uk>,
 <1a1601d3741f$1324e6b0$396eb410$@freeswitch.org>
Message-ID: <1513179689650.20901@itec-support.co.uk>

​Raised a JIRA but as we don't yet know the exact cause we can't replicate this so hasn't been reproduced on master yet.


Thanks,

S

________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Ken Rice <krice at freeswitch.org>
Sent: 13 December 2017 14:31
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] FreeSWITCH crash due to Signal 6 Abort while executing assert in nua_stack.c

Bug reports go to https://freeswitch.org/jira not the mailing list

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes
Sent: Wednesday, December 13, 2017 6:11 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSWITCH crash due to Signal 6 Abort while executing assert in nua_stack.c


​FYI.



FreeSWITCH version 1.6.19

Debian 8.9



________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Shaun Stokes <shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>>
Sent: 13 December 2017 11:48
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] FreeSWITCH crash due to Signal 6 Abort while executing assert in nua_stack.c


Hi,



Recently experienced a crash (Signal 6 Abort) on FreeSWITCH, daemon log as follows.


systemd[1]: freeswitch.service: main process exited, code=killed, status=6/ABRT
fs_cli[5866]: [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
systemd[1]: freeswitch.service: control process exited, code=exited status=255
systemd[1]: Unit freeswitch.service entered failed state.
systemd[1]: freeswitch.service holdoff time over, scheduling restart.
systemd[1]: Stopping freeswitch...
systemd[1]: Starting freeswitch...



We don't have the FreeSWITCH logs for this particular incident, back-trace as follows.


================================================================================
# bt full
================================================================================
#0  0x00007f6b05f79067 in __GI_raise (sig=sig at entry=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:56
        resultvar = 0
        pid = 20383
        selftid = 20402
#1  0x00007f6b05f7a448 in __GI_abort () at abort.c:89
        save_stage = 2
        act = {__sigaction_handler = {sa_handler = 0x7fff30339e75, sa_sigaction = 0x7fff30339e75}, sa_mask = {__val = {140097639576617, 140097517785411, 959, 4, 140097512340528, 0, 140096967610088, 4294967296, 0, 0, 0, 21474836480, 140097639576143, 140097512340680, 140097676636160, 140097639591704}}, sa_flags = -20442716, sa_restorer = 0x7f6afec8139a <__PRETTY_FUNCTION__.16073>}
        sigs = {__val = {32, 0 <repeats 15 times>}}
#2  0x00007f6b05f72266 in __assert_fail_base (fmt=0x7f6b060aaf18 "%s%s%s:%u: %s%sAssertion `%s' failed.\n%n", assertion=assertion at entry=0x7f6afec811a4 "*nh->nh_prev == nh", file=file at entry=0x7f6afec81143 "nua_stack.c", line=line at entry=959, function=function at entry=0x7f6afec8139a <__PRETTY_FUNCTION__.16073> "nh_remove") at assert.c:92
        str = 0x7f67e44770f0 ""
        total = 4096
#3  0x00007f6b05f72312 in __GI___assert_fail (assertion=assertion at entry=0x7f6afec811a4 "*nh->nh_prev == nh", file=file at entry=0x7f6afec81143 "nua_stack.c", line=line at entry=959, function=function at entry=0x7f6afec8139a <__PRETTY_FUNCTION__.16073> "nh_remove") at assert.c:101
No locals.
#4  0x00007f6afebf3a36 in nh_remove (nua=0x7f6aec019e70, nh=0x7f6adfb244a0) at nua_stack.c:959
No locals.
#5  nh_destroy (nua=0x7f6aec019e70, nh=0x7f6adfb244a0) at nua_stack.c:998
        nh = 0x7f6adfb244a0
        nua = 0x7f6aec019e70
#6  0x00007f6afebf4ba5 in nua_stack_destroy_handle (tags=<optimized out>, nh=<optimized out>, nua=<optimized out>) at nua_stack.c:660
No locals.
#7  nua_stack_signal (nua=0x7f6aec019e70, msg=0x31, ee=0x7f6ac401fba8) at nua_stack.c:661
        nh = 0x7f6adfb244a0
        tags = 0x7f6ac401fbd0
        event = nua_r_destroy
        error = 0
        __func__ = "nua_stack_signal"
#8  0x00007f6afec276d2 in su_base_port_execute_msgs (queue=0x7f67ea5624b0) at su_base_port.c:280
        root = <optimized out>
        f = <optimized out>
        msg = 0x0
        n = 12
#9  0x00007f6afec27bcd in su_base_port_run (self=0x7f6adc0008c0) at su_base_port.c:335
        tout = 15000
        tout2 = 0
        __PRETTY_FUNCTION__ = "su_base_port_run"
#10 0x00007f6afec2c900 in su_pthread_port_clone_main (varg=0x7f6afea2e750) at su_pthread_port.c:343
        arg = 0x0
        task = {{sut_port = 0x7f6adc0008c0, sut_root = 0x7f6adc001130}}
        zap = 1
#11 0x00007f6b062f7064 in start_thread (arg=0x7f6afe750700) at pthread_create.c:309
        __res = <optimized out>
        pd = 0x7f6afe750700
        now = <optimized out>
        unwind_buf = {cancel_jmp_buf = {{jmp_buf = {140097512343296, 1338962068899872702, 0, 140097676709984, 27192672, 140097512343296, -1422840606975082562, -1422294471635779650}, mask_was_saved = 0}}, priv = {pad = {0x0, 0x0, 0x0, 0x0}, data = {prev = 0x0, cleanup = 0x0, canceltype = 0}}}
        not_first_call = <optimized out>
        pagesize_m1 = <optimized out>
        sp = <optimized out>
        freesize = <optimized out>
        __PRETTY_FUNCTION__ = "start_thread"
#12 0x00007f6b0602c62d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111
No locals.

Essentially it failed on 'assert(*nh->nh_prev == nh);​' while executing this function in 'nua_stack.c'.
###############
/** @internal Remove a handle from list of handles */
static
void nh_remove(nua_t *nua, nua_handle_t *nh)
{
  assert(nh_is_inserted(nh)); assert(*nh->nh_prev == nh);

  if (nh->nh_next)
    nh->nh_next->nh_prev = nh->nh_prev;
  else
    nua->nua_handles_tail = nh->nh_prev;

  *nh->nh_prev = nh->nh_next;

  nh->nh_prev = NULL;
  nh->nh_next = NULL;
}
###############


It's difficult for us to understand the exact cause with-out a copy of the logs,​ if anyone has seem something similar or might be able to point us in the right direction we would be greatful.

Thanks,
S





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This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
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This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
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______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
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From philipp at zeitschel.net  Wed Dec 13 09:12:55 2017
From: philipp at zeitschel.net (Philipp Zeitschel)
Date: Wed, 13 Dec 2017 09:12:55 +0000
Subject: [Freeswitch-users] Jittering only on internal Lines
Message-ID: <2c5b65d33f304677aa551d8e359480aa@zeitschel.net>

Hi,

Hope somebody can help
I'm running 1.6.19~36~7a77e0b-1~jessie+1
The problem is that I got jittering on the internal line. If I get a call from outside (leg a) I can hear leg a totally clear only on leg b (my internal telephone) I get sound cuts.
If I make a total internal call both side have jitters, internal is no nat, the phones are in the same /24 Network with freeswitch.
I have no glue how to debug this, can you guide me in the right direction?
Here is ja call log:

nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (1)
nua_stack.c:271 nua_stack_event() nua(0x7fe6d8059c20): event i_invite 100 Trying
nua_session.c:4139 signal_call_state_change() nua(0x7fe6d8059c20): call state changed: init -> received, received offer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fe6d805e1a0, [0x7fe7040b88c8], [0x7fe7040b88d0], [(nil)]) called
nua_stack.c:271 nua_stack_event() nua(0x7fe6d8059c20): event i_state 100 Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2017-12-13 10:05:04.668947 [NOTICE] switch_channel.c:1104 New Channel sofia/external/+xxx at tmobile.de [d220a9b4-8c44-49ce-a226-18625641c2ed]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:584 (sofia/external/+xxx at tmobile.de) Running State Change CS_NEW (Cur 1 Tot 205)
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:04.668947 [DEBUG] sofia.c:9873 sofia/external/+xxx at tmobile.de receiving invite from 217.0.23.100:5060 version: 1.6.19 -36-7a77e0b 64bit
nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil)
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:04.668947 [DEBUG] sofia.c:7084 Channel sofia/external/+xxx at tmobile.de entering state [received][100]
2017-12-13 10:05:04.668947 [DEBUG] sofia.c:7094 Remote SDP:
v=0
o=- 629563164 271544510 IN IP4 217.0.23.100
s=-
c=IN IP4 217.0.4.164
t=0 0
a=sendrecv
m=audio 25732 RTP/AVP 9 8 100
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=maxptime:40
a=ptime:20

2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 100 at 8000
2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:3061 Set Codec sofia/external/+xxx at tmobile.de G722/8000 20 ms 160 samples 64000 bits 1 channels
2017-12-13 10:05:04.668947 [DEBUG] switch_core_codec.c:111 sofia/external/+xxx at tmobile.de Original read codec set to G722:9
2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 100 at 8000
2017-12-13 10:05:04.668947 [DEBUG] switch_core_media.c:4767 sofia/external/+xxx at tmobile.de Set 2833 dtmf send payload to 100 recv payload to 100
2017-12-13 10:05:04.668947 [DEBUG] sofia.c:7507 (sofia/external/+xxx at tmobile.de) State Change CS_NEW -> CS_INIT
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:603 (sofia/external/+xxx at tmobile.de) State NEW
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:584 (sofia/external/+xxx at tmobile.de) Running State Change CS_INIT (Cur 1 Tot 205)
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:627 (sofia/external/+xxx at tmobile.de) State INIT
2017-12-13 10:05:04.668947 [DEBUG] mod_sofia.c:90 sofia/external/+xxx at tmobile.de SOFIA INIT
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:40 sofia/external/+xxx at tmobile.de Standard INIT
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:48 (sofia/external/+xxx at tmobile.de) State Change CS_INIT -> CS_ROUTING
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:627 (sofia/external/+xxx at tmobile.de) State INIT going to sleep
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:584 (sofia/external/+xxx at tmobile.de) Running State Change CS_ROUTING (Cur 1 Tot 205)
2017-12-13 10:05:04.668947 [DEBUG] switch_channel.c:2249 (sofia/external/+xxx at tmobile.de) Callstate Change DOWN -> RINGING
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:643 (sofia/external/+xxx at tmobile.de) State ROUTING
2017-12-13 10:05:04.668947 [DEBUG] mod_sofia.c:143 sofia/external/+xxx at tmobile.de SOFIA ROUTING
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:236 sofia/external/+xxx at tmobile.de Standard ROUTING
2017-12-13 10:05:04.668947 [INFO] mod_dialplan_xml.c:637 Processing +xxx <+xxx>->xxx in context public
Dialplan: sofia/external/+xxx at tmobile.de parsing [public->xxx] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [xxx] destination_number(xxx) =~ /^(xxx)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de Action set(call_direction=inbound) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(call_direction=inbound)
2017-12-13 10:05:04.668947 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [call_direction]=[inbound]
Dialplan: sofia/external/+xxx at tmobile.de Action set(domain_uuid=404d503a-ef07-4c72-9b52-1ac9e32a3a71) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(domain_uuid=404d503a-ef07-4c72-9b52-1ac9e32a3a71)
2017-12-13 10:05:04.668947 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [domain_uuid]=[404d503a-ef07-4c72-9b52-1ac9e32a3a71]
Dialplan: sofia/external/+xxx at tmobile.de Action set(domain_name=sip.xxx) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(domain_name=sip.xxx)
2017-12-13 10:05:04.668947 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [domain_name]=[sip.xxx]
Dialplan: sofia/external/+xxx at tmobile.de Action transfer(50 XML sip.xxx)
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:286 (sofia/external/+xxx at tmobile.de) State Change CS_ROUTING -> CS_EXECUTE
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:643 (sofia/external/+xxx at tmobile.de) State ROUTING going to sleep
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:584 (sofia/external/+xxx at tmobile.de) Running State Change CS_EXECUTE (Cur 1 Tot 205)
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:650 (sofia/external/+xxx at tmobile.de) State EXECUTE
2017-12-13 10:05:04.668947 [DEBUG] mod_sofia.c:198 sofia/external/+xxx at tmobile.de SOFIA EXECUTE
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:328 sofia/external/+xxx at tmobile.de Standard EXECUTE
EXECUTE sofia/external/+xxx at tmobile.de transfer(50 XML sip.xxx)
2017-12-13 10:05:04.668947 [DEBUG] switch_ivr.c:2165 (sofia/external/+xxx at tmobile.de) State Change CS_EXECUTE -> CS_ROUTING
2017-12-13 10:05:04.668947 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/+xxx at tmobile.de to XML[50 at sip.xxx]
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:650 (sofia/external/+xxx at tmobile.de) State EXECUTE going to sleep
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:584 (sofia/external/+xxx at tmobile.de) Running State Change CS_ROUTING (Cur 1 Tot 205)
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:643 (sofia/external/+xxx at tmobile.de) State ROUTING
2017-12-13 10:05:04.668947 [DEBUG] mod_sofia.c:143 sofia/external/+xxx at tmobile.de SOFIA ROUTING
2017-12-13 10:05:04.668947 [DEBUG] switch_core_state_machine.c:236 sofia/external/+xxx at tmobile.de Standard ROUTING
2017-12-13 10:05:04.668947 [INFO] mod_dialplan_xml.c:637 Processing +xxx <+xxx>->50 in context sip.xxx
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->user_exists] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [user_exists] () =~ // break=on-false
Dialplan: sofia/external/+xxx at tmobile.de Action set(user_exists=${user_exists id ${destination_number} ${domain_name}}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(user_exists=true)
2017-12-13 10:05:04.668947 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [user_exists]=[true]
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [user_exists] ${user_exists}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de Action set(extension_uuid=${user_data ${destination_number}@${domain_name} var extension_uuid}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(extension_uuid=55f069a7-483d-4793-ba37-bb75b3e368ea)
2017-12-13 10:05:04.708841 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [extension_uuid]=[55f069a7-483d-4793-ba37-bb75b3e368ea]
Dialplan: sofia/external/+xxx at tmobile.de Action set(hold_music=${user_data ${destination_number}@${domain_name} var hold_music}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(hold_music=local_stream://default)
2017-12-13 10:05:04.708841 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [hold_music]=[local_stream://default]
Dialplan: sofia/external/+xxx at tmobile.de Action set(forward_all_enabled=${user_data ${destination_number}@${domain_name} var forward_all_enabled}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(forward_all_enabled=)
2017-12-13 10:05:04.708841 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [forward_all_enabled]=[UNDEF]
Dialplan: sofia/external/+xxx at tmobile.de Action set(forward_all_destination=${user_data ${destination_number}@${domain_name} var forward_all_destination}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(forward_all_destination=)
2017-12-13 10:05:04.728848 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [forward_all_destination]=[UNDEF]
Dialplan: sofia/external/+xxx at tmobile.de Action set(forward_busy_enabled=${user_data ${destination_number}@${domain_name} var forward_busy_enabled}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(forward_busy_enabled=)
2017-12-13 10:05:04.728848 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [forward_busy_enabled]=[UNDEF]
Dialplan: sofia/external/+xxx at tmobile.de Action set(forward_busy_destination=${user_data ${destination_number}@${domain_name} var forward_busy_destination}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(forward_busy_destination=)
2017-12-13 10:05:04.748842 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [forward_busy_destination]=[UNDEF]
Dialplan: sofia/external/+xxx at tmobile.de Action set(forward_no_answer_enabled=${user_data ${destination_number}@${domain_name} var forward_no_answer_enabled}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(forward_no_answer_enabled=)
2017-12-13 10:05:04.748842 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [forward_no_answer_enabled]=[UNDEF]
Dialplan: sofia/external/+xxx at tmobile.de Action set(forward_no_answer_destination=${user_data ${destination_number}@${domain_name} var forward_no_answer_destination}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(forward_no_answer_destination=)
2017-12-13 10:05:04.748842 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [forward_no_answer_destination]=[UNDEF]
Dialplan: sofia/external/+xxx at tmobile.de Action set(forward_user_not_registered_enabled=${user_data ${destination_number}@${domain_name} var forward_user_not_registered_enabled}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(forward_user_not_registered_enabled=)
2017-12-13 10:05:04.768845 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [forward_user_not_registered_enabled]=[UNDEF]
Dialplan: sofia/external/+xxx at tmobile.de Action set(forward_user_not_registered_destination=${user_data ${destination_number}@${domain_name} var forward_user_not_registered_destination}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(forward_user_not_registered_destination=)
2017-12-13 10:05:04.768845 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [forward_user_not_registered_destination]=[UNDEF]
Dialplan: sofia/external/+xxx at tmobile.de Action set(do_not_disturb=${user_data ${destination_number}@${domain_name} var do_not_disturb}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(do_not_disturb=)
2017-12-13 10:05:04.768845 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [do_not_disturb]=[UNDEF]
Dialplan: sofia/external/+xxx at tmobile.de Action set(call_timeout=${user_data ${destination_number}@${domain_name} var call_timeout}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(call_timeout=30)
2017-12-13 10:05:04.788848 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [call_timeout]=[30]
Dialplan: sofia/external/+xxx at tmobile.de Action set(missed_call_app=${user_data ${destination_number}@${domain_name} var missed_call_app}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(missed_call_app=)
2017-12-13 10:05:04.788848 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [missed_call_app]=[UNDEF]
Dialplan: sofia/external/+xxx at tmobile.de Action set(missed_call_data=${user_data ${destination_number}@${domain_name} var missed_call_data}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(missed_call_data=)
2017-12-13 10:05:04.788848 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [missed_call_data]=[UNDEF]
Dialplan: sofia/external/+xxx at tmobile.de Action set(toll_allow=${user_data ${destination_number}@${domain_name} var toll_allow}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(toll_allow=domestic,international,local)
2017-12-13 10:05:04.808842 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [toll_allow]=[domestic,international,local]
Dialplan: sofia/external/+xxx at tmobile.de Action set(call_screen_enabled=${user_data ${destination_number}@${domain_name} var call_screen_enabled}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(call_screen_enabled=false)
2017-12-13 10:05:04.808842 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [call_screen_enabled]=[false]
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->call-direction] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [call-direction] ${call_direction}(inbound) =~ /^(inbound|outbound|local)$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->variables] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [variables] () =~ // break=on-false
Dialplan: sofia/external/+xxx at tmobile.de Action export(origination_callee_id_name=${destination_number})
Dialplan: sofia/external/+xxx at tmobile.de Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->user_record] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [user_record] () =~ // break=on-false
Dialplan: sofia/external/+xxx at tmobile.de Action set(user_record=${user_data ${destination_number}@${domain_name} var user_record}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(user_record=all)
2017-12-13 10:05:04.828844 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [user_record]=[all]
Dialplan: sofia/external/+xxx at tmobile.de Action set(from_user_exists=${user_exists id ${sip_from_user} ${sip_from_host}}) INLINE
2017-12-13 10:05:04.848838 [DEBUG] freeswitch_lua.cpp:365 DBH handle 0x7fe6f032c110 Connected.
2017-12-13 10:05:04.848838 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0x7fe6f032c110 released.
EXECUTE sofia/external/+xxx at tmobile.de set(from_user_exists=false)
2017-12-13 10:05:04.848838 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [from_user_exists]=[false]
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [user_record] ${user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [user_record] ${user_record}(all) =~ /^all$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Action set(record_session=true) INLINE
EXECUTE sofia/external/+xxx at tmobile.de set(record_session=true)
2017-12-13 10:05:04.848838 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [record_session]=[true]
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [user_record] ${user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [user_record] ${call_direction}(inbound) =~ /^inbound$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${user_record}(all) =~ /^inbound$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [user_record] ${user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${call_direction}(inbound) =~ /^outbound$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${user_record}(all) =~ /^outbound$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [user_record] ${user_exists}(true) =~ /^true$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${call_direction}(inbound) =~ /^local$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${user_record}(all) =~ /^local$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${from_user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${from_user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${from_user_record}() =~ /^all$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${from_user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [user_record] ${call_direction}(inbound) =~ /^inbound$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${from_user_record}() =~ /^inbound$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${from_user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${call_direction}(inbound) =~ /^outbound$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${from_user_record}() =~ /^outbound$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${from_user_exists}(false) =~ /^true$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${call_direction}(inbound) =~ /^local$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [user_record] ${from_user_record}() =~ /^local$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [user_record] ${record_session}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de Action export(nolocal:api_on_answer=uuid_record ${uuid} start ${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext})
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->redial] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [redial] destination_number(50) =~ /^(redial|\*870)$/ break=on-true
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [redial] () =~ // break=never
Dialplan: sofia/external/+xxx at tmobile.de Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->speed_dial] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [speed_dial] destination_number(50) =~ /^\*0(.*)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->Telekom_4000733.0d] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [Telekom_4000733.0d] username(+xxx) =~ /50/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->Telekom_9236276.0d] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [Telekom_9236276.0d] username(+xxx) =~ /100/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->Telekom_9236276.0d-fritz] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [Telekom_9236276.0d-fritz] username(+xxx) =~ /1001/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->Telekom_9236277.0d] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [Telekom_9236277.0d] username(+xxx) =~ /10/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->Telekom_9236277.110] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [Telekom_9236277.110] destination_number(50) =~ /^(110)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->Telekom_9236277.112] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [Telekom_9236277.112] destination_number(50) =~ /^(112)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->Telekom_9236277.19222] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [Telekom_9236277.19222] destination_number(50) =~ /^(19222)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->Telekom_9727575.0d] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [Telekom_9727575.0d] username(+xxx) =~ /30/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [Telekom_9727575.0d] destination_number(50) =~ /^0(\d*)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->Telekom_9727577.0d] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [Telekom_9727577.0d] username(+xxx) =~ /20/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->agent_status] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [agent_status] destination_number(50) =~ /^\*22$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->agent_status_id] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [agent_status_id] destination_number(50) =~ /^\*23$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->group-intercept] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [group-intercept] destination_number(50) =~ /^\*8$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->page-extension] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [page-extension] destination_number(50) =~ /^\*8(\d{2,7})$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->eavesdrop] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [eavesdrop] destination_number(50) =~ /^\*33(\d{2,7})$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->call_privacy] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [call_privacy] destination_number(50) =~ /^\*67(\d+)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->call_return] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [call_return] destination_number(50) =~ /^\*69$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->extension_queue] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [extension_queue] destination_number(50) =~ /^\*800(.*)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->intercept-ext] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [intercept-ext] destination_number(50) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->intercept-ext-polycom] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [intercept-ext-polycom] destination_number(50) =~ /^\*97(\d+)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->dx] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [dx] destination_number(50) =~ /^dx$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->att_xfer] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [att_xfer] destination_number(50) =~ /^att_xfer$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->extension-to-voicemail] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [extension-to-voicemail] ${user_exists}(true) =~ /^true$/ break=on-false
2017-12-13 10:05:04.848838 [ERR] switch_regex.c:104 COMPILE ERROR: 1 [nothing to repeat][^+xxx$]
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [extension-to-voicemail] username(+xxx) =~ /^+xxx$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->fax] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [fax] destination_number(50) =~ /^100$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->send_to_voicemail] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [send_to_voicemail] destination_number(50) =~ /^\*99(\d{2,10})$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->vmain] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [vmain] destination_number(50) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->xfer_vm] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [xfer_vm] destination_number(50) =~ /^xfer_vm$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->is_transfer] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [is_transfer] destination_number(50) =~ /^is_transfer$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->vmain_user] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [vmain_user] destination_number(50) =~ /^\*97$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->cf] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [cf] destination_number(50) =~ /^cf$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->delay_echo] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [delay_echo] destination_number(50) =~ /^\*9195$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->echo] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [echo] destination_number(50) =~ /^\*9196$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->is_zrtp_secure] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [is_zrtp_secure] ${zrtp_secure_media_confirmed}() =~ /^true$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de ANTI-Action eval(not_secure)
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->milliwatt] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [milliwatt] destination_number(50) =~ /^\*9197$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->is_secure] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [is_secure] ${sip_via_protocol}(udp) =~ /tls/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->tone_stream] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [tone_stream] destination_number(50) =~ /^\*9198$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->hold_music] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [hold_music] destination_number(50) =~ /^\*9664$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->recordings] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [recordings] destination_number(50) =~ /^\*(732)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->directory] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [directory] destination_number(50) =~ /^\*411$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->wake-up] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [wake-up] destination_number(50) =~ /^\*(925)$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->valet_park] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [valet_park] destination_number(50) =~ /^(park\+)?(\*59[0-9][0-9])$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [valet_park] ${sip_h_Referred-By}() =~ /sip:(.*)@.*/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [valet_park] destination_number(50) =~ /^(park\+)?(\*59[0-9][0-9])$/ break=never
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [valet_park] destination_number(50) =~ /^(park\+)?(\*59[0-9][0-9])$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->operator] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [operator] destination_number(50) =~ /^0$|^operator$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->operator-forward] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [operator-forward] destination_number(50) =~ /^\*000$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->do-not-disturb] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [do-not-disturb] destination_number(50) =~ /^\*77$/ break=on-true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [do-not-disturb] destination_number(50) =~ /^\*78$|\*363$/ break=on-true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [do-not-disturb] destination_number(50) =~ /^\*79$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->call-forward] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [call-forward] destination_number(50) =~ /^\*72$/ break=on-true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [call-forward] destination_number(50) =~ /^\*73$/ break=on-true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [call-forward] destination_number(50) =~ /^\*74$/ break=on-true
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->call forward all] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [call forward all] ${user_exists}(true) =~ /^true/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [call forward all] ${forward_all_enabled}() =~ /^true/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->follow-me] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [follow-me] destination_number(50) =~ /^\*21$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->clear_sip_auto_answer] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [clear_sip_auto_answer] ${click_to_call}() =~ /true/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->talking clock date and time] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [talking clock date and time] destination_number(50) =~ /^\*9172$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->talking clock time] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [talking clock time] destination_number(50) =~ /^\*9170$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->talking clock date] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [talking clock date] destination_number(50) =~ /^\*9171$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->call_screen] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (FAIL) [call_screen] ${call_screen_enabled}(false) =~ /^true$/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->local_extension] continue=true
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [local_extension] ${user_exists}(true) =~ /true/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de Action export(dialed_extension=${destination_number}) INLINE
EXECUTE sofia/external/+xxx at tmobile.de export(dialed_extension=50)
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [dialed_extension]=[50]
Dialplan: sofia/external/+xxx at tmobile.de Action limit(hash ${domain_name} ${destination_number} ${limit_max} ${limit_destination})
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [local_extension] () =~ // break=on-false
Dialplan: sofia/external/+xxx at tmobile.de Action set(hangup_after_bridge=true)
Dialplan: sofia/external/+xxx at tmobile.de Action set(continue_on_fail=true)
Dialplan: sofia/external/+xxx at tmobile.de Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
Dialplan: sofia/external/+xxx at tmobile.de Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/external/+xxx at tmobile.de Action set(called_party_call_group=${user_data(${dialed_extension}@${domain_name} var call_group)})
Dialplan: sofia/external/+xxx at tmobile.de Action hash(insert/${domain_name}-last_dial/${called_party_call_group}/${uuid})
Dialplan: sofia/external/+xxx at tmobile.de Action set(api_hangup_hook=lua app.lua hangup)
Dialplan: sofia/external/+xxx at tmobile.de Action export(domain_name=${domain_name})
Dialplan: sofia/external/+xxx at tmobile.de Action bridge(user/${destination_number}@${domain_name})
Dialplan: sofia/external/+xxx at tmobile.de Action lua(app.lua failure_handler)
Dialplan: sofia/external/+xxx at tmobile.de parsing [sip.xxx->voicemail] continue=false
Dialplan: sofia/external/+xxx at tmobile.de Regex (PASS) [voicemail] ${user_exists}(true) =~ /true/ break=on-false
Dialplan: sofia/external/+xxx at tmobile.de Action answer()
Dialplan: sofia/external/+xxx at tmobile.de Action sleep(1000)
Dialplan: sofia/external/+xxx at tmobile.de Action set(voicemail_action=save)
Dialplan: sofia/external/+xxx at tmobile.de Action set(voicemail_id=${destination_number})
Dialplan: sofia/external/+xxx at tmobile.de Action set(voicemail_profile=default)
Dialplan: sofia/external/+xxx at tmobile.de Action lua(app.lua voicemail)
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:286 (sofia/external/+xxx at tmobile.de) State Change CS_ROUTING -> CS_EXECUTE
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:643 (sofia/external/+xxx at tmobile.de) State ROUTING going to sleep
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:584 (sofia/external/+xxx at tmobile.de) Running State Change CS_EXECUTE (Cur 1 Tot 205)
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:650 (sofia/external/+xxx at tmobile.de) State EXECUTE
2017-12-13 10:05:04.848838 [DEBUG] mod_sofia.c:198 sofia/external/+xxx at tmobile.de SOFIA EXECUTE
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:328 sofia/external/+xxx at tmobile.de Standard EXECUTE
EXECUTE sofia/external/+xxx at tmobile.de export(origination_callee_id_name=50)
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [origination_callee_id_name]=[50]
EXECUTE sofia/external/+xxx at tmobile.de set(RFC2822_DATE=Wed, 13 Dec 2017 10:05:04 +0100)
2017-12-13 10:05:04.848838 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [RFC2822_DATE]=[Wed, 13 Dec 2017 10:05:04 +0100]
EXECUTE sofia/external/+xxx at tmobile.de export(nolocal:api_on_answer=uuid_record d220a9b4-8c44-49ce-a226-18625641c2ed start /var/lib/freeswitch/recordings/sip.xxx/archive/2017/Dec/13/d220a9b4-8c44-49ce-a226-18625641c2ed.mp3)
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) (REMOTE ONLY) [api_on_answer]=[uuid_record d220a9b4-8c44-49ce-a226-18625641c2ed start /var/lib/freeswitch/recordings/sip.xxx/archive/2017/Dec/13/d220a9b4-8c44-49ce-a226-18625641c2ed.mp3]
EXECUTE sofia/external/+xxx at tmobile.de hash(insert/sip.xxx-last_dial/+xxx/50)
EXECUTE sofia/external/+xxx at tmobile.de eval(not_secure)
EXECUTE sofia/external/+xxx at tmobile.de limit(hash sip.xxx 50  )
2017-12-13 10:05:04.848838 [DEBUG] switch_limit.c:126 incr called: sip.xxx_50 max:-1, interval:0
2017-12-13 10:05:04.848838 [DEBUG] mod_hash.c:194 Usage for sip.xxx_50 is now 1
EXECUTE sofia/external/+xxx at tmobile.de set(hangup_after_bridge=true)
2017-12-13 10:05:04.848838 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [hangup_after_bridge]=[true]
EXECUTE sofia/external/+xxx at tmobile.de set(continue_on_fail=true)
2017-12-13 10:05:04.848838 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [continue_on_fail]=[true]
EXECUTE sofia/external/+xxx at tmobile.de hash(insert/sip.xxx-call_return/50/+xxx)
EXECUTE sofia/external/+xxx at tmobile.de hash(insert/sip.xxx-last_dial_ext/50/d220a9b4-8c44-49ce-a226-18625641c2ed)
EXECUTE sofia/external/+xxx at tmobile.de set(called_party_call_group=)
2017-12-13 10:05:04.848838 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [called_party_call_group]=[UNDEF]
EXECUTE sofia/external/+xxx at tmobile.de hash(insert/sip.xxx-last_dial//d220a9b4-8c44-49ce-a226-18625641c2ed)
EXECUTE sofia/external/+xxx at tmobile.de set(api_hangup_hook=lua app.lua hangup)
2017-12-13 10:05:04.848838 [DEBUG] mod_dptools.c:1548 SET sofia/external/+xxx at tmobile.de [api_hangup_hook]=[lua app.lua hangup]
EXECUTE sofia/external/+xxx at tmobile.de export(domain_name=sip.xxx)
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [domain_name]=[sip.xxx]
EXECUTE sofia/external/+xxx at tmobile.de bridge(user/50 at sip.xxx)
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1250 sofia/external/+xxx at tmobile.de EXPORTING[export_vars] [dialed_extension]=[50] to event
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1250 sofia/external/+xxx at tmobile.de EXPORTING[export_vars] [origination_callee_id_name]=[50] to event
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1250 sofia/external/+xxx at tmobile.de EXPORTING[export_vars] [api_on_answer]=[uuid_record d220a9b4-8c44-49ce-a226-18625641c2ed start /var/lib/freeswitch/recordings/sip.xxx/archive/2017/Dec/13/d220a9b4-8c44-49ce-a226-18625641c2ed.mp3] to event
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1250 sofia/external/+xxx at tmobile.de EXPORTING[export_vars] [domain_name]=[sip.xxx] to event
2017-12-13 10:05:04.848838 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1250 sofia/external/+xxx at tmobile.de EXPORTING[export_vars] [dialed_extension]=[50] to event
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1250 sofia/external/+xxx at tmobile.de EXPORTING[export_vars] [origination_callee_id_name]=[50] to event
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1250 sofia/external/+xxx at tmobile.de EXPORTING[export_vars] [api_on_answer]=[uuid_record d220a9b4-8c44-49ce-a226-18625641c2ed start /var/lib/freeswitch/recordings/sip.xxx/archive/2017/Dec/13/d220a9b4-8c44-49ce-a226-18625641c2ed.mp3] to event
2017-12-13 10:05:04.848838 [DEBUG] switch_channel.c:1250 sofia/external/+xxx at tmobile.de EXPORTING[export_vars] [domain_name]=[sip.xxx] to event
2017-12-13 10:05:04.848838 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables
2017-12-13 10:05:04.848838 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/50 at 10.1.0.5:5060 [b30aff2d-f7a1-492f-af42-d10814d2a663]
2017-12-13 10:05:04.848838 [DEBUG] mod_sofia.c:4819 (sofia/internal/50 at 10.1.0.5:5060) State Change CS_NEW -> CS_INIT
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/50 at 10.1.0.5:5060) Running State Change CS_INIT (Cur 2 Tot 206)
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/50 at 10.1.0.5:5060) State INIT
2017-12-13 10:05:04.848838 [DEBUG] mod_sofia.c:90 sofia/internal/50 at 10.1.0.5:5060 SOFIA INIT
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2017-12-13 10:05:04.848838 [DEBUG] sofia_glue.c:1295 sofia/internal/50 at 10.1.0.5:5060 sending invite version: 1.6.19 -36-7a77e0b 64bit
Local SDP:
v=0
o=FreeSWITCH 1513134686 1513134687 IN IP4 10.1.0.20
s=FreeSWITCH
c=IN IP4 10.1.0.20
t=0 0
m=audio 21218 RTP/AVP 9 8 0 3 101 13
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv

nua.c:633 nua_invite() nua: nua_invite: entering
nua_stack.c:529 nua_signal() nua(0x7fe6f0083660): sent signal r_invite
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:40 sofia/internal/50 at 10.1.0.5:5060 Standard INIT
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/50 at 10.1.0.5:5060) State Change CS_INIT -> CS_ROUTING
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:627 (sofia/internal/50 at 10.1.0.5:5060) State INIT going to sleep
nua_stack.c:569 nua_stack_signal() nua(0x7fe6f0083660): recv signal r_invite
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fe6c8001930, 0x7fe6c8001130, 0x7fe6f0083660) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c800dcc0, ...) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c800dcc0, ...) called
soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fe6c800dcc0, (nil), 0x7fe6f0082fbf, -1) called
soa.c:890 soa_set_capability_sdp() soa_set_capability_sdp(static::0x7fe6c800dcc0, (nil), 0x7fe6f0082fbf, -1) called
nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fe6f0083660): adding session usage
nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fe6c8012930)
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7fe6c800dcc0) called
soa.c:1426 soa_generate_offer() soa_generate_offer(static::0x7fe6c800dcc0, 0) called
soa_static.c:1148 offer_answer_step() soa_static_offer_answer_action(0x7fe6c800dcc0, soa_generate_offer): called
soa_static.c:1189 offer_answer_step() soa_static(0x7fe6c800dcc0, soa_generate_offer): generating local description
soa_static.c:1217 offer_answer_step() soa_static(0x7fe6c800dcc0, soa_generate_offer): upgrade with local description
soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fe70403aa70, (nil), ""): called
soa_static.c:1446 offer_answer_step() soa_static(0x7fe6c800dcc0, soa_generate_offer): storing local description
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe6c800dcc0, [(nil)], [0x7fe70403cbf8], [0x7fe70403cbf4]) called
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
tport.c:3257 tport_tsend() tport_tsend(0x7fe6c8004420) tpn = */10.1.0.5:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 10.1.0.5:5060
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fe6c8004420): not found by name */10.1.0.5:5060
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/50 at 10.1.0.5:5060) Running State Change CS_ROUTING (Cur 2 Tot 206)
tport.c:3594 tport_vsend() tport_vsend(0x7fe6c8004420): 1235 bytes of 1235 to udp/10.1.0.5:5060
tport.c:3492 tport_send_msg() tport_vsend returned 1235
nta.c:8304 outgoing_send() nta: sent INVITE (116241888) to */10.1.0.5:5060
tport.c:4160 tport_pend() tport_pend(0x7fe6c8004420): pending 0x7fe6c800f190 for udp/10.1.0.20:5060 (already 0)
nta.c:1348 set_timeout() nta: timer shortened to 1000 ms
nua_session.c:4139 signal_call_state_change() nua(0x7fe6f0083660): call state changed: init -> calling, sent offer
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe6c800dcc0, [0x7fe70403cbd8], [0x7fe70403cbe0], [(nil)]) called
nua_stack.c:269 nua_stack_event() nua(0x7fe6f0083660): event i_state INVITE sent
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/50 at 10.1.0.5:5060) State ROUTING
2017-12-13 10:05:04.848838 [DEBUG] mod_sofia.c:143 sofia/internal/50 at 10.1.0.5:5060 SOFIA ROUTING
2017-12-13 10:05:04.848838 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/50 at 10.1.0.5:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/50 at 10.1.0.5:5060) State ROUTING going to sleep
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/50 at 10.1.0.5:5060) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 206)
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:04.848838 [DEBUG] sofia.c:7084 Channel sofia/internal/50 at 10.1.0.5:5060 entering state [calling][0]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/50 at 10.1.0.5:5060) State CONSUME_MEDIA
2017-12-13 10:05:04.848838 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/50 at 10.1.0.5:5060) State CONSUME_MEDIA going to sleep
nta.c:1296 agent_timer() nta: timer set next to 4458 ms
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fe6c8004420): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe6c8004420)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe6c8004420) msg 0x7fe6c8013890 from (udp/10.1.0.20:5060) has 377 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fe6c8004420): msg 0x7fe6c8013890 (377 bytes) from udp/10.1.0.5:5060/sip next=(nil)
nta.c:3299 agent_recv_response() nta: received 100 Trying for INVITE (116241888)
nta.c:3366 agent_recv_response() nta: 100 Trying is going to a transaction
nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 35.308 ms
tport.c:4222 tport_release() tport_release(0x7fe6c8004420): 0x7fe6c800f190 by 0x7fe6c8015010 with 0x7fe6c8013890 (preliminary)
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fe6c8004420): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe6c8004420)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe6c8004420) msg 0x7fe6c8013890 from (udp/10.1.0.20:5060) has 416 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fe6c8004420): msg 0x7fe6c8013890 (416 bytes) from udp/10.1.0.5:5060/sip next=(nil)
nta.c:3299 agent_recv_response() nta: received 180 Ringing for INVITE (116241888)
nta.c:3366 agent_recv_response() nta: 180 Ringing is going to a transaction
tport.c:4222 tport_release() tport_release(0x7fe6c8004420): 0x7fe6c800f190 by 0x7fe6c8015010 with 0x7fe6c8013890 (preliminary)
nua_stack.c:271 nua_stack_event() nua(0x7fe6f0083660): event r_invite 180 Ringing
nua_session.c:4139 signal_call_state_change() nua(0x7fe6f0083660): call state changed: calling -> proceeding
nua_stack.c:271 nua_stack_event() nua(0x7fe6f0083660): event i_state 180 Ringing
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:04.988827 [DEBUG] sofia.c:7084 Channel sofia/internal/50 at 10.1.0.5:5060 entering state [proceeding][180]
2017-12-13 10:05:04.988827 [NOTICE] sofia.c:7192 Ring-Ready sofia/internal/50 at 10.1.0.5:5060!
2017-12-13 10:05:04.988827 [DEBUG] switch_channel.c:3346 (sofia/internal/50 at 10.1.0.5:5060) Callstate Change DOWN -> RINGING
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:05.008827 [INFO] switch_ivr_originate.c:1215 Sending early media
2017-12-13 10:05:05.008827 [DEBUG] switch_core_media.c:6878 AUDIO RTP [sofia/external/+xxx at tmobile.de] 10.1.0.20 port 29612 -> 217.0.4.164 port 25732 codec: 9 ms: 20
2017-12-13 10:05:05.008827 [DEBUG] switch_rtp.c:4111 Starting timer [soft] 160 bytes per 20ms
2017-12-13 10:05:05.028830 [DEBUG] switch_core_media.c:7179 sofia/external/+xxx at tmobile.de Set 2833 dtmf send payload to 100
2017-12-13 10:05:05.028830 [DEBUG] switch_core_media.c:7186 sofia/external/+xxx at tmobile.de Set 2833 dtmf receive payload to 100
2017-12-13 10:05:05.028830 [DEBUG] switch_core_media.c:7209 sofia/external/+xxx at tmobile.de Set rtp dtmf delay to 40
2017-12-13 10:05:05.028830 [DEBUG] mod_sofia.c:2364 Ring SDP:
v=0
o=FreeSWITCH 1513126293 1513126294 IN IP4 87.138.220.32
s=FreeSWITCH
c=IN IP4 87.138.220.32
t=0 0
m=audio 29612 RTP/AVP 9 100
a=rtpmap:9 G722/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=sendrecv

2017-12-13 10:05:05.028830 [NOTICE] mod_sofia.c:2367 Pre-Answer sofia/external/+xxx at tmobile.de!
2017-12-13 10:05:05.028830 [DEBUG] switch_channel.c:3474 (sofia/external/+xxx at tmobile.de) Callstate Change RINGING -> EARLY
nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:529 nua_signal() nua(0x7fe6d8059c20): sent signal r_respond
2017-12-13 10:05:05.028830 [DEBUG] switch_ivr_originate.c:1273 Raw Codec Activation Success L16 at 16000hz 1 channel 20ms
nua_stack.c:573 nua_stack_signal() nua(0x7fe6d8059c20): recv signal r_respond 183 Session Progress
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
2017-12-13 10:05:05.028830 [DEBUG] switch_core_codec.c:223 sofia/external/+xxx at tmobile.de Push codec L16:100
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6d805e1a0, ...) called
soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fe6d805e1a0, (nil), 0x7fe6f010e30c, -1) called
soa.c:890 soa_set_capability_sdp() soa_set_capability_sdp(static::0x7fe6d805e1a0, (nil), 0x7fe6f010e30c, -1) called
nua_session.c:2320 nua_invite_server_respond() nua: nua_invite_server_respond: entering
soa.c:1515 soa_generate_answer() soa_generate_answer(static::0x7fe6d805e1a0) called
2017-12-13 10:05:05.028830 [DEBUG] switch_ivr_originate.c:1342 Play Ringback Tone [%(1000,4000,425)]
soa_static.c:1148 offer_answer_step() soa_static_offer_answer_action(0x7fe6d805e1a0, soa_generate_answer): called
soa_static.c:1189 offer_answer_step() soa_static(0x7fe6d805e1a0, soa_generate_answer): generating local description
soa_static.c:1230 offer_answer_step() soa_static(0x7fe6d805e1a0, soa_generate_answer): upgrade with remote description
soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fe7040b6ab0, 0x7fe6d8059f10, ""): called
soa_static.c:1446 offer_answer_step() soa_static(0x7fe6d805e1a0, soa_generate_answer): storing local description
soa.c:1730 soa_activate() soa_activate(static::0x7fe6d805e1a0, (nil)) called
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe6d805e1a0, [(nil)], [0x7fe7040b8c38], [0x7fe7040b8c34]) called
tport.c:3257 tport_tsend() tport_tsend(0x7fe6d8004660) tpn = UDP/217.0.23.100:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 217.0.23.100:5060
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fe6d8004660): not found by name UDP/217.0.23.100:5060
tport.c:3594 tport_vsend() tport_vsend(0x7fe6d8004660): 1080 bytes of 1080 to udp/217.0.23.100:5060
tport.c:3492 tport_send_msg() tport_vsend returned 1080
nta.c:6791 incoming_reply() nta: sent 183 Session Progress for INVITE (1)
nua_session.c:4139 signal_call_state_change() nua(0x7fe6d8059c20): call state changed: received -> early, sent answer
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe6d805e1a0, [0x7fe7040b8ce8], [0x7fe7040b8cf0], [(nil)]) called
soa.c:616 soa_get_params() soa_get_params(static::0x7fe6d805e1a0, ...) called
nua_stack.c:271 nua_stack_event() nua(0x7fe6d8059c20): event i_state 183 Session Progress
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:05.048828 [DEBUG] sofia.c:7084 Channel sofia/external/+xxx at tmobile.de entering state [early][183]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:05.448891 [DEBUG] switch_rtp.c:7271 Correct audio ip/port confirmed.
nta.c:1296 agent_timer() nta: timer set next to 14237 ms
nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate BYE (116241861)
nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7fe7040b8d40)
nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta.c:1296 agent_timer() nta: timer set next to 55697 ms
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fe6c8004420): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe6c8004420)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe6c8004420) msg 0x7fe6c80178a0 from (udp/10.1.0.20:5060) has 687 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fe6c8004420): msg 0x7fe6c80178a0 (687 bytes) from udp/10.1.0.5:5060/sip next=(nil)
nta.c:3299 agent_recv_response() nta: received 200 OK for INVITE (116241888)
nta.c:3366 agent_recv_response() nta: 200 OK is going to a transaction
tport.c:4222 tport_release() tport_release(0x7fe6c8004420): 0x7fe6c800f190 by 0x7fe6c8015010 with 0x7fe6c80178a0
soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fe6c800dcc0, (nil), 0x7fe6c8018085, 186) called
soa.c:1595 soa_process_answer() soa_process_answer(static::0x7fe6c800dcc0) called
soa_static.c:1148 offer_answer_step() soa_static_offer_answer_action(0x7fe6c800dcc0, soa_process_answer): called
soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fe6c80144e0, 0x7fe6c8013890, ""): called
soa_static.c:1304 offer_answer_step() soa_static(0x7fe6c800dcc0, soa_process_answer): upgrade codecs with remote description
soa_static.c:1446 offer_answer_step() soa_static(0x7fe6c800dcc0, soa_process_answer): storing local description
soa.c:1730 soa_activate() soa_activate(static::0x7fe6c800dcc0, (nil)) called
nua_session.c:988 nua_session_client_response() nua(0x7fe6f0083660): INVITE: processed SDP answer in 200 OK
nua_stack.c:271 nua_stack_event() nua(0x7fe6f0083660): event r_invite 200 OK
nua_session.c:4139 signal_call_state_change() nua(0x7fe6f0083660): call state changed: proceeding -> completing, received answer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fe6c800dcc0, [0x7fe70403c5e8], [0x7fe70403c5f0], [(nil)]) called
soa.c:616 soa_get_params() soa_get_params(static::0x7fe6c800dcc0, ...) called
nua_stack.c:271 nua_stack_event() nua(0x7fe6f0083660): event i_state 200 OK
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:15.288844 [DEBUG] sofia.c:7084 Channel sofia/internal/50 at 10.1.0.5:5060 entering state [completing][200]
2017-12-13 10:05:15.288844 [DEBUG] sofia.c:7094 Remote SDP:
v=0
o=50 5016 49 IN IP4 10.1.0.5
s=Mapping
c=IN IP4 10.1.0.5
t=0 0
m=audio 5016 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

nua.c:639 nua_ack() nua: nua_ack: entering
nua_stack.c:529 nua_signal() nua(0x7fe6f0083660): sent signal r_ack
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua_stack.c:569 nua_stack_signal() nua(0x7fe6f0083660): recv signal r_ack
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c800dcc0, ...) called
soa.c:1730 soa_activate() soa_activate(static::0x7fe6c800dcc0, (nil)) called
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
tport.c:3257 tport_tsend() tport_tsend(0x7fe6c8004420) tpn = */10.1.0.5:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 10.1.0.5:5060
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fe6c8004420): not found by name */10.1.0.5:5060
tport.c:3594 tport_vsend() tport_vsend(0x7fe6c8004420): 371 bytes of 371 to udp/10.1.0.5:5060
tport.c:3492 tport_send_msg() tport_vsend returned 371
nta.c:8304 outgoing_send() nta: sent ACK (116241888) to */10.1.0.5:5060
nua_session.c:4139 signal_call_state_change() nua(0x7fe6f0083660): call state changed: completing -> ready
nua_stack.c:271 nua_stack_event() nua(0x7fe6f0083660): event i_state 200 ACK sent
nua_stack.c:271 nua_stack_event() nua(0x7fe6f0083660): event i_active 200 Call active
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:15.288844 [DEBUG] sofia.c:7084 Channel sofia/internal/50 at 10.1.0.5:5060 entering state [ready][200]
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [G722:9:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 101 at 8000
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:3061 Set Codec sofia/internal/50 at 10.1.0.5:5060 G722/8000 20 ms 160 samples 64000 bits 1 channels
2017-12-13 10:05:15.288844 [DEBUG] switch_core_codec.c:111 sofia/internal/50 at 10.1.0.5:5060 Original read codec set to G722:9
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 101 at 8000
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:4767 sofia/internal/50 at 10.1.0.5:5060 Set 2833 dtmf send payload to 101 recv payload to 101
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:6878 AUDIO RTP [sofia/internal/50 at 10.1.0.5:5060] 10.1.0.20 port 21218 -> 10.1.0.5 port 5016 codec: 9 ms: 20
2017-12-13 10:05:15.288844 [DEBUG] switch_rtp.c:4111 Starting timer [soft] 160 bytes per 20ms
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:7179 sofia/internal/50 at 10.1.0.5:5060 Set 2833 dtmf send payload to 101
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:7186 sofia/internal/50 at 10.1.0.5:5060 Set 2833 dtmf receive payload to 101
2017-12-13 10:05:15.288844 [DEBUG] switch_core_media.c:7209 sofia/internal/50 at 10.1.0.5:5060 Set rtp dtmf delay to 40
2017-12-13 10:05:15.288844 [NOTICE] sofia.c:8218 Channel [sofia/internal/50 at 10.1.0.5:5060] has been answered
2017-12-13 10:05:15.288844 [DEBUG] switch_channel.c:3571 sofia/internal/50 at 10.1.0.5:5060 process uuid_record d220a9b4-8c44-49ce-a226-18625641c2ed start /var/lib/freeswitch/recordings/sip.xxx/archive/2017/Dec/13/d220a9b4-8c44-49ce-a226-18625641c2ed.mp3: uuid_record(d220a9b4-8c44-49ce-a226-18625641c2ed start /var/lib/freeswitch/recordings/sip.xxx/archive/2017/Dec/13/d220a9b4-8c44-49ce-a226-18625641c2ed.mp3)

2017-12-13 10:05:15.308840 [DEBUG] switch_core_codec.c:248 sofia/external/+xxx at tmobile.de Restore previous codec G722:9.
2017-12-13 10:05:15.308840 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/external/+xxx at tmobile.de.
2017-12-13 10:05:15.308840 [DEBUG] mod_sofia.c:850 Local SDP sofia/external/+xxx at tmobile.de:
v=0
o=FreeSWITCH 1513126293 1513126295 IN IP4 87.138.220.32
s=FreeSWITCH
c=IN IP4 87.138.220.32
t=0 0
m=audio 29612 RTP/AVP 9 100
a=rtpmap:9 G722/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=sendrecv

nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:573 nua_stack_signal() nua(0x7fe6d8059c20): recv signal r_respond 200 OK
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
nua_stack.c:529 nua_signal() nua(0x7fe6d8059c20): sent signal r_respond
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6d805e1a0, ...) called
soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fe6d805e1a0, (nil), 0x7fe6f00854ea, -1) called
nua_session.c:2320 nua_invite_server_respond() nua: nua_invite_server_respond: entering
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe6d805e1a0, [(nil)], [0x7fe7040b8c38], [0x7fe7040b8c34]) called
tport.c:3257 tport_tsend() tport_tsend(0x7fe6d8004660) tpn = UDP/217.0.23.100:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 217.0.23.100:5060
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fe6d8004660): not found by name UDP/217.0.23.100:5060
tport.c:3594 tport_vsend() tport_vsend(0x7fe6d8004660): 1074 bytes of 1074 to udp/217.0.23.100:5060
tport.c:3492 tport_send_msg() tport_vsend returned 1074
nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (1)
nta.c:1348 set_timeout() nta: timer shortened to 500 ms
nua_session.c:4139 signal_call_state_change() nua(0x7fe6d8059c20): call state changed: early -> completed, sent answer
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe6d805e1a0, [0x7fe7040b8ce8], [0x7fe7040b8cf0], [(nil)]) called
soa.c:616 soa_get_params() soa_get_params(static::0x7fe6d805e1a0, ...) called
nua_stack.c:271 nua_stack_event() nua(0x7fe6d8059c20): event i_state 200 OK
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2017-12-13 10:05:15.308840 [NOTICE] switch_ivr_originate.c:3647 Channel [sofia/external/+xxx at tmobile.de] has been answered
2017-12-13 10:05:15.308840 [DEBUG] switch_core_media_bug.c:945 Attaching BUG to sofia/external/+xxx at tmobile.de
2017-12-13 10:05:15.308840 [DEBUG] switch_channel.c:3773 (sofia/internal/50 at 10.1.0.5:5060) Callstate Change RINGING -> ACTIVE
2017-12-13 10:05:15.308840 [DEBUG] switch_channel.c:3773 (sofia/external/+xxx at tmobile.de) Callstate Change EARLY -> ACTIVE
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:15.308840 [DEBUG] sofia.c:7084 Channel sofia/external/+xxx at tmobile.de entering state [completed][200]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:15.308840 [DEBUG] switch_ivr_originate.c:3705 Originate Resulted in Success: [sofia/internal/50 at 10.1.0.5:5060]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:15.308840 [DEBUG] switch_ivr_originate.c:3705 Originate Resulted in Success: [sofia/internal/50 at 10.1.0.5:5060]
2017-12-13 10:05:15.308840 [DEBUG] switch_ivr_bridge.c:1614 (sofia/internal/50 at 10.1.0.5:5060) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2017-12-13 10:05:15.308840 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/50 at 10.1.0.5:5060) Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 206)
2017-12-13 10:05:15.308840 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/50 at 10.1.0.5:5060) State EXCHANGE_MEDIA
2017-12-13 10:05:15.308840 [DEBUG] mod_sofia.c:631 SOFIA EXCHANGE_MEDIA
2017-12-13 10:05:15.328842 [DEBUG] switch_ivr_async.c:1500 No silence detection configured; assuming start of speech
2017-12-13 10:05:15.348847 [DEBUG] switch_rtp.c:7271 Correct audio ip/port confirmed.
2017-12-13 10:05:15.348847 [DEBUG] switch_core_io.c:448 Setting BUG Codec G722:9
2017-12-13 10:05:15.408848 [DEBUG] switch_rtp.c:7271 Correct audio ip/port confirmed.
nta.c:6996 _nta_incoming_timer() nta: timer G fired, retransmitting 200 reply
tport.c:3257 tport_tsend() tport_tsend(0x7fe6d8004660) tpn = UDP/217.0.23.100:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 217.0.23.100:5060
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fe6d8004660): not found by name UDP/217.0.23.100:5060
tport.c:3594 tport_vsend() tport_vsend(0x7fe6d8004660): 1074 bytes of 1074 to udp/217.0.23.100:5060
tport.c:3492 tport_send_msg() tport_vsend returned 1074
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
nta.c:1296 agent_timer() nta: timer set next to 1000 ms
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fe6d8004660): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe6d8004660)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe6d8004660) msg 0x7fe6d8061980 from (udp/10.1.0.20:5080) has 550 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fe6d8004660): msg 0x7fe6d8061980 (550 bytes) from udp/217.0.23.100:5080/sip next=(nil)
nta.c:2880 agent_recv_request() nta: received ACK sip:+4991314000733 at 87.138.220.32:5080;transport=udp SIP/2.0 (CSeq 1)
nta.c:3019 agent_recv_request() nta: ACK (1) is going to INVITE (1)
nua_session.c:2569 process_ack_or_cancel() nua: process_ack_or_cancel: entering
soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0x7fe6d805e1a0) called
nua_stack.c:271 nua_stack_event() nua(0x7fe6d8059c20): event i_ack 200 OK
nua_session.c:4139 signal_call_state_change() nua(0x7fe6d8059c20): call state changed: completed -> ready
nua_stack.c:271 nua_stack_event() nua(0x7fe6d8059c20): event i_state 200 OK
nua_stack.c:271 nua_stack_event() nua(0x7fe6d8059c20): event i_active 200 Call active
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_dialog.c:564 nua_dialog_usage_set_refresh_range() nua(): refresh session after 1768 seconds (in [1768..1768])
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2017-12-13 10:05:15.908861 [DEBUG] sofia.c:7084 Channel sofia/external/+xxx at tmobile.de entering state [ready][200]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nta.c:1296 agent_timer() nta: timer set next to 4099 ms
2017-12-13 10:05:17.368873 [INFO] mod_shout.c:332 LAME 3.99.5 64bits (http://lame.sf.net)
2017-12-13 10:05:17.368873 [INFO] mod_shout.c:332 polyphase lowpass filter disabled
nta.c:7159 _nta_incoming_timer() nta: timer J fired, terminate 401 response
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fe70403cc60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free
nta.c:1296 agent_timer() nta: timer set next to 37 ms
nta.c:7159 _nta_incoming_timer() nta: timer J fired, terminate 200 response
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fe70403cc60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
nta.c:1296 agent_timer() nta: timer set next to 15997 ms
nta.c:7134 _nta_incoming_timer() nta: timer I fired, terminate 200 response
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fe7040b8c60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta.c:1289 agent_timer() nta: timer not set
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fe6c8004420): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe6c8004420)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe6c8004420) msg 0x7fe6c800d4f0 from (udp/10.1.0.20:5060) has 463 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fe6c8004420): msg 0x7fe6c800d4f0 (463 bytes) from udp/10.1.0.4:5060/sip next=(nil)
nta.c:2880 agent_recv_request() nta: received REGISTER sip:sip.xxx SIP/2.0 (CSeq 10060)
nta.c:3085 agent_recv_request() nta: REGISTER (10060) going to a default leg
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
nua_stack.c:899 nh_create() nua: nh_create: entering
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fe6c8001930, 0x7fe6c8001130, 0x7fe6c800ff60) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c800ae10, ...) called
nua_stack.c:271 nua_stack_event() nua(0x7fe6c800ff60): event i_register 100 Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:529 nua_signal() nua(0x7fe6c800ff60): sent signal r_respond
2017-12-13 10:05:22.088943 [WARNING] sofia_reg.c:1792 SIP auth challenge (REGISTER) on sofia profile 'internal' for [20 at sip.xxx] from ip 10.1.0.4
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:573 nua_stack_signal() nua(0x7fe6c800ff60): recv signal r_respond 401 Unauthorized
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
nua_stack.c:529 nua_signal() nua(0x7fe6c800ff60): sent signal r_destroy
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c800ae10, ...) called
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
tport.c:3257 tport_tsend() tport_tsend(0x7fe6c8004420) tpn = UDP/10.1.0.4:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 10.1.0.4:5060
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fe6c8004420): not found by name UDP/10.1.0.4:5060
tport.c:3594 tport_vsend() tport_vsend(0x7fe6c8004420): 586 bytes of 586 to udp/10.1.0.4:5060
tport.c:3492 tport_send_msg() tport_vsend returned 586
nta.c:6791 incoming_reply() nta: sent 401 Unauthorized for REGISTER (10060)
nua_stack.c:569 nua_stack_signal() nua(0x7fe6c800ff60): recv signal r_destroy
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
soa.c:356 soa_destroy() soa_destroy(static::0x7fe6c800ae10) called
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fe6c8004420): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe6c8004420)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe6c8004420) msg 0x7fe6c8006140 from (udp/10.1.0.20:5060) has 736 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fe6c8004420): msg 0x7fe6c8006140 (736 bytes) from udp/10.1.0.4:5060/sip next=(nil)
nta.c:2880 agent_recv_request() nta: received REGISTER sip:sip.xxx SIP/2.0 (CSeq 10061)
nta.c:3085 agent_recv_request() nta: REGISTER (10061) going to a default leg
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
nua_stack.c:899 nh_create() nua: nh_create: entering
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fe6c8001930, 0x7fe6c8001130, 0x7fe6c8010200) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c8012550, ...) called
nua_stack.c:271 nua_stack_event() nua(0x7fe6c8010200): event i_register 100 Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:529 nua_signal() nua(0x7fe6c8010200): sent signal r_respond
nua_stack.c:573 nua_stack_signal() nua(0x7fe6c8010200): recv signal r_respond 200 OK
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c8012550, ...) called
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:529 nua_signal() nua(0x7fe6c8010200): sent signal r_destroy
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
tport.c:3257 tport_tsend() tport_tsend(0x7fe6c8004420) tpn = UDP/10.1.0.4:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 10.1.0.4:5060
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fe6c8004420): not found by name UDP/10.1.0.4:5060
tport.c:3594 tport_vsend() tport_vsend(0x7fe6c8004420): 532 bytes of 532 to udp/10.1.0.4:5060
tport.c:3492 tport_send_msg() tport_vsend returned 532
nta.c:6791 incoming_reply() nta: sent 200 OK for REGISTER (10061)
nua_stack.c:569 nua_stack_signal() nua(0x7fe6c8010200): recv signal r_destroy
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
soa.c:356 soa_destroy() soa_destroy(static::0x7fe6c8012550) called
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fe6c8004420): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe6c8004420)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe6c8004420) msg 0x7fe6c80182f0 from (udp/10.1.0.20:5060) has 466 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fe6c8004420): msg 0x7fe6c80182f0 (466 bytes) from udp/10.1.0.4:5060/sip next=(nil)
nta.c:2880 agent_recv_request() nta: received REGISTER sip:sip.xxx SIP/2.0 (CSeq 10060)
nta.c:3085 agent_recv_request() nta: REGISTER (10060) going to a default leg
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
nua_stack.c:899 nh_create() nua: nh_create: entering
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fe6c8001930, 0x7fe6c8001130, 0x7fe6c801a190) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c801a770, ...) called
nua_stack.c:271 nua_stack_event() nua(0x7fe6c801a190): event i_register 100 Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:529 nua_signal() nua(0x7fe6c801a190): sent signal r_respond
nua_stack.c:573 nua_stack_signal() nua(0x7fe6c801a190): recv signal r_respond 401 Unauthorized
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c801a770, ...) called
2017-12-13 10:05:22.168877 [WARNING] sofia_reg.c:1792 SIP auth challenge (REGISTER) on sofia profile 'internal' for [30 at sip.xxx] from ip 10.1.0.4
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:529 nua_signal() nua(0x7fe6c801a190): sent signal r_destroy
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
tport.c:3257 tport_tsend() tport_tsend(0x7fe6c8004420) tpn = UDP/10.1.0.4:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 10.1.0.4:5060
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fe6c8004420): not found by name UDP/10.1.0.4:5060
tport.c:3594 tport_vsend() tport_vsend(0x7fe6c8004420): 589 bytes of 589 to udp/10.1.0.4:5060
tport.c:3492 tport_send_msg() tport_vsend returned 589
nta.c:6791 incoming_reply() nta: sent 401 Unauthorized for REGISTER (10060)
nua_stack.c:569 nua_stack_signal() nua(0x7fe6c801a190): recv signal r_destroy
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
soa.c:356 soa_destroy() soa_destroy(static::0x7fe6c801a770) called
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fe6c8004420): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe6c8004420)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe6c8004420) msg 0x7fe6c8019f60 from (udp/10.1.0.20:5060) has 737 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fe6c8004420): msg 0x7fe6c8019f60 (737 bytes) from udp/10.1.0.4:5060/sip next=(nil)
nta.c:2880 agent_recv_request() nta: received REGISTER sip:sip.xxx SIP/2.0 (CSeq 10061)
nta.c:3085 agent_recv_request() nta: REGISTER (10061) going to a default leg
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
nua_stack.c:899 nh_create() nua: nh_create: entering
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fe6c8001930, 0x7fe6c8001130, 0x7fe6c801cdf0) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c801d520, ...) called
nua_stack.c:271 nua_stack_event() nua(0x7fe6c801cdf0): event i_register 100 Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:573 nua_stack_signal() nua(0x7fe6c801cdf0): recv signal r_respond 200 OK
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c801d520, ...) called
nua_stack.c:529 nua_signal() nua(0x7fe6c801cdf0): sent signal r_respond
tport.c:3257 tport_tsend() tport_tsend(0x7fe6c8004420) tpn = UDP/10.1.0.4:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 10.1.0.4:5060
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fe6c8004420): not found by name UDP/10.1.0.4:5060
nua_stack.c:529 nua_signal() nua(0x7fe6c801cdf0): sent signal r_destroy
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
tport.c:3594 tport_vsend() tport_vsend(0x7fe6c8004420): 533 bytes of 533 to udp/10.1.0.4:5060
tport.c:3492 tport_send_msg() tport_vsend returned 533
nta.c:6791 incoming_reply() nta: sent 200 OK for REGISTER (10061)
nua_stack.c:569 nua_stack_signal() nua(0x7fe6c801cdf0): recv signal r_destroy
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
soa.c:356 soa_destroy() soa_destroy(static::0x7fe6c801d520) called
nta.c:7159 _nta_incoming_timer() nta: timer J fired, terminate 200 response
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fe70403cc60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/5 term, 1/5 free
nta.c:1296 agent_timer() nta: timer set next to 11157 ms
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fe6c8004420): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe6c8004420)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe6c8004420) msg 0x7fe6c8007f80 from (udp/10.1.0.20:5060) has 467 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fe6c8004420): msg 0x7fe6c8007f80 (467 bytes) from udp/10.1.0.5:5060/sip next=(nil)
nta.c:2880 agent_recv_request() nta: received REGISTER sip:sip.xxx SIP/2.0 (CSeq 11232)
nta.c:3085 agent_recv_request() nta: REGISTER (11232) going to a default leg
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
nua_stack.c:899 nh_create() nua: nh_create: entering
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fe6c8001930, 0x7fe6c8001130, 0x7fe6c801d010) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c8008410, ...) called
nua_stack.c:271 nua_stack_event() nua(0x7fe6c801d010): event i_register 100 Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:529 nua_signal() nua(0x7fe6c801d010): sent signal r_respond
2017-12-13 10:05:46.788911 [WARNING] sofia_reg.c:1792 SIP auth challenge (REGISTER) on sofia profile 'internal' for [50 at sip.xxx] from ip 10.1.0.5
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:573 nua_stack_signal() nua(0x7fe6c801d010): recv signal r_respond 401 Unauthorized
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c8008410, ...) called
nua_stack.c:529 nua_signal() nua(0x7fe6c801d010): sent signal r_destroy
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
tport.c:3257 tport_tsend() tport_tsend(0x7fe6c8004420) tpn = UDP/10.1.0.5:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 10.1.0.5:5060
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fe6c8004420): not found by name UDP/10.1.0.5:5060
tport.c:3594 tport_vsend() tport_vsend(0x7fe6c8004420): 590 bytes of 590 to udp/10.1.0.5:5060
tport.c:3492 tport_send_msg() tport_vsend returned 590
nta.c:6791 incoming_reply() nta: sent 401 Unauthorized for REGISTER (11232)
nua_stack.c:569 nua_stack_signal() nua(0x7fe6c801d010): recv signal r_destroy
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
soa.c:356 soa_destroy() soa_destroy(static::0x7fe6c8008410) called
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7fe6c8004420): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe6c8004420)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe6c8004420) msg 0x7fe6c8008410 from (udp/10.1.0.20:5060) has 737 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fe6c8004420): msg 0x7fe6c8008410 (737 bytes) from udp/10.1.0.5:5060/sip next=(nil)
nta.c:2880 agent_recv_request() nta: received REGISTER sip:sip.xxx SIP/2.0 (CSeq 11233)
nta.c:3085 agent_recv_request() nta: REGISTER (11233) going to a default leg
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
nua_stack.c:899 nh_create() nua: nh_create: entering
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fe6c8001930, 0x7fe6c8001130, 0x7fe6c80201c0) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c80209f0, ...) called
nua_stack.c:271 nua_stack_event() nua(0x7fe6c80201c0): event i_register 100 Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:573 nua_stack_signal() nua(0x7fe6c80201c0): recv signal r_respond 200 OK
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fe6c80209f0, ...) called
nua_stack.c:529 nua_signal() nua(0x7fe6c80201c0): sent signal r_respond
tport.c:3257 tport_tsend() tport_tsend(0x7fe6c8004420) tpn = UDP/10.1.0.5:5060
tport.c:4046 tport_resolve() tport_resolve addrinfo = 10.1.0.5:5060
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7fe6c8004420): not found by name UDP/10.1.0.5:5060
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:529 nua_signal() nua(0x7fe6c80201c0): sent signal r_destroy
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
tport.c:3594 tport_vsend() tport_vsend(0x7fe6c8004420): 534 bytes of 534 to udp/10.1.0.5:5060
tport.c:3492 tport_send_msg() tport_vsend returned 534
nta.c:6791 incoming_reply() nta: sent 200 OK for REGISTER (11233)
nua_stack.c:569 nua_stack_signal() nua(0x7fe6c80201c0): recv signal r_destroy
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
soa.c:356 soa_destroy() soa_destroy(static::0x7fe6c80209f0) called
nta.c:9101 outgoing_timer_dk() nta: timer D fired, terminate INVITE (116241888)
nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7fe70403cd40)
nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/1 term, 1/2 free
nta.c:1296 agent_timer() nta: timer set next to 3 ms
nta.c:8982 outgoing_timer_bf() nta: timer F fired, terminating ACK (116241888)
nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7fe70403cd40)
nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 1/1 free
nta.c:1296 agent_timer() nta: timer set next to 6827 ms
nta.c:7159 _nta_incoming_timer() nta: timer J fired, terminate 401 response
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fe70403cc60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/6 term, 1/6 free
nta.c:1296 agent_timer() nta: timer set next to 35 ms
nta.c:7159 _nta_incoming_timer() nta: timer J fired, terminate 200 response
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fe70403cc60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/5 term, 1/5 free
nta.c:1296 agent_timer() nta: timer set next to 28 ms
nta.c:7159 _nta_incoming_timer() nta: timer J fired, terminate 401 response
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fe70403cc60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/4 term, 1/4 free
nta.c:1296 agent_timer() nta: timer set next to 37 ms
nta.c:7159 _nta_incoming_timer() nta: timer J fired, terminate 200 response
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fe70403cc60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/3 term, 1/3 free
nta.c:1296 agent_timer() nta: timer set next to 24575 ms
nta.c:7159 _nta_incoming_timer() nta: timer J fired, terminate 401 response
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fe70403cc60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
nta.c:1296 agent_timer() nta: timer set next to 73 ms
nta.c:7159 _nta_incoming_timer() nta: timer J fired, terminate 200 response
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fe70403cc60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta.c:1289 agent_timer() nta: timer not set


Regards

Philipp

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From enrico.vergata at sewan.fr  Wed Dec 13 15:02:25 2017
From: enrico.vergata at sewan.fr (Enrico Vergata)
Date: Wed, 13 Dec 2017 15:02:25 +0000
Subject: [Freeswitch-users] DTMF error
Message-ID: <DB6PR0602MB27906F6195B3D2D4D8E191B9EE350@DB6PR0602MB2790.eurprd06.prod.outlook.com>

Hello FS-users,

I'am trying to execute a python script, launched from the dialplan.

Everything works fine but at a given time I need to execute a TTS request to the gTTS module within python ( google translator API) to gather a .wav file to be played.
That's right after a DTMF is needed to select the good "if" condition.

FS performs good the task when it interact with Jisti but FS stop detect RTP DTMF Event when does not matter which other telephone/softphone is used.

Disabling the TTS request, FS detects well the RTP DTMF Event for both extension: Jitsi and the others.


I don't really know what might be the cause?

Anyway the script keeps carrying on and the media files are all well played.

I'm using FreeSWITCH Version 1.9.0 in Centos 7, I've replicated the script in Lua that has shown the same behavior with the exception that even with Jisti   FS stop detecting DTMF events.


Thanks you in advance for any help and idea.

Best regards,
Enrico





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From bogdan at opensips.org  Wed Dec 13 17:05:09 2017
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 13 Dec 2017 19:05:09 +0200
Subject: [Freeswitch-users] OpenSIPS Summit 2018 - Call For Papers
Message-ID: <ccb5acd9-00e0-190f-e7c3-5c365d1b966d@opensips.org>

<http://www.opensips.org/events/Summit-2018Amsterdam>


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-- 
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From mjlopez at smartic.es  Wed Dec 13 17:51:26 2017
From: mjlopez at smartic.es (=?iso-8859-1?Q?Miguel_Jes=FAs_L=F3pez_Valverde?=)
Date: Wed, 13 Dec 2017 18:51:26 +0100
Subject: [Freeswitch-users] Problem with personalized locutions and use of
	say with type shor_date_time.
Message-ID: <031601d3743b$007f94d0$017ebe70$@smartic.es>

Hello Freeswitch users:

 

I send this email to you with a new problem found.

 

I have a set of complete locutions created with a TTS for Spanish locutions
installed in Freeswitch, along with those brought by default in English.

 

When using the phrases in Spanish, I am only having problems when using the
directive:

<action function = "say" data = "$ 1" method = "pronounced" type =
"short_date_time" />

but with other directives "say" with different methods, I'm not having any
problems. For example, when accessing voicemail using these locutions set in
Spanish, I get the following error that prevents access to recorded
messages:

 

2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:250 Handle
play-file:[voicemail/vm-new.wav] (es_CB:es)

2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:1498 Codec
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:53.172924 [DEBUG] switch_ivr_play_say.c:1942 done playing
file /usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-new.wav

2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:250 Handle
play-file:[voicemail/vm-message_number.wav] (es_CB:es)

2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:1498 Codec
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:54.832899 [DEBUG] switch_ivr_play_say.c:1942 done playing
file
/usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-message_number.w
av

2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1]
(es_CB:es)

2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:1498 Codec
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:55.872903 [DEBUG] switch_ivr_play_say.c:1942 done playing
file /usr/local/freeswitch/sounds/es/colabora/sonia/digits/1.wav

2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:70 No language
specified - Using [es_CB]

2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:250 Handle
say:[1513180033] (es_CB:es)

2017-12-13 16:58:55.972959 [ERR] mod_say_es.c:471 Unknown Say type=[18]

 

By changing this language by default English, it works correctly ...

 

2017-12-13 17:18:05.327117 [DEBUG] switch_ivr_play_say.c:70 No language
specified - Using [en]

2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:250 Handle
play-file:[voicemail/vm-new.wav] (en:en)

2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:1498 Codec
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:05.667117 [DEBUG] switch_ivr_play_say.c:1942 done playing
file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav

2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:250 Handle
play-file:[voicemail/vm-message_number.wav] (en:en)

2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:1498 Codec
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:06.587140 [DEBUG] switch_ivr_play_say.c:1942 done playing
file
/usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-message_number.wav

2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1]
(en:en)

2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:1498 Codec
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:07.407128 [DEBUG] switch_ivr_play_say.c:1942 done playing
file file_string://digits/1.wav

2017-12-13 17:18:07.507133 [DEBUG] switch_ivr_play_say.c:70 No language
specified - Using [en]

2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:250 Handle
say:[1513180033] (en:en)

2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:1498 Codec
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:10.427138 [DEBUG] switch_ivr_play_say.c:1942 done playing
file
file_string://time/today.wav!time/at.wav!digits/3.wav!digits/40.wav!digits/7
.wav!time/p-m.wav

2017-12-13 17:18:10.547147 [DEBUG] switch_ivr_play_say.c:1498 Codec
Activated L16 at 8000hz 1 channels 20ms

 

It is proven that I have all the locutions in Spanish, including those in
the "time" and "digits" folders.

 

Does anyone know how I can get a proper voicemail operation with
personalized locutions when using the function "say" with "type ="
short_date_time "" ?.

 

Thank you very much and best regards.

 



---
El software de antivirus Avast ha analizado este correo electrónico en busca de virus.
https://www.avast.com/antivirus
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From SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com  Wed Dec 13 18:11:49 2017
From: SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com (Karanath sachidanandan, Sanooj)
Date: Wed, 13 Dec 2017 18:11:49 +0000
Subject: [Freeswitch-users] [EXTERNAL] Re:  FS connecting to Avaya
In-Reply-To: <CAERdjibo-K921xiPSUPRq_rXEjiesUGr4CSmvE+YvV=+YngryA@mail.gmail.com>
References: <E338CEDA-8B96-4201-A6A5-38B85D3C2529@homedepot.com>
 <CAERdjibo-K921xiPSUPRq_rXEjiesUGr4CSmvE+YvV=+YngryA@mail.gmail.com>
Message-ID: <77695D16-34C1-4D3F-8946-6363C0F56820@homedepot.com>

Thanks Tom for pointers , we are looking more to peer with CM . Will look more into the direction you suggest .
AES option was to connect to CM using JTAPI , but couldn’t make much progress in that dirction

Regards
Sanooj


From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Tom Lynn <tom at tomlynn.com>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: Sunday, December 10, 2017 at 7:32 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [EXTERNAL] Re: [Freeswitch-users] FS connecting to Avaya

Are you planning to peer directly to Communication Manager or using Avaya Session Manager in between?  It can be done both ways.  If the requirement is to transact only with Communication Manager a direct connection may work out.

In that case I'd suggest you'll need to add a node-name (which is like a host table entry) for FS and a signaling group of type SIP, along with a trunk group whose channels are controlled by said signaling group.  Assume TCP signaling to begin with and pay attention to your port numbers.  Some may already be used by other signaling groups.

Codecs are assigned based on a network-region construct which is used to model your voip network against your physical topology, supporting call admission control and where DSP channels are sourced.  Since Avaya doesn't use it's server host to mix audio, you'll have to know that the network region you choose either has DSPs or a logical path to a region which does is available.  The network region is assigned to the SIP signaling group in a field called Far End Network region.

To point calls at the trunk group you'll have to set up a route pattern and then place a dial string match in either an ARS analysis table (PSTN) or an AAR analysis table (private network).

How are you thinking about using AES?  I don't see an easy interface, but if you program TSAPI, it may be useful to get call events from the Avaya side.

Tom Lynn


On Fri, Dec 8, 2017 at 2:12 PM, Karanath sachidanandan, Sanooj <SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com<mailto:SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com>> wrote:
Hi All,
Is there any documentation on connecting FS to AVAYA CM/AES ? or any pointers ?

Regards
Sanooj


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From igorolhovskiy at gmail.com  Thu Dec 14 00:00:13 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Thu, 14 Dec 2017 02:00:13 +0200
Subject: [Freeswitch-users] FS not reacting to BYE
In-Reply-To: <4f36450f-7499-4c7f-994c-e3cce8eb6170@Spark>
References: <4f36450f-7499-4c7f-994c-e3cce8eb6170@Spark>
Message-ID: <294110ad-760b-45d8-a8ae-476c0eaf0146@Spark>

Seems it’s really strange issue.

Call was made with api_on_answer variable and in this api lua script was called.
lua script just make curl request with session:execute(«curl», «https://xxxx»)
Only in this case FS not has a problem with accepting BYE.

When I redo curl request with bgapi - all act as designed.

Regards, Igor

On Dec 13, 2017, 3:48 PM +0200, Igor Olhovskiy <igorolhovskiy at gmail.com>, wrote:
> Hi all!
>
>
> Got into issue with one of providers, and can’t get some things...
> For what reason FS is not reacting to BYE on external profile?
>
> FS -> Prov
> INVITE sip:3809XXXXXXXX at sip.telecomax.net SIP/2.0
> Via: SIP/2.0/UDP 138.68.65.164:5080;rport;branch=z9hG4bKt7cUUFUKyea3r
> Max-Forwards: 69
> From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
> To: <sip:3809XXXXXXXX at sip.telecomax.net>
> Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
> CSeq: 116249601 INVITE
> Contact: <sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c>
> User-Agent: FreeSWITCH
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 270
> X-Serialnumber: 000413719560
> P-Key-Flags: resolution="31x13", keys="4"
> X-accountcode: fusion.contactise.com
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1513140493 1513140494 IN IP4 138.68.65.164
> s=FreeSWITCH
> c=IN IP4 138.68.65.164
> t=0 0
> m=audio 30838 RTP/AVP 0 8 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> FS <- Prov
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP 138.68.65.164:5080;branch=z9hG4bKUg6KXacQUQ0Nm;received=138.68.65.164;rport=5080
> From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
> To: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
> Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
> CSeq: 116249602 INVITE
> User-Agent: Free World Web Client
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:3809XXXXXXXX at 80.242.134.195>
> Content-Type: application/sdp
> Content-Length: 268
>
> v=0
> o=root 21659 21659 IN IP4 80.242.134.195
> s=session
> c=IN IP4 80.242.134.195
> t=0 0
> m=audio 13494 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> FS <- Prov
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 138.68.65.164:5080;branch=z9hG4bKUg6KXacQUQ0Nm;received=138.68.65.164;rport=5080
> From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
> To: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
> Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
> CSeq: 116249602 INVITE
> User-Agent: Free World Web Client
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:3809XXXXXXXX at 80.242.134.195>
> Content-Type: application/sdp
> Content-Length: 268
>
> v=0
> o=root 21659 21660 IN IP4 80.242.134.195
> s=session
> c=IN IP4 80.242.134.195
> t=0 0
> m=audio 13494 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> FS -> Prov
> ACK sip:3809XXXXXXXX at 80.242.134.195 SIP/2.0
> Via: SIP/2.0/UDP 138.68.65.164:5080;rport;branch=z9hG4bKvSZcZ5vtr0p8F
> Max-Forwards: 70
> From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
> To: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
> Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
> CSeq: 116249602 ACK
> Contact: <sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c>
> Proxy-Authorization: Digest username="456XXXXXXX", realm="Free World Web Client", nonce="6d32caff", algorithm=MD5, uri="sip:3809XXXXXXXX at sip.telecomax.net", response="c0aa24f024fc447594290b580bcedbfa"
> Content-Length: 0
>
> <- This BYE is received, but no react to it in CLI at all.
>
> FS <- Prov
> BYE sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c SIP/2.0
> Via: SIP/2.0/UDP 80.242.134.195:5060;branch=z9hG4bK06dce488;rport
> From: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
> To: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
> Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
> CSeq: 102 BYE
> User-Agent: Free World Web Client
> Max-Forwards: 70
> X-Asterisk-HangupCause: Normal Clearing
> X-Asterisk-HangupCauseCode: 16
> Content-Length: 0
>
>
>
>
>
>
> I've set up sofia debug all 9 and received this
>    ------------------------------------------------------------------------
> BYE sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c SIP/2.0
>    Via: SIP/2.0/UDP 80.242.134.195:5060;branch=z9hG4bK06dce488;rport
>    From: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as77b4319c
>    To: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
>    Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
>    CSeq: 102 BYE
>    User-Agent: Free World Web Client
>    Max-Forwards: 70
>    X-Asterisk-HangupCause: Normal Clearing
>    X-Asterisk-HangupCauseCode: 16
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> tport.c:3023 tport_deliver() tport_deliver(0x7ff1e0004650): msg 0x7ff1e0025700 (536 bytes) from udp/80.242.134.195:5080/sip next=(nil)
> nta.c:2880 agent_recv_request() nta: received BYE sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c SIP/2.0 (CSeq 102)
> nta.c:3248 agent_aliases() nta: canonizing sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080 with contact
> nta.c:3060 agent_recv_request() nta: BYE (102) going to existing leg
> nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
> nua_stack.c:271 nua_stack_event() nua(0x7ff204000b60): event i_bye 100 Trying
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> nta.c:7134 _nta_incoming_timer() nta: timer I fired, terminate 200 response
> nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7ff20dc4ac60)
> nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
> nta.c:1289 agent_timer() nta: timer not set
> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7ff1e8004410): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7ff1e8004410)
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7ff1e8004410) msg 0x7ff1e80c90c0 from (udp/138.68.65.164:5060) has 4 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7ff1e8004410): bad msg 0x7ff1e80c90c0 (4 bytes) from udp/176.104.56.91:5060/sip next=(nil)
> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7ff1e0004650): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7ff1e0004650)
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7ff1e0004650) msg 0x7ff1e00582d0 from (udp/138.68.65.164:5080) has 536 bytes, veclen = 1
>
>
>
> And this is it….
> Also there is one-way sound, but I think it’s somehow related to this.
>
> FS 1.6.19 (FusionPBX)
>
> Regards, Igor
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From me at anatoli.ws  Thu Dec 14 02:02:01 2017
From: me at anatoli.ws (Anatoli)
Date: Wed, 13 Dec 2017 23:02:01 -0300
Subject: [Freeswitch-users] Problem with personalized locutions and use
 of say with type shor_date_time.
In-Reply-To: <031601d3743b$007f94d0$017ebe70$@smartic.es>
References: <031601d3743b$007f94d0$017ebe70$@smartic.es>
Message-ID: <e29102da-de85-40f0-ae83-e3b6eafc6263@anatoli.ws>

Hi Miguel Jesús,

That's because mod_say_es doesn't implement all say types, but most of 
the TTS configs use them (e.g. lang/en/vm/sounds.xml:396 <action 
function="say" data="$1" method="pronounced" type="short_date_time"/>). 
So, if you use as the base the English configs (because the Spanish base 
lacks voicemail or conference configs), mod_say_es won't work.

The Spanish sounds/prompts are broken. You may try to use mod_say_es_ar 
which is based on the latest mod_say_en, but it has some issues with 
some prompts anyway (though easier to solve). Basically, the problem is 
that the mod_say_es is outdated, the TTS configs are not in sync with it 
and the available Spanish prompts (sounds_es_mx_maria and 
sounds_es_ar_mario, I couldn't find others) are also not in sync with 
the configs and the say modules, and are incomplete (e.g. no "un" sound 
for "usted tiene /un/ mensaje nuevo" ("you have /one/ new message"), 
some words lack singular or plural forms, some prompts are completely 
absent (e.g. "entered the conference"), etc.).

I had to mix sounds from the 2 sets (MX and AR) and even borrow some 
sounds from the asterisk sounds. Even this way the result is not great 
(masculine and feminine voices mixed in the same sentence, lack of 
singular/plural forms for some words, etc.), but at least it works and 
the users can interact with the system.

The Spanish support in FS is not production ready. What should be done?

 1. mod_say_es should be updated to be in sync and as complete as the
    rest of the say modules (should be based on mod_say_en as this is
    the reference). And there is NO need for country-specific Spanish
    *say modules*, i.e. the grammar (the sentence composition) is the
    same in all regions. Only the prompts should be recorded by native
    speakers of each country.
 2. The Spanish TTS configs should be updated accordingly (again, NO
    country-specific configs).
 3. The list of necessary sounds (i.e. the prompts (texts) to
    country-adapt and record) should be defined (either in English or in
    neutral Spanish).
 4. A country-neutral set of Spanish sounds should be created.

Once this is ready, the community could provide the prompts for 
country-specific dialects. And I'm ready to co-sponsor the Spanish base 
(the 4 items list above) and to provide the correct es_AR prompts. If 
someone can implement the Spanish base (at least the first 3 items 
above), please let us know your price, I guess we could crowd-fund it.

Regards,
Anatoli

*From:* Miguel Jesús López Valverde
*Sent:* Wednesday, December 13, 2017 14:51
*To:* Freeswitch-users
*Subject:* [Freeswitch-users] Problem with personalized locutions and 
use of say with type shor_date_time.

Hello Freeswitch users:

I send this email to you with a new problem found.

I have a set of complete locutions created with a TTS for Spanish 
locutions installed in Freeswitch, along with those brought by default 
in English.

When using the phrases in Spanish, I am only having problems when using 
the directive:

<action function = "say" data = "$ 1" method = "pronounced" type = 
"short_date_time" />

but with other directives "say" with different methods, I'm not having 
any problems. For example, when accessing voicemail using these 
locutions set in Spanish, I get the following error that prevents access 
to recorded messages:

2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:250 Handle 
play-file:[voicemail/vm-new.wav] (es_CB:es)

2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:53.172924 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file 
/usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-new.wav

2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:250 Handle 
play-file:[voicemail/vm-message_number.wav] (es_CB:es)

2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:54.832899 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file 
/usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-message_number.wav

2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:250 Handle 
say:[1] (es_CB:es)

2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:55.872903 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file /usr/local/freeswitch/sounds/es/colabora/sonia/digits/1.wav

2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:70 No language 
specified - Using [es_CB]

*2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:250 Handle 
say:[1513180033] (es_CB:es)*

*2017-12-13 16:58:55.972959 [ERR] mod_say_es.c:471 Unknown Say type=[18]*

By changing this language by default English, it works correctly ...

2017-12-13 17:18:05.327117 [DEBUG] switch_ivr_play_say.c:70 No language 
specified - Using [en]

2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:250 Handle 
play-file:[voicemail/vm-new.wav] (en:en)

2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:05.667117 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav

2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:250 Handle 
play-file:[voicemail/vm-message_number.wav] (en:en)

2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:06.587140 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file 
/usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-message_number.wav

2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:250 Handle 
say:[1] (en:en)

2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:07.407128 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file file_string://digits/1.wav

2017-12-13 17:18:07.507133 [DEBUG] switch_ivr_play_say.c:70 No language 
specified - Using [en]

2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:250 Handle 
say:[1513180033] (en:en)

2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:10.427138 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file 
file_string://time/today.wav!time/at.wav!digits/3.wav!digits/40.wav!digits/7.wav!time/p-m.wav

2017-12-13 17:18:10.547147 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

It is proven that I have all the locutions in Spanish, including those 
in the "time" and "digits" folders.

Does anyone know how I can get a proper voicemail operation with 
personalized locutions when using the function "say" with "type =" 
short_date_time "" ?.

Thank you very much and best regards.


<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> 
	Libre de virus. www.avast.com 
<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> 


<#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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From gregor at infomedia.si  Thu Dec 14 09:05:40 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 14 Dec 2017 10:05:40 +0100
Subject: [Freeswitch-users] Random number in variable
Message-ID: <CAC7ttFGLKLQvY2PBcHCCwKO=TmuEmDhhhu2VzaJ3mgGMr==8Rg@mail.gmail.com>

Hi, Guys!

One hint, please. Is there a way to get random integer in FS variable
without lua in Dialplan like:

<action application="set" data="
​myvar=
​some function to get random number betwee 100-200 or curl fetch?
"/>

​Thank you, Gregor​
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From aviv at ironsip.com  Thu Dec 14 09:17:34 2017
From: aviv at ironsip.com (Aviv Shaham)
Date: Thu, 14 Dec 2017 02:17:34 -0700
Subject: [Freeswitch-users] Random number in variable
In-Reply-To: <CAC7ttFGLKLQvY2PBcHCCwKO=TmuEmDhhhu2VzaJ3mgGMr==8Rg@mail.gmail.com>
References: <CAC7ttFGLKLQvY2PBcHCCwKO=TmuEmDhhhu2VzaJ3mgGMr==8Rg@mail.gmail.com>
Message-ID: <1513243054.3303550.1204723424.64EE41AE@webmail.messagingengine.com>

https://wiki.freeswitch.org/wiki/Mod_expr has it


On Thu, Dec 14, 2017, at 02:05 AM, Gregor Nanger wrote:
> Hi, Guys!
> 
> One hint, please. Is there a way to get random integer in FS variable
> without lua in Dialplan like:> 
> <action application="set" data="
> myvar=
> some function to get random number betwee 100-200 or curl fetch?
> "/>
> 
> 
> Thank you, Gregor
>  
> ___________________________________________________________________-
> ________> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org

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From kamil.nigmatullin at gmail.com  Thu Dec 14 09:34:06 2017
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Thu, 14 Dec 2017 15:34:06 +0600
Subject: [Freeswitch-users] H248
Message-ID: <CA+Xtiec=tOqFCY-buFGR0bdxCH8DoWa1D8uBWKzvcbB0t39Qmg@mail.gmail.com>

Hello,

One of our clients have old equipment that seems can send h248 over TCP/IP
and they need to convert it to SIP. There is not much info I found in the
Net so please tell me if it is possible with FS or maybe Yate or others?
There was some module called mod_sigtran but I cannot find it in recent FS
confluence

Thank you

-- 
Kamil Nigmatullin
Tel: 77272323748
mob: 7 (707) 2517003
Skype: kamil.nigmatullin
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From s.safarov at gmail.com  Thu Dec 14 09:59:58 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 14 Dec 2017 09:59:58 +0000
Subject: [Freeswitch-users] FS not reacting to BYE
In-Reply-To: <294110ad-760b-45d8-a8ae-476c0eaf0146@Spark>
References: <4f36450f-7499-4c7f-994c-e3cce8eb6170@Spark>
 <294110ad-760b-45d8-a8ae-476c0eaf0146@Spark>
Message-ID: <CAHtxdDe1cjtwMjpJCCoXaoAmiSaGec0=9fL+waxwCZ5hCGajng@mail.gmail.com>

Look https://freeswitch.org/jira/browse/FS-7399

чт, 14 дек. 2017 г. в 3:06, Igor Olhovskiy <igorolhovskiy at gmail.com>:

> Seems it’s really strange issue.
>
> Call was made with api_on_answer variable and in this api lua script was
> called.
> lua script just make curl request with session:execute(«curl», «
> https://xxxx»)
> Only in this case FS not has a problem with accepting BYE.
>
> When I redo curl request with bgapi - all act as designed.
>
> Regards, Igor
>
> On Dec 13, 2017, 3:48 PM +0200, Igor Olhovskiy <igorolhovskiy at gmail.com>,
> wrote:
>
> Hi all!
>
>
> Got into issue with one of providers, and can’t get some things...
> For what reason FS is not reacting to BYE on external profile?
>
> FS -> Prov
> INVITE sip:3809XXXXXXXX at sip.telecomax.net SIP/2.0
> Via: SIP/2.0/UDP 138.68.65.164:5080;rport;branch=z9hG4bKt7cUUFUKyea3r
> Max-Forwards: 69
> From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
> To: <sip:3809XXXXXXXX at sip.telecomax.net>
> Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
> CSeq: 116249601 INVITE
> Contact: <sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080
> ;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c>
> User-Agent: FreeSWITCH
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 270
> X-Serialnumber: 000413719560
> P-Key-Flags: resolution="31x13", keys="4"
> X-accountcode: fusion.contactise.com
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net
> >;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1513140493 1513140494 IN IP4 138.68.65.164
> s=FreeSWITCH
> c=IN IP4 138.68.65.164
> t=0 0
> m=audio 30838 RTP/AVP 0 8 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> FS <- Prov
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP 138.68.65.164:5080
> ;branch=z9hG4bKUg6KXacQUQ0Nm;received=138.68.65.164;rport=5080
> From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
> To: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
> Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
> CSeq: 116249602 INVITE
> User-Agent: Free World Web Client
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:3809XXXXXXXX at 80.242.134.195>
> Content-Type: application/sdp
> Content-Length: 268
>
> v=0
> o=root 21659 21659 IN IP4 80.242.134.195
> s=session
> c=IN IP4 80.242.134.195
> t=0 0
> m=audio 13494 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> FS <- Prov
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 138.68.65.164:5080
> ;branch=z9hG4bKUg6KXacQUQ0Nm;received=138.68.65.164;rport=5080
> From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
> To: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
> Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
> CSeq: 116249602 INVITE
> User-Agent: Free World Web Client
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:3809XXXXXXXX at 80.242.134.195>
> Content-Type: application/sdp
> Content-Length: 268
>
> v=0
> o=root 21659 21660 IN IP4 80.242.134.195
> s=session
> c=IN IP4 80.242.134.195
> t=0 0
> m=audio 13494 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> FS -> Prov
> ACK sip:3809XXXXXXXX at 80.242.134.195 SIP/2.0
> Via: SIP/2.0/UDP 138.68.65.164:5080;rport;branch=z9hG4bKvSZcZ5vtr0p8F
> Max-Forwards: 70
> From: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
> To: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
> Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
> CSeq: 116249602 ACK
> Contact: <sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080
> ;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c>
> Proxy-Authorization: Digest username="456XXXXXXX", realm="Free World Web
> Client", nonce="6d32caff", algorithm=MD5, uri="
> sip:3809XXXXXXXX at sip.telecomax.net",
> response="c0aa24f024fc447594290b580bcedbfa"
> Content-Length: 0
>
> <- This BYE is received, but no react to it in CLI at all.
>
> FS <- Prov
> BYE sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c
> SIP/2.0
> Via: SIP/2.0/UDP 80.242.134.195:5060;branch=z9hG4bK06dce488;rport
> From: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as5cec60a1
> To: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
> Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
> CSeq: 102 BYE
> User-Agent: Free World Web Client
> Max-Forwards: 70
> X-Asterisk-HangupCause: Normal Clearing
> X-Asterisk-HangupCauseCode: 16
> Content-Length: 0
>
>
>
>
>
>
> I've set up sofia debug all 9 and received this
>    ------------------------------------------------------------------------
> BYE sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c
> SIP/2.0
>    Via: SIP/2.0/UDP 80.242.134.195:5060;branch=z9hG4bK06dce488;rport
>    From: <sip:3809XXXXXXXX at sip.telecomax.net>;tag=as77b4319c
>    To: "456XXXXXXX" <sip:456XXXXXXX at sip.telecomax.net>;tag=gFjZSmajjg36F
>    Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359
>    CSeq: 102 BYE
>    User-Agent: Free World Web Client
>    Max-Forwards: 70
>    X-Asterisk-HangupCause: Normal Clearing
>    X-Asterisk-HangupCauseCode: 16
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> tport.c:3023 tport_deliver() tport_deliver(0x7ff1e0004650): msg
> 0x7ff1e0025700 (536 bytes) from udp/80.242.134.195:5080/sip next=(nil)
> nta.c:2880 agent_recv_request() nta: received BYE
> sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c
> SIP/2.0 (CSeq 102)
> nta.c:3248 agent_aliases() nta: canonizing
> sip:gw+3b179ac8-000f-453a-8059-7922501ae04c at 138.68.65.164:5080 with
> contact
> nta.c:3060 agent_recv_request() nta: BYE (102) going to existing leg
> nua_server.c:102 nua_stack_process_request() nua:
> nua_stack_process_request: entering
> nua_stack.c:271 nua_stack_event() nua(0x7ff204000b60): event i_bye 100
> Trying
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nta.c:7134 _nta_incoming_timer() nta: timer I fired, terminate 200 response
> nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil),
> 0x7ff20dc4ac60)
> nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout,
> 1/1 term, 1/1 free
> nta.c:1289 agent_timer() nta: timer not set
> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7ff1e8004410): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7ff1e8004410)
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7ff1e8004410) msg
> 0x7ff1e80c90c0 from (udp/138.68.65.164:5060) has 4 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7ff1e8004410): bad msg
> 0x7ff1e80c90c0 (4 bytes) from udp/176.104.56.91:5060/sip next=(nil)
> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7ff1e0004650): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7ff1e0004650)
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7ff1e0004650) msg
> 0x7ff1e00582d0 from (udp/138.68.65.164:5080) has 536 bytes, veclen = 1
>
>
>
> And this is it….
> Also there is one-way sound, but I think it’s somehow related to this.
>
> FS 1.6.19 (FusionPBX)
>
> Regards, Igor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mjlopez at smartic.es  Thu Dec 14 11:53:41 2017
From: mjlopez at smartic.es (=?UTF-8?Q?Miguel_Jes=C3=BAs_L=C3=B3pez_Valverde?=)
Date: Thu, 14 Dec 2017 12:53:41 +0100
Subject: [Freeswitch-users] Problem with personalized locutions and use
	of say with type shor_date_time.
In-Reply-To: <e29102da-de85-40f0-ae83-e3b6eafc6263@anatoli.ws>
References: <031601d3743b$007f94d0$017ebe70$@smartic.es>
 <e29102da-de85-40f0-ae83-e3b6eafc6263@anatoli.ws>
Message-ID: <03e401d374d2$308ff010$91afd030$@smartic.es>

Hello Anatoli, thank you very much for your answer.

 

                At this time, the sets of locutions in Neutral-Spanish that I have available have been obtained with a TTS only with laboratory interest. I do not consider them suitable for a more professional or commercial use.

 

On the other hand, we are evaluating making recordings through some Spanish studies that have made us offers, based on the list available by default of locutions in English. For your information, the prices are approx. about 1800 Euros per group of locutions, (initially we would only make a group in female voice) considering that the studies that have offered us maintain a high level of quality in our point of view, since they have already done other work for us with optimal results .

 

Please, indicate if you have a more complete list than the one used by us to facilitate these studies, (the list that we have handled has been the one of locutions in English given in the version 1.6.18), so that we can pass it to assess.

 

In case of getting to make the recordings of the locutions, we could assess some form of cooperation to facilitate them to the community, (shared payment, training agreements or certification courses, etc ..).

 

Please, indicate if you are interested because, in this case, it may also be of interest to us to contract the completion of the locutions.

 

Again, Thank you very much and receive greeting.

 

 

 

De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Anatoli
Enviado el: jueves, 14 de diciembre de 2017 3:02
Para: freeswitch-users at lists.freeswitch.org; consulting at freeswitch.org
Asunto: Re: [Freeswitch-users] Problem with personalized locutions and use of say with type shor_date_time.

 

Hi Miguel Jesús,

That's because mod_say_es doesn't implement all say types, but most of the TTS configs use them (e.g. lang/en/vm/sounds.xml:396 <action function="say" data="$1" method="pronounced" type="short_date_time"/>). So, if you use as the base the English configs (because the Spanish base lacks voicemail or conference configs), mod_say_es won't work.

The Spanish sounds/prompts are broken. You may try to use mod_say_es_ar which is based on the latest mod_say_en, but it has some issues with some prompts anyway (though easier to solve). Basically, the problem is that the mod_say_es is outdated, the TTS configs are not in sync with it and the available Spanish prompts (sounds_es_mx_maria and sounds_es_ar_mario, I couldn't find others) are also not in sync with the configs and the say modules, and are incomplete (e.g. no "un" sound for "usted tiene un mensaje nuevo" ("you have one new message"), some words lack singular or plural forms, some prompts are completely absent (e.g. "entered the conference"), etc.).

I had to mix sounds from the 2 sets (MX and AR) and even borrow some sounds from the asterisk sounds. Even this way the result is not great (masculine and feminine voices mixed in the same sentence, lack of singular/plural forms for some words, etc.), but at least it works and the users can interact with the system.

The Spanish support in FS is not production ready. What should be done?

1.	mod_say_es should be updated to be in sync and as complete as the rest of the say modules (should be based on mod_say_en as this is the reference). And there is NO need for country-specific Spanish *say modules*, i.e. the grammar (the sentence composition) is the same in all regions. Only the prompts should be recorded by native speakers of each country.
2.	The Spanish TTS configs should be updated accordingly (again, NO country-specific configs).
3.	The list of necessary sounds (i.e. the prompts (texts) to country-adapt and record) should be defined (either in English or in neutral Spanish).
4.	A country-neutral set of Spanish sounds should be created.

Once this is ready, the community could provide the prompts for country-specific dialects. And I'm ready to co-sponsor the Spanish base (the 4 items list above) and to provide the correct es_AR prompts. If someone can implement the Spanish base (at least the first 3 items above), please let us know your price, I guess we could crowd-fund it.

Regards,
Anatoli

From: Miguel Jesús López Valverde
Sent: Wednesday, December 13, 2017 14:51
To: Freeswitch-users
Subject: [Freeswitch-users] Problem with personalized locutions and use of say with type shor_date_time.

 

Hello Freeswitch users:

 

I send this email to you with a new problem found.

 

I have a set of complete locutions created with a TTS for Spanish locutions installed in Freeswitch, along with those brought by default in English.

 

When using the phrases in Spanish, I am only having problems when using the directive:

<action function = "say" data = "$ 1" method = "pronounced" type = "short_date_time" />

but with other directives "say" with different methods, I'm not having any problems. For example, when accessing voicemail using these locutions set in Spanish, I get the following error that prevents access to recorded messages:

 

2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-new.wav] (es_CB:es)

2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:53.172924 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-new.wav

2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-message_number.wav] (es_CB:es)

2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:54.832899 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-message_number.wav

2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1] (es_CB:es)

2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:55.872903 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/es/colabora/sonia/digits/1.wav

2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [es_CB]

2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1513180033] (es_CB:es)

2017-12-13 16:58:55.972959 [ERR] mod_say_es.c:471 Unknown Say type=[18]

 

By changing this language by default English, it works correctly ...

 

2017-12-13 17:18:05.327117 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en]

2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-new.wav] (en:en)

2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:05.667117 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav

2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-message_number.wav] (en:en)

2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:06.587140 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-message_number.wav

2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1] (en:en)

2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:07.407128 [DEBUG] switch_ivr_play_say.c:1942 done playing file file_string://digits/1.wav

2017-12-13 17:18:07.507133 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en]

2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1513180033] (en:en)

2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:10.427138 [DEBUG] switch_ivr_play_say.c:1942 done playing file file_string://time/today.wav!time/at.wav!digits/3.wav!digits/40.wav!digits/7.wav!time/p-m.wav

2017-12-13 17:18:10.547147 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

 

It is proven that I have all the locutions in Spanish, including those in the "time" and "digits" folders.

 

Does anyone know how I can get a proper voicemail operation with personalized locutions when using the function "say" with "type =" short_date_time "" ?.

 

Thank you very much and best regards.

 

 


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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com
 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mmeek at livexchange.com  Thu Dec 14 12:01:00 2017
From: mmeek at livexchange.com (Matthew Meek)
Date: Thu, 14 Dec 2017 12:01:00 +0000
Subject: [Freeswitch-users] bridge_generate_comfort_noise not honoring
 integer value for volume level
Message-ID: <MWHPR11MB1486A0000B461F759B86A5CEB60A0@MWHPR11MB1486.namprd11.prod.outlook.com>

I can successfully inject whitenoise on the bridged leg (audio from A leg towards the bridged call) in the dial plan by setting bridge_generate_comfort_noise=true. The whitenoise is quite loud and I would like to make it quieter or silent (I am just trying to keep the far end firewall port open during long periods of A leg not sending packets due to hold).  I have tried values of -1, 40,1400, and 4000 instead of true and they produce the same amount of whitenoise volume. The source code in switch_ivr_bridge.c looks like it indeed honors an integer value, but somehow the end result is always the same. If I do not set the variable I get no whitenoise, so the variable is indeed controlling whitenoise, just not the volume level of whitenoise. The codec is PCMU.

I can repeat this with 1.2, 1.4, and 1.6 branches of FS.

Matthew Meek
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From gregor at infomedia.si  Thu Dec 14 12:05:26 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 14 Dec 2017 13:05:26 +0100
Subject: [Freeswitch-users] Random number in variable
In-Reply-To: <1513243054.3303550.1204723424.64EE41AE@webmail.messagingengine.com>
References: <CAC7ttFGLKLQvY2PBcHCCwKO=TmuEmDhhhu2VzaJ3mgGMr==8Rg@mail.gmail.com>
 <1513243054.3303550.1204723424.64EE41AE@webmail.messagingengine.com>
Message-ID: <CAC7ttFEfJ7hspK9zLx5FmDe2DL2aOyj5B_OwxWKp9BGgGMijfg@mail.gmail.com>

:-) thank you, Aviv

2017-12-14 10:17 GMT+01:00 Aviv Shaham <aviv at ironsip.com>:

> https://wiki.freeswitch.org/wiki/Mod_expr has it
>
>
> On Thu, Dec 14, 2017, at 02:05 AM, Gregor Nanger wrote:
>
> Hi, Guys!
>
> One hint, please. Is there a way to get random integer in FS variable
> without lua in Dialplan like:
>
> <action application="set" data="
> myvar=
> some function to get random number betwee 100-200 or curl fetch?
> "/>
>
>
> Thank you, Gregor
>
> *_________________________________________________________________________*
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From ssinyagin at gmail.com  Thu Dec 14 13:17:05 2017
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 14 Dec 2017 14:17:05 +0100
Subject: [Freeswitch-users] Random number in variable
In-Reply-To: <CAC7ttFEfJ7hspK9zLx5FmDe2DL2aOyj5B_OwxWKp9BGgGMijfg@mail.gmail.com>
References: <CAC7ttFGLKLQvY2PBcHCCwKO=TmuEmDhhhu2VzaJ3mgGMr==8Rg@mail.gmail.com>
 <1513243054.3303550.1204723424.64EE41AE@webmail.messagingengine.com>
 <CAC7ttFEfJ7hspK9zLx5FmDe2DL2aOyj5B_OwxWKp9BGgGMijfg@mail.gmail.com>
Message-ID: <CACOYK=gYvd3vBf2LZ-LfwfEKF7-N5B49x4f9CCrL_XO1MFVW4g@mail.gmail.com>

either the random function, or a global counter if you want some more
predictable randomness:

        <action application="set_global"
data="global_fsqa_counter=${expr(mod(${global_fsqa_counter}+1,23))}"/>
        <action application="set"
data="silence_pause=${expr(${global_fsqa_counter}*3+1000)}"/>
        <action application="playback"
data="silence_stream://${silence_pause}"/>

this was for a test appliance, to make sure the same pieces of playback
audio don't get cut into 20ms frames in the same way.




On Thu, Dec 14, 2017 at 1:05 PM, Gregor Nanger <gregor at infomedia.si> wrote:

> :-) thank you, Aviv
>
> 2017-12-14 10:17 GMT+01:00 Aviv Shaham <aviv at ironsip.com>:
>
>> https://wiki.freeswitch.org/wiki/Mod_expr has it
>>
>>
>> On Thu, Dec 14, 2017, at 02:05 AM, Gregor Nanger wrote:
>>
>> Hi, Guys!
>>
>> One hint, please. Is there a way to get random integer in FS variable
>> without lua in Dialplan like:
>>
>> <action application="set" data="
>> myvar=
>> some function to get random number betwee 100-200 or curl fetch?
>> "/>
>>
>>
>> Thank you, Gregor
>>
>>
>> *_________________________________________________________________________*
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gregor at infomedia.si  Thu Dec 14 13:21:06 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 14 Dec 2017 14:21:06 +0100
Subject: [Freeswitch-users] Random number in variable
In-Reply-To: <CACOYK=gYvd3vBf2LZ-LfwfEKF7-N5B49x4f9CCrL_XO1MFVW4g@mail.gmail.com>
References: <CAC7ttFGLKLQvY2PBcHCCwKO=TmuEmDhhhu2VzaJ3mgGMr==8Rg@mail.gmail.com>
 <1513243054.3303550.1204723424.64EE41AE@webmail.messagingengine.com>
 <CAC7ttFEfJ7hspK9zLx5FmDe2DL2aOyj5B_OwxWKp9BGgGMijfg@mail.gmail.com>
 <CACOYK=gYvd3vBf2LZ-LfwfEKF7-N5B49x4f9CCrL_XO1MFVW4g@mail.gmail.com>
Message-ID: <CAC7ttFFvDwY_CPw81hajZty7Sm_NOC1gZVFBH8PMaKtcvEusFQ@mail.gmail.com>

Thank you

2017-12-14 14:17 GMT+01:00 Stanislav Sinyagin <ssinyagin at gmail.com>:

> either the random function, or a global counter if you want some more
> predictable randomness:
>
>         <action application="set_global" data="global_fsqa_counter=${
> expr(mod(${global_fsqa_counter}+1,23))}"/>
>         <action application="set" data="silence_pause=${expr(${
> global_fsqa_counter}*3+1000)}"/>
>         <action application="playback" data="silence_stream://${
> silence_pause}"/>
>
> this was for a test appliance, to make sure the same pieces of playback
> audio don't get cut into 20ms frames in the same way.
>
>
>
>
> On Thu, Dec 14, 2017 at 1:05 PM, Gregor Nanger <gregor at infomedia.si>
> wrote:
>
>> :-) thank you, Aviv
>>
>> 2017-12-14 10:17 GMT+01:00 Aviv Shaham <aviv at ironsip.com>:
>>
>>> https://wiki.freeswitch.org/wiki/Mod_expr has it
>>>
>>>
>>> On Thu, Dec 14, 2017, at 02:05 AM, Gregor Nanger wrote:
>>>
>>> Hi, Guys!
>>>
>>> One hint, please. Is there a way to get random integer in FS variable
>>> without lua in Dialplan like:
>>>
>>> <action application="set" data="
>>> myvar=
>>> some function to get random number betwee 100-200 or curl fetch?
>>> "/>
>>>
>>>
>>> Thank you, Gregor
>>>
>>>
>>> *_________________________________________________________________________*
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Gregor Nanger
>>
>> *CTO*
>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>> • www.infomedia.si
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From me at anatoli.ws  Thu Dec 14 18:10:03 2017
From: me at anatoli.ws (Anatoli)
Date: Thu, 14 Dec 2017 15:10:03 -0300
Subject: [Freeswitch-users] Problem with personalized locutions and use
 of say with type shor_date_time.
In-Reply-To: <03e401d374d2$308ff010$91afd030$@smartic.es>
References: <031601d3743b$007f94d0$017ebe70$@smartic.es>
 <e29102da-de85-40f0-ae83-e3b6eafc6263@anatoli.ws>
 <03e401d374d2$308ff010$91afd030$@smartic.es>
Message-ID: <8e3e0258-248b-268d-df8c-6bbda2fa7321@anatoli.ws>

Miguel Jesús,

Nice to know you're also interested in high-quality Spanish support in 
FS. After trying for some time to make Spanish sounds work, in my 
opinion, without first completing the Spanish base (the first 3 items 
from my last email) there's no sense to record the prompts. You can't 
take the English base and just record Spanish prompts based on it 
blindly, it should be carefully adapted, otherwise dates, numbers and a 
lot of other elements won't work or would sound defective. And to get 
the complete list of the prompts (texts) to record, one should complete 
the first 2 items of my list.

The closest adaptation to the English base is mod_say_es_ar, but I guess 
someone should perform a thorough evaluation of the current code and 
configs (also evaluate the implementation of mod_say_es as there are 
some useful language details already implemented) and put everything in 
sync with the English base (i.e. make it as complete as mod_say_en and 
sounds_en_callie).

I don't know if someone from the community could perform this adaptation 
or we should directly ask the Professional FreeSWITCH Consulting 
Services (CC'ing them).

Regards,
Anatoli

*From:* Miguel Jesús López Valverde
*Sent:* Thursday, December 14, 2017 08:53
*To:* 'Freeswitch Users Help', Consulting
*Cc:* Me
*Subject:* RE: [Freeswitch-users] Problem with personalized locutions 
and use of say with type shor_date_time.

Hello Anatoli, thank you very much for your answer.

At this time, the sets of locutions in Neutral-Spanish that I have 
available have been obtained with a TTS only with laboratory interest. I 
do not consider them suitable for a more professional or commercial use.

On the other hand, we are evaluating making recordings through some 
Spanish studies that have made us offers, based on the list available by 
default of locutions in English. For your information, the prices are 
approx. about 1800 Euros per group of locutions, (initially we would 
only make a group in female voice) considering that the studies that 
have offered us maintain a high level of quality in our point of view, 
since they have already done other work for us with optimal results .

Please, indicate if you have a more complete list than the one used by 
us to facilitate these studies, (the list that we have handled has been 
the one of locutions in English given in the version 1.6.18), so that we 
can pass it to assess.

In case of getting to make the recordings of the locutions, we could 
assess some form of cooperation to facilitate them to the community, 
(shared payment, training agreements or certification courses, etc ..).

Please, indicate if you are interested because, in this case, it may 
also be of interest to us to contract the completion of the locutions.

Again, Thank you very much and receive greeting.

*De:*FreeSWITCH-users 
[mailto:freeswitch-users-bounces at lists.freeswitch.org] *En nombre de 
*Anatoli
*Enviado el:* jueves, 14 de diciembre de 2017 3:02
*Para:* freeswitch-users at lists.freeswitch.org; consulting at freeswitch.org
*Asunto:* Re: [Freeswitch-users] Problem with personalized locutions and 
use of say with type shor_date_time.

Hi Miguel Jesús,

That's because mod_say_es doesn't implement all say types, but most of 
the TTS configs use them (e.g. lang/en/vm/sounds.xml:396 <action 
function="say" data="$1" method="pronounced" type="short_date_time"/>). 
So, if you use as the base the English configs (because the Spanish base 
lacks voicemail or conference configs), mod_say_es won't work.

The Spanish sounds/prompts are broken. You may try to use mod_say_es_ar 
which is based on the latest mod_say_en, but it has some issues with 
some prompts anyway (though easier to solve). Basically, the problem is 
that the mod_say_es is outdated, the TTS configs are not in sync with it 
and the available Spanish prompts (sounds_es_mx_maria and 
sounds_es_ar_mario, I couldn't find others) are also not in sync with 
the configs and the say modules, and are incomplete (e.g. no "un" sound 
for "usted tiene /un/ mensaje nuevo" ("you have /one/ new message"), 
some words lack singular or plural forms, some prompts are completely 
absent (e.g. "entered the conference"), etc.).

I had to mix sounds from the 2 sets (MX and AR) and even borrow some 
sounds from the asterisk sounds. Even this way the result is not great 
(masculine and feminine voices mixed in the same sentence, lack of 
singular/plural forms for some words, etc.), but at least it works and 
the users can interact with the system.

The Spanish support in FS is not production ready. What should be done?

 1. mod_say_es should be updated to be in sync and as complete as the
    rest of the say modules (should be based on mod_say_en as this is
    the reference). And there is NO need for country-specific Spanish
    *say modules*, i.e. the grammar (the sentence composition) is the
    same in all regions. Only the prompts should be recorded by native
    speakers of each country.
 2. The Spanish TTS configs should be updated accordingly (again, NO
    country-specific configs).
 3. The list of necessary sounds (i.e. the prompts (texts) to
    country-adapt and record) should be defined (either in English or in
    neutral Spanish).
 4. A country-neutral set of Spanish sounds should be created.

Once this is ready, the community could provide the prompts for 
country-specific dialects. And I'm ready to co-sponsor the Spanish base 
(the 4 items list above) and to provide the correct es_AR prompts. If 
someone can implement the Spanish base (at least the first 3 items 
above), please let us know your price, I guess we could crowd-fund it.

Regards,
Anatoli

*From:*Miguel Jesús López Valverde
*Sent:* Wednesday, December 13, 2017 14:51
*To:* Freeswitch-users
*Subject:* [Freeswitch-users] Problem with personalized locutions and 
use of say with type shor_date_time.

Hello Freeswitch users:

I send this email to you with a new problem found.

I have a set of complete locutions created with a TTS for Spanish 
locutions installed in Freeswitch, along with those brought by default 
in English.

When using the phrases in Spanish, I am only having problems when using 
the directive:

<action function = "say" data = "$ 1" method = "pronounced" type = 
"short_date_time" />

but with other directives "say" with different methods, I'm not having 
any problems. For example, when accessing voicemail using these 
locutions set in Spanish, I get the following error that prevents access 
to recorded messages:

2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:250 Handle 
play-file:[voicemail/vm-new.wav] (es_CB:es)

2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:53.172924 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file 
/usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-new.wav

2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:250 Handle 
play-file:[voicemail/vm-message_number.wav] (es_CB:es)

2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:54.832899 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file 
/usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-message_number.wav

2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:250 Handle 
say:[1] (es_CB:es)

2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:55.872903 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file /usr/local/freeswitch/sounds/es/colabora/sonia/digits/1.wav

2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:70 No language 
specified - Using [es_CB]

*2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:250 Handle 
say:[1513180033] (es_CB:es)*

*2017-12-13 16:58:55.972959 [ERR] mod_say_es.c:471 Unknown Say type=[18]*

By changing this language by default English, it works correctly ...

2017-12-13 17:18:05.327117 [DEBUG] switch_ivr_play_say.c:70 No language 
specified - Using [en]

2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:250 Handle 
play-file:[voicemail/vm-new.wav] (en:en)

2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:05.667117 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav

2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:250 Handle 
play-file:[voicemail/vm-message_number.wav] (en:en)

2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:06.587140 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file 
/usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-message_number.wav

2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:250 Handle 
say:[1] (en:en)

2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:07.407128 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file file_string://digits/1.wav

2017-12-13 17:18:07.507133 [DEBUG] switch_ivr_play_say.c:70 No language 
specified - Using [en]

2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:250 Handle 
say:[1513180033] (en:en)

2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:10.427138 [DEBUG] switch_ivr_play_say.c:1942 done 
playing file 
file_string://time/today.wav!time/at.wav!digits/3.wav!digits/40.wav!digits/7.wav!time/p-m.wav

2017-12-13 17:18:10.547147 [DEBUG] switch_ivr_play_say.c:1498 Codec 
Activated L16 at 8000hz 1 channels 20ms

It is proven that I have all the locutions in Spanish, including those 
in the "time" and "digits" folders.

Does anyone know how I can get a proper voicemail operation with 
personalized locutions when using the function "say" with "type =" 
short_date_time "" ?.

Thank you very much and best regards.

<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient>

	

Libre de virus. www.avast.com 
<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> 





_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org <mailto:consulting at freeswitch.org>

http://www.freeswitchsolutions.com

Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

http://www.cluecon.com

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org 
<mailto:FreeSWITCH-users at lists.freeswitch.org>

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

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From igorolhovskiy at gmail.com  Thu Dec 14 18:10:10 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Thu, 14 Dec 2017 20:10:10 +0200
Subject: [Freeswitch-users] FS or JSSIP/SIP.JS bug?
Message-ID: <140491db-5137-43fb-8787-42d81eb87f24@Spark>

Hi!

Run into a bug, when using «#» (hash) symbol in numbers using sip-over-wss with jssip or sip.js.

Idea is, when you call some number, that contains # symbol, Freeswitch in a case of 2nd leg answer sends 200 OK with full dialed number in Contact field.
And when jssip or sipjs tries to parse Contact field with hash symbol, they can’t do it (maybe cause this symbol is not urlencoded)

So, is this a FS bug or more to JS sip libraries?

Regards, Igor
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From mike at jerris.com  Thu Dec 14 18:21:58 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Dec 2017 13:21:58 -0500
Subject: [Freeswitch-users] FS or JSSIP/SIP.JS bug?
In-Reply-To: <140491db-5137-43fb-8787-42d81eb87f24@Spark>
References: <140491db-5137-43fb-8787-42d81eb87f24@Spark>
Message-ID: <2F243360-506C-4866-911D-B335855EA867@jerris.com>

Does FreeSWITCH master show the same issue?  I added a bit back some more url encoding to more fields we generate to make sure we don’t send things that need to be encoded unencoded.  I don’t recall if  # is one of the chars that actually must be or not.

> On Dec 14, 2017, at 1:10 PM, Igor Olhovskiy <igorolhovskiy at gmail.com> wrote:
> 
> Hi!
> 
> Run into a bug, when using «#» (hash) symbol in numbers using sip-over-wss with jssip or sip.js.
> 
> Idea is, when you call some number, that contains # symbol, Freeswitch in a case of 2nd leg answer sends 200 OK with full dialed number in Contact field.
> And when jssip or sipjs tries to parse Contact field with hash symbol, they can’t do it (maybe cause this symbol is not urlencoded)
> 
> So, is this a FS bug or more to JS sip libraries?
> 
> Regards, Igor
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Thu Dec 14 18:58:10 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Dec 2017 12:58:10 -0600
Subject: [Freeswitch-users] bridge_generate_comfort_noise not honoring
 integer value for volume level
In-Reply-To: <MWHPR11MB1486A0000B461F759B86A5CEB60A0@MWHPR11MB1486.namprd11.prod.outlook.com>
References: <MWHPR11MB1486A0000B461F759B86A5CEB60A0@MWHPR11MB1486.namprd11.prod.outlook.com>
Message-ID: <CAKbxfG_9Qr+wZo1nqte1zjt=PDeRgsnaB+vv=WzoNut+sRxsVA@mail.gmail.com>

The higher the number the softer it will get.  You say 4000 didn't work?
-1 should be absolute silence.

On Thu, Dec 14, 2017 at 6:01 AM, Matthew Meek <mmeek at livexchange.com> wrote:

> I can successfully inject whitenoise on the bridged leg (audio from A leg
> towards the bridged call) in the dial plan by setting
> bridge_generate_comfort_noise=true. The whitenoise is quite loud and I
> would like to make it quieter or silent (I am just trying to keep the far
> end firewall port open during long periods of A leg not sending packets due
> to hold).  I have tried values of -1, 40,1400, and 4000 instead of true and
> they produce the same amount of whitenoise volume. The source code in
> switch_ivr_bridge.c looks like it indeed honors an integer value, but
> somehow the end result is always the same. If I do not set the variable I
> get no whitenoise, so the variable is indeed controlling whitenoise, just
> not the volume level of whitenoise. The codec is PCMU.
>
>
>
> I can repeat this with 1.2, 1.4, and 1.6 branches of FS.
>
>
>
> Matthew Meek
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II
Founder, FreeSWITCH.


https://youtu.be/l_hOxzCt6X4
https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From infos at madovsky.org  Thu Dec 14 20:24:51 2017
From: infos at madovsky.org (Madovsky)
Date: Thu, 14 Dec 2017 12:24:51 -0800
Subject: [Freeswitch-users] about sip hold and H264
Message-ID: <c030ad15-40d2-5e91-a044-894c8f506be3@madovsky.org>

Dunno if it's a problem of configuration or an issue:

if I hold the a-leg with a call with video H264 and unhold audio is ok 
but video received from the B-leg freezes.

I have to hold the b-leg and unhold to let the a-leg continue to receive 
the b-leg video frames.

is there any special configuration to solve it or I report a jira

Thanks

Franck



From igorolhovskiy at gmail.com  Thu Dec 14 22:23:28 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Fri, 15 Dec 2017 00:23:28 +0200
Subject: [Freeswitch-users] FS or JSSIP/SIP.JS bug?
In-Reply-To: <2F243360-506C-4866-911D-B335855EA867@jerris.com>
References: <140491db-5137-43fb-8787-42d81eb87f24@Spark>
 <2F243360-506C-4866-911D-B335855EA867@jerris.com>
Message-ID: <587a66cf-71e9-4df3-adf8-991c10527cb9@Spark>

Will test against master, but anyway, according to
https://tools.ietf.org/html/rfc3986#section-1.2.3
«#» is a special symbol for URI scheme (and Contact holds URI) and must be encoded.

Regards, Igor

On Dec 14, 2017, 8:22 PM +0200, Michael Jerris <mike at jerris.com>, wrote:
> Does FreeSWITCH master show the same issue?  I added a bit back some more url encoding to more fields we generate to make sure we don’t send things that need to be encoded unencoded.  I don’t recall if  # is one of the chars that actually must be or not.
>
> > On Dec 14, 2017, at 1:10 PM, Igor Olhovskiy <igorolhovskiy at gmail.com> wrote:
> >
> > Hi!
> >
> > Run into a bug, when using «#» (hash) symbol in numbers using sip-over-wss with jssip or sip.js.
> >
> > Idea is, when you call some number, that contains # symbol, Freeswitch in a case of 2nd leg answer sends 200 OK with full dialed number in Contact field.
> > And when jssip or sipjs tries to parse Contact field with hash symbol, they can’t do it (maybe cause this symbol is not urlencoded)
> >
> > So, is this a FS bug or more to JS sip libraries?
> >
> > Regards, Igor
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
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From andrew.keil at visytel.com  Fri Dec 15 03:52:03 2017
From: andrew.keil at visytel.com (Andrew Keil)
Date: Fri, 15 Dec 2017 03:52:03 +0000
Subject: [Freeswitch-users] Re Event Priority setting and effect?
Message-ID: <SY3PR01MB1243C45C822EB90EC2EFAD05F30B0@SY3PR01MB1243.ausprd01.prod.outlook.com>

To FreeSWITCH Users,

I noticed the following inside FreeSWITCH Confluence:

setPriority
setPriority([$number])
Sets the priority of an event to $number in case it's fired.

I assume this matches the Lua function: event:setPriority(x)

>From switch_types.h:

typedef enum {
                SWITCH_PRIORITY_NORMAL,
                SWITCH_PRIORITY_LOW,
                SWITCH_PRIORITY_HIGH
} switch_priority_t;


I have a couple of questions:

If I have setup the remote end of an ESL connection to FreeSWITCH to listen for the following:

event plain CUSTOM MY-EVENT

log 5

event plain BACKGROUND_JOB

  1.  If I fire a CUSTOM event (MY-EVENT) from a Lua script on FreeSWITCH does using event:setPriority(x) have any effect on packet ordering received at the remote ESL end (ie. If I set the Priority to 2 (for SWITCH_PRIORITY_HIGH) will the CUSTOM event come before any "log" events or "BACKGROUND_JOB" events)?
  2.  Can I confirm that the only options available to my Lua script using event:setPriority is 0 (for SWITCH_PRIORITY_NORMAL (which I assume is the default)), 1 (for SWITCH_PRIORITY_LOW) and 2 (for SWITCH_PRIORITY_HIGH)?

Thanks in advance,

Andrew Keil

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From andrew.keil at visytel.com  Fri Dec 15 06:26:25 2017
From: andrew.keil at visytel.com (Andrew Keil)
Date: Fri, 15 Dec 2017 06:26:25 +0000
Subject: [Freeswitch-users] Re Event Priority setting and effect?
In-Reply-To: <SY3PR01MB1243C45C822EB90EC2EFAD05F30B0@SY3PR01MB1243.ausprd01.prod.outlook.com>
References: <SY3PR01MB1243C45C822EB90EC2EFAD05F30B0@SY3PR01MB1243.ausprd01.prod.outlook.com>
Message-ID: <SY3PR01MB12434FB33E2ED2E337C6D2F6F30B0@SY3PR01MB1243.ausprd01.prod.outlook.com>

To FreeSWITCH Users,

Quick follow-up question to add to my questions below....

How do I use event:setPriority(...) inside my Lua script since everything I try returns the error "Error in Event::setPriority (arg 2), expected 'switch_priority_t' ...."

Thanks,

Andrew

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Keil
Sent: Friday, 15 December 2017 2:52 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Re Event Priority setting and effect?

To FreeSWITCH Users,

I noticed the following inside FreeSWITCH Confluence:

setPriority
setPriority([$number])
Sets the priority of an event to $number in case it's fired.

I assume this matches the Lua function: event:setPriority(x)

>From switch_types.h:

typedef enum {
                SWITCH_PRIORITY_NORMAL,
                SWITCH_PRIORITY_LOW,
                SWITCH_PRIORITY_HIGH
} switch_priority_t;


I have a couple of questions:

If I have setup the remote end of an ESL connection to FreeSWITCH to listen for the following:

event plain CUSTOM MY-EVENT

log 5

event plain BACKGROUND_JOB

  1.  If I fire a CUSTOM event (MY-EVENT) from a Lua script on FreeSWITCH does using event:setPriority(x) have any effect on packet ordering received at the remote ESL end (ie. If I set the Priority to 2 (for SWITCH_PRIORITY_HIGH) will the CUSTOM event come before any "log" events or "BACKGROUND_JOB" events)?
  2.  Can I confirm that the only options available to my Lua script using event:setPriority is 0 (for SWITCH_PRIORITY_NORMAL (which I assume is the default)), 1 (for SWITCH_PRIORITY_LOW) and 2 (for SWITCH_PRIORITY_HIGH)?

Thanks in advance,

Andrew Keil

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From mjlopez at smartic.es  Fri Dec 15 17:09:33 2017
From: mjlopez at smartic.es (=?UTF-8?Q?Miguel_Jes=C3=BAs_L=C3=B3pez_Valverde?=)
Date: Fri, 15 Dec 2017 18:09:33 +0100
Subject: [Freeswitch-users] Problem with personalized locutions and use
	of say with type shor_date_time.
In-Reply-To: <8e3e0258-248b-268d-df8c-6bbda2fa7321@anatoli.ws>
References: <031601d3743b$007f94d0$017ebe70$@smartic.es>
 <e29102da-de85-40f0-ae83-e3b6eafc6263@anatoli.ws>
 <03e401d374d2$308ff010$91afd030$@smartic.es>
 <8e3e0258-248b-268d-df8c-6bbda2fa7321@anatoli.ws>
Message-ID: <052f01d375c7$7b4e7f40$71eb7dc0$@smartic.es>

Hello again Anatoli.

 

Yes, we are really interested in having high quality voiceovers in FS for Spanish language. At the moment we are going to carry out laboratory tests with the adaptation of the TTS locutions that we have. We do not have the facility to solve the points you indicated as 1, 2 and 3, but we will be happy to find collaboration agreements with the community about the locutions.

If soon you can send us a complete list of locutions, we can request new offers, conditions and terms through our suppliers.

 

I remain aware of your comments and I thank you very much for your response and help.

 

 

De: Anatoli [mailto:me at anatoli.ws] 
Enviado el: jueves, 14 de diciembre de 2017 19:10
Para: Miguel Jesús López Valverde <mjlopez at smartic.es>; 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>; consulting at freeswitch.org
Asunto: Re: [Freeswitch-users] Problem with personalized locutions and use of say with type shor_date_time.

 

Miguel Jesús,

Nice to know you're also interested in high-quality Spanish support in FS. After trying for some time to make Spanish sounds work, in my opinion, without first completing the Spanish base (the first 3 items from my last email) there's no sense to record the prompts. You can't take the English base and just record Spanish prompts based on it blindly, it should be carefully adapted, otherwise dates, numbers and a lot of other elements won't work or would sound defective. And to get the complete list of the prompts (texts) to record, one should complete the first 2 items of my list.

The closest adaptation to the English base is mod_say_es_ar, but I guess someone should perform a thorough evaluation of the current code and configs (also evaluate the implementation of mod_say_es as there are some useful language details already implemented) and put everything in sync with the English base (i.e. make it as complete as mod_say_en and sounds_en_callie).

I don't know if someone from the community could perform this adaptation or we should directly ask the Professional FreeSWITCH Consulting Services (CC'ing them).

Regards,
Anatoli

From: Miguel Jesús López Valverde
Sent: Thursday, December 14, 2017 08:53
To: 'Freeswitch Users Help', Consulting
Cc: Me
Subject: RE: [Freeswitch-users] Problem with personalized locutions and use of say with type shor_date_time.

 

Hello Anatoli, thank you very much for your answer.

 

                At this time, the sets of locutions in Neutral-Spanish that I have available have been obtained with a TTS only with laboratory interest. I do not consider them suitable for a more professional or commercial use.

 

On the other hand, we are evaluating making recordings through some Spanish studies that have made us offers, based on the list available by default of locutions in English. For your information, the prices are approx. about 1800 Euros per group of locutions, (initially we would only make a group in female voice) considering that the studies that have offered us maintain a high level of quality in our point of view, since they have already done other work for us with optimal results .

 

Please, indicate if you have a more complete list than the one used by us to facilitate these studies, (the list that we have handled has been the one of locutions in English given in the version 1.6.18), so that we can pass it to assess.

 

In case of getting to make the recordings of the locutions, we could assess some form of cooperation to facilitate them to the community, (shared payment, training agreements or certification courses, etc ..).

 

Please, indicate if you are interested because, in this case, it may also be of interest to us to contract the completion of the locutions.

 

Again, Thank you very much and receive greeting.

 

 

 

De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Anatoli
Enviado el: jueves, 14 de diciembre de 2017 3:02
Para: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> ; consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
Asunto: Re: [Freeswitch-users] Problem with personalized locutions and use of say with type shor_date_time.

 

Hi Miguel Jesús,

That's because mod_say_es doesn't implement all say types, but most of the TTS configs use them (e.g. lang/en/vm/sounds.xml:396 <action function="say" data="$1" method="pronounced" type="short_date_time"/>). So, if you use as the base the English configs (because the Spanish base lacks voicemail or conference configs), mod_say_es won't work.

The Spanish sounds/prompts are broken. You may try to use mod_say_es_ar which is based on the latest mod_say_en, but it has some issues with some prompts anyway (though easier to solve). Basically, the problem is that the mod_say_es is outdated, the TTS configs are not in sync with it and the available Spanish prompts (sounds_es_mx_maria and sounds_es_ar_mario, I couldn't find others) are also not in sync with the configs and the say modules, and are incomplete (e.g. no "un" sound for "usted tiene un mensaje nuevo" ("you have one new message"), some words lack singular or plural forms, some prompts are completely absent (e.g. "entered the conference"), etc.).

I had to mix sounds from the 2 sets (MX and AR) and even borrow some sounds from the asterisk sounds. Even this way the result is not great (masculine and feminine voices mixed in the same sentence, lack of singular/plural forms for some words, etc.), but at least it works and the users can interact with the system.

The Spanish support in FS is not production ready. What should be done?

1.	mod_say_es should be updated to be in sync and as complete as the rest of the say modules (should be based on mod_say_en as this is the reference). And there is NO need for country-specific Spanish *say modules*, i.e. the grammar (the sentence composition) is the same in all regions. Only the prompts should be recorded by native speakers of each country.
2.	The Spanish TTS configs should be updated accordingly (again, NO country-specific configs).
3.	The list of necessary sounds (i.e. the prompts (texts) to country-adapt and record) should be defined (either in English or in neutral Spanish).
4.	A country-neutral set of Spanish sounds should be created.

Once this is ready, the community could provide the prompts for country-specific dialects. And I'm ready to co-sponsor the Spanish base (the 4 items list above) and to provide the correct es_AR prompts. If someone can implement the Spanish base (at least the first 3 items above), please let us know your price, I guess we could crowd-fund it.

Regards,
Anatoli

From: Miguel Jesús López Valverde
Sent: Wednesday, December 13, 2017 14:51
To: Freeswitch-users
Subject: [Freeswitch-users] Problem with personalized locutions and use of say with type shor_date_time.

 

Hello Freeswitch users:

 

I send this email to you with a new problem found.

 

I have a set of complete locutions created with a TTS for Spanish locutions installed in Freeswitch, along with those brought by default in English.

 

When using the phrases in Spanish, I am only having problems when using the directive:

<action function = "say" data = "$ 1" method = "pronounced" type = "short_date_time" />

but with other directives "say" with different methods, I'm not having any problems. For example, when accessing voicemail using these locutions set in Spanish, I get the following error that prevents access to recorded messages:

 

2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-new.wav] (es_CB:es)

2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:53.172924 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-new.wav

2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-message_number.wav] (es_CB:es)

2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:54.832899 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-message_number.wav

2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1] (es_CB:es)

2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 16:58:55.872903 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/es/colabora/sonia/digits/1.wav

2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [es_CB]

2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1513180033] (es_CB:es)

2017-12-13 16:58:55.972959 [ERR] mod_say_es.c:471 Unknown Say type=[18]

 

By changing this language by default English, it works correctly ...

 

2017-12-13 17:18:05.327117 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en]

2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-new.wav] (en:en)

2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:05.667117 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav

2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-message_number.wav] (en:en)

2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:06.587140 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-message_number.wav

2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1] (en:en)

2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:07.407128 [DEBUG] switch_ivr_play_say.c:1942 done playing file file_string://digits/1.wav

2017-12-13 17:18:07.507133 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en]

2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1513180033] (en:en)

2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

2017-12-13 17:18:10.427138 [DEBUG] switch_ivr_play_say.c:1942 done playing file file_string://time/today.wav!time/at.wav!digits/3.wav!digits/40.wav!digits/7.wav!time/p-m.wav

2017-12-13 17:18:10.547147 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms

 

It is proven that I have all the locutions in Spanish, including those in the "time" and "digits" folders.

 

Does anyone know how I can get a proper voicemail operation with personalized locutions when using the function "say" with "type =" short_date_time "" ?.

 

Thank you very much and best regards.

 

 


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Libre de virus.  <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> www.avast.com 







_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com
 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

 



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From mjlopez at smartic.es  Fri Dec 15 17:32:33 2017
From: mjlopez at smartic.es (=?iso-8859-1?Q?Miguel_Jes=FAs_L=F3pez_Valverde?=)
Date: Fri, 15 Dec 2017 18:32:33 +0100
Subject: [Freeswitch-users] how to block requests with From Ip equal to
	server interface IP?
Message-ID: <054601d375ca$b1af84f0$150e8ed0$@smartic.es>

Good afternoon everyone

 

I get a new query regarding a type of attack that our freeswitch servers
receive constantly in case someone knows how to block them.

 

These are INVITE or REGISTER requests in which the FROM: field arrives with
the ip and port equal to the public interface of the server, so the
different protection options that I have tried have not blocked these
requests:

 

- IpTables can not filter by the information From the INVITE message.

- Fail2Ban is equally limited than IpTables.

- ACLs have not resolved to filter these requests.

 

Does anyone know any way to block these requests?

 

I send here a trace with an INVITE message where you can see a request of
this type.

 

Thanks and best regards.

 

U 2017/12/14 18:32:55.156886 185.107.94.121:11120 -> 182.30.1.194:5060

INVITE sip:390239297988@ 182.30.1.194:5060;transport=UDP SIP/2.0.

Via: SIP/2.0/UDP
122.221.117.131:5060;branch=z9hG4bK-524287-1---xi3qy2kz737ni404.

Max-Forwards: 70.

Contact: <sip:15714000000 at 122.221.117.131:5060;transport=UDP>.

To: <sip:390239297988@ 182.30.1.194;transport=UDP>.

From: <sip:15714000000@ 182.30.1.194;transport=UDP>;tag=hlzg2jcv.

Call-ID: KaQqH51mAcFv34qN8cGyv3...

CSeq: 1 INVITE.

Content-Type: application/sdp.

User-Agent: Z 3.14.38765 rv2.8.3.

Allow-Events: presence, kpml, talk.

Content-Length: 0.

.

 



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From lists at telefaks.de  Fri Dec 15 17:45:56 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Fri, 15 Dec 2017 18:45:56 +0100
Subject: [Freeswitch-users] how to block requests with From Ip equal to
 server interface IP?
In-Reply-To: <054601d375ca$b1af84f0$150e8ed0$@smartic.es>
References: <054601d375ca$b1af84f0$150e8ed0$@smartic.es>
Message-ID: <5A340A54.6050506@telefaks.de>

Hello Miguel,

see here
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-April/071796.html
You will need to change the line
search="friendly-scanner"
to
search="Z 3.14.38765 rv2.8.3"

This worked for me.
Best regards Peter


On 12/15/17 18:32, Miguel Jesús López Valverde wrote:
>
> Good afternoon everyone
>
>  
>
> I get a new query regarding a type of attack that our freeswitch
> servers receive constantly in case someone knows how to block them.
>
>  
>
> These are INVITE or REGISTER requests in which the FROM: field arrives
> with the ip and port equal to the public interface of the server, so
> the different protection options that I have tried have not blocked
> these requests:
>
>  
>
> - IpTables can not filter by the information From the INVITE message.
>
> - Fail2Ban is equally limited than IpTables.
>
> - ACLs have not resolved to filter these requests.
>
>  
>
> Does anyone know any way to block these requests?
>
>  
>
> I send here a trace with an INVITE message where you can see a request
> of this type.
>
>  
>
> Thanks and best regards.
>
>  
>
> U 2017/12/14 18:32:55.156886 185.107.94.121:11120 -> 182.30.1.194:5060
>
> INVITE sip:390239297988@ 182.30.1.194:5060;transport=UDP SIP/2.0.
>
> Via: SIP/2.0/UDP
> 122.221.117.131:5060;branch=z9hG4bK-524287-1---xi3qy2kz737ni404.
>
> Max-Forwards: 70.
>
> Contact: <sip:15714000000 at 122.221.117.131:5060;transport=UDP>.
>
> To: <sip:390239297988@ 182.30.1.194;transport=UDP>.
>
> From: <sip:15714000000@ 182.30.1.194;transport=UDP>;tag=hlzg2jcv.
>
> Call-ID: KaQqH51mAcFv34qN8cGyv3...
>
> CSeq: 1 INVITE.
>
> Content-Type: application/sdp.
>
> User-Agent: Z 3.14.38765 rv2.8.3.
>
> Allow-Events: presence, kpml, talk.
>
> Content-Length: 0.
>
> .
>
>  
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient>
> 	Libre de virus. www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient>
>
>
> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

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From koki.roul at gmail.com  Fri Dec 15 21:11:52 2017
From: koki.roul at gmail.com (Lyubo Popov)
Date: Fri, 15 Dec 2017 19:11:52 -0200
Subject: [Freeswitch-users] How to reject a call and not log it in CDRs that
 contains unsupported ANCII characters?
Message-ID: <CAMSNS4ifnUuKqYG_4VMJrcnSvo1F4cxyphpZLc-Z9-UqKhh0QQ@mail.gmail.com>

Hello all,

Maybe someone can help me with this problem and will be greatly
appreciated. We are getting calls with CallerID like this one  ‘hi'or‘x’='x'.
Later when our billing start parsing the CDRs it will complain because of
the first character "`". My question I suppose is, how to prevent such
calls to get added to the CDRs? We want to reject the call that has non
numeric CallerID and not get it added in the CDRs. This is what we have in
the dialplan.

<extension name="Routing">
        <condition field="${radius_auth_result}" expression="0"/>

        <!--
        <condition field="${h323-redirect-number}" expression="^(.+)$"
break="never">
            <action application="set" data="destination_number=$1" />
        </condition>
        -->
        <condition field="caller_id_number" expression="^([0-9]+)$">
            <anti-action application="hangup"/>
</condition>
        <condition field="destination_number" expression="^(.+)$">
            <!--<action application="info"/>-->
            <action application="export"
data="nolocal:h323-call-origin=originate"/>
            <action application="set"
data="sip_h_X-accountcode=${accountcode}" />
            <action application="set" data="call_direction=outbound" />
            <action application="set" data="hangup_after_bridge=true"/>
            <action application="set" data="continue_on_fail=true"/>
            <action application="set" data="inherit_codec=true" />
            <action application="set" data="call_timeout=20"/>
            <action application="set"
data="fail_on_single_reject=USER_BUSY" />
            <action application="set"
data="origination_caller_id_name=${sip_req_user}"/>
            <action application="set"
data="origination_caller_id_number=${sip_from_user}"/>
            <action application="set" data="execute_on_answer=sched_hangup
+${h323-credit-time} alloted_timeout" />
            <action application="bridge"
data="{sip_invite_from_uri=sip:${sip_from_user}@
${sip_network_ip}}sofia/internal/${destination_number}@x.x.x.x:5060" />
            <action application="hangup" data="${bridge_hangup_cause}"/>
        </condition>
    </extension>

Thank you all!

L.Popov

<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>
Virus-free.
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From s.safarov at gmail.com  Sat Dec 16 05:38:20 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 16 Dec 2017 05:38:20 +0000
Subject: [Freeswitch-users] how to block requests with From Ip equal to
 server interface IP?
In-Reply-To: <5A340A54.6050506@telefaks.de>
References: <054601d375ca$b1af84f0$150e8ed0$@smartic.es>
 <5A340A54.6050506@telefaks.de>
Message-ID: <CAHtxdDd-tihN5LDnsxbWUY_uPaqJxj8jy24=7jK+GMrhyYmEhw@mail.gmail.com>

if you use domain names, then you can place FreeSwitch behind Kamailio and
implement filter on Kamailio side.
Filter logic:
1) if To header like \d+\.\d+\.\d+\.\d+ then drop packet;
2) if From header like \d+\.\d+\.\d+\.\d+ then drop packet;

Example of such filter logic you can find here
https://github.com/2600hz/kazoo-configs-kamailio/blob/master/kamailio/traffic-filter-role.cfg


пт, 15 дек. 2017 г. в 20:46, Peter Steinbach <lists at telefaks.de>:

> Hello Miguel,
>
> see here
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-April/071796.html
> You will need to change the line
> search="friendly-scanner"
> to
> search="Z 3.14.38765 rv2.8.3"
>
> This worked for me.
> Best regards Peter
>
>
>
> On 12/15/17 18:32, Miguel Jesús López Valverde wrote:
>
> Good afternoon everyone
>
>
>
> I get a new query regarding a type of attack that our freeswitch servers
> receive constantly in case someone knows how to block them.
>
>
>
> These are INVITE or REGISTER requests in which the FROM: field arrives
> with the ip and port equal to the public interface of the server, so the
> different protection options that I have tried have not blocked these
> requests:
>
>
>
> - IpTables can not filter by the information From the INVITE message.
>
> - Fail2Ban is equally limited than IpTables.
>
> - ACLs have not resolved to filter these requests.
>
>
>
> Does anyone know any way to block these requests?
>
>
>
> I send here a trace with an INVITE message where you can see a request of
> this type.
>
>
>
> Thanks and best regards.
>
>
>
> U 2017/12/14 18:32:55.156886 185.107.94.121:11120 -> 182.30.1.194:5060
>
> INVITE sip:390239297988@ 182.30.1.194:5060;transport=UDP SIP/2.0.
>
> Via: SIP/2.0/UDP 122.221.117.131:5060
> ;branch=z9hG4bK-524287-1---xi3qy2kz737ni404.
>
> Max-Forwards: 70.
>
> Contact: <sip:15714000000 at 122.221.117.131:5060;transport=UDP>
> <sip:15714000000 at 122.221.117.131:5060;transport=UDP>.
>
> To: <sip:390239297988@ 182.30.1.194;transport=UDP>.
>
> From: <sip:15714000000@ 182.30.1.194;transport=UDP>
> <sip:15714000000 at 182.30.1.194;transport=UDP>;tag=hlzg2jcv.
>
> Call-ID: KaQqH51mAcFv34qN8cGyv3...
>
> CSeq: 1 INVITE.
>
> Content-Type: application/sdp.
>
> User-Agent: Z 3.14.38765 rv2.8.3.
>
> Allow-Events: presence, kpml, talk.
>
> Content-Length: 0.
>
> .
>
>
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> Libre
> de virus. www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient>
> <#m_-1760520282224978735_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From steveayre at gmail.com  Sun Dec 17 22:21:25 2017
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 17 Dec 2017 22:21:25 +0000
Subject: [Freeswitch-users] how to block requests with From Ip equal to
 server interface IP?
In-Reply-To: <054601d375ca$b1af84f0$150e8ed0$@smartic.es>
References: <054601d375ca$b1af84f0$150e8ed0$@smartic.es>
Message-ID: <CAFiqYunaa+MoDwrvv200TJuV6EG=BEPCG-1n1NLy8H0BQ-g3uA@mail.gmail.com>

Having your server IP in the From header is not necessarily incorrect. They
aren't necessarily saying that they are sending from that IP, it could be
them saying they are a user registered on your server. I'd block it based
on something else such as user-agent or fail2ban.


On 15 December 2017 at 17:32, Miguel Jesús López Valverde <
mjlopez at smartic.es> wrote:

> Good afternoon everyone
>
>
>
> I get a new query regarding a type of attack that our freeswitch servers
> receive constantly in case someone knows how to block them.
>
>
>
> These are INVITE or REGISTER requests in which the FROM: field arrives
> with the ip and port equal to the public interface of the server, so the
> different protection options that I have tried have not blocked these
> requests:
>
>
>
> - IpTables can not filter by the information From the INVITE message.
>
> - Fail2Ban is equally limited than IpTables.
>
> - ACLs have not resolved to filter these requests.
>
>
>
> Does anyone know any way to block these requests?
>
>
>
> I send here a trace with an INVITE message where you can see a request of
> this type.
>
>
>
> Thanks and best regards.
>
>
>
> U 2017/12/14 18:32:55.156886 185.107.94.121:11120 -> 182.30.1.194:5060
>
> INVITE sip:390239297988@ 182.30.1.194:5060;transport=UDP SIP/2.0.
>
> Via: SIP/2.0/UDP 122.221.117.131:5060;branch=z9hG4bK-524287-1---
> xi3qy2kz737ni404.
>
> Max-Forwards: 70.
>
> Contact: <sip:15714000000 <(571)%20400-0000>@122.221.117.
> 131:5060;transport=UDP>.
>
> To: <sip:390239297988@ 182.30.1.194;transport=UDP>.
>
> From: <sip:15714000000 <(571)%20400-0000>@ 182.30.1.194;transport=UDP>;
> tag=hlzg2jcv.
>
> Call-ID: KaQqH51mAcFv34qN8cGyv3...
>
> CSeq: 1 INVITE.
>
> Content-Type: application/sdp.
>
> User-Agent: Z 3.14.38765 rv2.8.3.
>
> Allow-Events: presence, kpml, talk.
>
> Content-Length: 0.
>
> .
>
>
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> Libre
> de virus. www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient>
> <#m_-8919491363007729199_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mmeek at livexchange.com  Mon Dec 18 07:39:37 2017
From: mmeek at livexchange.com (Matthew Meek)
Date: Mon, 18 Dec 2017 07:39:37 +0000
Subject: [Freeswitch-users] bridge_generate_comfort_noise not honoring
 integer value for volume level
In-Reply-To: <CAKbxfG_9Qr+wZo1nqte1zjt=PDeRgsnaB+vv=WzoNut+sRxsVA@mail.gmail.com>
References: <MWHPR11MB1486A0000B461F759B86A5CEB60A0@MWHPR11MB1486.namprd11.prod.outlook.com>
 <CAKbxfG_9Qr+wZo1nqte1zjt=PDeRgsnaB+vv=WzoNut+sRxsVA@mail.gmail.com>
Message-ID: <MWHPR11MB148608B3A839FAA35140F0E2B60E0@MWHPR11MB1486.namprd11.prod.outlook.com>

The values of -2,-1  50, 1400, and 4000 all have the same volume of whitenoise.  I am setting it with:

<action application="set" data="bridge_generate_comfort_noise=4000"/>

I can see in the code where this should work (although -1 is doubtful to be silence as the if block checks for silence_val < -1 instead of <= -1), but in practice the value has no effect. I am not sure if the value is over-ridden somewhere else.

-Matthew Meek


The higher the number the softer it will get.  You say 4000 didn't work?  -1 should be absolute silence.

On Thu, Dec 14, 2017 at 6:01 AM, Matthew Meek <mmeek at livexchange.com<mailto:mmeek at livexchange.com>> wrote:

I can successfully inject whitenoise on the bridged leg (audio from A leg towards the bridged call) in the dial plan by setting bridge_generate_comfort_noise=true. The whitenoise is quite loud and I would like to make it quieter or silent (I am just trying to keep the far end firewall port open during long periods of A leg not sending packets due to hold).  I have tried values of -1, 40,1400, and 4000 instead of true and they produce the same amount of whitenoise volume. The source code in switch_ivr_bridge.c looks like it indeed honors an integer value, but somehow the end result is always the same. If I do not set the variable I get no whitenoise, so the variable is indeed controlling whitenoise, just not the volume level of whitenoise. The codec is PCMU.



I can repeat this with 1.2, 1.4, and 1.6 branches of FS.



Matthew Meek

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II
Founder, FreeSWITCH.


https://youtu.be/l_hOxzCt6X4
https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From mjlopez at smartic.es  Mon Dec 18 13:03:49 2017
From: mjlopez at smartic.es (=?UTF-8?Q?Miguel_Jes=C3=BAs_L=C3=B3pez_Valverde?=)
Date: Mon, 18 Dec 2017 14:03:49 +0100
Subject: [Freeswitch-users] how to block requests with From Ip equal to
	server interface IP?
In-Reply-To: <5A340A54.6050506@telefaks.de>
References: <054601d375ca$b1af84f0$150e8ed0$@smartic.es>
 <5A340A54.6050506@telefaks.de>
Message-ID: <04ed01d37800$a6d11610$f4734230$@smartic.es>

Thank you very much, Peter!.

 

 

De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Peter Steinbach
Enviado el: viernes, 15 de diciembre de 2017 18:46
Para: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Asunto: Re: [Freeswitch-users] how to block requests with From Ip equal to server interface IP?

 

Hello Miguel,

see here
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-April/071796.html
You will need to change the line 
search="friendly-scanner"
to
search="Z 3.14.38765 rv2.8.3"

This worked for me.
Best regards Peter


On 12/15/17 18:32, Miguel Jesús López Valverde wrote:

Good afternoon everyone

 

I get a new query regarding a type of attack that our freeswitch servers receive constantly in case someone knows how to block them.

 

These are INVITE or REGISTER requests in which the FROM: field arrives with the ip and port equal to the public interface of the server, so the different protection options that I have tried have not blocked these requests:

 

- IpTables can not filter by the information From the INVITE message.

- Fail2Ban is equally limited than IpTables.

- ACLs have not resolved to filter these requests.

 

Does anyone know any way to block these requests?

 

I send here a trace with an INVITE message where you can see a request of this type.

 

Thanks and best regards.

 

U 2017/12/14 18:32:55.156886 185.107.94.121:11120 -> 182.30.1.194:5060

INVITE sip:390239297988@ 182.30.1.194:5060;transport=UDP SIP/2.0.

Via: SIP/2.0/UDP 122.221.117.131:5060;branch=z9hG4bK-524287-1---xi3qy2kz737ni404.

Max-Forwards: 70.

Contact:  <mailto:sip:15714000000 at 122.221.117.131:5060;transport=UDP> <sip:15714000000 at 122.221.117.131:5060;transport=UDP>.

To: <sip:390239297988@ 182.30.1.194;transport=UDP <sip:390239297988@%20182.30.1.194;transport=UDP> >.

From:  <mailto:sip:15714000000 at 182.30.1.194;transport=UDP> <sip:15714000000@ 182.30.1.194;transport=UDP>;tag=hlzg2jcv.

Call-ID: KaQqH51mAcFv34qN8cGyv3...

CSeq: 1 INVITE.

Content-Type: application/sdp.

User-Agent: Z 3.14.38765 rv2.8.3.

Allow-Events: presence, kpml, talk.

Content-Length: 0.

.

 

 


 <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> 

Libre de virus.  <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> www.avast.com 






_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com
 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org






-- 
With kind regards
Peter Steinbach 
 
Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de <http://www.telefaks.de> 
 


---
El software de antivirus Avast ha analizado este correo electrónico en busca de virus.
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From allen at praecom.com  Sun Dec 17 13:41:17 2017
From: allen at praecom.com (=?utf-8?B?YWxsZW5AcHJhZWNvbS5jb20=?=)
Date: Sun, 17 Dec 2017 07:41:17 -0600
Subject: [Freeswitch-users]
	=?utf-8?q?FreeSWITCH-users_Digest=2C_Vol_138?=
	=?utf-8?q?=2C_Issue39?=
Message-ID: <1689589622-20590@mail.praecom.com>

You will probably have to set up media resources (medpro / media server)  as well. 

Sent from my HTC

----- Reply message -----
From: freeswitch-users-request at lists.freeswitch.org
To: <freeswitch-users at lists.freeswitch.org>
Subject: FreeSWITCH-users Digest, Vol 138, Issue39
Date: Wed, Dec 13, 2017 12:12 PM

Send FreeSWITCH-users mailing list submissions to
freeswitch-users at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
freeswitch-users-request at lists.freeswitch.org

You can reach the person managing the list at
freeswitch-users-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."


Today's Topics:

1. Re: [EXTERNAL] Re:  FS connecting to Avaya
(Karanath sachidanandan, Sanooj)


----------------------------------------------------------------------

Message: 1
Date: Wed, 13 Dec 2017 18:11:49 +0000
From: "Karanath sachidanandan, Sanooj"
<SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] [EXTERNAL] Re:  FS connecting to Avaya
Message-ID: <77695D16-34C1-4D3F-8946-6363C0F56820 at homedepot.com>
Content-Type: text/plain; charset="utf-8"

Thanks Tom for pointers , we are looking more to peer with CM . Will look more into the direction you suggest .
AES option was to connect to CM using JTAPI , but couldn’t make much progress in that dirction

Regards
Sanooj


From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Tom Lynn <tom at tomlynn.com>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: Sunday, December 10, 2017 at 7:32 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [EXTERNAL] Re: [Freeswitch-users] FS connecting to Avaya

Are you planning to peer directly to Communication Manager or using Avaya Session Manager in between?  It can be done both ways.  If the requirement is to transact only with Communication Manager a direct connection may work out.

In that case I'd suggest you'll need to add a node-name (which is like a host table entry) for FS and a signaling group of type SIP, along with a trunk group whose channels are controlled by said signaling group.  Assume TCP signaling to begin with and pay attention to your port numbers.  Some may already be used by other signaling groups.

Codecs are assigned based on a network-region construct which is used to model your voip network against your physical topology, supporting call admission control and where DSP channels are sourced.  Since Avaya doesn't use it's server host to mix audio, you'll have to know that the network region you choose either has DSPs or a logical path to a region which does is available.  The network region is assigned to the SIP signaling group in a field called Far End Network region.

To point calls at the trunk group you'll have to set up a route pattern and then place a dial string match in either an ARS analysis table (PSTN) or an AAR analysis table (private network).

How are you thinking about using AES?  I don't see an easy interface, but if you program TSAPI, it may be useful to get call events from the Avaya side.

Tom Lynn


On Fri, Dec 8, 2017 at 2:12 PM, Karanath sachidanandan, Sanooj <SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com<mailto:SANOOJ_KARANATH_SACHIDANANDAN at homedepot.com>> wrote:
Hi All,
Is there any documentation on connecting FS to AVAYA CM/AES ? or any pointers ?

Regards
Sanooj


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------------------------------

Subject: Digest Footer

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From richard.screene at thisisdrum.com  Fri Dec 15 13:51:34 2017
From: richard.screene at thisisdrum.com (Richard Screene)
Date: Fri, 15 Dec 2017 13:51:34 +0000
Subject: [Freeswitch-users] Play MP4 file into conference
Message-ID: <A0336BF3-F34D-414C-8AD1-14443E39D925@thisisdrum.com>

I am having problems attempting to play a MP4 video into a conference using mod_av.

I have built the 1.6.19 branch from source after removing the -pedantic compile flag to allow mod_av to compile.

My simple dial plan is:
```
    <extension name="test">
      <condition field="destination_number" expression="test">
        <action application="conference" data="conf1"/>
      </condition>
    </extension>
```

From fs_cli I then attempt to play the video using the command: "conference conf1 play /tmp/sample.mp4"

No error messages are visible in the console and I can hear the audio, but I do not see the video stream.

Strangely, when I do "uuid_broadcast <uuid> /tmp/sample.mp4" I get the audio and video (but obviously only to a single caller)

If I try on the code from the master branch then I do not video from either conference..play or uuid_broadcast.

Has anyone got any ideas why I cannot play video into a conference?

Also, is there any equivalent of uuid_fileman for files played into a conference?  I would like pause and seek functionality to affect the video of all participants in the conference.

Many thanks,
  Richard
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From kathleen at freeswitch.com  Mon Dec 18 19:07:41 2017
From: kathleen at freeswitch.com (Kathleen King)
Date: Mon, 18 Dec 2017 11:07:41 -0800
Subject: [Freeswitch-users] how to block requests with From Ip equal to
 server interface IP?
In-Reply-To: <054601d375ca$b1af84f0$150e8ed0$@smartic.es>
References: <054601d375ca$b1af84f0$150e8ed0$@smartic.es>
Message-ID: <CAOStUPZBhT8SNK90KnKYBRt5Za3_xvVL0xjOEoshj=4aB_S+-w@mail.gmail.com>

Hello,

We are going to have Brian West answer your question on the ClueCon weekly
call this week in our community corner. If you would like to join us live
on Wednesday at noon central time you can dial 888 at
https://conference.freeswitch.org/vc/ or watch it live on Youtube here:
https://youtu.be/F4OkiQ_okQI


Please let me know if you have any other questions.

[image: freeswitch logo giant.jpg] <https://freeswitch.com>

Kathleen King | Public Relations / Administrative Assistant

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045

Email: Kathleen at freeswitch.com

Mobile: 703-859-3757

Website: https://www.FreeSWITCH.com <https://www.freeswitch.com>

[image: color-facebook-96.png] <https://www.facebook.com/freeswitch/> [image:
color-twitter-96.png]
<https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>

On Fri, Dec 15, 2017 at 9:32 AM, Miguel Jesús López Valverde <
mjlopez at smartic.es> wrote:

> Good afternoon everyone
>
>
>
> I get a new query regarding a type of attack that our freeswitch servers
> receive constantly in case someone knows how to block them.
>
>
>
> These are INVITE or REGISTER requests in which the FROM: field arrives
> with the ip and port equal to the public interface of the server, so the
> different protection options that I have tried have not blocked these
> requests:
>
>
>
> - IpTables can not filter by the information From the INVITE message.
>
> - Fail2Ban is equally limited than IpTables.
>
> - ACLs have not resolved to filter these requests.
>
>
>
> Does anyone know any way to block these requests?
>
>
>
> I send here a trace with an INVITE message where you can see a request of
> this type.
>
>
>
> Thanks and best regards.
>
>
>
> U 2017/12/14 18:32:55.156886 185.107.94.121:11120 -> 182.30.1.194:5060
>
> INVITE sip:390239297988@ 182.30.1.194:5060;transport=UDP SIP/2.0.
>
> Via: SIP/2.0/UDP 122.221.117.131:5060;branch=z9hG4bK-524287-1---
> xi3qy2kz737ni404.
>
> Max-Forwards: 70.
>
> Contact: <sip:15714000000 <(571)%20400-0000>@122.221.117.
> 131:5060;transport=UDP>.
>
> To: <sip:390239297988@ 182.30.1.194;transport=UDP>.
>
> From: <sip:15714000000 <(571)%20400-0000>@ 182.30.1.194;transport=UDP>;
> tag=hlzg2jcv.
>
> Call-ID: KaQqH51mAcFv34qN8cGyv3...
>
> CSeq: 1 INVITE.
>
> Content-Type: application/sdp.
>
> User-Agent: Z 3.14.38765 rv2.8.3.
>
> Allow-Events: presence, kpml, talk.
>
> Content-Length: 0.
>
> .
>
>
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> Libre
> de virus. www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient>
> <#m_-638193062881438373_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Dec 18 20:44:11 2017
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Dec 2017 12:44:11 -0800
Subject: [Freeswitch-users] How to reject a call and not log it in CDRs
 that contains unsupported ANCII characters?
In-Reply-To: <CAMSNS4ifnUuKqYG_4VMJrcnSvo1F4cxyphpZLc-Z9-UqKhh0QQ@mail.gmail.com>
References: <CAMSNS4ifnUuKqYG_4VMJrcnSvo1F4cxyphpZLc-Z9-UqKhh0QQ@mail.gmail.com>
Message-ID: <CAKzWOxWbrNgrYD8X3Vv5dL8wDiebsguaR861qqTh1BF9+Q_hSQ@mail.gmail.com>

Hi Lyubo,

In this case it may be better to see if your CDR parser can skip the
non-numeric caller id values, perhaps by adding a validation check prior to
performing the parse action. As a rule of thumb, if your CDR parser can be
tripped up by the data it is parsing then it needs to be hardened. I'm sure
many here would highly recommend sanitizing/validation as a best practice,
particularly when handling data that comes from the public Internet.
Another consideration is that you may actually want to have a record of
these kinds of attacks in case there is a need to investigate an incident
or otherwise analyze attack patterns.

I would recommend that you change the behavior of the parser from
"complaining" to "keeping the CDR database clean but logging invalid input
for future reference."

Hope this helps,
-MC


On Fri, Dec 15, 2017 at 1:11 PM, Lyubo Popov <koki.roul at gmail.com> wrote:

> Hello all,
>
> Maybe someone can help me with this problem and will be greatly
> appreciated. We are getting calls with CallerID like this one  ‘hi'or‘x’='x'.
> Later when our billing start parsing the CDRs it will complain because of
> the first character "`". My question I suppose is, how to prevent such
> calls to get added to the CDRs? We want to reject the call that has non
> numeric CallerID and not get it added in the CDRs. This is what we have in
> the dialplan.
>
> <extension name="Routing">
>         <condition field="${radius_auth_result}" expression="0"/>
>
>         <!--
>         <condition field="${h323-redirect-number}" expression="^(.+)$"
> break="never">
>             <action application="set" data="destination_number=$1" />
>         </condition>
>         -->
>         <condition field="caller_id_number" expression="^([0-9]+)$">
>             <anti-action application="hangup"/>
> </condition>
>         <condition field="destination_number" expression="^(.+)$">
>             <!--<action application="info"/>-->
>             <action application="export" data="nolocal:h323-call-
> origin=originate"/>
>             <action application="set" data="sip_h_X-accountcode=${accountcode}"
> />
>             <action application="set" data="call_direction=outbound" />
>             <action application="set" data="hangup_after_bridge=true"/>
>             <action application="set" data="continue_on_fail=true"/>
>             <action application="set" data="inherit_codec=true" />
>             <action application="set" data="call_timeout=20"/>
>             <action application="set" data="fail_on_single_reject=USER_BUSY"
> />
>             <action application="set" data="origination_caller_id_
> name=${sip_req_user}"/>
>             <action application="set" data="origination_caller_id_
> number=${sip_from_user}"/>
>             <action application="set" data="execute_on_answer=sched_hangup
> +${h323-credit-time} alloted_timeout" />
>             <action application="bridge" data="{sip_invite_from_uri=
> sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${
> destination_number}@x.x.x.x:5060" />
>             <action application="hangup" data="${bridge_hangup_cause}"/>
>         </condition>
>     </extension>
>
> Thank you all!
>
> L.Popov
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon> Virus-free.
> www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
> <#m_-7626445720286026976_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From gregor at infomedia.si  Mon Dec 18 22:37:36 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Mon, 18 Dec 2017 23:37:36 +0100
Subject: [Freeswitch-users] How to reject a call and not log it in CDRs
 that contains unsupported ANCII characters?
In-Reply-To: <CAKzWOxWbrNgrYD8X3Vv5dL8wDiebsguaR861qqTh1BF9+Q_hSQ@mail.gmail.com>
References: <CAMSNS4ifnUuKqYG_4VMJrcnSvo1F4cxyphpZLc-Z9-UqKhh0QQ@mail.gmail.com>
 <CAKzWOxWbrNgrYD8X3Vv5dL8wDiebsguaR861qqTh1BF9+Q_hSQ@mail.gmail.com>
Message-ID: <CAC7ttFGjy9-LpWVKBo=kmKYUX3xn6aaznoB11W0awPzsv9gd=A@mail.gmail.com>

​Hi, Lyubo!

I had similar case. Use process_cdr variable in dialplan, if you do not
want to store in CDR.

<action application="set" data="process_cdr=
​false​
"/>

​Hope it helps, Gregor​


2017-12-18 21:44 GMT+01:00 Michael Collins <msc at freeswitch.org>:

> Hi Lyubo,
>
> In this case it may be better to see if your CDR parser can skip the
> non-numeric caller id values, perhaps by adding a validation check prior to
> performing the parse action. As a rule of thumb, if your CDR parser can be
> tripped up by the data it is parsing then it needs to be hardened. I'm sure
> many here would highly recommend sanitizing/validation as a best practice,
> particularly when handling data that comes from the public Internet.
> Another consideration is that you may actually want to have a record of
> these kinds of attacks in case there is a need to investigate an incident
> or otherwise analyze attack patterns.
>
> I would recommend that you change the behavior of the parser from
> "complaining" to "keeping the CDR database clean but logging invalid input
> for future reference."
>
> Hope this helps,
> -MC
>
>
> On Fri, Dec 15, 2017 at 1:11 PM, Lyubo Popov <koki.roul at gmail.com> wrote:
>
>> Hello all,
>>
>> Maybe someone can help me with this problem and will be greatly
>> appreciated. We are getting calls with CallerID like this one  ‘hi'or‘x’='x'.
>> Later when our billing start parsing the CDRs it will complain because of
>> the first character "`". My question I suppose is, how to prevent such
>> calls to get added to the CDRs? We want to reject the call that has non
>> numeric CallerID and not get it added in the CDRs. This is what we have in
>> the dialplan.
>>
>> <extension name="Routing">
>>         <condition field="${radius_auth_result}" expression="0"/>
>>
>>         <!--
>>         <condition field="${h323-redirect-number}" expression="^(.+)$"
>> break="never">
>>             <action application="set" data="destination_number=$1" />
>>         </condition>
>>         -->
>>         <condition field="caller_id_number" expression="^([0-9]+)$">
>>             <anti-action application="hangup"/>
>> </condition>
>>         <condition field="destination_number" expression="^(.+)$">
>>             <!--<action application="info"/>-->
>>             <action application="export" data="nolocal:h323-call-origin
>> =originate"/>
>>             <action application="set" data="sip_h_X-accountcode=${accountcode}"
>> />
>>             <action application="set" data="call_direction=outbound" />
>>             <action application="set" data="hangup_after_bridge=true"/>
>>             <action application="set" data="continue_on_fail=true"/>
>>             <action application="set" data="inherit_codec=true" />
>>             <action application="set" data="call_timeout=20"/>
>>             <action application="set" data="fail_on_single_reject=USER_BUSY"
>> />
>>             <action application="set" data="origination_caller_id_na
>> me=${sip_req_user}"/>
>>             <action application="set" data="origination_caller_id_nu
>> mber=${sip_from_user}"/>
>>             <action application="set" data="execute_on_answer=sched_hangup
>> +${h323-credit-time} alloted_timeout" />
>>             <action application="bridge" data="{sip_invite_from_uri=sip
>> :${sip_from_user}@${sip_network_ip}}sofia/internal/${destina
>> tion_number}@x.x.x.x:5060" />
>>             <action application="hangup" data="${bridge_hangup_cause}"/>
>>         </condition>
>>     </extension>
>>
>> Thank you all!
>>
>> L.Popov
>>
>>
>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon> Virus-free.
>> www.avast.com
>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
>> <#m_2040158889290273727_m_-7626445720286026976_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From alihaider.4189 at gmail.com  Tue Dec 19 00:09:13 2017
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Tue, 19 Dec 2017 05:09:13 +0500
Subject: [Freeswitch-users] Help me to install and run
Message-ID: <923ADAF6-30A4-40E2-B009-0CCF31566C91@gmail.com>

Hi Everyone 

Hiiii I need  help anyone help me how to install and run FreeSWITCH in Linux 16.04 Lts as well as Windows 
On windows I have already done steps till Visual studio soln when I build On visual studio 2013 it shows 182 error I can’t under stand how to remove it  it is file missing error but I don’t know how to add files in git
Sent from my iPhone

From joel at gogii.net  Tue Dec 19 01:38:14 2017
From: joel at gogii.net (Joel Serrano)
Date: Tue, 19 Dec 2017 01:38:14 +0000
Subject: [Freeswitch-users] Help me to install and run
In-Reply-To: <923ADAF6-30A4-40E2-B009-0CCF31566C91@gmail.com>
References: <923ADAF6-30A4-40E2-B009-0CCF31566C91@gmail.com>
Message-ID: <CAMtXxQn0njn+YxDCd9tEvHHcNo5SCgi1Eh+bTAyqJqsRhQc9jQ@mail.gmail.com>

Hi,

You have a complete walkthrough in the wiki..

For ubuntu:

https://freeswitch.org/confluence/pages/viewpage.action?pageId=10683647

And windows:

https://freeswitch.org/confluence/pages/viewpage.action?pageId=1966780


Have you followed those steps and run into any issues?

Joel.


On Mon, Dec 18, 2017 at 17:11 Ali Haider <alihaider.4189 at gmail.com> wrote:

> Hi Everyone
>
> Hiiii I need  help anyone help me how to install and run FreeSWITCH in
> Linux 16.04 Lts as well as Windows
> On windows I have already done steps till Visual studio soln when I build
> On visual studio 2013 it shows 182 error I can’t under stand how to remove
> it  it is file missing error but I don’t know how to add files in git
> Sent from my iPhone
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From lists at telefaks.de  Tue Dec 19 11:49:08 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 19 Dec 2017 12:49:08 +0100
Subject: [Freeswitch-users] MWI not sent
Message-ID: <5A38FCB4.5000105@telefaks.de>

I have a Yealink T46G registered at our Freeswitch machine. MWI is
enabled on the Yealink Phone, I see a "message-summary" entry in
sip_subscriptions. So I think, presence is setup fine

Content of the sip_subscriptions table:
"proto" "sip_user" "sip_host" "sub_to_user" "sub_to_host"
"presence_hosts" "event" "contact" "call_id" "full_from" "full_via"
"expires" "user_agent" "accept" "profile_name" "hostname" "network_port"
"network_ip" "version" "orig_proto" "full_to" "id"
"sip" "200" "fs00.my.domain" "200" "fs00.my.domain" "" "message-summary"
"""Peter3 Buero"" <sip:200 at 192.168.1.72:5060>"
"0_3056780279 at 192.168.1.72" """Peter3 Buero""
<sip:200 at fs00.my.domain:5060>;tag=1082071637" "SIP/2.0/UDP
192.168.1.72:5060;branch=z9hG4bK2245575863" "1513683832" "Yealink
SIP-T46G 28.80.0.130" "application/simple-message-summary " "internal"
"fs01.my.domain" "5060" "192.168.1.72" "-1" ""
"<sip:200 at fs00.my.domain:5060>;tag=WdayfEpssASN" "2821206"

Events are enabled on the freeswitch console
/events MESSAGE_QUERY MESSAGE_WAITING

When a voicemail is recorded, I see a number of recurring events on the
console (see below), but no NOTIFY message is sent to the phone.
However, after a while (after 10min or even after hours) the phone
receives its message.
When I delete all VM messages, the MWI is never sent to the phone,
although messages are prepared (MWI-Messages-Waiting: no), as I can see
them on the console.

If I set the parameter “send-message-query-on-register“ in internal
sofia profile, then MWI is updated with every register to the phone.
Alos on every restart of the phone. But as we cannot use short intervals
intervals due to some reason, this is not the solution.

Anybody has a hint, how to solve this?


Example messages:
RECV EVENT
Event-Name: MESSAGE_QUERY
Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
FreeSWITCH-Hostname: fs01.my.domain
FreeSWITCH-Switchname: fs01.my.domain
FreeSWITCH-IPv4: 192.168.1.9
FreeSWITCH-IPv6: ::1
Event-Date-Local: 2017-12-19 12:27:40
Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
Event-Date-Timestamp: 1513682860739342
Event-Calling-File: sofia_presence.c
Event-Calling-Function: sofia_presence_sub_reg_callback
Event-Calling-Line-Number: 1736
Event-Sequence: 16929
Message-Account: sip:200 at fs00.my.domain
VM-Sofia-Profile: internal
VM-sub-call-id: 0_3056780279 at 192.168.1.72


RECV EVENT
Event-Name: MESSAGE_WAITING
Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
FreeSWITCH-Hostname: fs01.my.domain
FreeSWITCH-Switchname: fs01.my.domain
FreeSWITCH-IPv4: 192.168.1.9
FreeSWITCH-IPv6: ::1
Event-Date-Local: 2017-12-19 12:27:40
Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
Event-Date-Timestamp: 1513682860899354
Event-Calling-File: mod_voicemail.c
Event-Calling-Function: actual_message_query_handler
Event-Calling-Line-Number: 4036
Event-Sequence: 16931
MWI-Messages-Waiting: yes
MWI-Message-Account: sip:200 at fs00.my.domain
MWI-Voice-Message: 1/0 (0/0)
Sofia-Profile: internal
sub-call-id: 0_3056780279 at 192.168.1.72

-- 
With kind regards
Peter Steinbach 



From lists at telefaks.de  Tue Dec 19 13:23:40 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 19 Dec 2017 14:23:40 +0100
Subject: [Freeswitch-users] Lose Race instead of missed call
In-Reply-To: <5A2EB2B5.9030907@telefaks.de>
References: <5A2EB2B5.9030907@telefaks.de>
Message-ID: <5A3912DC.10903@telefaks.de>

Hello.

I did some further testiumng and figured out, that as soon I use
enterprise originate with :_: , I do not see any missed calls. If I use
a comma instead, missed calls are shown on the phone. But then as a
drawback only one phone rings, if I have more than physical phone
registered to one user extension.

Anybody has an idea how to overcome this?

Best rgeards
Peter

On 12/11/17 17:30, Peter Steinbach wrote:
> Hello,
>
> we have some Problems with the right signalling of missed calls when
> calling multiple phones in parallel
>
> Here's the scenario:
> Phone no 49 is calling a group with 2170 and 3275 with the following
> dialstring
>
> <action application="bridge"
> data="{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59eee5e0-c0bc-0135-7dc9-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/2170 at flex.mydomain.de:_:{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59f07af0-c0bc-0135-7dca-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/3275 at flex.mydomain.de
> <mailto:customer_id=261%7Duser/3275 at flex.mydomain.de>" />
>
> Destination dialstrings are seperated by ":_:" ("Enterprise
> Origination"). We use curly brackets instead of "<" as we sometimes
> have to insert asserted identy tags into the dialstring.
>
> We checked 3 versions of Freeswitch for this
>
>   * Version  Feb 2016 shows missed calls on both phones. Even if one
>     phone answers, the other phone one still shows a missed call
>     (reason for upgrading to newer Freeswitch)
>   * Version  Aug 2017 never shows missed call, see logs and hangup
>     message below
>   * Version  10/Dec 2017(yesterday) never shows missed call, as above
>
> So for the 2 never Freeswitch Versions, here are the logs at hangup
> 2017-12-11 17:07:46.022450 [DEBUG] sofia.c:7283 Channel
> sofia/internal/49 at flex.mydomain.de:5060 entering state [terminated][487]
> 2017-12-11 17:07:46.022450 [NOTICE] sofia.c:8474 Hangup
> sofia/internal/*49*@flex.mydomain.de:5060 [CS_EXECUTE]
> *[ORIGINATOR_CANCEL]*
> 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup
> sofia/internal/*2170*@94.xx.xxx.xx:42170 [CS_CONSUME_MEDIA] *[LOSE_RACE]*
> 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852
> Originate Cancelled by originator termination Cause: 487
> [ORIGINATOR_CANCEL]
> 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot
> create outgoing channel of type [user] cause: [LOSE_RACE]
> 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863
> Originate Resulted in Error Cause: 502 [LOSE_RACE]
> 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:584
> (sofia/internal/2170 at 94.xx.xxx.xx:42170) Running State Change
> CS_HANGUP (Cur 3 Tot 22)
> 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:850
> (sofia/internal/2170 at 94.xx.xxx.xx:42170) Callstate Change RINGING ->
> HANGUP
> 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:852
> (sofia/internal/2170 at 94.xx.xxx.xx:42170) State HANGUP
> 2017-12-11 17:07:46.042331 [DEBUG] mod_sofia.c:449 Channel
> sofia/internal/2170 at 94.xx.xxx.xx:42170 hanging up, cause: LOSE_RACE
> 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627 Hangup
> sofia/internal/*3275*@94.xx.xxx.xx:43275 [CS_CONSUME_MEDIA] [*LOSE_RACE*]
> 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852
> Originate Cancelled by originator termination Cause: 487
> [ORIGINATOR_CANCEL]
> 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866 Cannot
> create outgoing channel of type [user] cause: [LOSE_RACE]
> 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863
> Originate Resulted in Error Cause: 502 [LOSE_RACE]
>
> Here is the Cancel message for one of the called phones:
> U 2017/12/11 17:07:46.045120 144.xx.xxx.xx:5060 -> 94.xx.xxx.xx:42170
> *CANCEL sip:2170*@94.xx.xxx.xx:42170 SIP/2.0.
> Via: SIP/2.0/UDP 144.xx.xxx.xx;rport;branch=z9hG4bK07r3ce94cvjpp.
> Max-Forwards: 70.
> From: "Test" <sip:49 at flex.mydomain.de>;tag=mQS1eFDpp1peS.
> To: <sip:2170 at 94.xx.xxx.xx:42170>.
> Call-ID: 4238b5a7-5930-1236-b2ab-00505600a1a5.
> CSeq: 116168167 CANCEL.
> *Reason: SIP;cause=200;text="Call completed elsewhere"*.
> Content-Length: 0.
>
> Any hints why this happens, or anyone has this scenario working?
>
> -- 
> With kind regards
> Peter Steinbach 
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

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From alihaider.4189 at gmail.com  Tue Dec 19 04:51:08 2017
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Tue, 19 Dec 2017 09:51:08 +0500
Subject: [Freeswitch-users] =?utf-8?q?=28no_subject=29?=
Message-ID: <F48F9953-F444-4549-9727-6B37704EA2CA@gmail.com>

Hiii jeol 
https://freeswitch.org/confluence/pages/viewpage.action?pageId=1966780
I’m followed these steps on windows but after installation of visual studio 2013 which was may be required 182 error is found for missing files 
Please tell me which version of visual studio is installed to build soln and how to remove error


Sent from my iPhone
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From Pascal.Hari at csa.ch  Tue Dec 19 12:58:16 2017
From: Pascal.Hari at csa.ch (Pascal Hari)
Date: Tue, 19 Dec 2017 12:58:16 +0000
Subject: [Freeswitch-users] mod_verto with standart firewall ports
Message-ID: <e4065ee653a54523a3a3b03dbfe508e5@CSASVEX001.csa.ch>

Hello

I'm new to FreeSWITCH and made a setup where everything is behind the same NAT. I have a linux debian 8 server where FreeSWITCH is running on and I'm able to connect different clients (SIP and WebRTC). I'm using the demonstration configuration which come with the FreeSWITCH installation. I mostly followed the instruction "Quick Start FreeSWITCH Demo With Verto Communicator<https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-Scriptinstallfreeswitchdemowithverto_comunicator>" but used apache2 instead of nginx.

Now I want that same setup but on a server in the web with public IP. I'm running a linux debian 8 in azure and installed everything accordingly to the "behind the Nat" setup.
I use real (trusted) certificates and can reach the WebRTC client over https. If I'm in the public network of our company I can log in, dial a number and connect to FreeSWITCH (according to the dialplan). If I'm in the corp (internal) network of our company I cannot log in. I only get the message "Waiting for server reconnection". Even in FreeSWITCH (fs_cli) I don't see a request.
For me this sound like a firewall issue and many post say to open ports (16384-32768, 5066, ...). I don't have this option because I can't change our companies firewall. And I'm anyway looking for a solution to provide a WebRTC client which can be used in a normal company network with relatively strict firewall rules. Workarounds like VPN-tunnel are also not a wanted solution because I want a really easy setup on the client (WebRTC) side.

Am I missing a step? Is my guess with the firewall right or could it be something else? Is there a way to configure FreeSWITCH and verto communicator to use other ports?
Thanks already for suggestions.

Best regards,

Pascal Hari
SW Developer

CSA Engineering AG
______________________________________________________________________________________________
Confidentiality Note: This message is intended only for the use of the named recipient(s) and may contain confidential and/or privileged information. If you are not the/an intended recipient, please contact the sender and delete this message. Any unauthorized use of the information contained in this message is prohibited.

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From mike at jerris.com  Tue Dec 19 15:09:23 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 Dec 2017 10:09:23 -0500
Subject: [Freeswitch-users] MWI not sent
In-Reply-To: <5A38FCB4.5000105@telefaks.de>
References: <5A38FCB4.5000105@telefaks.de>
Message-ID: <6EA801C8-F2B8-45DE-95B7-FA9CE9CD64B5@jerris.com>

if you do a voicemail to the user very soon after it registers (within seconds) does it work?

> On Dec 19, 2017, at 6:49 AM, Peter Steinbach <lists at telefaks.de> wrote:
> 
> I have a Yealink T46G registered at our Freeswitch machine. MWI is
> enabled on the Yealink Phone, I see a "message-summary" entry in
> sip_subscriptions. So I think, presence is setup fine
> 
> Content of the sip_subscriptions table:
> "proto" "sip_user" "sip_host" "sub_to_user" "sub_to_host"
> "presence_hosts" "event" "contact" "call_id" "full_from" "full_via"
> "expires" "user_agent" "accept" "profile_name" "hostname" "network_port"
> "network_ip" "version" "orig_proto" "full_to" "id"
> "sip" "200" "fs00.my.domain" "200" "fs00.my.domain" "" "message-summary"
> """Peter3 Buero"" <sip:200 at 192.168.1.72:5060>"
> "0_3056780279 at 192.168.1.72" """Peter3 Buero""
> <sip:200 at fs00.my.domain:5060>;tag=1082071637" "SIP/2.0/UDP
> 192.168.1.72:5060;branch=z9hG4bK2245575863" "1513683832" "Yealink
> SIP-T46G 28.80.0.130" "application/simple-message-summary " "internal"
> "fs01.my.domain" "5060" "192.168.1.72" "-1" ""
> "<sip:200 at fs00.my.domain:5060>;tag=WdayfEpssASN" "2821206"
> 
> Events are enabled on the freeswitch console
> /events MESSAGE_QUERY MESSAGE_WAITING
> 
> When a voicemail is recorded, I see a number of recurring events on the
> console (see below), but no NOTIFY message is sent to the phone.
> However, after a while (after 10min or even after hours) the phone
> receives its message.
> When I delete all VM messages, the MWI is never sent to the phone,
> although messages are prepared (MWI-Messages-Waiting: no), as I can see
> them on the console.
> 
> If I set the parameter “send-message-query-on-register“ in internal
> sofia profile, then MWI is updated with every register to the phone.
> Alos on every restart of the phone. But as we cannot use short intervals
> intervals due to some reason, this is not the solution.
> 
> Anybody has a hint, how to solve this?
> 
> 
> Example messages:
> RECV EVENT
> Event-Name: MESSAGE_QUERY
> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
> FreeSWITCH-Hostname: fs01.my.domain
> FreeSWITCH-Switchname: fs01.my.domain
> FreeSWITCH-IPv4: 192.168.1.9
> FreeSWITCH-IPv6: ::1
> Event-Date-Local: 2017-12-19 12:27:40
> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
> Event-Date-Timestamp: 1513682860739342
> Event-Calling-File: sofia_presence.c
> Event-Calling-Function: sofia_presence_sub_reg_callback
> Event-Calling-Line-Number: 1736
> Event-Sequence: 16929
> Message-Account: sip:200 at fs00.my.domain
> VM-Sofia-Profile: internal
> VM-sub-call-id: 0_3056780279 at 192.168.1.72
> 
> 
> RECV EVENT
> Event-Name: MESSAGE_WAITING
> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
> FreeSWITCH-Hostname: fs01.my.domain
> FreeSWITCH-Switchname: fs01.my.domain
> FreeSWITCH-IPv4: 192.168.1.9
> FreeSWITCH-IPv6: ::1
> Event-Date-Local: 2017-12-19 12:27:40
> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
> Event-Date-Timestamp: 1513682860899354
> Event-Calling-File: mod_voicemail.c
> Event-Calling-Function: actual_message_query_handler
> Event-Calling-Line-Number: 4036
> Event-Sequence: 16931
> MWI-Messages-Waiting: yes
> MWI-Message-Account: sip:200 at fs00.my.domain
> MWI-Voice-Message: 1/0 (0/0)
> Sofia-Profile: internal
> sub-call-id: 0_3056780279 at 192.168.1.72



From mike at jerris.com  Tue Dec 19 15:11:38 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 Dec 2017 10:11:38 -0500
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <F48F9953-F444-4549-9727-6B37704EA2CA@gmail.com>
References: <F48F9953-F444-4549-9727-6B37704EA2CA@gmail.com>
Message-ID: <41EDC8B7-312C-42FB-B4E5-D8C7351275B8@jerris.com>

looks like there is some out of date info on compilers supported there.  Try using the latest msvc compiler instead of older ones.

> On Dec 18, 2017, at 11:51 PM, Ali Haider <alihaider.4189 at gmail.com> wrote:
> 
> Hiii jeol 
> https://freeswitch.org/confluence/pages/viewpage.action?pageId=1966780 <https://freeswitch.org/confluence/pages/viewpage.action?pageId=1966780>
> I’m followed these steps on windows but after installation of visual studio 2013 which was may be required 182 error is found for missing files 
> Please tell me which version of visual studio is installed to build soln and how to remove error
> 
> 
> Sent from my iPhone
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From lists at telefaks.de  Tue Dec 19 15:26:12 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 19 Dec 2017 16:26:12 +0100
Subject: [Freeswitch-users] MWI not sent
In-Reply-To: <6EA801C8-F2B8-45DE-95B7-FA9CE9CD64B5@jerris.com>
References: <5A38FCB4.5000105@telefaks.de>
 <6EA801C8-F2B8-45DE-95B7-FA9CE9CD64B5@jerris.com>
Message-ID: <5A392F94.50002@telefaks.de>

Hello Michael,

with parameter “send-message-query-on-register“ in internal sofia profile it works during registration, without this parameter is does not.

Best regards
Peter



On 12/19/17 16:09, Michael Jerris wrote:
> if you do a voicemail to the user very soon after it registers (within seconds) does it work?
>
>> On Dec 19, 2017, at 6:49 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>
>> I have a Yealink T46G registered at our Freeswitch machine. MWI is
>> enabled on the Yealink Phone, I see a "message-summary" entry in
>> sip_subscriptions. So I think, presence is setup fine
>>
>> Content of the sip_subscriptions table:
>> "proto" "sip_user" "sip_host" "sub_to_user" "sub_to_host"
>> "presence_hosts" "event" "contact" "call_id" "full_from" "full_via"
>> "expires" "user_agent" "accept" "profile_name" "hostname" "network_port"
>> "network_ip" "version" "orig_proto" "full_to" "id"
>> "sip" "200" "fs00.my.domain" "200" "fs00.my.domain" "" "message-summary"
>> """Peter3 Buero"" <sip:200 at 192.168.1.72:5060>"
>> "0_3056780279 at 192.168.1.72" """Peter3 Buero""
>> <sip:200 at fs00.my.domain:5060>;tag=1082071637" "SIP/2.0/UDP
>> 192.168.1.72:5060;branch=z9hG4bK2245575863" "1513683832" "Yealink
>> SIP-T46G 28.80.0.130" "application/simple-message-summary " "internal"
>> "fs01.my.domain" "5060" "192.168.1.72" "-1" ""
>> "<sip:200 at fs00.my.domain:5060>;tag=WdayfEpssASN" "2821206"
>>
>> Events are enabled on the freeswitch console
>> /events MESSAGE_QUERY MESSAGE_WAITING
>>
>> When a voicemail is recorded, I see a number of recurring events on the
>> console (see below), but no NOTIFY message is sent to the phone.
>> However, after a while (after 10min or even after hours) the phone
>> receives its message.
>> When I delete all VM messages, the MWI is never sent to the phone,
>> although messages are prepared (MWI-Messages-Waiting: no), as I can see
>> them on the console.
>>
>> If I set the parameter “send-message-query-on-register“ in internal
>> sofia profile, then MWI is updated with every register to the phone.
>> Alos on every restart of the phone. But as we cannot use short intervals
>> intervals due to some reason, this is not the solution.
>>
>> Anybody has a hint, how to solve this?
>>
>>
>> Example messages:
>> RECV EVENT
>> Event-Name: MESSAGE_QUERY
>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>> FreeSWITCH-Hostname: fs01.my.domain
>> FreeSWITCH-Switchname: fs01.my.domain
>> FreeSWITCH-IPv4: 192.168.1.9
>> FreeSWITCH-IPv6: ::1
>> Event-Date-Local: 2017-12-19 12:27:40
>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>> Event-Date-Timestamp: 1513682860739342
>> Event-Calling-File: sofia_presence.c
>> Event-Calling-Function: sofia_presence_sub_reg_callback
>> Event-Calling-Line-Number: 1736
>> Event-Sequence: 16929
>> Message-Account: sip:200 at fs00.my.domain
>> VM-Sofia-Profile: internal
>> VM-sub-call-id: 0_3056780279 at 192.168.1.72
>>
>>
>> RECV EVENT
>> Event-Name: MESSAGE_WAITING
>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>> FreeSWITCH-Hostname: fs01.my.domain
>> FreeSWITCH-Switchname: fs01.my.domain
>> FreeSWITCH-IPv4: 192.168.1.9
>> FreeSWITCH-IPv6: ::1
>> Event-Date-Local: 2017-12-19 12:27:40
>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>> Event-Date-Timestamp: 1513682860899354
>> Event-Calling-File: mod_voicemail.c
>> Event-Calling-Function: actual_message_query_handler
>> Event-Calling-Line-Number: 4036
>> Event-Sequence: 16931
>> MWI-Messages-Waiting: yes
>> MWI-Message-Account: sip:200 at fs00.my.domain
>> MWI-Voice-Message: 1/0 (0/0)
>> Sofia-Profile: internal
>> sub-call-id: 0_3056780279 at 192.168.1.72
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de



From mike at jerris.com  Tue Dec 19 15:37:52 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 Dec 2017 10:37:52 -0500
Subject: [Freeswitch-users] MWI not sent
In-Reply-To: <5A392F94.50002@telefaks.de>
References: <5A38FCB4.5000105@telefaks.de>
 <6EA801C8-F2B8-45DE-95B7-FA9CE9CD64B5@jerris.com>
 <5A392F94.50002@telefaks.de>
Message-ID: <07BAB46B-B5DC-4F5D-9F5A-2CEC2932167B@jerris.com>

without that setting, when sending a voicemail to the user right after they register?

> On Dec 19, 2017, at 10:26 AM, Peter Steinbach <lists at telefaks.de> wrote:
> 
> Hello Michael,
> 
> with parameter “send-message-query-on-register“ in internal sofia profile it works during registration, without this parameter is does not.
> 
> Best regards
> Peter
> 
> 
> 
> On 12/19/17 16:09, Michael Jerris wrote:
>> if you do a voicemail to the user very soon after it registers (within seconds) does it work?
>> 
>>> On Dec 19, 2017, at 6:49 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>> 
>>> I have a Yealink T46G registered at our Freeswitch machine. MWI is
>>> enabled on the Yealink Phone, I see a "message-summary" entry in
>>> sip_subscriptions. So I think, presence is setup fine
>>> 
>>> Content of the sip_subscriptions table:
>>> "proto" "sip_user" "sip_host" "sub_to_user" "sub_to_host"
>>> "presence_hosts" "event" "contact" "call_id" "full_from" "full_via"
>>> "expires" "user_agent" "accept" "profile_name" "hostname" "network_port"
>>> "network_ip" "version" "orig_proto" "full_to" "id"
>>> "sip" "200" "fs00.my.domain" "200" "fs00.my.domain" "" "message-summary"
>>> """Peter3 Buero"" <sip:200 at 192.168.1.72:5060>"
>>> "0_3056780279 at 192.168.1.72" """Peter3 Buero""
>>> <sip:200 at fs00.my.domain:5060>;tag=1082071637" "SIP/2.0/UDP
>>> 192.168.1.72:5060;branch=z9hG4bK2245575863" "1513683832" "Yealink
>>> SIP-T46G 28.80.0.130" "application/simple-message-summary " "internal"
>>> "fs01.my.domain" "5060" "192.168.1.72" "-1" ""
>>> "<sip:200 at fs00.my.domain:5060>;tag=WdayfEpssASN" "2821206"
>>> 
>>> Events are enabled on the freeswitch console
>>> /events MESSAGE_QUERY MESSAGE_WAITING
>>> 
>>> When a voicemail is recorded, I see a number of recurring events on the
>>> console (see below), but no NOTIFY message is sent to the phone.
>>> However, after a while (after 10min or even after hours) the phone
>>> receives its message.
>>> When I delete all VM messages, the MWI is never sent to the phone,
>>> although messages are prepared (MWI-Messages-Waiting: no), as I can see
>>> them on the console.
>>> 
>>> If I set the parameter “send-message-query-on-register“ in internal
>>> sofia profile, then MWI is updated with every register to the phone.
>>> Alos on every restart of the phone. But as we cannot use short intervals
>>> intervals due to some reason, this is not the solution.
>>> 
>>> Anybody has a hint, how to solve this?
>>> 
>>> 
>>> Example messages:
>>> RECV EVENT
>>> Event-Name: MESSAGE_QUERY
>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>> FreeSWITCH-Hostname: fs01.my.domain
>>> FreeSWITCH-Switchname: fs01.my.domain
>>> FreeSWITCH-IPv4: 192.168.1.9
>>> FreeSWITCH-IPv6: ::1
>>> Event-Date-Local: 2017-12-19 12:27:40
>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>> Event-Date-Timestamp: 1513682860739342
>>> Event-Calling-File: sofia_presence.c
>>> Event-Calling-Function: sofia_presence_sub_reg_callback
>>> Event-Calling-Line-Number: 1736
>>> Event-Sequence: 16929
>>> Message-Account: sip:200 at fs00.my.domain
>>> VM-Sofia-Profile: internal
>>> VM-sub-call-id: 0_3056780279 at 192.168.1.72
>>> 
>>> 
>>> RECV EVENT
>>> Event-Name: MESSAGE_WAITING
>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>> FreeSWITCH-Hostname: fs01.my.domain
>>> FreeSWITCH-Switchname: fs01.my.domain
>>> FreeSWITCH-IPv4: 192.168.1.9
>>> FreeSWITCH-IPv6: ::1
>>> Event-Date-Local: 2017-12-19 12:27:40
>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>> Event-Date-Timestamp: 1513682860899354
>>> Event-Calling-File: mod_voicemail.c
>>> Event-Calling-Function: actual_message_query_handler
>>> Event-Calling-Line-Number: 4036
>>> Event-Sequence: 16931
>>> MWI-Messages-Waiting: yes
>>> MWI-Message-Account: sip:200 at fs00.my.domain
>>> MWI-Voice-Message: 1/0 (0/0)
>>> Sofia-Profile: internal
>>> sub-call-id: 0_3056780279 at 192.168.1.72



From bipin at xbipin.com  Tue Dec 19 19:50:15 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Tue, 19 Dec 2017 23:50:15 +0400
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <F48F9953-F444-4549-9727-6B37704EA2CA@gmail.com>
References: <F48F9953-F444-4549-9727-6B37704EA2CA@gmail.com>
Message-ID: <160705398d8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Hi,

Use visual studio 2015 with the sdk mentioned in that link and compile and 
it should be fine, we are using it since long and compile the latest code 
and works fine in production.




On December 19, 2017 7:07:23 PM Ali Haider <alihaider.4189 at gmail.com> wrote:

> Hiii jeol
> https://freeswitch.org/confluence/pages/viewpage.action?pageId=1966780
> I’m followed these steps on windows but after installation of visual studio 
> 2013 which was may be required 182 error is found for missing files
> Please tell me which version of visual studio is installed to build soln 
> and how to remove error
>
>
> Sent from my iPhone
>
>
> ----------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mario_fs at mgtech.com  Tue Dec 19 21:25:29 2017
From: mario_fs at mgtech.com (Mario)
Date: Tue, 19 Dec 2017 13:25:29 -0800
Subject: [Freeswitch-users] MWI not sent
In-Reply-To: <07BAB46B-B5DC-4F5D-9F5A-2CEC2932167B@jerris.com>
References: <5A38FCB4.5000105@telefaks.de>
 <6EA801C8-F2B8-45DE-95B7-FA9CE9CD64B5@jerris.com>
 <5A392F94.50002@telefaks.de>
 <07BAB46B-B5DC-4F5D-9F5A-2CEC2932167B@jerris.com>
Message-ID: <EFD8104C-4675-4723-8978-09EBF3014C3C@mgtech.com>

Could be related to this? https://freeswitch.org/jira/browse/FS-10683?filter=-2

> On Dec 19, 2017, at 7:37 AM, Michael Jerris <mike at jerris.com> wrote:
> 
> without that setting, when sending a voicemail to the user right after they register?
> 
>> On Dec 19, 2017, at 10:26 AM, Peter Steinbach <lists at telefaks.de> wrote:
>> 
>> Hello Michael,
>> 
>> with parameter “send-message-query-on-register“ in internal sofia profile it works during registration, without this parameter is does not.
>> 
>> Best regards
>> Peter
>> 
>> 
>> 
>> On 12/19/17 16:09, Michael Jerris wrote:
>>> if you do a voicemail to the user very soon after it registers (within seconds) does it work?
>>> 
>>>> On Dec 19, 2017, at 6:49 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>>> 
>>>> I have a Yealink T46G registered at our Freeswitch machine. MWI is
>>>> enabled on the Yealink Phone, I see a "message-summary" entry in
>>>> sip_subscriptions. So I think, presence is setup fine
>>>> 
>>>> Content of the sip_subscriptions table:
>>>> "proto" "sip_user" "sip_host" "sub_to_user" "sub_to_host"
>>>> "presence_hosts" "event" "contact" "call_id" "full_from" "full_via"
>>>> "expires" "user_agent" "accept" "profile_name" "hostname" "network_port"
>>>> "network_ip" "version" "orig_proto" "full_to" "id"
>>>> "sip" "200" "fs00.my.domain" "200" "fs00.my.domain" "" "message-summary"
>>>> """Peter3 Buero"" <sip:200 at 192.168.1.72:5060>"
>>>> "0_3056780279 at 192.168.1.72" """Peter3 Buero""
>>>> <sip:200 at fs00.my.domain:5060>;tag=1082071637" "SIP/2.0/UDP
>>>> 192.168.1.72:5060;branch=z9hG4bK2245575863" "1513683832" "Yealink
>>>> SIP-T46G 28.80.0.130" "application/simple-message-summary " "internal"
>>>> "fs01.my.domain" "5060" "192.168.1.72" "-1" ""
>>>> "<sip:200 at fs00.my.domain:5060>;tag=WdayfEpssASN" "2821206"
>>>> 
>>>> Events are enabled on the freeswitch console
>>>> /events MESSAGE_QUERY MESSAGE_WAITING
>>>> 
>>>> When a voicemail is recorded, I see a number of recurring events on the
>>>> console (see below), but no NOTIFY message is sent to the phone.
>>>> However, after a while (after 10min or even after hours) the phone
>>>> receives its message.
>>>> When I delete all VM messages, the MWI is never sent to the phone,
>>>> although messages are prepared (MWI-Messages-Waiting: no), as I can see
>>>> them on the console.
>>>> 
>>>> If I set the parameter “send-message-query-on-register“ in internal
>>>> sofia profile, then MWI is updated with every register to the phone.
>>>> Alos on every restart of the phone. But as we cannot use short intervals
>>>> intervals due to some reason, this is not the solution.
>>>> 
>>>> Anybody has a hint, how to solve this?
>>>> 
>>>> 
>>>> Example messages:
>>>> RECV EVENT
>>>> Event-Name: MESSAGE_QUERY
>>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>>> FreeSWITCH-Hostname: fs01.my.domain
>>>> FreeSWITCH-Switchname: fs01.my.domain
>>>> FreeSWITCH-IPv4: 192.168.1.9
>>>> FreeSWITCH-IPv6: ::1
>>>> Event-Date-Local: 2017-12-19 12:27:40
>>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>>> Event-Date-Timestamp: 1513682860739342
>>>> Event-Calling-File: sofia_presence.c
>>>> Event-Calling-Function: sofia_presence_sub_reg_callback
>>>> Event-Calling-Line-Number: 1736
>>>> Event-Sequence: 16929
>>>> Message-Account: sip:200 at fs00.my.domain
>>>> VM-Sofia-Profile: internal
>>>> VM-sub-call-id: 0_3056780279 at 192.168.1.72
>>>> 
>>>> 
>>>> RECV EVENT
>>>> Event-Name: MESSAGE_WAITING
>>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>>> FreeSWITCH-Hostname: fs01.my.domain
>>>> FreeSWITCH-Switchname: fs01.my.domain
>>>> FreeSWITCH-IPv4: 192.168.1.9
>>>> FreeSWITCH-IPv6: ::1
>>>> Event-Date-Local: 2017-12-19 12:27:40
>>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>>> Event-Date-Timestamp: 1513682860899354
>>>> Event-Calling-File: mod_voicemail.c
>>>> Event-Calling-Function: actual_message_query_handler
>>>> Event-Calling-Line-Number: 4036
>>>> Event-Sequence: 16931
>>>> MWI-Messages-Waiting: yes
>>>> MWI-Message-Account: sip:200 at fs00.my.domain
>>>> MWI-Voice-Message: 1/0 (0/0)
>>>> Sofia-Profile: internal
>>>> sub-call-id: 0_3056780279 at 192.168.1.72
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Tue Dec 19 21:35:01 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Dec 2017 15:35:01 -0600
Subject: [Freeswitch-users] bridge_generate_comfort_noise not honoring
 integer value for volume level
In-Reply-To: <MWHPR11MB148608B3A839FAA35140F0E2B60E0@MWHPR11MB1486.namprd11.prod.outlook.com>
References: <MWHPR11MB1486A0000B461F759B86A5CEB60A0@MWHPR11MB1486.namprd11.prod.outlook.com>
 <CAKbxfG_9Qr+wZo1nqte1zjt=PDeRgsnaB+vv=WzoNut+sRxsVA@mail.gmail.com>
 <MWHPR11MB148608B3A839FAA35140F0E2B60E0@MWHPR11MB1486.namprd11.prod.outlook.com>
Message-ID: <CAKbxfG_YkzCjJJ5RLH=VBQaFCNXeiLd5_1x4=vKyF5p5f_R9hA@mail.gmail.com>

did you try 8000 or 16000

you may need to add some debugging to the code and see.  If this is a bug,
you really should be reporting it to JIRA as we don't really monitor the
discussion list for bug reports.


On Mon, Dec 18, 2017 at 1:39 AM, Matthew Meek <mmeek at livexchange.com> wrote:

> The values of -2,-1  50, 1400, and 4000 all have the same volume of
> whitenoise.  I am setting it with:
>
>
>
> <action application="set" data="bridge_generate_comfort_noise=4000"/>
>
>
>
> I can see in the code where this should work (although -1 is doubtful to
> be silence as the if block checks for silence_val < -1 instead of <= -1),
> but in practice the value has no effect. I am not sure if the value is
> over-ridden somewhere else.
>
>
>
> -Matthew Meek
>
>
>
>
>
> The higher the number the softer it will get.  You say 4000 didn't work?
> -1 should be absolute silence.
>
>
>
> On Thu, Dec 14, 2017 at 6:01 AM, Matthew Meek <mmeek at livexchange.com>
> wrote:
>
> I can successfully inject whitenoise on the bridged leg (audio from A leg
> towards the bridged call) in the dial plan by setting
> bridge_generate_comfort_noise=true. The whitenoise is quite loud and I
> would like to make it quieter or silent (I am just trying to keep the far
> end firewall port open during long periods of A leg not sending packets due
> to hold).  I have tried values of -1, 40,1400, and 4000 instead of true and
> they produce the same amount of whitenoise volume. The source code in
> switch_ivr_bridge.c looks like it indeed honors an integer value, but
> somehow the end result is always the same. If I do not set the variable I
> get no whitenoise, so the variable is indeed controlling whitenoise, just
> not the volume level of whitenoise. The codec is PCMU.
>
>
>
> I can repeat this with 1.2, 1.4, and 1.6 branches of FS.
>
>
>
> Matthew Meek
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> Anthony Minessale II
>
> Founder, FreeSWITCH.
>
>
>
>
>
> https://youtu.be/l_hOxzCt6X4
>
> https://www.youtube.com/watch?v=oAxXgyx5jUw
>
> https://www.youtube.com/watch?v=9XXgW34t40s
>
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II
Founder, FreeSWITCH.
http://freeswitch.com


https://youtu.be/l_hOxzCt6X4
https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From mario_fs at mgtech.com  Tue Dec 19 21:36:57 2017
From: mario_fs at mgtech.com (Mario)
Date: Tue, 19 Dec 2017 13:36:57 -0800
Subject: [Freeswitch-users] MWI not sent
In-Reply-To: <EFD8104C-4675-4723-8978-09EBF3014C3C@mgtech.com>
References: <5A38FCB4.5000105@telefaks.de>
 <6EA801C8-F2B8-45DE-95B7-FA9CE9CD64B5@jerris.com>
 <5A392F94.50002@telefaks.de>
 <07BAB46B-B5DC-4F5D-9F5A-2CEC2932167B@jerris.com>
 <EFD8104C-4675-4723-8978-09EBF3014C3C@mgtech.com>
Message-ID: <51ABC32D-7445-48D8-A020-9A86EC38B2C0@mgtech.com>

BTW, this sounds very similar in that even though in my case I am using the MWI-Account parameter, if I leave the phone alone it sometimes will show MWI after several minutes (30?). It also may show up after a VM is left, but always lost after a restart of the phone. In my case though, if the VM matches the phone account then there is no problem, only using MWI-Account which I need. This symptoms make them sound related.
Mario G

https://freeswitch.org/jira/browse/FS-10683



> On Dec 19, 2017, at 1:25 PM, Mario <mario_fs at mgtech.com> wrote:
> 
> https://freeswitch.org/jira/browse/FS-10683?filter=-2 <https://freeswitch.org/jira/browse/FS-10683?filter=-2>
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From lpopov at blasterphone.com  Wed Dec 20 01:58:20 2017
From: lpopov at blasterphone.com (Lyubo Popov)
Date: Tue, 19 Dec 2017 23:58:20 -0200
Subject: [Freeswitch-users] How to reject a call and not log it in CDRs
 that contains unsupported ANCII characters?
In-Reply-To: <CAC7ttFGjy9-LpWVKBo=kmKYUX3xn6aaznoB11W0awPzsv9gd=A@mail.gmail.com>
References: <CAMSNS4ifnUuKqYG_4VMJrcnSvo1F4cxyphpZLc-Z9-UqKhh0QQ@mail.gmail.com>
 <CAKzWOxWbrNgrYD8X3Vv5dL8wDiebsguaR861qqTh1BF9+Q_hSQ@mail.gmail.com>
 <CAC7ttFGjy9-LpWVKBo=kmKYUX3xn6aaznoB11W0awPzsv9gd=A@mail.gmail.com>
Message-ID: <CAMSNS4h1tWRK3mynO1=kE-vfq2_K3vgtGR4R_ubUq0HtG=c+Yg@mail.gmail.com>

Thank you very much for the reply. The process_cdr variable is very useful
and exactly what I was looking for. Appreciated!

<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>
Virus-free.
www.avast.com
<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
<#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

On Mon, Dec 18, 2017 at 8:37 PM, Gregor Nanger <gregor at infomedia.si> wrote:

> ​Hi, Lyubo!
>
> I had similar case. Use process_cdr variable in dialplan, if you do not
> want to store in CDR.
>
> <action application="set" data="process_cdr=
> ​false​
> "/>
>
> ​Hope it helps, Gregor​
>
>
> 2017-12-18 21:44 GMT+01:00 Michael Collins <msc at freeswitch.org>:
>
>> Hi Lyubo,
>>
>> In this case it may be better to see if your CDR parser can skip the
>> non-numeric caller id values, perhaps by adding a validation check prior to
>> performing the parse action. As a rule of thumb, if your CDR parser can be
>> tripped up by the data it is parsing then it needs to be hardened. I'm sure
>> many here would highly recommend sanitizing/validation as a best practice,
>> particularly when handling data that comes from the public Internet.
>> Another consideration is that you may actually want to have a record of
>> these kinds of attacks in case there is a need to investigate an incident
>> or otherwise analyze attack patterns.
>>
>> I would recommend that you change the behavior of the parser from
>> "complaining" to "keeping the CDR database clean but logging invalid input
>> for future reference."
>>
>> Hope this helps,
>> -MC
>>
>>
>> On Fri, Dec 15, 2017 at 1:11 PM, Lyubo Popov <koki.roul at gmail.com> wrote:
>>
>>> Hello all,
>>>
>>> Maybe someone can help me with this problem and will be greatly
>>> appreciated. We are getting calls with CallerID like this one  ‘hi'or‘x’='x'.
>>> Later when our billing start parsing the CDRs it will complain because of
>>> the first character "`". My question I suppose is, how to prevent such
>>> calls to get added to the CDRs? We want to reject the call that has non
>>> numeric CallerID and not get it added in the CDRs. This is what we have in
>>> the dialplan.
>>>
>>> <extension name="Routing">
>>>         <condition field="${radius_auth_result}" expression="0"/>
>>>
>>>         <!--
>>>         <condition field="${h323-redirect-number}" expression="^(.+)$"
>>> break="never">
>>>             <action application="set" data="destination_number=$1" />
>>>         </condition>
>>>         -->
>>>         <condition field="caller_id_number" expression="^([0-9]+)$">
>>>             <anti-action application="hangup"/>
>>> </condition>
>>>         <condition field="destination_number" expression="^(.+)$">
>>>             <!--<action application="info"/>-->
>>>             <action application="export" data="nolocal:h323-call-origin
>>> =originate"/>
>>>             <action application="set" data="sip_h_X-accountcode=${accountcode}"
>>> />
>>>             <action application="set" data="call_direction=outbound" />
>>>             <action application="set" data="hangup_after_bridge=true"/>
>>>             <action application="set" data="continue_on_fail=true"/>
>>>             <action application="set" data="inherit_codec=true" />
>>>             <action application="set" data="call_timeout=20"/>
>>>             <action application="set" data="fail_on_single_reject=USER_BUSY"
>>> />
>>>             <action application="set" data="origination_caller_id_na
>>> me=${sip_req_user}"/>
>>>             <action application="set" data="origination_caller_id_nu
>>> mber=${sip_from_user}"/>
>>>             <action application="set" data="execute_on_answer=sched_hangup
>>> +${h323-credit-time} alloted_timeout" />
>>>             <action application="bridge" data="{sip_invite_from_uri=sip
>>> :${sip_from_user}@${sip_network_ip}}sofia/internal/${destina
>>> tion_number}@x.x.x.x:5060" />
>>>             <action application="hangup" data="${bridge_hangup_cause}"/>
>>>         </condition>
>>>     </extension>
>>>
>>> Thank you all!
>>>
>>> L.Popov
>>>
>>>
>>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon> Virus-free.
>>> www.avast.com
>>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
>>> <#m_2242263481946948631_m_2040158889290273727_m_-7626445720286026976_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Atenciosamente,
============================
Lyubo Popov
CEO - BlasterPhone LLC
Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
iNum: +883 510001-354111
Website: http://www.blastervoip.com.br/
============================
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From alexanderhenryperkins at gmail.com  Wed Dec 20 03:27:29 2017
From: alexanderhenryperkins at gmail.com (Alexander Perkins)
Date: Tue, 19 Dec 2017 21:27:29 -0600
Subject: [Freeswitch-users] Freeswitch and ODBC
Message-ID: <CALLkTp2jg6_0XpQop+4mdvCMxcRwKO0eBLZ1iZxPxCSkjf46=g@mail.gmail.com>

Hi All.  I am fairly new to FS and I am attempting to connect sofia it to
my MySQL database.  I have looked at a lot of documentation from
Confluence, but cannot seem to figure out if it is connecting or not.  I
have looked at the database to see if there are new tables there (I guess
FS automatically creates the tables), but I do not see them.

How do I know if FS is connecting to my database?  How do I create the
tables for SIP?

Thank you!
Alex
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From lists at telefaks.de  Wed Dec 20 09:20:28 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 20 Dec 2017 10:20:28 +0100
Subject: [Freeswitch-users] MWI not sent
In-Reply-To: <07BAB46B-B5DC-4F5D-9F5A-2CEC2932167B@jerris.com>
References: <5A38FCB4.5000105@telefaks.de>
 <6EA801C8-F2B8-45DE-95B7-FA9CE9CD64B5@jerris.com>
 <5A392F94.50002@telefaks.de>
 <07BAB46B-B5DC-4F5D-9F5A-2CEC2932167B@jerris.com>
Message-ID: <5A3A2B5C.8060402@telefaks.de>

Without that settings it takes upto hours to update the MWI information
on the phone (even if I set registration time to 1 min).

Best regards
Peter



On 12/19/17 16:37, Michael Jerris wrote:
> without that setting, when sending a voicemail to the user right after they register?
>
>> On Dec 19, 2017, at 10:26 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>
>> Hello Michael,
>>
>> with parameter “send-message-query-on-register“ in internal sofia profile it works during registration, without this parameter is does not.
>>
>> Best regards
>> Peter
>>
>>
>>
>> On 12/19/17 16:09, Michael Jerris wrote:
>>> if you do a voicemail to the user very soon after it registers (within seconds) does it work?
>>>
>>>> On Dec 19, 2017, at 6:49 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>>>
>>>> I have a Yealink T46G registered at our Freeswitch machine. MWI is
>>>> enabled on the Yealink Phone, I see a "message-summary" entry in
>>>> sip_subscriptions. So I think, presence is setup fine
>>>>
>>>> Content of the sip_subscriptions table:
>>>> "proto" "sip_user" "sip_host" "sub_to_user" "sub_to_host"
>>>> "presence_hosts" "event" "contact" "call_id" "full_from" "full_via"
>>>> "expires" "user_agent" "accept" "profile_name" "hostname" "network_port"
>>>> "network_ip" "version" "orig_proto" "full_to" "id"
>>>> "sip" "200" "fs00.my.domain" "200" "fs00.my.domain" "" "message-summary"
>>>> """Peter3 Buero"" <sip:200 at 192.168.1.72:5060>"
>>>> "0_3056780279 at 192.168.1.72" """Peter3 Buero""
>>>> <sip:200 at fs00.my.domain:5060>;tag=1082071637" "SIP/2.0/UDP
>>>> 192.168.1.72:5060;branch=z9hG4bK2245575863" "1513683832" "Yealink
>>>> SIP-T46G 28.80.0.130" "application/simple-message-summary " "internal"
>>>> "fs01.my.domain" "5060" "192.168.1.72" "-1" ""
>>>> "<sip:200 at fs00.my.domain:5060>;tag=WdayfEpssASN" "2821206"
>>>>
>>>> Events are enabled on the freeswitch console
>>>> /events MESSAGE_QUERY MESSAGE_WAITING
>>>>
>>>> When a voicemail is recorded, I see a number of recurring events on the
>>>> console (see below), but no NOTIFY message is sent to the phone.
>>>> However, after a while (after 10min or even after hours) the phone
>>>> receives its message.
>>>> When I delete all VM messages, the MWI is never sent to the phone,
>>>> although messages are prepared (MWI-Messages-Waiting: no), as I can see
>>>> them on the console.
>>>>
>>>> If I set the parameter “send-message-query-on-register“ in internal
>>>> sofia profile, then MWI is updated with every register to the phone.
>>>> Alos on every restart of the phone. But as we cannot use short intervals
>>>> intervals due to some reason, this is not the solution.
>>>>
>>>> Anybody has a hint, how to solve this?
>>>>
>>>>
>>>> Example messages:
>>>> RECV EVENT
>>>> Event-Name: MESSAGE_QUERY
>>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>>> FreeSWITCH-Hostname: fs01.my.domain
>>>> FreeSWITCH-Switchname: fs01.my.domain
>>>> FreeSWITCH-IPv4: 192.168.1.9
>>>> FreeSWITCH-IPv6: ::1
>>>> Event-Date-Local: 2017-12-19 12:27:40
>>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>>> Event-Date-Timestamp: 1513682860739342
>>>> Event-Calling-File: sofia_presence.c
>>>> Event-Calling-Function: sofia_presence_sub_reg_callback
>>>> Event-Calling-Line-Number: 1736
>>>> Event-Sequence: 16929
>>>> Message-Account: sip:200 at fs00.my.domain
>>>> VM-Sofia-Profile: internal
>>>> VM-sub-call-id: 0_3056780279 at 192.168.1.72
>>>>
>>>>
>>>> RECV EVENT
>>>> Event-Name: MESSAGE_WAITING
>>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>>> FreeSWITCH-Hostname: fs01.my.domain
>>>> FreeSWITCH-Switchname: fs01.my.domain
>>>> FreeSWITCH-IPv4: 192.168.1.9
>>>> FreeSWITCH-IPv6: ::1
>>>> Event-Date-Local: 2017-12-19 12:27:40
>>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>>> Event-Date-Timestamp: 1513682860899354
>>>> Event-Calling-File: mod_voicemail.c
>>>> Event-Calling-Function: actual_message_query_handler
>>>> Event-Calling-Line-Number: 4036
>>>> Event-Sequence: 16931
>>>> MWI-Messages-Waiting: yes
>>>> MWI-Message-Account: sip:200 at fs00.my.domain
>>>> MWI-Voice-Message: 1/0 (0/0)
>>>> Sofia-Profile: internal
>>>> sub-call-id: 0_3056780279 at 192.168.1.72
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de



From lists at telefaks.de  Wed Dec 20 09:23:59 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 20 Dec 2017 10:23:59 +0100
Subject: [Freeswitch-users] MWI not sent
In-Reply-To: <51ABC32D-7445-48D8-A020-9A86EC38B2C0@mgtech.com>
References: <5A38FCB4.5000105@telefaks.de>
 <6EA801C8-F2B8-45DE-95B7-FA9CE9CD64B5@jerris.com>
 <5A392F94.50002@telefaks.de>
 <07BAB46B-B5DC-4F5D-9F5A-2CEC2932167B@jerris.com>
 <EFD8104C-4675-4723-8978-09EBF3014C3C@mgtech.com>
 <51ABC32D-7445-48D8-A020-9A86EC38B2C0@mgtech.com>
Message-ID: <5A3A2C2F.5010208@telefaks.de>

In my case, it's different: When Phone is rebooted, then MWI info on the
phone is updated immediately. And MWI-Account parameter is set
(200 at my.domain) and matches phone account number (200 at my.domain).

 

On 12/19/17 22:36, Mario wrote:
> BTW, this sounds very similar in that even though in my case I am
> using the MWI-Account parameter, if I leave the phone alone it
> sometimes will show MWI after several minutes (30?). It also may show
> up after a VM is left, but always lost after a restart of the phone.
> In my case though, if the VM matches the phone account then there is
> no problem, only using MWI-Account which I need. This symptoms make
> them sound related.
> Mario G
>
> https://freeswitch.org/jira/browse/FS-10683
>
>
>
>> On Dec 19, 2017, at 1:25 PM, Mario <mario_fs at mgtech.com> wrote:
>>
>> https://freeswitch.org/jira/browse/FS-10683?filter=-2
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

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From gregor at infomedia.si  Wed Dec 20 09:29:43 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Wed, 20 Dec 2017 10:29:43 +0100
Subject: [Freeswitch-users] How to reject a call and not log it in CDRs
 that contains unsupported ANCII characters?
In-Reply-To: <CAMSNS4h1tWRK3mynO1=kE-vfq2_K3vgtGR4R_ubUq0HtG=c+Yg@mail.gmail.com>
References: <CAMSNS4ifnUuKqYG_4VMJrcnSvo1F4cxyphpZLc-Z9-UqKhh0QQ@mail.gmail.com>
 <CAKzWOxWbrNgrYD8X3Vv5dL8wDiebsguaR861qqTh1BF9+Q_hSQ@mail.gmail.com>
 <CAC7ttFGjy9-LpWVKBo=kmKYUX3xn6aaznoB11W0awPzsv9gd=A@mail.gmail.com>
 <CAMSNS4h1tWRK3mynO1=kE-vfq2_K3vgtGR4R_ubUq0HtG=c+Yg@mail.gmail.com>
Message-ID: <CAC7ttFFE-h6QQRJJokZypsgb2dX2wS0yaMtnb7ircwoBh8i_ow@mail.gmail.com>

Glad to help.
​ Just be carefull, if you also would like to use b_only. It only works if
you also have b leg loging enabled in conf file.​


On Wed, Dec 20, 2017, 03:01 Lyubo Popov <lpopov at blasterphone.com
<https://mltrk.io/link/mailto%3Alpopov%40blasterphone.com/jCUwHkcTexx2TytmrpF6>>
wrote:

> Thank you very much for the reply. The process_cdr variable is very
> useful and exactly what I was looking for. Appreciated!
>
>
> <https://mltrk.io/link/https%3A%2F%2Fwww.avast.com%2Fsig-email%3Futm_medium%3Demail%26utm_source%3Dlink%26utm_campaign%3Dsig-email%26utm_content%3Dwebmail%26utm_term%3Dicon/jCUwHkcTexx2TytmrpF6> Virus-free.
> www.avast.com
> <https://mltrk.io/link/https%3A%2F%2Fwww.avast.com%2Fsig-email%3Futm_medium%3Demail%26utm_source%3Dlink%26utm_campaign%3Dsig-email%26utm_content%3Dwebmail%26utm_term%3Dlink/jCUwHkcTexx2TytmrpF6>
> <https://mltrk.io/link/%23m_926543164795071026_m_-8463921194330766450_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2/jCUwHkcTexx2TytmrpF6>
>
> On Mon, Dec 18, 2017 at 8:37 PM, Gregor Nanger <gregor at infomedia.si
> <https://mltrk.io/link/mailto%3Agregor%40infomedia.si/jCUwHkcTexx2TytmrpF6>
> > wrote:
>
>> ​Hi, Lyubo!
>>
>> I had similar case. Use process_cdr variable in dialplan, if you do not
>> want to store in CDR.
>>
>> <action application="set" data="process_cdr=
>> ​false​
>> "/>
>>
>> ​Hope it helps, Gregor​
>>
>>
>> 2017-12-18 21:44 GMT+01:00 Michael Collins <msc at freeswitch.org
>> <https://mltrk.io/link/mailto%3Amsc%40freeswitch.org/jCUwHkcTexx2TytmrpF6>
>> >:
>>
>>> Hi Lyubo,
>>>
>>> In this case it may be better to see if your CDR parser can skip the
>>> non-numeric caller id values, perhaps by adding a validation check prior to
>>> performing the parse action. As a rule of thumb, if your CDR parser can be
>>> tripped up by the data it is parsing then it needs to be hardened. I'm sure
>>> many here would highly recommend sanitizing/validation as a best practice,
>>> particularly when handling data that comes from the public Internet.
>>> Another consideration is that you may actually want to have a record of
>>> these kinds of attacks in case there is a need to investigate an incident
>>> or otherwise analyze attack patterns.
>>>
>>> I would recommend that you change the behavior of the parser from
>>> "complaining" to "keeping the CDR database clean but logging invalid input
>>> for future reference."
>>>
>>> Hope this helps,
>>> -MC
>>>
>>>
>>> On Fri, Dec 15, 2017 at 1:11 PM, Lyubo Popov <koki.roul at gmail.com
>>> <https://mltrk.io/link/mailto%3Akoki.roul%40gmail.com/jCUwHkcTexx2TytmrpF6>
>>> > wrote:
>>>
>>>> Hello all,
>>>>
>>>> Maybe someone can help me with this problem and will be greatly
>>>> appreciated. We are getting calls with CallerID like this one  ‘hi'or‘x’='x'.
>>>> Later when our billing start parsing the CDRs it will complain because of
>>>> the first character "`". My question I suppose is, how to prevent such
>>>> calls to get added to the CDRs? We want to reject the call that has non
>>>> numeric CallerID and not get it added in the CDRs. This is what we have in
>>>> the dialplan.
>>>>
>>>> <extension name="Routing">
>>>>         <condition field="${radius_auth_result}" expression="0"/>
>>>>
>>>>         <!--
>>>>         <condition field="${h323-redirect-number}" expression="^(.+)$"
>>>> break="never">
>>>>             <action application="set" data="destination_number=$1" />
>>>>         </condition>
>>>>         -->
>>>>         <condition field="caller_id_number" expression="^([0-9]+)$">
>>>>             <anti-action application="hangup"/>
>>>> </condition>
>>>>         <condition field="destination_number" expression="^(.+)$">
>>>>             <!--<action application="info"/>-->
>>>>             <action application="export" data="nolocal:h323-call-
>>>> origin=originate"/>
>>>>             <action application="set" data="sip_h_X-accountcode=${accountcode}"
>>>> />
>>>>             <action application="set" data="call_direction=outbound" />
>>>>             <action application="set" data="hangup_after_bridge=true"/>
>>>>             <action application="set" data="continue_on_fail=true"/>
>>>>             <action application="set" data="inherit_codec=true" />
>>>>             <action application="set" data="call_timeout=20"/>
>>>>             <action application="set" data="fail_on_single_reject=USER_BUSY"
>>>> />
>>>>             <action application="set" data="origination_caller_id_
>>>> name=${sip_req_user}"/>
>>>>             <action application="set" data="origination_caller_id_
>>>> number=${sip_from_user}"/>
>>>>             <action application="set" data="execute_on_answer=sched_hangup
>>>> +${h323-credit-time} alloted_timeout" />
>>>>             <action application="bridge" data="{sip_invite_from_uri=
>>>> sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${
>>>> destination_number}@x.x.x.x:5060" />
>>>>             <action application="hangup" data="${bridge_hangup_cause}"/
>>>> >
>>>>         </condition>
>>>>     </extension>
>>>>
>>>> Thank you all!
>>>>
>>>> L.Popov
>>>>
>>>>
>>>> <https://mltrk.io/link/https%3A%2F%2Fwww.avast.com%2Fsig-email%3Futm_medium%3Demail%26utm_source%3Dlink%26utm_campaign%3Dsig-email%26utm_content%3Dwebmail%26utm_term%3Dicon/jCUwHkcTexx2TytmrpF6> Virus-free.
>>>> www.avast.com
>>>> <https://mltrk.io/link/https%3A%2F%2Fwww.avast.com%2Fsig-email%3Futm_medium%3Demail%26utm_source%3Dlink%26utm_campaign%3Dsig-email%26utm_content%3Dwebmail%26utm_term%3Dlink/jCUwHkcTexx2TytmrpF6>
>>>> <https://mltrk.io/link/%23m_926543164795071026_m_-8463921194330766450_m_2242263481946948631_m_2040158889290273727_m_-7626445720286026976_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2/jCUwHkcTexx2TytmrpF6>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> <https://mltrk.io/link/mailto%3Aconsulting%40freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>>>
>>>> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitchsolutions.com/jCUwHkcTexx2TytmrpF6>
>>>> http://www
>>>> <https://mltrk.io/link/http%3A%2F%2Fwww/jCUwHkcTexx2TytmrpF6>.
>>>> freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>>>
>>>> <https://mltrk.io/link/http%3A%2F%2Fconfluence.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>>> http://confluence.freeswitch
>>>> <https://mltrk.io/link/http%3A%2F%2Fconfluence.freeswitch/jCUwHkcTexx2TytmrpF6>
>>>> .org
>>>> http://www.cluecon.com
>>>> <https://mltrk.io/link/http%3A%2F%2Fwww.cluecon.com/jCUwHkcTexx2TytmrpF6>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <https://mltrk.io/link/mailto%3AFreeSWITCH-users%40lists.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>>>
>>>> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users/jCUwHkcTexx2TytmrpF6>
>>>> http://lists.freeswitch.org/
>>>> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2F/jCUwHkcTexx2TytmrpF6>
>>>> mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users/jCUwHkcTexx2TytmrpF6>
>>>> http://lists
>>>> <https://mltrk.io/link/http%3A%2F%2Flists/jCUwHkcTexx2TytmrpF6>.
>>>> freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <https://mltrk.io/link/mailto%3Aconsulting%40freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>>
>>> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitchsolutions.com/jCUwHkcTexx2TytmrpF6>
>>> http://www <https://mltrk.io/link/http%3A%2F%2Fwww/jCUwHkcTexx2TytmrpF6>
>>> .freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>>
>>> <https://mltrk.io/link/http%3A%2F%2Fconfluence.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>> http://confluence.freeswitch
>>> <https://mltrk.io/link/http%3A%2F%2Fconfluence.freeswitch/jCUwHkcTexx2TytmrpF6>
>>> .org
>>> http://www.cluecon.com
>>> <https://mltrk.io/link/http%3A%2F%2Fwww.cluecon.com/jCUwHkcTexx2TytmrpF6>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <https://mltrk.io/link/mailto%3AFreeSWITCH-users%40lists.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>>
>>> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users/jCUwHkcTexx2TytmrpF6>
>>> http://lists.freeswitch.org/
>>> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2F/jCUwHkcTexx2TytmrpF6>
>>> mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:
>>> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users/jCUwHkcTexx2TytmrpF6>
>>> http://lists
>>> <https://mltrk.io/link/http%3A%2F%2Flists/jCUwHkcTexx2TytmrpF6>.
>>> freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>>
>>
>>
>>
>> --
>> Gregor Nanger
>>
>> *CTO*
>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>> • www.infomedia.si
>> <https://mltrk.io/link/http%3A%2F%2Fwww.infomedia.si/jCUwHkcTexx2TytmrpF6>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <https://mltrk.io/link/mailto%3Aconsulting%40freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>
>> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitchsolutions.com/jCUwHkcTexx2TytmrpF6>
>> http://www <https://mltrk.io/link/http%3A%2F%2Fwww/jCUwHkcTexx2TytmrpF6>.
>> freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>
>> <https://mltrk.io/link/http%3A%2F%2Fconfluence.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>> http://confluence.freeswitch
>> <https://mltrk.io/link/http%3A%2F%2Fconfluence.freeswitch/jCUwHkcTexx2TytmrpF6>
>> .org
>> http://www.cluecon.com
>> <https://mltrk.io/link/http%3A%2F%2Fwww.cluecon.com/jCUwHkcTexx2TytmrpF6>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <https://mltrk.io/link/mailto%3AFreeSWITCH-users%40lists.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>
>> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users/jCUwHkcTexx2TytmrpF6>
>> http://lists.freeswitch.org/
>> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2F/jCUwHkcTexx2TytmrpF6>
>> mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:
>> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users/jCUwHkcTexx2TytmrpF6>
>> http://lists
>> <https://mltrk.io/link/http%3A%2F%2Flists/jCUwHkcTexx2TytmrpF6>.
>> freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>>
>
>
>
> --
> Atenciosamente,
> ============================
> Lyubo Popov
> CEO - BlasterPhone LLC
> Tel: 4003-1556 ( Outside Brazil 55 11 4003-1556)
> iNum: +883 510001-354111
> <https://mltrk.io/link/tel%3A%2B883%2520510%2520001%2520354%2520111/jCUwHkcTexx2TytmrpF6>
> Website: http://www.blastervoip.com.br/
> <https://mltrk.io/link/http%3A%2F%2Fwww.blastervoip.com.br%2F/jCUwHkcTexx2TytmrpF6>
> ============================
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <https://mltrk.io/link/mailto%3Aconsulting%40freeswitch.org/jCUwHkcTexx2TytmrpF6>
>
> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitchsolutions.com/jCUwHkcTexx2TytmrpF6>
> http://www <https://mltrk.io/link/http%3A%2F%2Fwww/jCUwHkcTexx2TytmrpF6>.
> freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>
> <https://mltrk.io/link/http%3A%2F%2Fconfluence.freeswitch.org/jCUwHkcTexx2TytmrpF6>
> http://confluence.freeswitch
> <https://mltrk.io/link/http%3A%2F%2Fconfluence.freeswitch/jCUwHkcTexx2TytmrpF6>
> .org
> http://www.cluecon.com
> <https://mltrk.io/link/http%3A%2F%2Fwww.cluecon.com/jCUwHkcTexx2TytmrpF6>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <https://mltrk.io/link/mailto%3AFreeSWITCH-users%40lists.freeswitch.org/jCUwHkcTexx2TytmrpF6>
>
> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Flistinfo%2Ffreeswitch-users/jCUwHkcTexx2TytmrpF6>
> http://lists.freeswitch.org/
> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2F/jCUwHkcTexx2TytmrpF6>
> mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:
> <https://mltrk.io/link/http%3A%2F%2Flists.freeswitch.org%2Fmailman%2Foptions%2Ffreeswitch-users/jCUwHkcTexx2TytmrpF6>
> http://lists
> <https://mltrk.io/link/http%3A%2F%2Flists/jCUwHkcTexx2TytmrpF6>.
> freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> <https://mltrk.io/link/http%3A%2F%2Fwww.freeswitch.org/jCUwHkcTexx2TytmrpF6>
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From alihaider.4189 at gmail.com  Wed Dec 20 16:17:23 2017
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Wed, 20 Dec 2017 21:17:23 +0500
Subject: [Freeswitch-users] Any one have code
Message-ID: <B6DDB0FD-4A17-488A-A5CE-2CB402DD03FE@gmail.com>

Hiii
 Anyone have code of audio call and video call from laptop to mobile

Sent from my iPhone


From mike at jerris.com  Wed Dec 20 16:30:21 2017
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Dec 2017 11:30:21 -0500
Subject: [Freeswitch-users] MWI not sent
In-Reply-To: <5A3A2B5C.8060402@telefaks.de>
References: <5A38FCB4.5000105@telefaks.de>
 <6EA801C8-F2B8-45DE-95B7-FA9CE9CD64B5@jerris.com>
 <5A392F94.50002@telefaks.de>
 <07BAB46B-B5DC-4F5D-9F5A-2CEC2932167B@jerris.com>
 <5A3A2B5C.8060402@telefaks.de>
Message-ID: <956FAC69-094D-49D4-B298-537698EA51A1@jerris.com>

Do you have multiple sip profiles that something at that domain is registered to?

> On Dec 20, 2017, at 4:20 AM, Peter Steinbach <lists at telefaks.de> wrote:
> 
> Without that settings it takes upto hours to update the MWI information
> on the phone (even if I set registration time to 1 min).
> 
> Best regards
> Peter
> 
> 
> 
> On 12/19/17 16:37, Michael Jerris wrote:
>> without that setting, when sending a voicemail to the user right after they register?
>> 
>>> On Dec 19, 2017, at 10:26 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>> 
>>> Hello Michael,
>>> 
>>> with parameter “send-message-query-on-register“ in internal sofia profile it works during registration, without this parameter is does not.
>>> 
>>> Best regards
>>> Peter
>>> 
>>> 
>>> 
>>> On 12/19/17 16:09, Michael Jerris wrote:
>>>> if you do a voicemail to the user very soon after it registers (within seconds) does it work?
>>>> 
>>>>> On Dec 19, 2017, at 6:49 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>>>> 
>>>>> I have a Yealink T46G registered at our Freeswitch machine. MWI is
>>>>> enabled on the Yealink Phone, I see a "message-summary" entry in
>>>>> sip_subscriptions. So I think, presence is setup fine
>>>>> 
>>>>> Content of the sip_subscriptions table:
>>>>> "proto" "sip_user" "sip_host" "sub_to_user" "sub_to_host"
>>>>> "presence_hosts" "event" "contact" "call_id" "full_from" "full_via"
>>>>> "expires" "user_agent" "accept" "profile_name" "hostname" "network_port"
>>>>> "network_ip" "version" "orig_proto" "full_to" "id"
>>>>> "sip" "200" "fs00.my.domain" "200" "fs00.my.domain" "" "message-summary"
>>>>> """Peter3 Buero"" <sip:200 at 192.168.1.72:5060>"
>>>>> "0_3056780279 at 192.168.1.72" """Peter3 Buero""
>>>>> <sip:200 at fs00.my.domain:5060>;tag=1082071637" "SIP/2.0/UDP
>>>>> 192.168.1.72:5060;branch=z9hG4bK2245575863" "1513683832" "Yealink
>>>>> SIP-T46G 28.80.0.130" "application/simple-message-summary " "internal"
>>>>> "fs01.my.domain" "5060" "192.168.1.72" "-1" ""
>>>>> "<sip:200 at fs00.my.domain:5060>;tag=WdayfEpssASN" "2821206"
>>>>> 
>>>>> Events are enabled on the freeswitch console
>>>>> /events MESSAGE_QUERY MESSAGE_WAITING
>>>>> 
>>>>> When a voicemail is recorded, I see a number of recurring events on the
>>>>> console (see below), but no NOTIFY message is sent to the phone.
>>>>> However, after a while (after 10min or even after hours) the phone
>>>>> receives its message.
>>>>> When I delete all VM messages, the MWI is never sent to the phone,
>>>>> although messages are prepared (MWI-Messages-Waiting: no), as I can see
>>>>> them on the console.
>>>>> 
>>>>> If I set the parameter “send-message-query-on-register“ in internal
>>>>> sofia profile, then MWI is updated with every register to the phone.
>>>>> Alos on every restart of the phone. But as we cannot use short intervals
>>>>> intervals due to some reason, this is not the solution.
>>>>> 
>>>>> Anybody has a hint, how to solve this?
>>>>> 
>>>>> 
>>>>> Example messages:
>>>>> RECV EVENT
>>>>> Event-Name: MESSAGE_QUERY
>>>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>>>> FreeSWITCH-Hostname: fs01.my.domain
>>>>> FreeSWITCH-Switchname: fs01.my.domain
>>>>> FreeSWITCH-IPv4: 192.168.1.9
>>>>> FreeSWITCH-IPv6: ::1
>>>>> Event-Date-Local: 2017-12-19 12:27:40
>>>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>>>> Event-Date-Timestamp: 1513682860739342
>>>>> Event-Calling-File: sofia_presence.c
>>>>> Event-Calling-Function: sofia_presence_sub_reg_callback
>>>>> Event-Calling-Line-Number: 1736
>>>>> Event-Sequence: 16929
>>>>> Message-Account: sip:200 at fs00.my.domain
>>>>> VM-Sofia-Profile: internal
>>>>> VM-sub-call-id: 0_3056780279 at 192.168.1.72
>>>>> 
>>>>> 
>>>>> RECV EVENT
>>>>> Event-Name: MESSAGE_WAITING
>>>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>>>> FreeSWITCH-Hostname: fs01.my.domain
>>>>> FreeSWITCH-Switchname: fs01.my.domain
>>>>> FreeSWITCH-IPv4: 192.168.1.9
>>>>> FreeSWITCH-IPv6: ::1
>>>>> Event-Date-Local: 2017-12-19 12:27:40
>>>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>>>> Event-Date-Timestamp: 1513682860899354
>>>>> Event-Calling-File: mod_voicemail.c
>>>>> Event-Calling-Function: actual_message_query_handler
>>>>> Event-Calling-Line-Number: 4036
>>>>> Event-Sequence: 16931
>>>>> MWI-Messages-Waiting: yes
>>>>> MWI-Message-Account: sip:200 at fs00.my.domain
>>>>> MWI-Voice-Message: 1/0 (0/0)
>>>>> Sofia-Profile: internal
>>>>> sub-call-id: 0_3056780279 at 192.168.1.72
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> -- 
> With kind regards
> Peter Steinbach 
> 
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From wirefastny at gmail.com  Wed Dec 20 11:13:13 2017
From: wirefastny at gmail.com (Neil Youngman)
Date: Wed, 20 Dec 2017 11:13:13 +0000
Subject: [Freeswitch-users] Getting started on Mint 17.3
Message-ID: <CAHi=CNoa7vBgWg34023-E4yi07Z5JasExr_yTDrXNhDQg4-MZw@mail.gmail.com>

I had a few issues following the Debian "first steps" on Mint 17.3. Some
issues were definitely specific to Mint 17.3, others probably apply
elsewhere as well.

Firstly after the apt-get install, the guide suggests that Freeswitch
should be up and running. It wasn't.

Attempting to start Freeswitch from the command line I got an error telling
me Freeswitch wasn't configured and suggesting a command to copy the
vanilla configuration. Unfortunately the suggested command only works if
the /etc/freeswitch directory doesn't exist, but it did exist and instead
of creating a copy of the vanilla configuration in /etc/freeswitch, it
created a /etc/freeswitch/vanilla subdirectory and Freeswitch still
wouldn't start.

There don't seem to be any suggested softphones for Linux. It seemed likely
that linphone would run in Linux, however the version of linphone that is
provided in the Mint 17.3 repositories has a known bug and the current
version, installed from PPA crashes when I try to dial.

After some digging around for alternatives I tried Ekiga, but I was unable
to get that to talk to Freeswitch.

In the end I was able to install linphone from PPA on a Mint 18 laptop and
that enabled me to do a minimal smoke test.

I'm mostly posting this for information in case anyone else hits the same
issues, but I would be happy to help with tweaking the "first steps" page.

Should I post a bug for the suggested configuration copy command not always
being suitable?

If anyone knows the correct way to set up ekiga it would be good to have
that information.

Neil Youngman
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From alihaider.4189 at gmail.com  Wed Dec 20 17:04:15 2017
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Wed, 20 Dec 2017 22:04:15 +0500
Subject: [Freeswitch-users] =?utf-8?q?=28no_subject=29?=
Message-ID: <CAB2bS9TdO-qi9Mg9YAofDXO6dFjgCcPHp1vd0f5MBTDLCn_WpA@mail.gmail.com>

hiiii everyone
any one expert of freeswitch
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From krice at freeswitch.org  Wed Dec 20 17:11:43 2017
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 20 Dec 2017 11:11:43 -0600
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAB2bS9TdO-qi9Mg9YAofDXO6dFjgCcPHp1vd0f5MBTDLCn_WpA@mail.gmail.com>
References: <CAB2bS9TdO-qi9Mg9YAofDXO6dFjgCcPHp1vd0f5MBTDLCn_WpA@mail.gmail.com>
Message-ID: <291201d379b5$9c117e70$d4347b50$@freeswitch.org>

Yes there are many FreeSWITCH experts on this list, but it is best to actually ask a question, or start at https://freeswitch.org/confluence  then ask a question.

 

If you need professional help you can always email consulting at freeswitch.org <mailto:consulting at freeswitch.org>  

 

K

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ali Haider
Sent: Wednesday, December 20, 2017 11:04 AM
To: FreeSWITCH-users at lists.freeswitch.org
Subject: [Freeswitch-users] (no subject)

 

hiiii everyone

any one expert of freeswitch

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From alihaider.4189 at gmail.com  Wed Dec 20 17:14:21 2017
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Wed, 20 Dec 2017 22:14:21 +0500
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <291201d379b5$9c117e70$d4347b50$@freeswitch.org>
References: <CAB2bS9TdO-qi9Mg9YAofDXO6dFjgCcPHp1vd0f5MBTDLCn_WpA@mail.gmail.com>
 <291201d379b5$9c117e70$d4347b50$@freeswitch.org>
Message-ID: <CAB2bS9S5rAzeCRJoyY9FODYiwyFRki=kyRNFLcrKuik5GnX6tA@mail.gmail.com>

can you help me or give me code for vice and video calling from laptop to
mobile

On 20 December 2017 at 22:11, Ken Rice <krice at freeswitch.org> wrote:

> Yes there are many FreeSWITCH experts on this list, but it is best to
> actually ask a question, or start at https://freeswitch.org/confluence
> then ask a question.
>
>
>
> If you need professional help you can always email
> consulting at freeswitch.org
>
>
>
> K
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Ali Haider
> *Sent:* Wednesday, December 20, 2017 11:04 AM
> *To:* FreeSWITCH-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] (no subject)
>
>
>
> hiiii everyone
>
> any one expert of freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Dec 20 17:49:10 2017
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Dec 2017 12:49:10 -0500
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAB2bS9S5rAzeCRJoyY9FODYiwyFRki=kyRNFLcrKuik5GnX6tA@mail.gmail.com>
References: <CAB2bS9TdO-qi9Mg9YAofDXO6dFjgCcPHp1vd0f5MBTDLCn_WpA@mail.gmail.com>
 <291201d379b5$9c117e70$d4347b50$@freeswitch.org>
 <CAB2bS9S5rAzeCRJoyY9FODYiwyFRki=kyRNFLcrKuik5GnX6tA@mail.gmail.com>
Message-ID: <CAF18B1D-2D1B-4E22-84AE-748A47AB381D@jerris.com>

Yes FreeSWITCH is capable of these sorts of features.  You can use the code subject to their open source licenses.  Check out the installation pages on confluence as Ken pointed out for where to get started.  Also there are some FreeSWITCH books that might help you get a good start on this.

Mike


> On Dec 20, 2017, at 12:14 PM, Ali Haider <alihaider.4189 at gmail.com> wrote:
> 
> can you help me or give me code for vice and video calling from laptop to mobile
> 
> On 20 December 2017 at 22:11, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
> Yes there are many FreeSWITCH experts on this list, but it is best to actually ask a question, or start at https://freeswitch.org/confluence <https://freeswitch.org/confluence>  then ask a question.
> 
>  If you need professional help you can always email consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Ali Haider
> 
> Sent: Wednesday, December 20, 2017 11:04 AM
> To: FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] (no subject)
> 
>  
> 
> hiiii everyone
> 
> any one expert of freeswitch
> 

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From alihaider.4189 at gmail.com  Wed Dec 20 18:29:01 2017
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Wed, 20 Dec 2017 23:29:01 +0500
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAF18B1D-2D1B-4E22-84AE-748A47AB381D@jerris.com>
References: <CAB2bS9TdO-qi9Mg9YAofDXO6dFjgCcPHp1vd0f5MBTDLCn_WpA@mail.gmail.com>
 <291201d379b5$9c117e70$d4347b50$@freeswitch.org>
 <CAB2bS9S5rAzeCRJoyY9FODYiwyFRki=kyRNFLcrKuik5GnX6tA@mail.gmail.com>
 <CAF18B1D-2D1B-4E22-84AE-748A47AB381D@jerris.com>
Message-ID: <CAB2bS9Q+=q3QcE9cDgo52EeBNCr9cqZSQB5TokpVxhcNimYqxQ@mail.gmail.com>

which book is more help me any name and edition

On 20 December 2017 at 22:49, Michael Jerris <mike at jerris.com> wrote:

> Yes FreeSWITCH is capable of these sorts of features.  You can use the
> code subject to their open source licenses.  Check out the installation
> pages on confluence as Ken pointed out for where to get started.  Also
> there are some FreeSWITCH books that might help you get a good start on
> this.
>
> Mike
>
>
> On Dec 20, 2017, at 12:14 PM, Ali Haider <alihaider.4189 at gmail.com> wrote:
>
> can you help me or give me code for vice and video calling from laptop to
> mobile
>
> On 20 December 2017 at 22:11, Ken Rice <krice at freeswitch.org> wrote:
>
>> Yes there are many FreeSWITCH experts on this list, but it is best to
>> actually ask a question, or start at https://freeswitch.org/confluence
>> then ask a question.
>>
>>  If you need professional help you can always email
>> consulting at freeswitch.org
>>
>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
>> es at lists.freeswitch.org] *On Behalf Of *Ali Haider
>>
>> *Sent:* Wednesday, December 20, 2017 11:04 AM
>> *To:* FreeSWITCH-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] (no subject)
>>
>>
>>
>> hiiii everyone
>>
>> any one expert of freeswitch
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at gmail.com  Wed Dec 20 18:31:08 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 20 Dec 2017 19:31:08 +0100
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAB2bS9Q+=q3QcE9cDgo52EeBNCr9cqZSQB5TokpVxhcNimYqxQ@mail.gmail.com>
References: <CAB2bS9TdO-qi9Mg9YAofDXO6dFjgCcPHp1vd0f5MBTDLCn_WpA@mail.gmail.com>
 <291201d379b5$9c117e70$d4347b50$@freeswitch.org>
 <CAB2bS9S5rAzeCRJoyY9FODYiwyFRki=kyRNFLcrKuik5GnX6tA@mail.gmail.com>
 <CAF18B1D-2D1B-4E22-84AE-748A47AB381D@jerris.com>
 <CAB2bS9Q+=q3QcE9cDgo52EeBNCr9cqZSQB5TokpVxhcNimYqxQ@mail.gmail.com>
Message-ID: <CALXCt0reg-U5TG1L5fxWLamP5-gWJM+-+zCiKiajOJBSNztvAA@mail.gmail.com>

You can find all FreeSWITCH books here: http://bfy.tw/Ffkv



On 20 December 2017 at 19:29, Ali Haider <alihaider.4189 at gmail.com> wrote:

> which book is more help me any name and edition
>
> On 20 December 2017 at 22:49, Michael Jerris <mike at jerris.com> wrote:
>
>> Yes FreeSWITCH is capable of these sorts of features.  You can use the
>> code subject to their open source licenses.  Check out the installation
>> pages on confluence as Ken pointed out for where to get started.  Also
>> there are some FreeSWITCH books that might help you get a good start on
>> this.
>>
>> Mike
>>
>>
>> On Dec 20, 2017, at 12:14 PM, Ali Haider <alihaider.4189 at gmail.com>
>> wrote:
>>
>> can you help me or give me code for vice and video calling from laptop to
>> mobile
>>
>> On 20 December 2017 at 22:11, Ken Rice <krice at freeswitch.org> wrote:
>>
>>> Yes there are many FreeSWITCH experts on this list, but it is best to
>>> actually ask a question, or start at https://freeswitch.org/confluence
>>> then ask a question.
>>>
>>>  If you need professional help you can always email
>>> consulting at freeswitch.org
>>>
>>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc
>>> es at lists.freeswitch.org] *On Behalf Of *Ali Haider
>>>
>>> *Sent:* Wednesday, December 20, 2017 11:04 AM
>>> *To:* FreeSWITCH-users at lists.freeswitch.org
>>> *Subject:* [Freeswitch-users] (no subject)
>>>
>>>
>>>
>>> hiiii everyone
>>>
>>> any one expert of freeswitch
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From infos at madovsky.org  Wed Dec 20 18:59:20 2017
From: infos at madovsky.org (Madovsky)
Date: Wed, 20 Dec 2017 10:59:20 -0800
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAB2bS9Q+=q3QcE9cDgo52EeBNCr9cqZSQB5TokpVxhcNimYqxQ@mail.gmail.com>
References: <CAB2bS9TdO-qi9Mg9YAofDXO6dFjgCcPHp1vd0f5MBTDLCn_WpA@mail.gmail.com>
 <291201d379b5$9c117e70$d4347b50$@freeswitch.org>
 <CAB2bS9S5rAzeCRJoyY9FODYiwyFRki=kyRNFLcrKuik5GnX6tA@mail.gmail.com>
 <CAF18B1D-2D1B-4E22-84AE-748A47AB381D@jerris.com>
 <CAB2bS9Q+=q3QcE9cDgo52EeBNCr9cqZSQB5TokpVxhcNimYqxQ@mail.gmail.com>
Message-ID: <933c24f2-5822-c73a-57af-cf3bbda07e2f@madovsky.org>

https://www.amazon.com/FreeSWITCH-1-6-Cookbook-Anthony-Minessale/dp/1785280910/ref=pd_sbs_14_1?_encoding=UTF8&pd_rd_i=1785280910&pd_rd_r=3PPFV4HAD22NK4V8GWDR&pd_rd_w=1S2PO&pd_rd_wg=ODvrr&psc=1&refRID=3PPFV4HAD22NK4V8GWDR


On 12/20/2017 10:29 AM, Ali Haider wrote:
> which book is more help me any name and edition
>
> On 20 December 2017 at 22:49, Michael Jerris <mike at jerris.com 
> <mailto:mike at jerris.com>> wrote:
>
>     Yes FreeSWITCH is capable of these sorts of features.  You can use
>     the code subject to their open source licenses.  Check out the
>     installation pages on confluence as Ken pointed out for where to
>     get started.  Also there are some FreeSWITCH books that might help
>     you get a good start on this.
>
>     Mike
>
>
>>     On Dec 20, 2017, at 12:14 PM, Ali Haider
>>     <alihaider.4189 at gmail.com <mailto:alihaider.4189 at gmail.com>> wrote:
>>
>>     can you help me or give me code for vice and video calling from
>>     laptop to mobile
>>
>>     On 20 December 2017 at 22:11, Ken Rice <krice at freeswitch.org
>>     <mailto:krice at freeswitch.org>> wrote:
>>
>>         Yes there are many FreeSWITCH experts on this list, but it is
>>         best to actually ask a question, or start at
>>         https://freeswitch.org/confluence
>>         <https://freeswitch.org/confluence> then ask a question.
>>
>>          If you need professional help you can always email
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>
>>         *From:* FreeSWITCH-users
>>         [mailto:freeswitch-users-bounces at lists.freeswitch.org
>>         <mailto:freeswitch-users-bounces at lists.freeswitch.org>] *On
>>         Behalf Of *Ali Haider
>>
>>         *Sent:* Wednesday, December 20, 2017 11:04 AM
>>         *To:* FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         *Subject:* [Freeswitch-users] (no subject)
>>
>>         hiiii everyone
>>
>>         any one expert of freeswitch
>>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From gregor at infomedia.si  Wed Dec 20 19:11:39 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Wed, 20 Dec 2017 19:11:39 +0000
Subject: [Freeswitch-users] File transoding
Message-ID: <CAC7ttFG-6eeDuPbbQc-GxZhY5gDG5-eMJvreWGjtk=5VTmzL9Q@mail.gmail.com>

Hi, guys!

I have file for playback in ulaw format. If call offers only alaw, will
file be transcoded or there will be error?

Best regards, Gregor
-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From alihaider.4189 at gmail.com  Wed Dec 20 20:22:13 2017
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Thu, 21 Dec 2017 01:22:13 +0500
Subject: [Freeswitch-users] how can i sortout these error when bulid on vs
	2015
Message-ID: <CAB2bS9Q=FkZObczONpzRUJwvuVk=s-_dJrCSDGmxVzCZF3dc-A@mail.gmail.com>

hiiiii
i got 85 error when bulid on vs 2015 how to fix these error help me






Severity Code Description Project File Line Suppression State
Error MSB4057 The target "v8:Rebuild" does not exist in the project.
[C:\FS_GIT\libs\v8-5.6.326\src\v8.sln] libv8
C:\FS_GIT\libs\v8-5.6.326\src\v8.sln.metaproj 1
Error C1001 An internal error has occurred in the compiler. libteletone
C:\FS_GIT\libs\libteletone\src\libteletone_generate.h 116
Error C1001 An internal error has occurred in the compiler. libpng
C:\FS_GIT\libs\win32\libpng\LINK 1
Error LNK1000 Internal error during IMAGE::Pass1 libpng
C:\FS_GIT\libs\win32\libpng\Win32\Debug\png.obj 1
Error MSB4057 The target "v8:Rebuild" does not exist in the project.
[C:\FS_GIT\libs\v8-5.6.326\src\v8.sln] libv8
C:\FS_GIT\libs\v8-5.6.326\src\v8.sln.metaproj 1
Error MSB3721 The command "C:\FS_GIT\libs\vsyasm.exe -Xvc -f Win32 -i
"C:\FS_GIT\libs\libx264\common\x86" -d "PREFIX" -d "STACK_ALIGNMENT=4" -d
"HIGH_BIT_DEPTH=0" -d "BIT_DEPTH=8" -d "WIN32=1" -d "ARCH_X86_64=0" -o
"Win32\Debug\\" -rnasm -pnasm   "..\..\libx264\common\x86\bitstream-a.asm"
"..\..\libx264\common\x86\cabac-a.asm"
"..\..\libx264\common\x86\const-a.asm"
"..\..\libx264\common\x86\cpu-a.asm"
"..\..\libx264\common\x86\dct-32.asm"
"..\..\libx264\common\x86\dct-a.asm"
"..\..\libx264\common\x86\deblock-a.asm"
"..\..\libx264\common\x86\mc-a.asm"  "..\..\libx264\common\x86\mc-a2.asm"
"..\..\libx264\common\x86\pixel-32.asm"
"..\..\libx264\common\x86\pixel-a.asm"
"..\..\libx264\common\x86\predict-a.asm"
"..\..\libx264\common\x86\quant-a.asm"
"..\..\libx264\common\x86\sad-a.asm"" exited with code 1. libx264
C:\FS_GIT\libs\win32\libx264\vsyasm.targets 45
Error C2737 'std::is_same_v': 'constexpr' object must be initialized
gsmlib C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\xtr1common 100
Error C2998 'const bool std::is_same_v': cannot be a template definition
gsmlib C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xtr1common
100
Error C2737 'std::is_integral_v': 'constexpr' object must be initialized
gsmlib C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xtr1common
245
Error C2998 'const bool std::is_integral_v': cannot be a template definition
gsmlib C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xtr1common
245
Error C2737 'std::is_floating_point_v': 'constexpr' object must be
initialized gsmlib C:\Program Files\Microsoft Visual Studio
14.0\VC\include\xtr1common 282
Error C2998 'const bool std::is_floating_point_v': cannot be a template
definition gsmlib C:\Program Files\Microsoft Visual Studio
14.0\VC\include\xtr1common 282
Error C2737 'std::is_arithmetic_v': 'constexpr' object must be initialized
gsmlib C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xtr1common
295
Error C2998 'const bool std::is_arithmetic_v': cannot be a template
definition gsmlib C:\Program Files\Microsoft Visual Studio
14.0\VC\include\xtr1common 295
Error C2737 'std::is_function_v': 'constexpr' object must be initialized
gsmlib C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xstddef 697
Error C2998 'const bool std::is_function_v': cannot be a template definition
gsmlib C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xstddef 697
Error C2275 '_To': illegal use of this type as an expression gsmlib C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 524
Error C2275 '_From': illegal use of this type as an expression gsmlib
C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 524
Error C3861 '__is_assignable': identifier not found gsmlib C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 524
Error C2975 '_Val': invalid template argument for 'std::integral_constant',
expected compile-time constant expression gsmlib C:\Program Files\Microsoft
Visual Studio 14.0\VC\include\type_traits 524
Error C2061 syntax error: identifier '__make_integer_seq' gsmlib C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 1245
Error C2065 '_Vals': undeclared identifier gsmlib C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 1249
Error C2975 '_Vals': invalid template argument for 'std::integer_sequence',
expected compile-time constant expression gsmlib C:\Program Files\Microsoft
Visual Studio 14.0\VC\include\type_traits 1249
Error C2061 syntax error: identifier 'make_integer_sequence' gsmlib C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 1252
Error C2631 'identity': a class or enum cannot be defined in an alias
template gsmlib C:\Program Files\Microsoft Visual Studio
14.0\VC\include\type_traits 1261
Error C1903 unable to recover from previous error(s); stopping compilation
gsmlib
C:\FS_GIT\src\mod\endpoints\mod_gsmopen\gsmlib\gsmlib-1.10-patched-13ubuntu\gsmlib\gsm_win32_serial.cc
1
Error C2737 'std::is_same_v': 'constexpr' object must be initialized
libcbt C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\xtr1common 100
Error C2998 'const bool std::is_same_v': cannot be a template definition
libcbt C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xtr1common
100
Error C2737 'std::is_integral_v': 'constexpr' object must be initialized
libcbt C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xtr1common
245
Error C2998 'const bool std::is_integral_v': cannot be a template definition
libcbt C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xtr1common
245
Error C2737 'std::is_floating_point_v': 'constexpr' object must be
initialized libcbt C:\Program Files\Microsoft Visual Studio
14.0\VC\include\xtr1common 282
Error C2998 'const bool std::is_floating_point_v': cannot be a template
definition libcbt C:\Program Files\Microsoft Visual Studio
14.0\VC\include\xtr1common 282
Error C2737 'std::is_arithmetic_v': 'constexpr' object must be initialized
libcbt C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xtr1common
295
Error C2998 'const bool std::is_arithmetic_v': cannot be a template
definition libcbt C:\Program Files\Microsoft Visual Studio
14.0\VC\include\xtr1common 295
Error C2737 'std::is_function_v': 'constexpr' object must be initialized
libcbt C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xstddef 697
Error C2998 'const bool std::is_function_v': cannot be a template definition
libcbt C:\Program Files\Microsoft Visual Studio 14.0\VC\include\xstddef 697
Error C2275 '_To': illegal use of this type as an expression libcbt C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 524
Error C2275 '_From': illegal use of this type as an expression libcbt
C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 524
Error C3861 '__is_assignable': identifier not found libcbt C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 524
Error C2975 '_Val': invalid template argument for 'std::integral_constant',
expected compile-time constant expression libcbt C:\Program Files\Microsoft
Visual Studio 14.0\VC\include\type_traits 524
Error C2061 syntax error: identifier '__make_integer_seq' libcbt C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 1245
Error C2065 '_Vals': undeclared identifier libcbt C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 1249
Error C2975 '_Vals': invalid template argument for 'std::integer_sequence',
expected compile-time constant expression libcbt C:\Program Files\Microsoft
Visual Studio 14.0\VC\include\type_traits 1249
Error C2061 syntax error: identifier 'make_integer_sequence' libcbt C:\Program
Files\Microsoft Visual Studio 14.0\VC\include\type_traits 1252
Error C2631 'identity': a class or enum cannot be defined in an alias
template libcbt C:\Program Files\Microsoft Visual Studio
14.0\VC\include\type_traits 1261
Error C1903 unable to recover from previous error(s); stopping compilation
libcbt C:\FS_GIT\src\mod\endpoints\mod_gsmopen\libctb-0.16\src\serportx.cpp
1
Error C1001 An internal error has occurred in the compiler. esl
C:\FS_GIT\libs\esl\src\esl.c 525
Error C1083 Cannot open include file: 'defines.h': No such file or directory
libcodec2 C:\FS_GIT\libs\libcodec2-2.59\src\codebookge.c 8
Error C1083 Cannot open include file: 'defines.h': No such file or directory
libcodec2 C:\FS_GIT\libs\libcodec2-2.59\src\codebookdt.c 8
Error C1083 Cannot open include file: 'defines.h': No such file or directory
libcodec2 C:\FS_GIT\libs\libcodec2-2.59\src\codebookvqanssi.c 8
Error C1083 Cannot open include file: 'defines.h': No such file or directory
libcodec2 C:\FS_GIT\libs\libcodec2-2.59\src\codebookjvm.c 8
Error C1083 Cannot open include file: 'defines.h': No such file or directory
libcodec2 C:\FS_GIT\libs\libcodec2-2.59\src\codebookjnd.c 8
Error C1083 Cannot open include file: 'defines.h': No such file or directory
libcodec2 C:\FS_GIT\libs\libcodec2-2.59\src\codebookvq.c 8
Error C1083 Cannot open include file: 'defines.h': No such file or directory
libcodec2 C:\FS_GIT\libs\libcodec2-2.59\src\codebookd.c 8
Error C1083 Cannot open include file: 'defines.h': No such file or directory
libcodec2 C:\FS_GIT\libs\libcodec2-2.59\src\codebook.c 8
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\pack.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\quantise.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\phase.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\lsp.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\interp.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\kiss_fft.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\fdmdv.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\fifo.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\codec2.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\sine.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\postfilter.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\nlp.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\lpc.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error C1083 Cannot open source file:
'C:\FS_GIT\libs\libcodec2-2.59\src\dump.c': No such file or directory
libcodec2 C:\FS_GIT\libs\win32\libcodec2\c1 1
Error LNK1104 cannot open file 'C:\FS_GIT\Win32\Debug\libteletone.lib'
FreeSwitchCoreLib C:\FS_GIT\w32\Library\LINK 1
Error C1001 An internal error has occurred in the compiler. mod_logfile
c:\fs_git\src\include\switch_loadable_module.h 533
Error C1001 An internal error has occurred in the compiler. mod_tone_stream
c:\fs_git\src\include\switch_loadable_module.h 533
Error C1001 An internal error has occurred in the compiler. mod_expr
c:\fs_git\src\include\switch_loadable_module.h 533
Error C1001 An internal error has occurred in the compiler. mod_snom
c:\fs_git\src\include\switch_loadable_module.h 533
Error LNK1181 cannot open input file
'C:\FS_GIT\Win32\Debug\FreeSwitchCore.lib' mod_shout
C:\FS_GIT\src\mod\formats\mod_shout\LINK 1
Error LNK1181 cannot open input file
'C:\FS_GIT\Win32\Debug\FreeSwitchCore.lib' mod_lua
C:\FS_GIT\src\mod\languages\mod_lua\LINK 1
Error C1001 An internal error has occurred in the compiler. mod_siren
c:\fs_git\src\include\switch_loadable_module.h 533
Error C1001 An internal error has occurred in the compiler. mod_say_zh
c:\fs_git\src\include\switch_loadable_module.h 533
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From lists at telefaks.de  Wed Dec 20 22:09:51 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 20 Dec 2017 23:09:51 +0100
Subject: [Freeswitch-users] MWI not sent
In-Reply-To: <956FAC69-094D-49D4-B298-537698EA51A1@jerris.com>
References: <5A38FCB4.5000105@telefaks.de>
 <6EA801C8-F2B8-45DE-95B7-FA9CE9CD64B5@jerris.com>
 <5A392F94.50002@telefaks.de>
 <07BAB46B-B5DC-4F5D-9F5A-2CEC2932167B@jerris.com>
 <5A3A2B5C.8060402@telefaks.de>
 <956FAC69-094D-49D4-B298-537698EA51A1@jerris.com>
Message-ID: <5A3ADFAF.4060406@telefaks.de>

We have several internal SIP profiles, internal, internalnat,
internalvpn which belong to the same default context. However in our
case, all phones which belong to the same domain, are registered to
internal profile.
Does this matter?

Best regards
Peter

On 12/20/17 17:30, Michael Jerris wrote:
> Do you have multiple sip profiles that something at that domain is registered to?
>
>> On Dec 20, 2017, at 4:20 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>
>> Without that settings it takes upto hours to update the MWI information
>> on the phone (even if I set registration time to 1 min).
>>
>> Best regards
>> Peter
>>
>>
>>
>> On 12/19/17 16:37, Michael Jerris wrote:
>>> without that setting, when sending a voicemail to the user right after they register?
>>>
>>>> On Dec 19, 2017, at 10:26 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>>>
>>>> Hello Michael,
>>>>
>>>> with parameter “send-message-query-on-register“ in internal sofia profile it works during registration, without this parameter is does not.
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>>
>>>>
>>>> On 12/19/17 16:09, Michael Jerris wrote:
>>>>> if you do a voicemail to the user very soon after it registers (within seconds) does it work?
>>>>>
>>>>>> On Dec 19, 2017, at 6:49 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>>>>>
>>>>>> I have a Yealink T46G registered at our Freeswitch machine. MWI is
>>>>>> enabled on the Yealink Phone, I see a "message-summary" entry in
>>>>>> sip_subscriptions. So I think, presence is setup fine
>>>>>>
>>>>>> Content of the sip_subscriptions table:
>>>>>> "proto" "sip_user" "sip_host" "sub_to_user" "sub_to_host"
>>>>>> "presence_hosts" "event" "contact" "call_id" "full_from" "full_via"
>>>>>> "expires" "user_agent" "accept" "profile_name" "hostname" "network_port"
>>>>>> "network_ip" "version" "orig_proto" "full_to" "id"
>>>>>> "sip" "200" "fs00.my.domain" "200" "fs00.my.domain" "" "message-summary"
>>>>>> """Peter3 Buero"" <sip:200 at 192.168.1.72:5060>"
>>>>>> "0_3056780279 at 192.168.1.72" """Peter3 Buero""
>>>>>> <sip:200 at fs00.my.domain:5060>;tag=1082071637" "SIP/2.0/UDP
>>>>>> 192.168.1.72:5060;branch=z9hG4bK2245575863" "1513683832" "Yealink
>>>>>> SIP-T46G 28.80.0.130" "application/simple-message-summary " "internal"
>>>>>> "fs01.my.domain" "5060" "192.168.1.72" "-1" ""
>>>>>> "<sip:200 at fs00.my.domain:5060>;tag=WdayfEpssASN" "2821206"
>>>>>>
>>>>>> Events are enabled on the freeswitch console
>>>>>> /events MESSAGE_QUERY MESSAGE_WAITING
>>>>>>
>>>>>> When a voicemail is recorded, I see a number of recurring events on the
>>>>>> console (see below), but no NOTIFY message is sent to the phone.
>>>>>> However, after a while (after 10min or even after hours) the phone
>>>>>> receives its message.
>>>>>> When I delete all VM messages, the MWI is never sent to the phone,
>>>>>> although messages are prepared (MWI-Messages-Waiting: no), as I can see
>>>>>> them on the console.
>>>>>>
>>>>>> If I set the parameter “send-message-query-on-register“ in internal
>>>>>> sofia profile, then MWI is updated with every register to the phone.
>>>>>> Alos on every restart of the phone. But as we cannot use short intervals
>>>>>> intervals due to some reason, this is not the solution.
>>>>>>
>>>>>> Anybody has a hint, how to solve this?
>>>>>>
>>>>>>
>>>>>> Example messages:
>>>>>> RECV EVENT
>>>>>> Event-Name: MESSAGE_QUERY
>>>>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>>>>> FreeSWITCH-Hostname: fs01.my.domain
>>>>>> FreeSWITCH-Switchname: fs01.my.domain
>>>>>> FreeSWITCH-IPv4: 192.168.1.9
>>>>>> FreeSWITCH-IPv6: ::1
>>>>>> Event-Date-Local: 2017-12-19 12:27:40
>>>>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>>>>> Event-Date-Timestamp: 1513682860739342
>>>>>> Event-Calling-File: sofia_presence.c
>>>>>> Event-Calling-Function: sofia_presence_sub_reg_callback
>>>>>> Event-Calling-Line-Number: 1736
>>>>>> Event-Sequence: 16929
>>>>>> Message-Account: sip:200 at fs00.my.domain
>>>>>> VM-Sofia-Profile: internal
>>>>>> VM-sub-call-id: 0_3056780279 at 192.168.1.72
>>>>>>
>>>>>>
>>>>>> RECV EVENT
>>>>>> Event-Name: MESSAGE_WAITING
>>>>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>>>>> FreeSWITCH-Hostname: fs01.my.domain
>>>>>> FreeSWITCH-Switchname: fs01.my.domain
>>>>>> FreeSWITCH-IPv4: 192.168.1.9
>>>>>> FreeSWITCH-IPv6: ::1
>>>>>> Event-Date-Local: 2017-12-19 12:27:40
>>>>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>>>>> Event-Date-Timestamp: 1513682860899354
>>>>>> Event-Calling-File: mod_voicemail.c
>>>>>> Event-Calling-Function: actual_message_query_handler
>>>>>> Event-Calling-Line-Number: 4036
>>>>>> Event-Sequence: 16931
>>>>>> MWI-Messages-Waiting: yes
>>>>>> MWI-Message-Account: sip:200 at fs00.my.domain
>>>>>> MWI-Voice-Message: 1/0 (0/0)
>>>>>> Sofia-Profile: internal
>>>>>> sub-call-id: 0_3056780279 at 192.168.1.72
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> -- 
>> With kind regards
>> Peter Steinbach 
>>
>> Telefaks Services GmbH
>> mailto:lists (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de



From mike at jerris.com  Wed Dec 20 22:33:21 2017
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Dec 2017 17:33:21 -0500
Subject: [Freeswitch-users] MWI not sent
In-Reply-To: <5A3ADFAF.4060406@telefaks.de>
References: <5A38FCB4.5000105@telefaks.de>
 <6EA801C8-F2B8-45DE-95B7-FA9CE9CD64B5@jerris.com>
 <5A392F94.50002@telefaks.de>
 <07BAB46B-B5DC-4F5D-9F5A-2CEC2932167B@jerris.com>
 <5A3A2B5C.8060402@telefaks.de>
 <956FAC69-094D-49D4-B298-537698EA51A1@jerris.com>
 <5A3ADFAF.4060406@telefaks.de>
Message-ID: <DD7323EB-6D38-4F96-9EE6-35367AEF2375@jerris.com>

It was a guess at what the problem might be, but isn’t based on your response.

> On Dec 20, 2017, at 5:09 PM, Peter Steinbach <lists at telefaks.de> wrote:
> 
> We have several internal SIP profiles, internal, internalnat,
> internalvpn which belong to the same default context. However in our
> case, all phones which belong to the same domain, are registered to
> internal profile.
> Does this matter?
> 
> Best regards
> Peter
> 
> On 12/20/17 17:30, Michael Jerris wrote:
>> Do you have multiple sip profiles that something at that domain is registered to?
>> 
>>> On Dec 20, 2017, at 4:20 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>> 
>>> Without that settings it takes upto hours to update the MWI information
>>> on the phone (even if I set registration time to 1 min).
>>> 
>>> Best regards
>>> Peter
>>> 
>>> 
>>> 
>>> On 12/19/17 16:37, Michael Jerris wrote:
>>>> without that setting, when sending a voicemail to the user right after they register?
>>>> 
>>>>> On Dec 19, 2017, at 10:26 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>>>> 
>>>>> Hello Michael,
>>>>> 
>>>>> with parameter “send-message-query-on-register“ in internal sofia profile it works during registration, without this parameter is does not.
>>>>> 
>>>>> Best regards
>>>>> Peter
>>>>> 
>>>>> 
>>>>> 
>>>>> On 12/19/17 16:09, Michael Jerris wrote:
>>>>>> if you do a voicemail to the user very soon after it registers (within seconds) does it work?
>>>>>> 
>>>>>>> On Dec 19, 2017, at 6:49 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>>>>>> 
>>>>>>> I have a Yealink T46G registered at our Freeswitch machine. MWI is
>>>>>>> enabled on the Yealink Phone, I see a "message-summary" entry in
>>>>>>> sip_subscriptions. So I think, presence is setup fine
>>>>>>> 
>>>>>>> Content of the sip_subscriptions table:
>>>>>>> "proto" "sip_user" "sip_host" "sub_to_user" "sub_to_host"
>>>>>>> "presence_hosts" "event" "contact" "call_id" "full_from" "full_via"
>>>>>>> "expires" "user_agent" "accept" "profile_name" "hostname" "network_port"
>>>>>>> "network_ip" "version" "orig_proto" "full_to" "id"
>>>>>>> "sip" "200" "fs00.my.domain" "200" "fs00.my.domain" "" "message-summary"
>>>>>>> """Peter3 Buero"" <sip:200 at 192.168.1.72:5060>"
>>>>>>> "0_3056780279 at 192.168.1.72" """Peter3 Buero""
>>>>>>> <sip:200 at fs00.my.domain:5060>;tag=1082071637" "SIP/2.0/UDP
>>>>>>> 192.168.1.72:5060;branch=z9hG4bK2245575863" "1513683832" "Yealink
>>>>>>> SIP-T46G 28.80.0.130" "application/simple-message-summary " "internal"
>>>>>>> "fs01.my.domain" "5060" "192.168.1.72" "-1" ""
>>>>>>> "<sip:200 at fs00.my.domain:5060>;tag=WdayfEpssASN" "2821206"
>>>>>>> 
>>>>>>> Events are enabled on the freeswitch console
>>>>>>> /events MESSAGE_QUERY MESSAGE_WAITING
>>>>>>> 
>>>>>>> When a voicemail is recorded, I see a number of recurring events on the
>>>>>>> console (see below), but no NOTIFY message is sent to the phone.
>>>>>>> However, after a while (after 10min or even after hours) the phone
>>>>>>> receives its message.
>>>>>>> When I delete all VM messages, the MWI is never sent to the phone,
>>>>>>> although messages are prepared (MWI-Messages-Waiting: no), as I can see
>>>>>>> them on the console.
>>>>>>> 
>>>>>>> If I set the parameter “send-message-query-on-register“ in internal
>>>>>>> sofia profile, then MWI is updated with every register to the phone.
>>>>>>> Alos on every restart of the phone. But as we cannot use short intervals
>>>>>>> intervals due to some reason, this is not the solution.
>>>>>>> 
>>>>>>> Anybody has a hint, how to solve this?
>>>>>>> 
>>>>>>> 
>>>>>>> Example messages:
>>>>>>> RECV EVENT
>>>>>>> Event-Name: MESSAGE_QUERY
>>>>>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>>>>>> FreeSWITCH-Hostname: fs01.my.domain
>>>>>>> FreeSWITCH-Switchname: fs01.my.domain
>>>>>>> FreeSWITCH-IPv4: 192.168.1.9
>>>>>>> FreeSWITCH-IPv6: ::1
>>>>>>> Event-Date-Local: 2017-12-19 12:27:40
>>>>>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>>>>>> Event-Date-Timestamp: 1513682860739342
>>>>>>> Event-Calling-File: sofia_presence.c
>>>>>>> Event-Calling-Function: sofia_presence_sub_reg_callback
>>>>>>> Event-Calling-Line-Number: 1736
>>>>>>> Event-Sequence: 16929
>>>>>>> Message-Account: sip:200 at fs00.my.domain
>>>>>>> VM-Sofia-Profile: internal
>>>>>>> VM-sub-call-id: 0_3056780279 at 192.168.1.72
>>>>>>> 
>>>>>>> 
>>>>>>> RECV EVENT
>>>>>>> Event-Name: MESSAGE_WAITING
>>>>>>> Core-UUID: 1ec11171-65e9-49e5-a043-379951acfef1
>>>>>>> FreeSWITCH-Hostname: fs01.my.domain
>>>>>>> FreeSWITCH-Switchname: fs01.my.domain
>>>>>>> FreeSWITCH-IPv4: 192.168.1.9
>>>>>>> FreeSWITCH-IPv6: ::1
>>>>>>> Event-Date-Local: 2017-12-19 12:27:40
>>>>>>> Event-Date-GMT: Tue, 19 Dec 2017 11:27:40 GMT
>>>>>>> Event-Date-Timestamp: 1513682860899354
>>>>>>> Event-Calling-File: mod_voicemail.c
>>>>>>> Event-Calling-Function: actual_message_query_handler
>>>>>>> Event-Calling-Line-Number: 4036
>>>>>>> Event-Sequence: 16931
>>>>>>> MWI-Messages-Waiting: yes
>>>>>>> MWI-Message-Account: sip:200 at fs00.my.domain
>>>>>>> MWI-Voice-Message: 1/0 (0/0)
>>>>>>> Sofia-Profile: internal
>>>>>>> sub-call-id: 0_3056780279 at 192.168.1.72
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> -- 
>>> With kind regards
>>> Peter Steinbach 
>>> 
>>> Telefaks Services GmbH
>>> mailto:lists (att) telefaks.de
>>> Internet: www.telefaks.de
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> -- 
> With kind regards
> Peter Steinbach 
> 
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From cong.wang.itsherpa at gmail.com  Thu Dec 21 10:16:08 2017
From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=)
Date: Thu, 21 Dec 2017 19:16:08 +0900
Subject: [Freeswitch-users] About Video Call
Message-ID: <135AF557-923F-4F02-B8A3-6F073F714B9F@gmail.com>

Hey all, 

My FreeSWITCH server could both support audio-only call and video call, so is there any way for FS to know if a call is a video call? 

Regards.

From bilaln018 at gmail.com  Thu Dec 21 11:15:15 2017
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Thu, 21 Dec 2017 16:15:15 +0500
Subject: [Freeswitch-users] File transoding
In-Reply-To: <CAC7ttFG-6eeDuPbbQc-GxZhY5gDG5-eMJvreWGjtk=5VTmzL9Q@mail.gmail.com>
References: <CAC7ttFG-6eeDuPbbQc-GxZhY5gDG5-eMJvreWGjtk=5VTmzL9Q@mail.gmail.com>
Message-ID: <CADSVerB9vJ554wZ2xkHRXuM8UT_r8JJXYSGRPd1D8Aae_Jd05A@mail.gmail.com>

My experience was freeswitch does transcode that, but you will not be able
to hear proper file/media.

Regards
Abbasi

On Thu, Dec 21, 2017 at 12:11 AM, Gregor Nanger <gregor at infomedia.si> wrote:

> Hi, guys!
>
> I have file for playback in ulaw format. If call offers only alaw, will
> file be transcoded or there will be error?
>
> Best regards, Gregor
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Thu Dec 21 14:15:22 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 Dec 2017 08:15:22 -0600
Subject: [Freeswitch-users] File transoding
In-Reply-To: <CAC7ttFG-6eeDuPbbQc-GxZhY5gDG5-eMJvreWGjtk=5VTmzL9Q@mail.gmail.com>
References: <CAC7ttFG-6eeDuPbbQc-GxZhY5gDG5-eMJvreWGjtk=5VTmzL9Q@mail.gmail.com>
Message-ID: <CAKbxfG8Gpb1DHPky+MrWr59hGF3h+gHfOSt-NUm-7ZLwRXW7zg@mail.gmail.com>

If you play the file via mod_sndfile it will yes, this would require the
file extension .al
Best practice is to store WAV with raw pcm so there is only one transcoding
op not 2.


On Wed, Dec 20, 2017 at 1:11 PM, Gregor Nanger <gregor at infomedia.si> wrote:

> Hi, guys!
>
> I have file for playback in ulaw format. If call offers only alaw, will
> file be transcoded or there will be error?
>
> Best regards, Gregor
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II
Founder, FreeSWITCH.
http://freeswitch.com


https://youtu.be/l_hOxzCt6X4
https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From gregor at infomedia.si  Thu Dec 21 14:46:41 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 21 Dec 2017 15:46:41 +0100
Subject: [Freeswitch-users] File transoding
In-Reply-To: <CAKbxfG8Gpb1DHPky+MrWr59hGF3h+gHfOSt-NUm-7ZLwRXW7zg@mail.gmail.com>
References: <CAC7ttFG-6eeDuPbbQc-GxZhY5gDG5-eMJvreWGjtk=5VTmzL9Q@mail.gmail.com>
 <CAKbxfG8Gpb1DHPky+MrWr59hGF3h+gHfOSt-NUm-7ZLwRXW7zg@mail.gmail.com>
Message-ID: <CAC7ttFEMP0F35gs5o=q2D+N4U+A3J_8ZKcm-ahprf7gQzCn00Q@mail.gmail.com>

​Hmm, does playback appplication use mod_sndfile?​

2017-12-21 15:15 GMT+01:00 Anthony Minessale <anthony.minessale at gmail.com>:

> If you play the file via mod_sndfile it will yes, this would require the
> file extension .al
> Best practice is to store WAV with raw pcm so there is only one
> transcoding op not 2.
>
>
> On Wed, Dec 20, 2017 at 1:11 PM, Gregor Nanger <gregor at infomedia.si>
> wrote:
>
>> Hi, guys!
>>
>> I have file for playback in ulaw format. If call offers only alaw, will
>> file be transcoded or there will be error?
>>
>> Best regards, Gregor
>> --
>> Gregor Nanger
>>
>> *CTO*
>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>> • www.infomedia.si
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
> Founder, FreeSWITCH.
> http://freeswitch.com
>
>
> https://youtu.be/l_hOxzCt6X4
> https://www.youtube.com/watch?v=oAxXgyx5jUw
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From lists at telefaks.de  Thu Dec 21 18:41:01 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 21 Dec 2017 19:41:01 +0100
Subject: [Freeswitch-users] Lose Race instead of missed call
In-Reply-To: <5A3912DC.10903@telefaks.de>
References: <5A2EB2B5.9030907@telefaks.de> <5A3912DC.10903@telefaks.de>
Message-ID: <5A3C003D.1020505@telefaks.de>

After some tests, I figured out, that it's the enterprise originate
which prevents signalling of missed calls. As soon as I use "," instead
of ":_:", the signalling of missed calls is fine on the phones.

Question: Is there any reason for this behaviour, or should we consider
this as a defect and open a Jira?

Best regards
Peter


On 12/19/17 14:23, Peter Steinbach wrote:
> Hello.
>
> I did some further testiumng and figured out, that as soon I use
> enterprise originate with :_: , I do not see any missed calls. If I
> use a comma instead, missed calls are shown on the phone. But then as
> a drawback only one phone rings, if I have more than physical phone
> registered to one user extension.
>
> Anybody has an idea how to overcome this?
>
> Best rgeards
> Peter
>
> On 12/11/17 17:30, Peter Steinbach wrote:
>> Hello,
>>
>> we have some Problems with the right signalling of missed calls when
>> calling multiple phones in parallel
>>
>> Here's the scenario:
>> Phone no 49 is calling a group with 2170 and 3275 with the following
>> dialstring
>>
>> <action application="bridge"
>> data="{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59eee5e0-c0bc-0135-7dc9-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/2170 at flex.mydomain.de:_:{default_language=de,ignore_early_media=true,global_to_originate_1=true,caller_cc=49,callee_cc=49,routing_flags=INT-ONSYSTEM-RELIA0-QUAL0-T38NOLAST-CALLG-CALLG5003-UUID,call_timeout=60,originate_timeout=60,origination_caller_id_number=49,effective_caller_id_number=49,caller_uuid=7f49c581-cf47-429d-85a0-365d00ff031c,origination_uuid=59f07af0-c0bc-0135-7dca-00505600a1a5,sip_invite_domain=flex.mydomain.de,customer_id=261}user/3275 at flex.mydomain.de
>> <mailto:customer_id=261%7Duser/3275 at flex.mydomain.de>" />
>>
>> Destination dialstrings are seperated by ":_:" ("Enterprise
>> Origination"). We use curly brackets instead of "<" as we sometimes
>> have to insert asserted identy tags into the dialstring.
>>
>> We checked 3 versions of Freeswitch for this
>>
>>   * Version  Feb 2016 shows missed calls on both phones. Even if one
>>     phone answers, the other phone one still shows a missed call
>>     (reason for upgrading to newer Freeswitch)
>>   * Version  Aug 2017 never shows missed call, see logs and hangup
>>     message below
>>   * Version  10/Dec 2017(yesterday) never shows missed call, as above
>>
>> So for the 2 never Freeswitch Versions, here are the logs at hangup
>> 2017-12-11 17:07:46.022450 [DEBUG] sofia.c:7283 Channel
>> sofia/internal/49 at flex.mydomain.de:5060 entering state [terminated][487]
>> 2017-12-11 17:07:46.022450 [NOTICE] sofia.c:8474 Hangup
>> sofia/internal/*49*@flex.mydomain.de:5060 [CS_EXECUTE]
>> *[ORIGINATOR_CANCEL]*
>> 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627
>> Hangup sofia/internal/*2170*@94.xx.xxx.xx:42170 [CS_CONSUME_MEDIA]
>> *[LOSE_RACE]*
>> 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852
>> Originate Cancelled by originator termination Cause: 487
>> [ORIGINATOR_CANCEL]
>> 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866
>> Cannot create outgoing channel of type [user] cause: [LOSE_RACE]
>> 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863
>> Originate Resulted in Error Cause: 502 [LOSE_RACE]
>> 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:584
>> (sofia/internal/2170 at 94.xx.xxx.xx:42170) Running State Change
>> CS_HANGUP (Cur 3 Tot 22)
>> 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:850
>> (sofia/internal/2170 at 94.xx.xxx.xx:42170) Callstate Change RINGING ->
>> HANGUP
>> 2017-12-11 17:07:46.042331 [DEBUG] switch_core_state_machine.c:852
>> (sofia/internal/2170 at 94.xx.xxx.xx:42170) State HANGUP
>> 2017-12-11 17:07:46.042331 [DEBUG] mod_sofia.c:449 Channel
>> sofia/internal/2170 at 94.xx.xxx.xx:42170 hanging up, cause: LOSE_RACE
>> 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:3627
>> Hangup sofia/internal/*3275*@94.xx.xxx.xx:43275 [CS_CONSUME_MEDIA]
>> [*LOSE_RACE*]
>> 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3852
>> Originate Cancelled by originator termination Cause: 487
>> [ORIGINATOR_CANCEL]
>> 2017-12-11 17:07:46.042331 [NOTICE] switch_ivr_originate.c:2866
>> Cannot create outgoing channel of type [user] cause: [LOSE_RACE]
>> 2017-12-11 17:07:46.042331 [DEBUG] switch_ivr_originate.c:3863
>> Originate Resulted in Error Cause: 502 [LOSE_RACE]
>>
>> Here is the Cancel message for one of the called phones:
>> U 2017/12/11 17:07:46.045120 144.xx.xxx.xx:5060 -> 94.xx.xxx.xx:42170
>> *CANCEL sip:2170*@94.xx.xxx.xx:42170 SIP/2.0.
>> Via: SIP/2.0/UDP 144.xx.xxx.xx;rport;branch=z9hG4bK07r3ce94cvjpp.
>> Max-Forwards: 70.
>> From: "Test" <sip:49 at flex.mydomain.de>;tag=mQS1eFDpp1peS.
>> To: <sip:2170 at 94.xx.xxx.xx:42170>.
>> Call-ID: 4238b5a7-5930-1236-b2ab-00505600a1a5.
>> CSeq: 116168167 CANCEL.
>> *Reason: SIP;cause=200;text="Call completed elsewhere"*.
>> Content-Length: 0.
>>
>> Any hints why this happens, or anyone has this scenario working?
>>
>> -- 
>> With kind regards
>> Peter Steinbach 
>>
>> Telefaks Services GmbH
>> mailto:lists (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> -- 
> With kind regards
> Peter Steinbach 
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de

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From sdevoy at bizfocused.com  Fri Dec 22 03:02:26 2017
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Fri, 22 Dec 2017 03:02:26 +0000
Subject: [Freeswitch-users] VTECH VSP725 MWI
Message-ID: <DM5PR01MB26342AC6A6B1781DC75A882BBB020@DM5PR01MB2634.prod.exchangelabs.com>

Hi All,

Does anyone have any pointers about getting the Message Waiting Lamp to work on a VTECH VSP725?

This is our first VTECH SIP device and it seems fairly simple except for the MWI.  The only thing I have found by searching has been to add: <param name="manage-presence" value="true"/> which had no effect.

Thanks,
Sean
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From bipin at xbipin.com  Fri Dec 22 06:46:27 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 22 Dec 2017 10:46:27 +0400
Subject: [Freeswitch-users] Sofia stops responding after a few days
Message-ID: <5c53627a-9596-840e-efa0-f978f784fc7f@xbipin.com>

hi,

I have 2 instances of FS running on a single windows box, first instance 
uses normal sip UDP profiles mainly used for routing calls to carriers 
and its running as a service since a few months without any issues. The 
second instance runs a sip TLS profile and accepts inbound registrations 
and forwards calls in sip UDP ahead, it uses xml_curl for directory 
users but the problem is every 2-3 days the inbound TLS profile stops 
responding to registrations, and when that happens i cant even stop and 
restart the service, have to kill it and start again. FS_CLI works but 
doesnt show any error, at first i though it could be the xml_curl 
causing the issue but later realized it never sends any requests when 
sofia stops responding.

i have been banging my head from the past week or so but not able to 
find the cause, could any1 help in guiding me what to check when this 
happens so can find the root cause, im using commercial certs for TLS 
and when its running there r no issues other than sofia stops responding 
every few days, it happens at random times.


-- 
Regards,
Bipin


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From mjlopez at smartic.es  Fri Dec 22 08:50:52 2017
From: mjlopez at smartic.es (=?iso-8859-1?Q?Miguel_Jes=FAs_L=F3pez_Valverde?=)
Date: Fri, 22 Dec 2017 09:50:52 +0100
Subject: [Freeswitch-users] Sofia stops responding after a few days
In-Reply-To: <5c53627a-9596-840e-efa0-f978f784fc7f@xbipin.com>
References: <5c53627a-9596-840e-efa0-f978f784fc7f@xbipin.com>
Message-ID: <008c01d37b01$f96b9f40$ec42ddc0$@smartic.es>

I had similar problems with FS installed under an Amazon EFS instance. When
FS did not attend registration requests and executed the "sofia profile
internal restart" command, it did not load the profile and it no longer
appeared before the "sofia status" query.

 

I checked that this instance was short of ram memory and I changed the
instance to a higher one with more memory. Since then I have not appreciated
this problem again.

 

Receive a greeting.

 

De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org]
En nombre de Bipin Patel
Enviado el: viernes, 22 de diciembre de 2017 7:46
Para: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Asunto: [Freeswitch-users] Sofia stops responding after a few days

 

hi,

I have 2 instances of FS running on a single windows box, first instance
uses normal sip UDP profiles mainly used for routing calls to carriers and
its running as a service since a few months without any issues. The second
instance runs a sip TLS profile and accepts inbound registrations and
forwards calls in sip UDP ahead, it uses xml_curl for directory users but
the problem is every 2-3 days the inbound TLS profile stops responding to
registrations, and when that happens i cant even stop and restart the
service, have to kill it and start again. FS_CLI works but doesnt show any
error, at first i though it could be the xml_curl causing the issue but
later realized it never sends any requests when sofia stops responding.

i have been banging my head from the past week or so but not able to find
the cause, could any1 help in guiding me what to check when this happens so
can find the root cause, im using commercial certs for TLS and when its
running there r no issues other than sofia stops responding every few days,
it happens at random times.



-- 
Regards,
Bipin



  _____  



---
El software de antivirus Avast ha analizado este correo electrónico en busca de virus.
https://www.avast.com/antivirus
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From bipin at xbipin.com  Fri Dec 22 15:09:58 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 22 Dec 2017 19:09:58 +0400
Subject: [Freeswitch-users] Sofia stops responding after a few days
In-Reply-To: <008c01d37b01$f96b9f40$ec42ddc0$@smartic.es>
References: <5c53627a-9596-840e-efa0-f978f784fc7f@xbipin.com>
 <008c01d37b01$f96b9f40$ec42ddc0$@smartic.es>
Message-ID: <1607ec61170.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Hi,

Thanks for the reply but in my case the server already has 8gb of ram and 
it's not at all busy and plus this instance of fs hardly has less than 10 
registrations at any given time.





On December 22, 2017 12:53:04 PM Miguel Jesús López Valverde 
<mjlopez at smartic.es> wrote:

> I had similar problems with FS installed under an Amazon EFS instance. When
> FS did not attend registration requests and executed the "sofia profile
> internal restart" command, it did not load the profile and it no longer
> appeared before the "sofia status" query.
>
>
>
> I checked that this instance was short of ram memory and I changed the
> instance to a higher one with more memory. Since then I have not appreciated
> this problem again.
>
>
>
> Receive a greeting.
>
>
>
> De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org]
> En nombre de Bipin Patel
> Enviado el: viernes, 22 de diciembre de 2017 7:46
> Para: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Asunto: [Freeswitch-users] Sofia stops responding after a few days
>
>
>
> hi,
>
> I have 2 instances of FS running on a single windows box, first instance
> uses normal sip UDP profiles mainly used for routing calls to carriers and
> its running as a service since a few months without any issues. The second
> instance runs a sip TLS profile and accepts inbound registrations and
> forwards calls in sip UDP ahead, it uses xml_curl for directory users but
> the problem is every 2-3 days the inbound TLS profile stops responding to
> registrations, and when that happens i cant even stop and restart the
> service, have to kill it and start again. FS_CLI works but doesnt show any
> error, at first i though it could be the xml_curl causing the issue but
> later realized it never sends any requests when sofia stops responding.
>
> i have been banging my head from the past week or so but not able to find
> the cause, could any1 help in guiding me what to check when this happens so
> can find the root cause, im using commercial certs for TLS and when its
> running there r no issues other than sofia stops responding every few days,
> it happens at random times.
>
>
>
> --
> Regards,
> Bipin
>
>
>
>   _____
>
>
>
> ---
> El software de antivirus Avast ha analizado este correo electrónico en 
> busca de virus.
> https://www.avast.com/antivirus
>
>
>
> ----------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From dujinfang at gmail.com  Sat Dec 23 00:01:29 2017
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 23 Dec 2017 08:01:29 +0800
Subject: [Freeswitch-users] About Video Call
In-Reply-To: <135AF557-923F-4F02-B8A3-6F073F714B9F@gmail.com>
References: <135AF557-923F-4F02-B8A3-6F073F714B9F@gmail.com>
Message-ID: <CAJs+-5p4KPkpMXqX2WptyQjTrFZAHcYsNKZK1hiEgNKXNPgrmw@mail.gmail.com>

yes, depending how you want to get that, you can get m=video in sdp and you
can get video releated event headers in ESL, and you will have CF_VIDEO in
C.

On Thu, Dec 21, 2017 at 6:16 PM, 王聡 <cong.wang.itsherpa at gmail.com> wrote:

> Hey all,
>
> My FreeSWITCH server could both support audio-only call and video call, so
> is there any way for FS to know if a call is a video call?
>
> Regards.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
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From dujinfang at gmail.com  Sat Dec 23 00:07:06 2017
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 23 Dec 2017 08:07:06 +0800
Subject: [Freeswitch-users] mod_verto with standart firewall ports
In-Reply-To: <e4065ee653a54523a3a3b03dbfe508e5@CSASVEX001.csa.ch>
References: <e4065ee653a54523a3a3b03dbfe508e5@CSASVEX001.csa.ch>
Message-ID: <CAJs+-5rEaYHJmF2q_0XsN9-Fmn1nSWsyOLNPca_QiSfhOcbxug@mail.gmail.com>

yes, you can change that range in switch.conf.xml but you need your
firewall open some ports anyway to get it work.

On Tue, Dec 19, 2017 at 8:58 PM, Pascal Hari <Pascal.Hari at csa.ch> wrote:

> Hello
>
>
>
> I’m new to FreeSWITCH and made a setup where everything is behind the same
> NAT. I have a linux debian 8 server where FreeSWITCH is running on and I’m
> able to connect different clients (SIP and WebRTC). I’m using the
> demonstration configuration which come with the FreeSWITCH installation. I
> mostly followed the instruction “Quick Start FreeSWITCH Demo With Verto
> Communicator
> <https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-Scriptinstallfreeswitchdemowithverto_comunicator>”
> but used apache2 instead of nginx.
>
>
>
> Now I want that same setup but on a server in the web with public IP. I’m
> running a linux debian 8 in azure and installed everything accordingly to
> the “behind the Nat” setup.
>
> I use real (trusted) certificates and can reach the WebRTC client over
> https. If I’m in the public network of our company I can log in, dial a
> number and connect to FreeSWITCH (according to the dialplan). If I’m in the
> corp (internal) network of our company I cannot log in. I only get the
> message “Waiting for server reconnection”. Even in FreeSWITCH (fs_cli) I
> don’t see a request.
>
> For me this sound like a firewall issue and many post say to open ports
> (16384-32768, 5066, …). I don’t have this option because I can’t change our
> companies firewall. And I’m anyway looking for a solution to provide a
> WebRTC client which can be used in a normal company network with relatively
> strict firewall rules. Workarounds like VPN-tunnel are also not a wanted
> solution because I want a really easy setup on the client (WebRTC) side.
>
>
>
> Am I missing a step? Is my guess with the firewall right or could it be
> something else? Is there a way to configure FreeSWITCH and verto
> communicator to use other ports?
>
> Thanks already for suggestions.
>
>
>
> Best regards,
>
>
>
> Pascal Hari
>
> SW Developer
>
>
>
> CSA Engineering AG
>
> ____________________________________________________________
> __________________________________
>
> Confidentiality Note: This message is intended only for the use of the
> named recipient(s) and may contain confidential and/or privileged
> information. If you are not the/an intended recipient, please contact the
> sender and delete this message. Any unauthorized use of the information
> contained in this message is prohibited.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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Blog: http://www.dujinfang.com
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From dujinfang at gmail.com  Sat Dec 23 00:13:05 2017
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 23 Dec 2017 08:13:05 +0800
Subject: [Freeswitch-users] Play MP4 file into conference
In-Reply-To: <A0336BF3-F34D-414C-8AD1-14443E39D925@thisisdrum.com>
References: <A0336BF3-F34D-414C-8AD1-14443E39D925@thisisdrum.com>
Message-ID: <CAJs+-5osBHL-gb-YAJdsKaOun_OiOOudFaQ1Umqc2m39yqmhMA@mail.gmail.com>

we use conference play for a while and don't have your problem, please try
master and if you think it's a bug fire a jira.


On Fri, Dec 15, 2017 at 9:51 PM, Richard Screene <
richard.screene at thisisdrum.com> wrote:

> I am having problems attempting to play a MP4 video into a conference
> using mod_av.
>
> I have built the 1.6.19 branch from source after removing the -pedantic
> compile flag to allow mod_av to compile.
>
> My simple dial plan is:
> ```
>     <extension name="test">
>       <condition field="destination_number" expression="test">
>         <action application="conference" data="conf1"/>
>       </condition>
>     </extension>
> ```
>
> From fs_cli I then attempt to play the video using the command:
> "conference conf1 play /tmp/sample.mp4"
>
> No error messages are visible in the console and I can hear the audio, but
> I do not see the video stream.
>
> Strangely, when I do "uuid_broadcast <uuid> /tmp/sample.mp4" I get the
> audio and video (but obviously only to a single caller)
>
> If I try on the code from the master branch then I do not video from
> either conference..play or uuid_broadcast.
>
> Has anyone got any ideas why I cannot play video into a conference?
>
> Also, is there any equivalent of uuid_fileman for files played into a
> conference?  I would like pause and seek functionality to affect the video
> of all participants in the conference.
>
> Many thanks,
>   Richard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
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From dujinfang at gmail.com  Sat Dec 23 00:19:28 2017
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 23 Dec 2017 08:19:28 +0800
Subject: [Freeswitch-users] How should I configure xml_curl bindings for
	phrases?
In-Reply-To: <CAEVfcab9V_hMj3gbCjuOx3d=7yoVGWXwD66o9cX_AzbDp8U8fA@mail.gmail.com>
References: <CAEVfcab9V_hMj3gbCjuOx3d=7yoVGWXwD66o9cX_AzbDp8U8fA@mail.gmail.com>
Message-ID: <CAJs+-5qfPXZ_qEqPHsJkLHuc=S6Dh7_HUdNP5O57jsBpTt5dPA@mail.gmail.com>

static struct xml_section_t SECTIONS[] = {
{"result", SWITCH_XML_SECTION_RESULT},
{"config", SWITCH_XML_SECTION_CONFIG},
{"directory", SWITCH_XML_SECTION_DIRECTORY},
{"dialplan", SWITCH_XML_SECTION_DIALPLAN},
{"languages", SWITCH_XML_SECTION_LANGUAGES},
{"chatplan", SWITCH_XML_SECTION_CHATPLAN},
{"channels", SWITCH_XML_SECTION_CHANNELS},
{NULL, 0}
};


On Wed, Dec 13, 2017 at 4:50 PM, Babak Yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> how can I use xml_curl to fetch phrases?
> is this right?
> <binding name="languages">
>     <param name="gateway-url" value="http://localhost:3300/fs/phrases"
> bindings="phrases" />
>   </binding>
> or I should use languages instead of phrases?
>
> and what should be the format of generated xml from server?
> thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
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From dujinfang at gmail.com  Sat Dec 23 00:21:28 2017
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 23 Dec 2017 08:21:28 +0800
Subject: [Freeswitch-users] Major change to macOS Xcode/CLT requirements
In-Reply-To: <0F9E5D80-4892-46DC-B27D-E24B7194139A@mgtech.com>
References: <0F9E5D80-4892-46DC-B27D-E24B7194139A@mgtech.com>
Message-ID: <CAJs+-5qO28eF641qPioQ6uZCaGUqethXPdwBxFR3n+xu-tEMXA@mail.gmail.com>

cool

On Wed, Dec 13, 2017 at 3:24 AM, Mario <mario_fs at mgtech.com> wrote:

> FYI for macOS folks running FreeSWITCH:
>
> Homebrew made major changes in the last few weeks that affect the
> installation instructions for FreeSwitch on macOS. It also affects Homebrew
> update/upgrade functions. I updated 3 pages of the macOS wiki and the MacOS
> FreeSwitch Installer, macFI. A brief summary:
>
> Homebrew always required the full Xcode instead of the standalone Apple
> Command Line Tools. That is now the opposite…
>
> When Apple updated Xcode it didn’t always provide the correct version of
> CLT for the previous macOS. For instance, 10.13 and 10.12 use Xcode 9.2,
> but 9.2 may install the CLT for 10.13 on 10.12 meaning the CLT was for
> building 10.13 apps. Homebrew would detect this as a showstopper. This has
> happened a couple of times, and that’s why we sometimes had to use an older
> Xcode for a macOS that was not the latest to build FreeSwitch. This didn’t
> just affect FreeSwitch but some Homebrew formulas.
>
> So…. The Homebrew folks decided to fix this once and for all: Now, when
> Homebrew is installed, it looks for the correct CLT. If the latest Xcode
> (9.2) is already installed on the latest macOS (10.13), nothing needs to
> happen because the CLT that is part of Xcode is already installed and
> correct. If Homebrew does not detect the CLT, it will locate them online,
> download and install the correct version for the macOS running!
>
> The big benefit is that if you don’t need Xcode you will save about 4GB of
> space, and of course it takes a lot less time installing FreeSwitch from
> scratch.
>
> Again, the wiki and macFI are updated, I tested them on 10.13, 10.12,
> 10.11, and 10.10.
> Mario G
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




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Blog: http://www.dujinfang.com
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From dujinfang at gmail.com  Sat Dec 23 00:30:17 2017
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 23 Dec 2017 08:30:17 +0800
Subject: [Freeswitch-users] Get digits through play_and_detect_speech
In-Reply-To: <CAGVOjVXNhbhnEydfNODcdfyAk03B-Q_hY_9rfNX6-aQZXe4phA@mail.gmail.com>
References: <CAGVOjVXNhbhnEydfNODcdfyAk03B-Q_hY_9rfNX6-aQZXe4phA@mail.gmail.com>
Message-ID: <CAJs+-5on_xmv9FUks+O-fKz0Sw3DynHntc5_KnBvqDmMtSdP1Q@mail.gmail.com>

run the info app after it

On Tue, Nov 21, 2017 at 10:51 PM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Greeting list,
>
> I am unable to find a way to get digits through play_and_detect_speech. I
> have tried lua callback function but it seems to work only for stream.
> Searching archive and google, gives no result.
>
> Could someone please give me some pointers?
>
> Best Regards,
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon> Virus-free.
> www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
> <#m_-1068209424349502188_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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Blog: http://www.dujinfang.com
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From dujinfang at gmail.com  Sat Dec 23 01:23:28 2017
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 23 Dec 2017 09:23:28 +0800
Subject: [Freeswitch-users] [OT] baresip
In-Reply-To: <CALXCt0oWvv_T+C_eH_EWJCmAgDgPzwX4C_iZ-ZRD01htPJ71EA@mail.gmail.com>
References: <8b028afd-74ce-ec34-9e4c-19f18a748e95@gmail.com>
 <CALXCt0oWvv_T+C_eH_EWJCmAgDgPzwX4C_iZ-ZRD01htPJ71EA@mail.gmail.com>
Message-ID: <CAJs+-5qqw77hYr2J9H_U6S7E07PPbR56mPMGbqvof=qrY-W65A@mail.gmail.com>

Thanks Alfred, I use it on Mac and it's really nice to have a command line
soft phone working video audio/video.

On Sun, Sep 24, 2017 at 1:24 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Thanks Alfred!
>
> -giovanni
>
>
> On 23 September 2017 at 19:30, Alfred E. Heggestad <
> alfred.heggestad at gmail.com> wrote:
>
>> Hi,
>>
>>
>> My apologies for this off-topic post, but I hope that this
>> can be of interest to the nice Freeswitch community.
>>
>>
>> We have released a new version of Baresip which is a
>> open source SIP client that is compatible with Freeswitch.
>> you can find the source code here:
>>
>>     https://github.com/alfredh/baresip/releases/tag/v0.5.5
>>
>>
>> We are doing regular interop-testing with Freeswitch,
>> mainly SIP and RTP related features.
>>
>>
>> From the Baresip project we send the best greetings to
>> the nice people on this list :)
>>
>>
>>
>>
>> /alfred
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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From d.mordovin at dwide.com  Sat Dec 23 11:38:42 2017
From: d.mordovin at dwide.com (Dmitry Mordovin)
Date: Sat, 23 Dec 2017 14:38:42 +0300
Subject: [Freeswitch-users] XML-CURL and dynamic IVR menus
Message-ID: <d440ecb1-2a5d-4e93-fa9c-6795d4aed322@dwide.com>

Hello!

I use xml-curl module - bindings="directory|dialplan"

I want use IVR in dialplan, so, return this XML:

<document type="freeswitch/xml">
   <section name="configuration">
     <configuration name="ivr.conf" description="IVR menus">
       <menus>
         <menu name="ivr-test" 
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" 
exit-sound="voicemail/vm-goodbye.wav" timeout ="3000"
                inter-digit-timeout="2000" max-failures="2" 
digit-len="3" phrase-lang="en">
           <entry action="menu-exec-app" digits="0" param="transfer 11 
XML voicemail" />
         </menu>
       </menus>
     </configuration>
   </section>
   <section name="dialplan" description="IVR call">
     <context name="public">
       <extension name="ivr_1">
         <condition field="destination_number" expression="^([1-9]\\d+)$">
           <action application="ivr" data="ivr-test"/>
           <action application="bridge" 
data="sofia/internal/1000%10.10.10.10"/>
         </condition>
       </extension>
     </context>
   </section>
</document>


As I see in log, FS ignore IVR configuration part and ERROR happen

[ERR] mod_dptools.c:2055 Unable to find menu


Does it possible attach IVR menu with dialplan response?

Or I need use another way to load dynamic IVR?





From vma at vallimamod.org  Sat Dec 23 16:06:58 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Sat, 23 Dec 2017 17:06:58 +0100
Subject: [Freeswitch-users] XML-CURL and dynamic IVR menus
In-Reply-To: <d440ecb1-2a5d-4e93-fa9c-6795d4aed322@dwide.com>
References: <d440ecb1-2a5d-4e93-fa9c-6795d4aed322@dwide.com>
Message-ID: <0681AE15-3CD5-44E4-8759-A102DCA4285A@vallimamod.org>

Hi,

The IVR menu you want to return is from configuration section so, the associated binding would be "configuration". 
If the url is the same for all sections, you can just put bindings="all" for simplicity.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.


> On 23 Dec 2017, at 12:38, Dmitry Mordovin <d.mordovin at dwide.com> wrote:
> 
> Hello!
> 
> I use xml-curl module - bindings="directory|dialplan"
> 
> I want use IVR in dialplan, so, return this XML:
> 
> <document type="freeswitch/xml">
>   <section name="configuration">
>     <configuration name="ivr.conf" description="IVR menus">
>       <menus>
>         <menu name="ivr-test" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" timeout ="3000"
>                inter-digit-timeout="2000" max-failures="2" digit-len="3" phrase-lang="en">
>           <entry action="menu-exec-app" digits="0" param="transfer 11 XML voicemail" />
>         </menu>
>       </menus>
>     </configuration>
>   </section>
>   <section name="dialplan" description="IVR call">
>     <context name="public">
>       <extension name="ivr_1">
>         <condition field="destination_number" expression="^([1-9]\\d+)$">
>           <action application="ivr" data="ivr-test"/>
>           <action application="bridge" data="sofia/internal/1000%10.10.10.10"/>
>         </condition>
>       </extension>
>     </context>
>   </section>
> </document>
> 
> 
> As I see in log, FS ignore IVR configuration part and ERROR happen
> 
> [ERR] mod_dptools.c:2055 Unable to find menu
> 
> 
> Does it possible attach IVR menu with dialplan response?
> 
> Or I need use another way to load dynamic IVR?
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Sat Dec 23 20:30:10 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 23 Dec 2017 20:30:10 +0000
Subject: [Freeswitch-users] Major change to macOS Xcode/CLT requirements
In-Reply-To: <CAJs+-5qO28eF641qPioQ6uZCaGUqethXPdwBxFR3n+xu-tEMXA@mail.gmail.com>
References: <0F9E5D80-4892-46DC-B27D-E24B7194139A@mgtech.com>
 <CAJs+-5qO28eF641qPioQ6uZCaGUqethXPdwBxFR3n+xu-tEMXA@mail.gmail.com>
Message-ID: <CAKbxfG-YrkY61cK+5j15Azfnoz0TkQXuAjFbRogpK-ivHSnxGw@mail.gmail.com>

Maybe we should have a fs package too!

On Fri, Dec 22, 2017 at 6:22 PM Seven Du <dujinfang at gmail.com> wrote:

> cool
>
> On Wed, Dec 13, 2017 at 3:24 AM, Mario <mario_fs at mgtech.com> wrote:
>
>> FYI for macOS folks running FreeSWITCH:
>>
>> Homebrew made major changes in the last few weeks that affect the
>> installation instructions for FreeSwitch on macOS. It also affects Homebrew
>> update/upgrade functions. I updated 3 pages of the macOS wiki and the MacOS
>> FreeSwitch Installer, macFI. A brief summary:
>>
>> Homebrew always required the full Xcode instead of the standalone Apple
>> Command Line Tools. That is now the opposite…
>>
>> When Apple updated Xcode it didn’t always provide the correct version of
>> CLT for the previous macOS. For instance, 10.13 and 10.12 use Xcode 9.2,
>> but 9.2 may install the CLT for 10.13 on 10.12 meaning the CLT was for
>> building 10.13 apps. Homebrew would detect this as a showstopper. This has
>> happened a couple of times, and that’s why we sometimes had to use an older
>> Xcode for a macOS that was not the latest to build FreeSwitch. This didn’t
>> just affect FreeSwitch but some Homebrew formulas.
>>
>> So…. The Homebrew folks decided to fix this once and for all: Now, when
>> Homebrew is installed, it looks for the correct CLT. If the latest Xcode
>> (9.2) is already installed on the latest macOS (10.13), nothing needs to
>> happen because the CLT that is part of Xcode is already installed and
>> correct. If Homebrew does not detect the CLT, it will locate them online,
>> download and install the correct version for the macOS running!
>>
>> The big benefit is that if you don’t need Xcode you will save about 4GB
>> of space, and of course it takes a lot less time installing FreeSwitch from
>> scratch.
>>
>> Again, the wiki and macFI are updated, I tested them on 10.13, 10.12,
>> 10.11, and 10.10.
>> Mario G
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Anthony Minessale II
Founder, FreeSWITCH.
http://freeswitch.com


https://youtu.be/l_hOxzCt6X4
https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From babak.freeswitch at gmail.com  Mon Dec 25 08:43:35 2017
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Mon, 25 Dec 2017 12:13:35 +0330
Subject: [Freeswitch-users] How should I configure xml_curl bindings for
	phrases?
In-Reply-To: <CAJs+-5qfPXZ_qEqPHsJkLHuc=S6Dh7_HUdNP5O57jsBpTt5dPA@mail.gmail.com>
References: <CAEVfcab9V_hMj3gbCjuOx3d=7yoVGWXwD66o9cX_AzbDp8U8fA@mail.gmail.com>
 <CAJs+-5qfPXZ_qEqPHsJkLHuc=S6Dh7_HUdNP5O57jsBpTt5dPA@mail.gmail.com>
Message-ID: <CAEVfcaadzPUFJau81gWeQysGacghcjs47Rq_+FbNA5qdw1HHRg@mail.gmail.com>

thanks
this is working:
<binding name="languages">
    <param name="gateway-url" value="http://localhost:3300/fs/phrases"
bindings="languages" />
  </binding>
but it is not fetching on reloadxml. it is fetched on first reference to a
phrase in dialplan

On Sat, Dec 23, 2017 at 3:49 AM, Seven Du <dujinfang at gmail.com> wrote:

> static struct xml_section_t SECTIONS[] = {
> {"result", SWITCH_XML_SECTION_RESULT},
> {"config", SWITCH_XML_SECTION_CONFIG},
> {"directory", SWITCH_XML_SECTION_DIRECTORY},
> {"dialplan", SWITCH_XML_SECTION_DIALPLAN},
> {"languages", SWITCH_XML_SECTION_LANGUAGES},
> {"chatplan", SWITCH_XML_SECTION_CHATPLAN},
> {"channels", SWITCH_XML_SECTION_CHANNELS},
> {NULL, 0}
> };
>
>
> On Wed, Dec 13, 2017 at 4:50 PM, Babak Yakhchali <
> babak.freeswitch at gmail.com> wrote:
>
>> Hi
>> how can I use xml_curl to fetch phrases?
>> is this right?
>> <binding name="languages">
>>     <param name="gateway-url" value="http://localhost:3300/fs/phrases"
>> bindings="phrases" />
>>   </binding>
>> or I should use languages instead of phrases?
>>
>> and what should be the format of generated xml from server?
>> thanks
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From udy786 at gmail.com  Mon Dec 25 08:48:18 2017
From: udy786 at gmail.com (Uday kumar)
Date: Mon, 25 Dec 2017 14:18:18 +0530
Subject: [Freeswitch-users] How should I configure xml_curl bindings for
	phrases?
In-Reply-To: <CAEVfcaadzPUFJau81gWeQysGacghcjs47Rq_+FbNA5qdw1HHRg@mail.gmail.com>
References: <CAEVfcab9V_hMj3gbCjuOx3d=7yoVGWXwD66o9cX_AzbDp8U8fA@mail.gmail.com>
 <CAJs+-5qfPXZ_qEqPHsJkLHuc=S6Dh7_HUdNP5O57jsBpTt5dPA@mail.gmail.com>
 <CAEVfcaadzPUFJau81gWeQysGacghcjs47Rq_+FbNA5qdw1HHRg@mail.gmail.com>
Message-ID: <CAOfqg5ao5az_5ECBo8P=JHVc8YtYhaqimNAGq_1AGiOO3NQH8A@mail.gmail.com>

Try to reload mod_xml_curl module.

On Mon, Dec 25, 2017 at 2:13 PM, Babak Yakhchali <babak.freeswitch at gmail.com
> wrote:

> thanks
> this is working:
> <binding name="languages">
>     <param name="gateway-url" value="http://localhost:3300/fs/phrases"
> bindings="languages" />
>   </binding>
> but it is not fetching on reloadxml. it is fetched on first reference to a
> phrase in dialplan
>
> On Sat, Dec 23, 2017 at 3:49 AM, Seven Du <dujinfang at gmail.com> wrote:
>
>> static struct xml_section_t SECTIONS[] = {
>> {"result", SWITCH_XML_SECTION_RESULT},
>> {"config", SWITCH_XML_SECTION_CONFIG},
>> {"directory", SWITCH_XML_SECTION_DIRECTORY},
>> {"dialplan", SWITCH_XML_SECTION_DIALPLAN},
>> {"languages", SWITCH_XML_SECTION_LANGUAGES},
>> {"chatplan", SWITCH_XML_SECTION_CHATPLAN},
>> {"channels", SWITCH_XML_SECTION_CHANNELS},
>> {NULL, 0}
>> };
>>
>>
>> On Wed, Dec 13, 2017 at 4:50 PM, Babak Yakhchali <
>> babak.freeswitch at gmail.com> wrote:
>>
>>> Hi
>>> how can I use xml_curl to fetch phrases?
>>> is this right?
>>> <binding name="languages">
>>>     <param name="gateway-url" value="http://localhost:3300/fs/phrases"
>>> bindings="phrases" />
>>>   </binding>
>>> or I should use languages instead of phrases?
>>>
>>> and what should be the format of generated xml from server?
>>> thanks
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj:  http://www.freeswitch.org.cn
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regard
Uday.
Mobile:- +91-9377579349
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From alihaider.4189 at gmail.com  Tue Dec 26 13:54:11 2017
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Tue, 26 Dec 2017 18:54:11 +0500
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <160705398d8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
References: <F48F9953-F444-4549-9727-6B37704EA2CA@gmail.com>
 <160705398d8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
Message-ID: <CAB2bS9SSeX=9N_p_v4O5ZXX7iqoX=nGmfQZFvLWRSqVwo4qRKA@mail.gmail.com>

Hiiiii
     Some error found after build package on vs 2015
now what to do

On 20 December 2017 at 00:50, Bipin Patel <bipin at xbipin.com> wrote:

> Hi,
>
> Use visual studio 2015 with the sdk mentioned in that link and compile and
> it should be fine, we are using it since long and compile the latest code
> and works fine in production.
>
> On December 19, 2017 7:07:23 PM Ali Haider <alihaider.4189 at gmail.com>
> wrote:
>
>> Hiii jeol
>> https://freeswitch.org/confluence/pages/viewpage.action?pageId=1966780
>> I’m followed these steps on windows but after installation of visual
>> studio 2013 which was may be required 182 error is found for missing files
>> Please tell me which version of visual studio is installed to build soln
>> and how to remove error
>>
>>
>> Sent from my iPhone
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From alihaider.4189 at gmail.com  Tue Dec 26 13:55:14 2017
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Tue, 26 Dec 2017 18:55:14 +0500
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAB2bS9SSeX=9N_p_v4O5ZXX7iqoX=nGmfQZFvLWRSqVwo4qRKA@mail.gmail.com>
References: <F48F9953-F444-4549-9727-6B37704EA2CA@gmail.com>
 <160705398d8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <CAB2bS9SSeX=9N_p_v4O5ZXX7iqoX=nGmfQZFvLWRSqVwo4qRKA@mail.gmail.com>
Message-ID: <CAB2bS9S3YunosX8mc3Ayh+WnOoPs7h5dEG8Cp14OBoTSCRgPxQ@mail.gmail.com>

156 error is found now what to do
help me how to resolve

On 26 December 2017 at 18:54, Ali Haider <alihaider.4189 at gmail.com> wrote:

> Hiiiii
>      Some error found after build package on vs 2015
> now what to do
>
> On 20 December 2017 at 00:50, Bipin Patel <bipin at xbipin.com> wrote:
>
>> Hi,
>>
>> Use visual studio 2015 with the sdk mentioned in that link and compile
>> and it should be fine, we are using it since long and compile the latest
>> code and works fine in production.
>>
>> On December 19, 2017 7:07:23 PM Ali Haider <alihaider.4189 at gmail.com>
>> wrote:
>>
>>> Hiii jeol
>>> https://freeswitch.org/confluence/pages/viewpage.action?pageId=1966780
>>> I’m followed these steps on windows but after installation of visual
>>> studio 2013 which was may be required 182 error is found for missing files
>>> Please tell me which version of visual studio is installed to build soln
>>> and how to remove error
>>>
>>>
>>> Sent from my iPhone
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From flavio at voffice.com.br  Tue Dec 26 16:18:49 2017
From: flavio at voffice.com.br (Flavio Goncalves)
Date: Tue, 26 Dec 2017 14:18:49 -0200
Subject: [Freeswitch-users] Cannot load shared object file mod_callcenter
Message-ID: <CAJkFROz0ZHU87UUFci3aEXOiZi+EB7EqG8aDsYcM=X_bYJkYuQ@mail.gmail.com>

Hi,

I'm using CentOS 7 and have installed FS using yum.

I'm getting the following message when loading the mod_callcenter

freeswitch at sippulse.com> load module mod_callcenter
+OK Reloading XML
-ERR [module load file routine returned an error]

2017-12-26 11:10:46.097700 [CRIT] switch_loadable_module.c:1522 Error
Loading module /usr/lib64/freeswitch/mod/module mod_callcenter.so
**/usr/lib64/freeswitch/mod/module mod_callcenter.so: cannot open shared
object file: No such file or directory**
`
What shared object file is missing?

Thanks

Flavio E. Goncalves
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From acheraime at gmail.com  Tue Dec 26 16:45:28 2017
From: acheraime at gmail.com (acheraime at gmail.com)
Date: Tue, 26 Dec 2017 11:45:28 -0500
Subject: [Freeswitch-users] Cannot load shared object file mod_callcenter
In-Reply-To: <CAJkFROz0ZHU87UUFci3aEXOiZi+EB7EqG8aDsYcM=X_bYJkYuQ@mail.gmail.com>
References: <CAJkFROz0ZHU87UUFci3aEXOiZi+EB7EqG8aDsYcM=X_bYJkYuQ@mail.gmail.com>
Message-ID: <80FC1C28-2541-4B9B-9088-1EE9E1AE752B@gmail.com>

Hi Flavio,
Install mod_callcenter via your package management first then try to load the module again.
“yum install freeswitch-mod-callcenter”


—
Adolphe 
Sent from my iPhone

> On Dec 26, 2017, at 11:18 AM, Flavio Goncalves <flavio at voffice.com.br> wrote:
> 
> Hi, 
> 
> I'm using CentOS 7 and have installed FS using yum. 
> 
> I'm getting the following message when loading the mod_callcenter
> 
> freeswitch at sippulse.com> load module mod_callcenter
> +OK Reloading XML
> -ERR [module load file routine returned an error]
> 
> 2017-12-26 11:10:46.097700 [CRIT] switch_loadable_module.c:1522 Error Loading module /usr/lib64/freeswitch/mod/module mod_callcenter.so
> **/usr/lib64/freeswitch/mod/module mod_callcenter.so: cannot open shared object file: No such file or directory**
> `
> What shared object file is missing?
> 
> Thanks
> 
> Flavio E. Goncalves
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From vma at vallimamod.org  Tue Dec 26 16:45:54 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Tue, 26 Dec 2017 17:45:54 +0100
Subject: [Freeswitch-users] Cannot load shared object file mod_callcenter
In-Reply-To: <CAJkFROz0ZHU87UUFci3aEXOiZi+EB7EqG8aDsYcM=X_bYJkYuQ@mail.gmail.com>
References: <CAJkFROz0ZHU87UUFci3aEXOiZi+EB7EqG8aDsYcM=X_bYJkYuQ@mail.gmail.com>
Message-ID: <98747169-5232-4CFF-868E-DCB4B5079A4B@vallimamod.org>

Hi,

The command is actually just "load mod_callcenter". From what you typed, FS is trying to load a module named "module mod_callcenter.so" :)
Hope this helps.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.


> On 26 Dec 2017, at 17:18, Flavio Goncalves <flavio at voffice.com.br> wrote:
> 
> Hi, 
> 
> I'm using CentOS 7 and have installed FS using yum. 
> 
> I'm getting the following message when loading the mod_callcenter
> 
> freeswitch at sippulse.com <mailto:freeswitch at sippulse.com>> load module mod_callcenter
> +OK Reloading XML
> -ERR [module load file routine returned an error]
> 
> 2017-12-26 11:10:46.097700 [CRIT] switch_loadable_module.c:1522 Error Loading module /usr/lib64/freeswitch/mod/module mod_callcenter.so
> **/usr/lib64/freeswitch/mod/module mod_callcenter.so: cannot open shared object file: No such file or directory**
> `
> What shared object file is missing?
> 
> Thanks
> 
> Flavio E. Goncalves
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From flavio at voffice.com.br  Tue Dec 26 16:54:19 2017
From: flavio at voffice.com.br (Flavio Goncalves)
Date: Tue, 26 Dec 2017 14:54:19 -0200
Subject: [Freeswitch-users] Cannot load shared object file mod_callcenter
In-Reply-To: <80FC1C28-2541-4B9B-9088-1EE9E1AE752B@gmail.com>
References: <CAJkFROz0ZHU87UUFci3aEXOiZi+EB7EqG8aDsYcM=X_bYJkYuQ@mail.gmail.com>
 <80FC1C28-2541-4B9B-9088-1EE9E1AE752B@gmail.com>
Message-ID: <CAJkFROyL2AZgtp3DVV4jk58ji1_od_1QstDW_+EDozPgqLu_kg@mail.gmail.com>

Sorry, my mistake. It is loading now.

Flavio E. Goncalves



2017-12-26 14:45 GMT-02:00 <acheraime at gmail.com>:

> Hi Flavio,
> Install mod_callcenter via your package management first then try to load
> the module again.
> “yum install freeswitch-mod-callcenter”
>
>
> —
> Adolphe
> Sent from my iPhone
>
> On Dec 26, 2017, at 11:18 AM, Flavio Goncalves <flavio at voffice.com.br>
> wrote:
>
> Hi,
>
> I'm using CentOS 7 and have installed FS using yum.
>
> I'm getting the following message when loading the mod_callcenter
>
> freeswitch at sippulse.com> load module mod_callcenter
> +OK Reloading XML
> -ERR [module load file routine returned an error]
>
> 2017-12-26 11:10:46.097700 [CRIT] switch_loadable_module.c:1522 Error
> Loading module /usr/lib64/freeswitch/mod/module mod_callcenter.so
> **/usr/lib64/freeswitch/mod/module mod_callcenter.so: cannot open shared
> object file: No such file or directory**
> `
> What shared object file is missing?
>
> Thanks
>
> Flavio E. Goncalves
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aqsyounas at gmail.com  Tue Dec 26 17:05:17 2017
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 26 Dec 2017 22:05:17 +0500
Subject: [Freeswitch-users] Get digits through play_and_detect_speech
In-Reply-To: <CAJs+-5on_xmv9FUks+O-fKz0Sw3DynHntc5_KnBvqDmMtSdP1Q@mail.gmail.com>
References: <CAGVOjVXNhbhnEydfNODcdfyAk03B-Q_hY_9rfNX6-aQZXe4phA@mail.gmail.com>
 <CAJs+-5on_xmv9FUks+O-fKz0Sw3DynHntc5_KnBvqDmMtSdP1Q@mail.gmail.com>
Message-ID: <CAGVOjVUtJTNRiARrmwMfso7ZsScznQ7MEUaOkUY6tabz=bm-Tw@mail.gmail.com>

Thanks for the answer, I was able to achieve the desired behavior through
the bind_digit_action app.

On 23 December 2017 at 05:30, Seven Du <dujinfang at gmail.com> wrote:

> run the info app after it
>
> On Tue, Nov 21, 2017 at 10:51 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Greeting list,
>>
>> I am unable to find a way to get digits through play_and_detect_speech. I
>> have tried lua callback function but it seems to work only for stream.
>> Searching archive and google, gives no result.
>>
>> Could someone please give me some pointers?
>>
>> Best Regards,
>>
>>
>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon> Virus-free.
>> www.avast.com
>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
>> <#m_-4469044929042307560_m_-1068209424349502188_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ryharris at airmail.cc  Fri Dec 22 16:55:42 2017
From: ryharris at airmail.cc (Ryan Harris)
Date: Fri, 22 Dec 2017 11:55:42 -0500
Subject: [Freeswitch-users] Sofia stops responding after a few days
In-Reply-To: <1607ec61170.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
References: <5c53627a-9596-840e-efa0-f978f784fc7f@xbipin.com>
 <008c01d37b01$f96b9f40$ec42ddc0$@smartic.es>
 <1607ec61170.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
Message-ID: <8a89ffbd-db08-1931-db1e-78b9e6b392e5@airmail.cc>

Hello,

This comment in the default sip_profiles/internal.xml has caught my eye
in the past:
https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/sip_profiles/internal.xml#63-82

Maybe you can enable the watchdog and see if you can get a useful core dump.


On 12/22/2017 10:09 AM, Bipin Patel wrote:
>
> Hi,
>
> Thanks for the reply but in my case the server already has 8gb of ram
> and it's not at all busy and plus this instance of fs hardly has less
> than 10 registrations at any given time.
>
> On December 22, 2017 12:53:04 PM Miguel Jesús López Valverde
> <mjlopez at smartic.es> wrote:
>
>> I had similar problems with FS installed under an Amazon EFS
>> instance. When FS did not attend registration requests and executed
>> the "sofia profile internal restart" command, it did not load the
>> profile and it no longer appeared before the "sofia status" query.
>>
>>  
>>
>> I checked that this instance was short of ram memory and I changed
>> the instance to a higher one with more memory. Since then I have not
>> appreciated this problem again.
>>
>>  
>>
>> Receive a greeting.
>>
>>  
>>
>> *De:*FreeSWITCH-users
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *En nombre de
>> *Bipin Patel
>> *Enviado el:* viernes, 22 de diciembre de 2017 7:46
>> *Para:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Asunto:* [Freeswitch-users] Sofia stops responding after a few days
>>
>>  
>>
>> hi,
>>
>> I have 2 instances of FS running on a single windows box, first
>> instance uses normal sip UDP profiles mainly used for routing calls
>> to carriers and its running as a service since a few months without
>> any issues. The second instance runs a sip TLS profile and accepts
>> inbound registrations and forwards calls in sip UDP ahead, it uses
>> xml_curl for directory users but the problem is every 2-3 days the
>> inbound TLS profile stops responding to registrations, and when that
>> happens i cant even stop and restart the service, have to kill it and
>> start again. FS_CLI works but doesnt show any error, at first i
>> though it could be the xml_curl causing the issue but later realized
>> it never sends any requests when sofia stops responding.
>>
>> i have been banging my head from the past week or so but not able to
>> find the cause, could any1 help in guiding me what to check when this
>> happens so can find the root cause, im using commercial certs for TLS
>> and when its running there r no issues other than sofia stops
>> responding every few days, it happens at random times.
>>
>> -- 
>> Regards,
>> Bipin
>>
>> ------------------------------------------------------------------------
>>
>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient>
>> 	Libre de virus. www.avast.com
>> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient>
>>
>>
>> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting%40freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users%40lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From melekoktay at gmail.com  Fri Dec 22 10:05:38 2017
From: melekoktay at gmail.com (Melek Oktay)
Date: Fri, 22 Dec 2017 12:05:38 +0200
Subject: [Freeswitch-users] Fwd: FreeSwitch blocked
In-Reply-To: <CAOopgPUrCTMEo02VsttFivuxhRo_D1y9zrUjZh63ef6FX-xT9w@mail.gmail.com>
References: <CAOopgPUrCTMEo02VsttFivuxhRo_D1y9zrUjZh63ef6FX-xT9w@mail.gmail.com>
Message-ID: <CAOopgPU3=3+yBd1kHA50RFe_90=i6SrhC2KdwuCRksvucg6KFA@mail.gmail.com>

Hi,

FreeSwitch software working well in a few days (~3 - 5 days), then new
incoming call requests are accepted since FreeSwitch is blocked !! Ongoing
calls continue their session, their calls seems not effected, but new calls
are not accepted.  I got FreeSwitch snapshot and analyzed it in GDB.

I have 601 therads & most of them are waiting

Thread 0x7f16bc55f700 (LWP 28544) pthread_cond_wait@@GLIBC_2.3.2 () at
../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_cond_wait.S:185

When i apply "*thread apply all bt*" in gdb, I see most of the threads try
to push events into queue (*switch_queue_push *)

Thread 600 (Thread 0x7f16bc55f700 (LWP 28544)):
#0  pthread_cond_wait@@GLIBC_2.3.2 () at ../nptl/sysdeps/unix/sysv/
linux/x86_64/pthread_cond_wait.S:185
#1  0x00007f180cf9b87d in apr_thread_cond_wait (cond=<optimized out>,
mutex=<optimized out>) at locks/unix/thread_cond.c:68
#2  0x00007f180cf92dd0 in apr_queue_push (queue=queue at entry=0x7f180db157a8,
data=data at entry=0x7f16d3d5ec20) at misc/apr_queue.c:166
#3  0x00007f180cc958fb in *switch_queue_push *(queue=*0x7f180db157a8*,
data=data at entry=*0x7f16d3d5ec20*) at src/switch_apr.c:1134
#4  0x00007f180cd17850 in switch_event_queue_dispatch_event
(eventp=0x7f16bc55ec48) at src/switch_event.c:384
#5  switch_event_fire_detailed (file=file at entry=0x7f180cfb07ea
"src/switch_channel.c", func=func at entry=0x7f180cfb2ba0 <__func__.18348>
"switch_channel_perform_set_running_state", line=line at entry=2260,
event=event at entry=0x7f16bc55ec48, user_data=user_data at entry=0x0) at
src/switch_event.c:1986
#6  0x00007f180cc9f118 in switch_channel_perform_set_running_state
(channel=0x7f17e3e7de00, state=CS_NEW, file=0x7f180cfbc590
"src/switch_core_state_machine.c", func=<optimized out>, line=543) at
src/switch_channel.c:2260
#7  0x00007f180ccc87d0 in switch_core_session_run (session=0x7f17e3e7fd28)
at src/switch_core_state_machine.c:543
#8  0x00007f180ccc36de in switch_core_session_thread (thread=<optimized
out>, obj=0x7f17e3e7fd28) at src/switch_core_session.c:1629
#9  0x00007f180ccbf47d in switch_core_session_thread_pool_worker
(thread=0x7f17e3e9abb0, obj=0x80) at src/switch_core_session.c:1692
#10 0x00007f180cfa1910 in dummy_worker (opaque=0x7f17e3e9abb0) at
threadproc/unix/thread.c:151
#11 0x00007f180c1e0064 in start_thread (arg=0x7f16bc55f700) at
pthread_create.c:309
#12 0x00007f180b8b862d in clone () at ../sysdeps/unix/sysv/linux/
x86_64/clone.S:111


More interesting thing is below, when I look up event type, approximately
all of  them are "SWITCH_EVENT_CHANNEL_STATE" and switch_queue (i think
sofia_module queue is used in this scenario ) *become full* !!! *nelts*
(number of elements ) and *bounds *values are equal, and there are 553
(full_waiters) waiters try to push , but no body try to consume it
(empty_waiters = 0)

(gdb) print *(switch_queue_t *) *0x7f180db157a8*
$1 = {
  data = 0x7f1805cfe038,
  nelts = 50000,
  in = 43000,
  out = 43000,
  bounds = 50000,
  full_waiters = 553,
  empty_waiters = 0,
  one_big_mutex = 0x7f180db157e8,
  not_empty = 0x7f180db15838,
  not_full = 0x7f180db15890,
  terminated = 0
}

(gdb) print *(switch_event_t *) *0x7f16d3d5ec20*
$1 = {
  event_id = SWITCH_EVENT_CHANNEL_STATE,
  priority = SWITCH_PRIORITY_NORMAL,
  owner = 0x0,
  subclass_name = 0x0,
  headers = 0x7f16d3d5f750,
  last_header = 0x7f16d3d601d0,
  body = 0x0,
  bind_user_data = 0x0,
  event_user_data = 0x0,
  key = 0,
  next = 0x0,
  flags = 0
}


Why i am gonna getting this state?

Any thoughts, tips, tricks would be much appreciated.

Regards,

Angel
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From sdevoy at bizfocused.com  Wed Dec 27 22:01:52 2017
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Wed, 27 Dec 2017 22:01:52 +0000
Subject: [Freeswitch-users] VTECH VSP725 MWI
In-Reply-To: <DM5PR01MB26342AC6A6B1781DC75A882BBB020@DM5PR01MB2634.prod.exchangelabs.com>
References: <DM5PR01MB26342AC6A6B1781DC75A882BBB020@DM5PR01MB2634.prod.exchangelabs.com>
Message-ID: <DM5PR01MB263442466C22A246A108FD5EBB070@DM5PR01MB2634.prod.exchangelabs.com>

I know it is the holidays, so I am going to bump this one.

Can anyone help with VTECH setup?  Nothing in the docs except a check that it works!

Thanks, and Happy Holidays to all.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy
Sent: Thursday, December 21, 2017 10:02 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] VTECH VSP725 MWI

Hi All,

Does anyone have any pointers about getting the Message Waiting Lamp to work on a VTECH VSP725?

This is our first VTECH SIP device and it seems fairly simple except for the MWI.  The only thing I have found by searching has been to add: <param name="manage-presence" value="true"/> which had no effect.

Thanks,
Sean
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From cong.wang.itsherpa at gmail.com  Fri Dec 29 08:01:30 2017
From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=)
Date: Fri, 29 Dec 2017 17:01:30 +0900
Subject: [Freeswitch-users] About Codec Negotiation
Message-ID: <46E4B695-3A2D-4367-84D4-47328A408C4E@gmail.com>

Hey all,

I had got an issue when I tried to bridge FS to Asterisk for a outgoing call. The late negotation had been set as true, but I got a codec error on Asterisk which showed:

[Dec 29 16:16:52] NOTICE[2095][C-00000002]: chan_sip.c:10465 process_sdp: No compatible codecs, not accepting this offer!

After this I checked the codec negotiation, like this:

SDP message from softphone:

43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:46.677190 [DEBUG] sofia.c:7094 Remote SDP:401e6cb7-3c4d-43ca-9bc9-a5055ad52485 v=0
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:46.677190 [DEBUG] sofia.c:7094 Remote SDP:
43e110bf-0dd2-4518-a5d8-3b3afd52f6de v=0
43e110bf-0dd2-4518-a5d8-3b3afd52f6de o=52bf0a8a88c515081c124a49cf6ce7bc 2651 2264 IN IP4 192.168.254.65
43e110bf-0dd2-4518-a5d8-3b3afd52f6de s=Talk
43e110bf-0dd2-4518-a5d8-3b3afd52f6de c=IN IP4 192.168.254.65
43e110bf-0dd2-4518-a5d8-3b3afd52f6de t=0 0
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
43e110bf-0dd2-4518-a5d8-3b3afd52f6de m=audio 7076 RTP/AVP 96 97 98 99 0 8 101 100 102
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:96 opus/48000/2
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:96 useinbandfec=1
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:97 SILK/16000
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:98 speex/16000
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:98 vbr=on
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:99 speex/8000
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:99 vbr=on
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:101 telephone-event/48000
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:100 telephone-event/16000
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:102 telephone-event/8000
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtcp-fb:* trr-int 5000

Negotiation steps:

43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[PCMU:0:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[PCMA:8:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[GSM:3:8000:20:13200:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[SPEEX:99:8000:20:24600:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[SPEEX:99:16000:20:42200:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[PCMU:0:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[PCMA:8:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[GSM:3:8000:20:13200:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[SPEEX:99:8000:20:24600:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[SPEEX:99:16000:20:42200:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[PCMU:0:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[PCMA:8:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[GSM:3:8000:20:13200:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[SPEEX:99:8000:20:24600:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[SPEEX:99:16000:20:42200:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [SPEEX:99:16000:20:42200:1] ++++ is saved as a match
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[PCMU:0:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[PCMA:8:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[GSM:3:8000:20:13200:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[SPEEX:99:8000:20:24600:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [SPEEX:99:8000:20:24600:1] ++++ is saved as a match
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[SPEEX:99:16000:20:42200:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[SPEEX:99:8000:20:24600:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[SPEEX:99:16000:20:42200:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[SPEEX:99:8000:20:24600:1]
43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[SPEEX:99:16000:20:42200:1]
Returned SDP message:

43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] mod_sofia.c:2364 Ring SDP:
43e110bf-0dd2-4518-a5d8-3b3afd52f6de v=0
43e110bf-0dd2-4518-a5d8-3b3afd52f6de o=FreeSWITCH 1514482587 1514482588 IN IP4 172.16.25.25
43e110bf-0dd2-4518-a5d8-3b3afd52f6de s=FreeSWITCH
43e110bf-0dd2-4518-a5d8-3b3afd52f6de c=IN IP4 210.148.155.42
43e110bf-0dd2-4518-a5d8-3b3afd52f6de t=0 0
43e110bf-0dd2-4518-a5d8-3b3afd52f6de m=audio 50360 RTP/AVP 98 100
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:98 speex/16000
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:100 telephone-event/16000
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:100 0-16
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=ptime:20
43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=sendrecv

I’m not clear what happened, but PCMU and PCMA codecs are gone even it passed the check. Next the SDP had sent to Asterisk:

c5439be6-10e0-431b-b5e3-7904a14a37c0 Local SDP:
c5439be6-10e0-431b-b5e3-7904a14a37c0 v=0
c5439be6-10e0-431b-b5e3-7904a14a37c0 o=FreeSWITCH 1514482871 1514482872 IN IP4 172.16.25.24
c5439be6-10e0-431b-b5e3-7904a14a37c0 s=FreeSWITCH
c5439be6-10e0-431b-b5e3-7904a14a37c0 c=IN IP4 172.16.25.24
c5439be6-10e0-431b-b5e3-7904a14a37c0 t=0 0
c5439be6-10e0-431b-b5e3-7904a14a37c0 m=audio 50076 RTP/AVP 102 101 13
c5439be6-10e0-431b-b5e3-7904a14a37c0 a=rtpmap:102 SPEEX/16000
c5439be6-10e0-431b-b5e3-7904a14a37c0 a=rtpmap:101 telephone-event/16000
c5439be6-10e0-431b-b5e3-7904a14a37c0 a=fmtp:101 0-16
c5439be6-10e0-431b-b5e3-7904a14a37c0 a=rtpmap:13 CN/16000
c5439be6-10e0-431b-b5e3-7904a14a37c0 a=ptime:20
c5439be6-10e0-431b-b5e3-7904a14a37c0 a=sendrecv

Asterisk can’t recognize these codecs, and it turned into a failure.

So what happened in the codec negotiation, and how could I let PCMU enabled?

PS. I had pasted some setting file and hope it helps.

Vars.xml:

…
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,GSM,speex at 8000h@20i,speex at 16000h@20i,VP8,H264,H263-1998"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,speex at 8000h@20i,speex at 16000h@20i”/>
…

Sip_profiles/external.xml:

…
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}”/>

    <param name="inbound-codec-negotiation" value=“generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="inbound-late-negotiation" value="true"/>
    <param name="inbound-zrtp-passthru" value="true"/> <!-- (also enables late negotiation) -->
…

Regards.
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From vma at vallimamod.org  Fri Dec 29 14:04:30 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Fri, 29 Dec 2017 15:04:30 +0100
Subject: [Freeswitch-users] About Codec Negotiation
In-Reply-To: <46E4B695-3A2D-4367-84D4-47328A408C4E@gmail.com>
References: <46E4B695-3A2D-4367-84D4-47328A408C4E@gmail.com>
Message-ID: <C37FEDAE-7661-49B8-854A-0175C7208D57@vallimamod.org>

Hi,

Have a look at https://freeswitch.org/jira/browse/FS-8321 (media_mix_inbound_outbound_codecs)
It may help with your issue.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.

> On 29 Dec 2017, at 09:01, 王聡 <cong.wang.itsherpa at gmail.com> wrote:
> 
> Hey all,
> 
> I had got an issue when I tried to bridge FS to Asterisk for a outgoing call. The late negotation had been set as true, but I got a codec error on Asterisk which showed:
> 
> [Dec 29 16:16:52] NOTICE[2095][C-00000002]: chan_sip.c:10465 process_sdp: No compatible codecs, not accepting this offer!
> 
> After this I checked the codec negotiation, like this:
> 
> SDP message from softphone:
> 
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:46.677190 [DEBUG] sofia.c:7094 Remote SDP:401e6cb7-3c4d-43ca-9bc9-a5055ad52485 v=0
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:46.677190 [DEBUG] sofia.c:7094 Remote SDP:
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de v=0
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de o=52bf0a8a88c515081c124a49cf6ce7bc 2651 2264 IN IP4 192.168.254.65
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de s=Talk
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de c=IN IP4 192.168.254.65
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de t=0 0
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de m=audio 7076 RTP/AVP 96 97 98 99 0 8 101 100 102
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:96 opus/48000/2
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:96 useinbandfec=1
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:97 SILK/16000
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:98 speex/16000
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:98 vbr=on
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:99 speex/8000
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:99 vbr=on
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:101 telephone-event/48000
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:100 telephone-event/16000
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:102 telephone-event/8000
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtcp-fb:* trr-int 5000
> 
> Negotiation steps:
> 
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[PCMU:0:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[PCMA:8:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[GSM:3:8000:20:13200:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[SPEEX:99:8000:20:24600:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [opus:96:48000:20:0:1]/[SPEEX:99:16000:20:42200:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[PCMU:0:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[PCMA:8:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[GSM:3:8000:20:13200:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[SPEEX:99:8000:20:24600:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [SILK:97:16000:20:0:1]/[SPEEX:99:16000:20:42200:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[PCMU:0:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[PCMA:8:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[GSM:3:8000:20:13200:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[SPEEX:99:8000:20:24600:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:98:16000:20:0:1]/[SPEEX:99:16000:20:42200:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [SPEEX:99:16000:20:42200:1] ++++ is saved as a match
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[PCMU:0:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[PCMA:8:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[GSM:3:8000:20:13200:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[SPEEX:99:8000:20:24600:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [SPEEX:99:8000:20:24600:1] ++++ is saved as a match
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:99:8000:20:0:1]/[SPEEX:99:16000:20:42200:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[SPEEX:99:8000:20:24600:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[SPEEX:99:16000:20:42200:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[SPEEX:99:8000:20:24600:1]
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[SPEEX:99:16000:20:42200:1]
> Returned SDP message:
> 
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de 2017-12-29 16:35:47.277188 [DEBUG] mod_sofia.c:2364 Ring SDP:
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de v=0
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de o=FreeSWITCH 1514482587 1514482588 IN IP4 172.16.25.25
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de s=FreeSWITCH
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de c=IN IP4 210.148.155.42
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de t=0 0
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de m=audio 50360 RTP/AVP 98 100
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:98 speex/16000
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=rtpmap:100 telephone-event/16000
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=fmtp:100 0-16
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=ptime:20
> 43e110bf-0dd2-4518-a5d8-3b3afd52f6de a=sendrecv
> 
> I’m not clear what happened, but PCMU and PCMA codecs are gone even it passed the check. Next the SDP had sent to Asterisk:
> 
> c5439be6-10e0-431b-b5e3-7904a14a37c0 Local SDP:
> c5439be6-10e0-431b-b5e3-7904a14a37c0 v=0
> c5439be6-10e0-431b-b5e3-7904a14a37c0 o=FreeSWITCH 1514482871 1514482872 IN IP4 172.16.25.24
> c5439be6-10e0-431b-b5e3-7904a14a37c0 s=FreeSWITCH
> c5439be6-10e0-431b-b5e3-7904a14a37c0 c=IN IP4 172.16.25.24
> c5439be6-10e0-431b-b5e3-7904a14a37c0 t=0 0
> c5439be6-10e0-431b-b5e3-7904a14a37c0 m=audio 50076 RTP/AVP 102 101 13
> c5439be6-10e0-431b-b5e3-7904a14a37c0 a=rtpmap:102 SPEEX/16000
> c5439be6-10e0-431b-b5e3-7904a14a37c0 a=rtpmap:101 telephone-event/16000
> c5439be6-10e0-431b-b5e3-7904a14a37c0 a=fmtp:101 0-16
> c5439be6-10e0-431b-b5e3-7904a14a37c0 a=rtpmap:13 CN/16000
> c5439be6-10e0-431b-b5e3-7904a14a37c0 a=ptime:20
> c5439be6-10e0-431b-b5e3-7904a14a37c0 a=sendrecv
> 
> Asterisk can’t recognize these codecs, and it turned into a failure.
> 
> So what happened in the codec negotiation, and how could I let PCMU enabled?
> 
> PS. I had pasted some setting file and hope it helps.
> 
> Vars.xml:
> 
> …
>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,GSM,speex at 8000h@20i,speex at 16000h@20i,VP8,H264,H263-1998"/>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,speex at 8000h@20i,speex at 16000h@20i”/>
> …
> 
> Sip_profiles/external.xml:
> 
> …
>     <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>     <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}”/>
> 
>     <param name="inbound-codec-negotiation" value=“generous"/>
>     <param name="nonce-ttl" value="60"/>
>     <param name="auth-calls" value="false"/>
>     <param name="inbound-late-negotiation" value="true"/>
>     <param name="inbound-zrtp-passthru" value="true"/> <!-- (also enables late negotiation) -->
> …
> 
> Regards.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Sun Dec 31 15:55:09 2017
From: infos at madovsky.org (Madovsky)
Date: Sun, 31 Dec 2017 07:55:09 -0800
Subject: [Freeswitch-users] uuid_record options
Message-ID: <1f1eebbd-56fd-eaf6-f98e-42c5756f88bf@madovsky.org>

Hi Folks,

is it possible to control the record settings with uuid_record

like width,height,quality etc.?

thanks and have a great new year.



From alihaider.4189 at gmail.com  Sun Dec 31 16:28:18 2017
From: alihaider.4189 at gmail.com (Ali Haider)
Date: Sun, 31 Dec 2017 21:28:18 +0500
Subject: [Freeswitch-users] error correction
Message-ID: <CAB2bS9Q4zgFzUO9bL5PD5Dgob52jzvxGJmnKEj5ZncbKYnNMkg@mail.gmail.com>

Hiiii Everyone
        Anyone who can help me fix the error that are being reviewed in
picture during build on visual studio 2015
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From mangudai_ at hotmail.com  Fri Dec 29 02:02:27 2017
From: mangudai_ at hotmail.com (yunqing)
Date: Fri, 29 Dec 2017 02:02:27 +0000
Subject: [Freeswitch-users] GetDigits in another thread
References: <mailman.357.1514306762.1298.freeswitch-users@lists.freeswitch.org>
Message-ID: <BMXPR01MB06157BC0334655DFDCA521EB8C050@BMXPR01MB0615.INDPRD01.PROD.OUTLOOK.COM>

Hi there,

i got a problem receiving digit within another thread,
i was following the example of "Lua Script to announce members position" from https://freeswitch.org/confluence/display/FREESWITCH/mod_callcenter,
everything works fine until i added GetDigits after the announcement,
the voice became sorta smeared, and GetDigits was not working at all.

the Lua Script was called from dialplan in unblocked mode like:
<action application="set" data="result=${luarun(callcenter-announce-position.lua ${uuid} example at default 10000)}"/>

and i want the caller to input dtmf after the prompt while still stay queuing, so what should i do? any ideas are appreciated.

________________________________
mangudai_ at hotmail.com
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