From aroumie at yahoo.com  Mon Feb  2 22:32:50 2015
From: aroumie at yahoo.com (Ali R.)
Date: Mon, 2 Feb 2015 19:32:50 +0000 (UTC)
Subject: [Freeswitch-dev] GDB
Message-ID: <1400802287.252488.1422905570453.JavaMail.yahoo@mail.yahoo.com>

Dear All,I have FS configured/compiled with "devel-bootstrap.sh" for better debugging. Is it possible to extract the parsed SIP message from a core dump (or as much SIP as possible.) traversing the back trace, I was able to see host, port, destination number which seems to be fine but I have a feeling there might be some structures with deep SIP info. I'm having random core dumps and I would like to get ideas at the call patterns that causes this crash. I will for sure report all details on this particular crash to JIRA for better tracking. In the mean time, any guidance is appreciated.?


#0 ?0x00007f6d8192f165 in raise () from /lib/x86_64-linux-gnu/libc.so.6#1 ?0x00007f6d819323e0 in abort () from /lib/x86_64-linux-gnu/libc.so.6#2 ?0x00007f6d8196a1cb in ?? () from /lib/x86_64-linux-gnu/libc.so.6#3 ?0x00007f6d81973a16 in ?? () from /lib/x86_64-linux-gnu/libc.so.6#4 ?0x00007f6d819772f8 in ?? () from /lib/x86_64-linux-gnu/libc.so.6#5 ?0x00007f6d819788a0 in malloc () from /lib/x86_64-linux-gnu/libc.so.6#6 ?0x00007f6d82bf78c2 in my_dup (s=0x7f6d54313360 "user=phone") at src/switch_event.c:113#7 ?0x00007f6d82bf8e20 in switch_event_add_header_string (event=0x7f6d555b3ae0, stack=11636, stack at entry=SWITCH_STACK_BOTTOM, header_name=0x6 <Address 0x6 out of bounds>, ?header_name at entry=0x7f6d77d9afbc "sip_contact_params", data=0xffffffffffffffff <Address 0xffffffffffffffff out of bounds>, data at entry=0x7f6d54313360 "user=phone") at src/switch_event.c:1182#8 ?0x00007f6d82b8ca02 in switch_channel_set_variable_var_check (channel=channel at entry=0x7f6d5492e380, varname=varname at entry=0x7f6d77d9afbc "sip_contact_params", value=0x7f6d54313360 "user=phone", var_check=var_check at entry=SWITCH_TRUE) at src/switch_channel.c:1385#9 ?0x00007f6d77c7462d in _url_set_chanvars (session=session at entry=0x7f6d55395d88, url=0x7f6d5432c488, user_var=user_var at entry=0x7f6d77d9afab "sip_contact_user",? ? host_var=host_var at entry=0x7f6d77d9af9a "sip_contact_host", port_var=port_var at entry=0x7f6d77d9af89 "sip_contact_port", uri_var=uri_var at entry=0x7f6d77d9af79 "sip_contact_uri",? ? params_var=params_var at entry=0x7f6d77d9afbc "sip_contact_params") at sofia.c:119#10 0x00007f6d77c8ec40 in sofia_handle_sip_i_invite (session=session at entry=0x7f6d55395d88, nua=nua at entry=0x7f6d782aae80, profile=profile at entry=0xb60550, nh=nh at entry=0x7f6d5498d830, sofia_private=0x7f6d552b0c70,? ? sip=sip at entry=0x7f6d540492c8, de=de at entry=0x7f6d54207b30, tags=tags at entry=0x7f6d554de4b0) at sofia.c:9434#11 0x00007f6d77c96bc1 in our_sofia_event_callback (event=nua_i_invite, status=100, phrase=0x7f6d554de4c0 "Trying", nua=0x7f6d782aae80, profile=0xb60550, nh=0x7f6d5498d830, sofia_private=0x7f6d552b0c70,? ? sip=0x7f6d540492c8, de=de at entry=0x7f6d54207b30, tags=0x7f6d554de4b0) at sofia.c:1491#12 0x00007f6d77c9a96c in sofia_process_dispatch_event (dep=dep at entry=0x7f6d427e2108) at sofia.c:1903#13 0x00007f6d77c6ace9 in sofia_receive_message (session=0x7f6d55395d88, msg=0x7f6d427e2a90) at mod_sofia.c:1115#14 0x00007f6d82bb14ae in switch_core_session_perform_receive_message (session=session at entry=0x7f6d55395d88, message=message at entry=0x7f6d427e2a90, file=file at entry=0x7f6d82caf290 "src/switch_ivr.c",? ? func=func at entry=0x7f6d82cafb60 "switch_ivr_parse_all_signal_data", line=line at entry=838) at src/switch_core_session.c:775#15 0x00007f6d82c35705 in switch_ivr_parse_all_signal_data (session=session at entry=0x7f6d55395d88) at src/switch_ivr.c:838#16 0x00007f6d82c35750 in switch_ivr_parse_all_messages (session=session at entry=0x7f6d55395d88) at src/switch_ivr.c:797#17 0x00007f6d82bb5fdf in switch_core_session_run (session=0x7f6d55395d88) at src/switch_core_state_machine.c:475#18 0x00007f6d82bb0f1e in switch_core_session_thread (thread=<optimized out>, obj=0x7f6d55395d88) at src/switch_core_session.c:1598#19 0x00007f6d82bad1b5 in switch_core_session_thread_pool_worker (thread=0x7f6d710c7d40, obj=<optimized out>) at src/switch_core_session.c:1690#20 0x00007f6d822e7b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0#21 0x00007f6d819d97bd in clone () from /lib/x86_64-linux-gnu/libc.so.6#22 0x0000000000000000 in ?? ()

Ali R.
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From krice at freeswitch.org  Fri Feb  6 18:01:51 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 06 Feb 2015 15:01:51 +0000
Subject: [Freeswitch-dev] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <54d4d75f66260_3e13e55330448ee@ip-10-184-131-145.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From william.king at quentustech.com  Sat Feb  7 05:08:40 2015
From: william.king at quentustech.com (William King)
Date: Fri, 06 Feb 2015 21:08:40 -0500
Subject: [Freeswitch-dev] GDB
In-Reply-To: <1400802287.252488.1422905570453.JavaMail.yahoo@mail.yahoo.com>
References: <1400802287.252488.1422905570453.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <54D573A8.8060104@quentustech.com>

Ali,

A call to malloc() called abort, so I'd bet what happened was your
system ran out of memory, or had heap memory corruption:
http://unixhelp.ed.ac.uk/CGI/man-cgi?malloc+3

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 2/2/15 2:32 PM, Ali R. wrote:
> Dear All,
> I have FS configured/compiled with "devel-bootstrap.sh" for better
> debugging. Is it possible to extract the parsed SIP message from a core
> dump (or as much SIP as possible.) traversing the back trace, I was able
> to see host, port, destination number which seems to be fine but I have
> a feeling there might be some structures with deep SIP info. I'm having
> random core dumps and I would like to get ideas at the call patterns
> that causes this crash. I will for sure report all details on this
> particular crash to JIRA for better tracking. In the mean time, any
> guidance is appreciated.
>  
> 
> 
> 
> #0  0x00007f6d8192f165 in raise () from /lib/x86_64-linux-gnu/libc.so.6
> #1  0x00007f6d819323e0 in abort () from /lib/x86_64-linux-gnu/libc.so.6
> #2  0x00007f6d8196a1cb in ?? () from /lib/x86_64-linux-gnu/libc.so.6
> #3  0x00007f6d81973a16 in ?? () from /lib/x86_64-linux-gnu/libc.so.6
> #4  0x00007f6d819772f8 in ?? () from /lib/x86_64-linux-gnu/libc.so.6
> #5  0x00007f6d819788a0 in malloc () from /lib/x86_64-linux-gnu/libc.so.6
> #6  0x00007f6d82bf78c2 in my_dup (s=0x7f6d54313360 "user=phone") at
> src/switch_event.c:113
> #7  0x00007f6d82bf8e20 in switch_event_add_header_string
> (event=0x7f6d555b3ae0, stack=11636, stack at entry=SWITCH_STACK_BOTTOM,
> header_name=0x6 <Address 0x6 out of bounds>,
>  header_name at entry=0x7f6d77d9afbc "sip_contact_params",
> data=0xffffffffffffffff <Address 0xffffffffffffffff out of bounds>,
> data at entry=0x7f6d54313360 "user=phone") at src/switch_event.c:1182
> #8  0x00007f6d82b8ca02 in switch_channel_set_variable_var_check
> (channel=channel at entry=0x7f6d5492e380,
> varname=varname at entry=0x7f6d77d9afbc "sip_contact_params",
> value=0x7f6d54313360 "user=phone",
> var_check=var_check at entry=SWITCH_TRUE) at src/switch_channel.c:1385
> #9  0x00007f6d77c7462d in _url_set_chanvars
> (session=session at entry=0x7f6d55395d88, url=0x7f6d5432c488,
> user_var=user_var at entry=0x7f6d77d9afab "sip_contact_user",
>     host_var=host_var at entry=0x7f6d77d9af9a "sip_contact_host",
> port_var=port_var at entry=0x7f6d77d9af89 "sip_contact_port",
> uri_var=uri_var at entry=0x7f6d77d9af79 "sip_contact_uri",
>     params_var=params_var at entry=0x7f6d77d9afbc "sip_contact_params") at
> sofia.c:119
> #10 0x00007f6d77c8ec40 in sofia_handle_sip_i_invite
> (session=session at entry=0x7f6d55395d88, nua=nua at entry=0x7f6d782aae80,
> profile=profile at entry=0xb60550, nh=nh at entry=0x7f6d5498d830,
> sofia_private=0x7f6d552b0c70,
>     sip=sip at entry=0x7f6d540492c8, de=de at entry=0x7f6d54207b30,
> tags=tags at entry=0x7f6d554de4b0) at sofia.c:9434
> #11 0x00007f6d77c96bc1 in our_sofia_event_callback (event=nua_i_invite,
> status=100, phrase=0x7f6d554de4c0 "Trying", nua=0x7f6d782aae80,
> profile=0xb60550, nh=0x7f6d5498d830, sofia_private=0x7f6d552b0c70,
>     sip=0x7f6d540492c8, de=de at entry=0x7f6d54207b30, tags=0x7f6d554de4b0)
> at sofia.c:1491
> #12 0x00007f6d77c9a96c in sofia_process_dispatch_event
> (dep=dep at entry=0x7f6d427e2108) at sofia.c:1903
> #13 0x00007f6d77c6ace9 in sofia_receive_message (session=0x7f6d55395d88,
> msg=0x7f6d427e2a90) at mod_sofia.c:1115
> #14 0x00007f6d82bb14ae in switch_core_session_perform_receive_message
> (session=session at entry=0x7f6d55395d88,
> message=message at entry=0x7f6d427e2a90, file=file at entry=0x7f6d82caf290
> "src/switch_ivr.c",
>     func=func at entry=0x7f6d82cafb60 "switch_ivr_parse_all_signal_data",
> line=line at entry=838) at src/switch_core_session.c:775
> #15 0x00007f6d82c35705 in switch_ivr_parse_all_signal_data
> (session=session at entry=0x7f6d55395d88) at src/switch_ivr.c:838
> #16 0x00007f6d82c35750 in switch_ivr_parse_all_messages
> (session=session at entry=0x7f6d55395d88) at src/switch_ivr.c:797
> #17 0x00007f6d82bb5fdf in switch_core_session_run
> (session=0x7f6d55395d88) at src/switch_core_state_machine.c:475
> #18 0x00007f6d82bb0f1e in switch_core_session_thread (thread=<optimized
> out>, obj=0x7f6d55395d88) at src/switch_core_session.c:1598
> #19 0x00007f6d82bad1b5 in switch_core_session_thread_pool_worker
> (thread=0x7f6d710c7d40, obj=<optimized out>) at
> src/switch_core_session.c:1690
> #20 0x00007f6d822e7b50 in start_thread () from
> /lib/x86_64-linux-gnu/libpthread.so.0
> #21 0x00007f6d819d97bd in clone () from /lib/x86_64-linux-gnu/libc.so.6
> #22 0x0000000000000000 in ?? ()
> 
> 
> Ali R.
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 


From ben at langfeld.co.uk  Mon Feb  9 15:18:10 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 9 Feb 2015 10:18:10 -0200
Subject: [Freeswitch-dev] Using FS with TLS and not TLS
In-Reply-To: <CANT=1X0M_wGBEg==hVaKL5k6hOFteDuFUpa2-FS2orGQMV0E1A@mail.gmail.com>
References: <CANT=1X0M_wGBEg==hVaKL5k6hOFteDuFUpa2-FS2orGQMV0E1A@mail.gmail.com>
Message-ID: <CAAyX+KHPpGEBfyDgyWVZeLPYTo2GW-RCYJ72y-M1GY0FWor40Q@mail.gmail.com>

Separate SIP profiles for the two cases.

On 28 January 2015 at 11:30, Y?cel ALTUNAY <is.yaltunay at gmail.com> wrote:

> Hi,
> I want to use freeswitch with TLS on my mobile phones and without TLS on
> my GSM gateways. I see some examples but i coudn't do it.
> Is someone has example to do this.
> Thank you.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From aeburriel at gmail.com  Tue Feb 10 00:45:42 2015
From: aeburriel at gmail.com (Antonio Eugenio Burriel)
Date: Mon, 9 Feb 2015 22:45:42 +0100
Subject: [Freeswitch-dev] Can't negotiate L16@16000h with Asterisk
Message-ID: <CAMwE4SFkT0Qf+u8Mvam3GEsag7TYAXHu_iR9d8O5N7wwQxNGmw@mail.gmail.com>

Yesterday I posted the subject "LIN16 at 16000h Freeswitch <->
Asterisk" in the user mailing list, but after having taken a peek at
the source, I thought it was better to share my analysis here:

Problem: Freeswitch can't negotiate L16 at 16000h codec with some peers
(like Aterisk) which assigns a dynamic PT (over 95) to L16.

Internally Freeswitch assigns a PT of 70 to L16; in
switch_core_media_negotiate_sdp() we have the following match check:
match = (!strcasecmp(rm_encoding, imp->iananame) &&
    ((map->rm_pt < 96 && imp->ianacode < 96) || (map->rm_pt > 95 &&
imp->ianacode > 95)) &&
    (remote_codec_rate == codec_rate || fmtp_remote_codec_rate ==
imp->actual_samples_per_second)) ? 1 : 0;
Which means that when our peer's codec is dynamic, our match must be
also dynamic and, if not, out match mustn't.

Except for L16 at 44100h, which has a PT of 10, its PT should be dynamic
(>95) according to RFC3551. But Freeswitch is internally assigning 70,
so it will never match.
Incidentally, Asterisk erroneously assigns to L16 at 8000h a PT of 10,
which passes the check.

I'm not quite sure which is the proper way to fix this.
If we change PT from 70 to something over 95, we'll break
interoperability with older Freeswitches and other peers, such
LN16 at 800h with Asterisk.

So maybe its cleaner to add a exception to the match check like:
match = (!strcasecmp(rm_encoding, imp->iananame) &&
    ((map->rm_pt < 96 && imp->ianacode < 96) || (map->rm_pt > 95 &&
(imp->ianacode > 95 || imp->ianacode == 70))) &&
    (remote_codec_rate == codec_rate || fmtp_remote_codec_rate ==
imp->actual_samples_per_second)) ? 1 : 0;

Any thoughts?


From brian at freeswitch.org  Tue Feb 10 01:11:45 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 9 Feb 2015 16:11:45 -0600
Subject: [Freeswitch-dev] Can't negotiate L16@16000h with Asterisk
In-Reply-To: <CAMwE4SFkT0Qf+u8Mvam3GEsag7TYAXHu_iR9d8O5N7wwQxNGmw@mail.gmail.com>
References: <CAMwE4SFkT0Qf+u8Mvam3GEsag7TYAXHu_iR9d8O5N7wwQxNGmw@mail.gmail.com>
Message-ID: <CAEJMVkB-ZyGT0n-ngMZFpGFaJQBVN-1z0s51xB9=8XizZVQiKw@mail.gmail.com>

Bug reports are filed at http://freeswitch.org/jira

On Mon, Feb 9, 2015 at 3:45 PM, Antonio Eugenio Burriel <aeburriel at gmail.com
> wrote:

> Yesterday I posted the subject "LIN16 at 16000h Freeswitch <->
> Asterisk" in the user mailing list, but after having taken a peek at
> the source, I thought it was better to share my analysis here:
>
> Problem: Freeswitch can't negotiate L16 at 16000h codec with some peers
> (like Aterisk) which assigns a dynamic PT (over 95) to L16.
>
> Internally Freeswitch assigns a PT of 70 to L16; in
> switch_core_media_negotiate_sdp() we have the following match check:
> match = (!strcasecmp(rm_encoding, imp->iananame) &&
>     ((map->rm_pt < 96 && imp->ianacode < 96) || (map->rm_pt > 95 &&
> imp->ianacode > 95)) &&
>     (remote_codec_rate == codec_rate || fmtp_remote_codec_rate ==
> imp->actual_samples_per_second)) ? 1 : 0;
> Which means that when our peer's codec is dynamic, our match must be
> also dynamic and, if not, out match mustn't.
>
> Except for L16 at 44100h, which has a PT of 10, its PT should be dynamic
> (>95) according to RFC3551. But Freeswitch is internally assigning 70,
> so it will never match.
> Incidentally, Asterisk erroneously assigns to L16 at 8000h a PT of 10,
> which passes the check.
>
> I'm not quite sure which is the proper way to fix this.
> If we change PT from 70 to something over 95, we'll break
> interoperability with older Freeswitches and other peers, such
> LN16 at 800h with Asterisk.
>
> So maybe its cleaner to add a exception to the match check like:
> match = (!strcasecmp(rm_encoding, imp->iananame) &&
>     ((map->rm_pt < 96 && imp->ianacode < 96) || (map->rm_pt > 95 &&
> (imp->ianacode > 95 || imp->ianacode == 70))) &&
>     (remote_codec_rate == codec_rate || fmtp_remote_codec_rate ==
> imp->actual_samples_per_second)) ? 1 : 0;
>
> Any thoughts?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From anthony.minessale at gmail.com  Tue Feb 10 01:42:23 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 9 Feb 2015 16:42:23 -0600
Subject: [Freeswitch-dev] Can't negotiate L16@16000h with Asterisk
In-Reply-To: <CAEJMVkB-ZyGT0n-ngMZFpGFaJQBVN-1z0s51xB9=8XizZVQiKw@mail.gmail.com>
References: <CAMwE4SFkT0Qf+u8Mvam3GEsag7TYAXHu_iR9d8O5N7wwQxNGmw@mail.gmail.com>
	<CAEJMVkB-ZyGT0n-ngMZFpGFaJQBVN-1z0s51xB9=8XizZVQiKw@mail.gmail.com>
Message-ID: <CAKbxfG-gAA4dm4JfU-SGxRMyCQFnM8-6P+OAK=c=3LHXrfxUJQ@mail.gmail.com>

Try changing the 70 to 100 everywhere in switch_pcm.  The value in the code
is not used unless its < 96
file results as a jira

On Mon, Feb 9, 2015 at 4:11 PM, Brian West <brian at freeswitch.org> wrote:

> Bug reports are filed at http://freeswitch.org/jira
>
> On Mon, Feb 9, 2015 at 3:45 PM, Antonio Eugenio Burriel <
> aeburriel at gmail.com> wrote:
>
>> Yesterday I posted the subject "LIN16 at 16000h Freeswitch <->
>> Asterisk" in the user mailing list, but after having taken a peek at
>> the source, I thought it was better to share my analysis here:
>>
>> Problem: Freeswitch can't negotiate L16 at 16000h codec with some peers
>> (like Aterisk) which assigns a dynamic PT (over 95) to L16.
>>
>> Internally Freeswitch assigns a PT of 70 to L16; in
>> switch_core_media_negotiate_sdp() we have the following match check:
>> match = (!strcasecmp(rm_encoding, imp->iananame) &&
>>     ((map->rm_pt < 96 && imp->ianacode < 96) || (map->rm_pt > 95 &&
>> imp->ianacode > 95)) &&
>>     (remote_codec_rate == codec_rate || fmtp_remote_codec_rate ==
>> imp->actual_samples_per_second)) ? 1 : 0;
>> Which means that when our peer's codec is dynamic, our match must be
>> also dynamic and, if not, out match mustn't.
>>
>> Except for L16 at 44100h, which has a PT of 10, its PT should be dynamic
>> (>95) according to RFC3551. But Freeswitch is internally assigning 70,
>> so it will never match.
>> Incidentally, Asterisk erroneously assigns to L16 at 8000h a PT of 10,
>> which passes the check.
>>
>> I'm not quite sure which is the proper way to fix this.
>> If we change PT from 70 to something over 95, we'll break
>> interoperability with older Freeswitches and other peers, such
>> LN16 at 800h with Asterisk.
>>
>> So maybe its cleaner to add a exception to the match check like:
>> match = (!strcasecmp(rm_encoding, imp->iananame) &&
>>     ((map->rm_pt < 96 && imp->ianacode < 96) || (map->rm_pt > 95 &&
>> (imp->ianacode > 95 || imp->ianacode == 70))) &&
>>     (remote_codec_rate == codec_rate || fmtp_remote_codec_rate ==
>> imp->actual_samples_per_second)) ? 1 : 0;
>>
>> Any thoughts?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?

? http://freeswitch.org/  ? http://cluecon.com/  ?
http://twitter.com/FreeSWITCH
? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
<http://freeswitch.org/g+>*

ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org  ? +19193869900
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From krice at freeswitch.org  Fri Feb 13 18:01:05 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 13 Feb 2015 15:01:05 +0000
Subject: [Freeswitch-dev] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <54de11b1dde7e_715b3cb32487936@ip-10-113-178-254.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From eschmidbauer at gmail.com  Thu Feb 19 17:43:11 2015
From: eschmidbauer at gmail.com (e schmidbauer)
Date: Thu, 19 Feb 2015 09:43:11 -0500
Subject: [Freeswitch-dev] valet parking new feature
Message-ID: <CAJwe+CCAr4Nnu_HHb-Qm2EDx1_VfVt4pQEmot0tbM2YGyfKT6A@mail.gmail.com>

Hi,
Would anyone want to test out new parking feature for polycoms?
The feature is outlined in this doc:

http://support.polycom.com/global/documents/support/technical/products/voice/Static_BLF_TB62475.pdf

Just pull clean FS branch and do a git apply with that diff like
git apply --ignore-whitespace ~/valet_sofia.c.diff

make mod_sofia-install
make mod_valet_parking-install

restart FS

Provisiong the polycom with:

attendant.resourceList.2.address="orbit+lot at mydomain.com"
attendant.resourceList.2.label="Orbit Lot"
attendant.resourceList.2.type="automata"

FreeSWITCH dialplan
Park in Orbit:
<extension name="parking_auto_in">
  <condition field="destination_number" expression="^\*7$">
    <action application="answer"/>
    <action application="valet_park" data="lot at mydomain.com auto in 701 799"/>
  </condition>
</extension>

Pick up from Orbit:

<extension name="parking_pickup">
  <condition field="destination_number" expression="^\*97(7[0-9][1-9])$">
    <action application="answer"/>
    <action application="valet_park" data="lot at mydomain.com $1"/>
  </condition>
</extension>

Please test and let me know what you think... I'd like to get this
feature pushed to valet_parking instead of forking the module

Thanks


From eschmidbauer at gmail.com  Thu Feb 19 17:47:53 2015
From: eschmidbauer at gmail.com (e schmidbauer)
Date: Thu, 19 Feb 2015 09:47:53 -0500
Subject: [Freeswitch-dev] valet parking new feature
In-Reply-To: <CAJwe+CCAr4Nnu_HHb-Qm2EDx1_VfVt4pQEmot0tbM2YGyfKT6A@mail.gmail.com>
References: <CAJwe+CCAr4Nnu_HHb-Qm2EDx1_VfVt4pQEmot0tbM2YGyfKT6A@mail.gmail.com>
Message-ID: <CAJwe+CAYngXD0nifJXsDvKcDD3QDbr+gSWp1SwSPo=FY-AwPjg@mail.gmail.com>

https://freeswitch.org/jira/secure/attachment/22755/valet_sofia.c.diff

On Thu, Feb 19, 2015 at 9:43 AM, e schmidbauer <eschmidbauer at gmail.com> wrote:
> Hi,
> Would anyone want to test out new parking feature for polycoms?
> The feature is outlined in this doc:
>
> http://support.polycom.com/global/documents/support/technical/products/voice/Static_BLF_TB62475.pdf
>
> Just pull clean FS branch and do a git apply with that diff like
> git apply --ignore-whitespace ~/valet_sofia.c.diff
>
> make mod_sofia-install
> make mod_valet_parking-install
>
> restart FS
>
> Provisiong the polycom with:
>
> attendant.resourceList.2.address="orbit+lot at mydomain.com"
> attendant.resourceList.2.label="Orbit Lot"
> attendant.resourceList.2.type="automata"
>
> FreeSWITCH dialplan
> Park in Orbit:
> <extension name="parking_auto_in">
>   <condition field="destination_number" expression="^\*7$">
>     <action application="answer"/>
>     <action application="valet_park" data="lot at mydomain.com auto in 701 799"/>
>   </condition>
> </extension>
>
> Pick up from Orbit:
>
> <extension name="parking_pickup">
>   <condition field="destination_number" expression="^\*97(7[0-9][1-9])$">
>     <action application="answer"/>
>     <action application="valet_park" data="lot at mydomain.com $1"/>
>   </condition>
> </extension>
>
> Please test and let me know what you think... I'd like to get this
> feature pushed to valet_parking instead of forking the module
>
> Thanks


From krice at freeswitch.org  Fri Feb 20 18:01:54 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 20 Feb 2015 15:01:54 +0000
Subject: [Freeswitch-dev] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <54e74c62d6c22_1fcb6f328478ed@ip-10-179-128-200.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From sanatulp at gmail.com  Mon Feb 23 10:30:53 2015
From: sanatulp at gmail.com (miguelo sana)
Date: Mon, 23 Feb 2015 08:30:53 +0100
Subject: [Freeswitch-dev] Silence in conference recording
Message-ID: <CAMp9Byiqbu5iw+TGH9Xp9zC7qCVTLOrDz7-3ZbpbWTwert7WDQ@mail.gmail.com>

Hi,

I use  the recording functionality of mod_conference to record conference
calls. The recording is save to a .wav file.
Now for the second time in a few months I experienced something strange.
The recording goes silent after 15 minutes of recording. But the recording
is not stopped.
There is nothing strange found in logging. And other recordings are not
having this issue.
Is there anyone who knows anything about this and how to fix or prevent
this.


Thanks.

Miguelo
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From brian at freeswitch.org  Mon Feb 23 19:05:48 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 23 Feb 2015 10:05:48 -0600
Subject: [Freeswitch-dev] Silence in conference recording
In-Reply-To: <CAMp9Byiqbu5iw+TGH9Xp9zC7qCVTLOrDz7-3ZbpbWTwert7WDQ@mail.gmail.com>
References: <CAMp9Byiqbu5iw+TGH9Xp9zC7qCVTLOrDz7-3ZbpbWTwert7WDQ@mail.gmail.com>
Message-ID: <CAEJMVkBVZW1edwhP17Zu0Em0NnNGrsKnOm0Mz2NJqcYJdXMVHg@mail.gmail.com>

Try latest master, collect all the log info and see if you can replicate
the issue, watch signaling too, Then file a JIRA and attach all the
appropriate logs and information.

Thanks,

On Mon, Feb 23, 2015 at 1:30 AM, miguelo sana <sanatulp at gmail.com> wrote:

> Hi,
>
> I use  the recording functionality of mod_conference to record conference
> calls. The recording is save to a .wav file.
> Now for the second time in a few months I experienced something strange.
> The recording goes silent after 15 minutes of recording. But the recording
> is not stopped.
> There is nothing strange found in logging. And other recordings are not
> having this issue.
> Is there anyone who knows anything about this and how to fix or prevent
> this.
>
>
> Thanks.
>
> Miguelo
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 

*Brian West*
brian at freeswitch.org


*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From krice at freeswitch.org  Fri Feb 27 18:01:01 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 27 Feb 2015 15:01:01 +0000
Subject: [Freeswitch-dev] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <54f086ad345a_7c2652d33469530@resque-worker-ip-10-158-132-148.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From jacob at hodgeshouse.net  Sat Feb 21 04:41:45 2015
From: jacob at hodgeshouse.net (Jacob Hodges)
Date: Sat, 21 Feb 2015 11:41:45 +1000
Subject: [Freeswitch-dev] Bare Minimum Freeswitch
Message-ID: <SNT150-W7906A183F75E0668C477F1B82B0@phx.gbl>

Hi all,
I'm quite new to freeswitch and have used it in the past to establish a SIP Trunk between Lync and another SIP provider. I'm now looking at Freeswitch as a potential replacement for a legacy PBX at my local church. 

I'm extremely wary of the default Freeswitch configuration as it seem to include a lot of functionality that I don't need and a lot configuration I don't understand. All I need is the following:

- 5 extensions that can dial each other and the PSTN

- Ability to do group pickup, paging and voicemail to email

- Ability to set day/night mode

- Ability to use SPA3102 as a gateway to the PSTN.


Can anyone tell me the minimum number of modules / configuration that is needed to accomplish this?


Thanks,

Jacob. 		 	   		  

From idokan at gmail.com  Tue Feb 24 11:28:07 2015
From: idokan at gmail.com (ik)
Date: Tue, 24 Feb 2015 10:28:07 +0200
Subject: [Freeswitch-dev] mod_rayo patching
Message-ID: <CAOccsKDkdXmHyc-QFSGb5NbN74vRNbM5t5iaZWO0g0dpp+sceA@mail.gmail.com>

Hello,

There is a bug in mod_rayo that I wish to patch myself, however I have
never touched any sources of FS before, not to mention rayo itself (and
it's been many years since I touched C).

Before I'll start to figure out the proper source code to touch, is there a
good guide to read about FS development that can help in the matter ?

Thanks
Ido
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