Re: [Freeswitch-users] mod_erlang_event compile problem
Hi Andrew, Everything is running on an Ubuntu Hardy Xen domu with kernel 2.6.24-23-xen. Erlang is version R12B5 and was compiled from source with options -- enable-hipe, --enable-smp-support en --enable-threads. FS is trunk version 12197. I did copy the configuration file to ~freeswitch/conf/autoload_configs Also, I just checked the 'empd -names', after both FS and an erl shell have been started: r...@erlyfs:~# epmd -names epmd: up and running on port 4369 with data: name ldr at port 57114 name freeswitch at port 8031 So that should be fine.. I also tried loading mod_erlang_event from modules.conf, and starting FS as root, but - not surprisingly - that didn't make any difference. I've been looking in wireshark, what exactly is going over the line, and the strange thing is, that erl opens a TCP connection, a SYN packet is sent to FS, after which FS immediately returns an RST/ACK packet and thus closes the connection.. I still don't see anything in the FS CLI. Is there anything I can do to get more verbose output from FS - esp info about why the connection was closed ? thanks, Leon On Feb 23, 2009, at 1:12 AM, Andrew Thompson wrote: Leon, I can't replicate your issue, at the very least I'd expect you to see the Ignorable error in ei_accept - probable bad client version, bad cookie or bad nodename warning. What OS/Erlang version are you using? Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Random problems with cepstral text to speech
freeswitch-users-boun...@lists.freeswitch.org wrote: From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Are you using cepstral 5.1? There is a known issue with that release and it's closed source so we cannot do much about it. Cepstral 4.x works fine. Yes 5.1, my fault. I have added an initial warning here on the wiki http://wiki.freeswitch.org/wiki/Mod_cepstral although it also speaks about 5.1 and Ubuntu... Hello, if Cepstral 4.x is the way to go does anybody know where to get the demo version? BRs, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] undefined symbol: krb5_auth_con_getrcache**
got this error when starting freeswitch -latest svn 2009-02-23 13:22:50 [CRIT] switch_loadable_module.c:840 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: krb5_auth_con_getrcache** no ccompile errors, krb5libs installed, config with and w/o libcurl. br /pekka ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] OpenZAP codec Question: Why l...@8000 codec for incoming calls
Hello,
today I found in FS logfile lines like this:
2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605
switch_ivr_originate() Raw Codec Activation Success l...@8000hz 1 channel
20ms
It looks like L16 codec is used for incoming calls:
2009-02-23 15:27:03 [DEBUG] switch_core_session.c:523
switch_core_session_perform_receive_message() Send signal
OpenZAP/1:18/2799 [BREAK]
2009-02-23 15:27:03 [NOTICE] switch_ivr_originate.c:1588
switch_ivr_originate() Pre-Answer OpenZAP/1:18/2799!
2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1605
switch_ivr_originate() Raw Codec Activation Success l...@8000hz 1 channel
20ms
2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1664
switch_ivr_originate() Play Ringback Tone [%(1000, 4000, 425.0, 0)]
2009-02-23 15:27:03 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state()
Channel
sofia/internal/sip:2...@85.16.245.254:5060;user=phone;transport=udp
entering state [proceeding]
2009-02-23 15:27:03 [NOTICE] sofia.c:2779 sofia_handle_sip_i_state()
Ring-Ready
sofia/internal/sip:2...@85.16.245.254:5060;user=phone;transport=udp!
2009-02-23 15:27:03 [DEBUG] switch_core_io.c:652
switch_core_session_write_frame() OpenZAP/1:18/2799 receive message
[TRANSCODING_NECESSARY]
2009-02-23 15:27:07 [DEBUG] Span:1 Q.931() Timer 0 of call 6 (CRV: 61,
State: 0) timed out
2009-02-23 15:27:12 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state()
Channel
sofia/internal/sip:2...@85.16.245.254:5060;user=phone;transport=udp
entering state [ready]
2009-02-23 15:27:12 [DEBUG] sofia.c:2729 sofia_handle_sip_i_state()
Remote SDP:
v=0^M
o=2799 121183017 121183017 IN IP4 85.16.245.254^M
s=ATA186 Call^M
c=IN IP4 85.16.245.254^M
t=0 0^M
m=audio 16384 RTP/AVP 8 101^M
a=rtpmap:8 PCMA/8000/1^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-15^M
2009-02-23 15:27:12 [DEBUG] sofia_glue.c:2549 sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-02-23 15:27:12 [DEBUG] sofia_glue.c:1684
sofia_glue_tech_set_codec() Set Codec
sofia/internal/sip:2...@85.16.245.254:5060;user=phone;transport=udp
PCMA/8000 20 ms 160 samples
The audio codec compare function finds slightly different codecs for A
and B party.
The dialplan for incoming calls via openzap is this. I set the codec to
use in extensions bridge line:
extension name=fp_Local_Extension
condition field=destination_number
expression=(491[0-9]|492[0-8])$
action application=ring_ready/
action application=set data=ringback=${de-ring}/
action application=export
data=nolocal:sip_secure_media=${user_data(${dialed_extensi...@${domain_name}
var sip_secure_media)}/
action application=bridge
data={absolute_codec_string=PCMA}user/$...@$${domain}/
/condition
/extension
In my vars.xml config I have these codecs configured:
X-PRE-PROCESS cmd=set data=global_codec_prefs=G722,PCMA/
X-PRE-PROCESS cmd=set data=outbound_codec_prefs=G722,PCMA/
So where can I disable the L16 codec, or why is a transcoding necessary?
regards
Helmut
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Re: [Freeswitch-users] Realm Value
I could compile and install FS 1.0.2 successsfully so, do I need to install ODBC-devel package for 1.0.3 version? Thanks, Message: 5 Date: Thu, 19 Feb 2009 14:35:29 -0800 From: Michael Collins m...@freeswitch.org Subject: Re: [Freeswitch-users] Realm Value To: freeswitch-users@lists.freeswitch.org Message-ID: 87f2f3b90902191435p1c9c03aend3303dfb01349...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 On Thu, Feb 19, 2009 at 12:17 PM, Raymond Chandler intralan...@freeswitch.org wrote: did you ./configure --enable-core-odbc-suport... those errors reek of that flag with no unixODBC-devel package installed -Ray Anthony described this as a false positive on detecting ODBC. If you are in Linux you can install the ODBC-devel package and be done with it. -MC ** ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Realm Value
You have something on your system thats causing the audio detect to see you have odbc installed.. easiest way to get around this is to just install the devel headers. http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011762.html /b On Feb 23, 2009, at 8:42 AM, Ali Al-Rubaie wrote: I could compile and install FS 1.0.2 successsfully so, do I need to install ODBC-devel package for 1.0.3 version? Thanks, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] OpenZAP codec Question: Why l...@8000 codec for incoming calls
On Feb 23, 2009, at 9:44 AM, Helmut Kuper wrote:
Hello,
today I found in FS logfile lines like this:
2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605
switch_ivr_originate() Raw Codec Activation Success l...@8000hz 1
channel
20ms
It looks like L16 codec is used for incoming calls:
2009-02-23 15:27:03 [DEBUG] switch_core_session.c:523
switch_core_session_perform_receive_message() Send signal
OpenZAP/1:18/2799 [BREAK]
2009-02-23 15:27:03 [NOTICE] switch_ivr_originate.c:1588
switch_ivr_originate() Pre-Answer OpenZAP/1:18/2799!
2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1605
switch_ivr_originate() Raw Codec Activation Success l...@8000hz 1
channel
20ms
2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1664
switch_ivr_originate() Play Ringback Tone [%(1000, 4000, 425.0, 0)]
2009-02-23 15:27:03 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state()
Channel
sofia/internal/sip:2...@85.16.245.254:5060;user=phone;transport=udp
entering state [proceeding]
2009-02-23 15:27:03 [NOTICE] sofia.c:2779 sofia_handle_sip_i_state()
Ring-Ready
sofia/internal/sip:2...@85.16.245.254:5060;user=phone;transport=udp!
2009-02-23 15:27:03 [DEBUG] switch_core_io.c:652
switch_core_session_write_frame() OpenZAP/1:18/2799 receive message
[TRANSCODING_NECESSARY]
2009-02-23 15:27:07 [DEBUG] Span:1 Q.931() Timer 0 of call 6 (CRV: 61,
State: 0) timed out
2009-02-23 15:27:12 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state()
Channel
sofia/internal/sip:2...@85.16.245.254:5060;user=phone;transport=udp
entering state [ready]
2009-02-23 15:27:12 [DEBUG] sofia.c:2729 sofia_handle_sip_i_state()
Remote SDP:
v=0^M
o=2799 121183017 121183017 IN IP4 85.16.245.254^M
s=ATA186 Call^M
c=IN IP4 85.16.245.254^M
t=0 0^M
m=audio 16384 RTP/AVP 8 101^M
a=rtpmap:8 PCMA/8000/1^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-15^M
2009-02-23 15:27:12 [DEBUG] sofia_glue.c:2549
sofia_glue_negotiate_sdp()
Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-02-23 15:27:12 [DEBUG] sofia_glue.c:1684
sofia_glue_tech_set_codec() Set Codec
sofia/internal/sip:2...@85.16.245.254:5060;user=phone;transport=udp
PCMA/8000 20 ms 160 samples
The audio codec compare function finds slightly different codecs for A
and B party.
The dialplan for incoming calls via openzap is this. I set the codec
to
use in extensions bridge line:
extension name=fp_Local_Extension
condition field=destination_number
expression=(491[0-9]|492[0-8])$
action application=ring_ready/
action application=set data=ringback=${de-ring}/
action application=export
data=nolocal:sip_secure_media=${user_data(${dialed_extensi...@$
{domain_name}
var sip_secure_media)}/
action application=bridge
data={absolute_codec_string=PCMA}user/$...@$${domain}/
/condition
/extension
In my vars.xml config I have these codecs configured:
X-PRE-PROCESS cmd=set data=global_codec_prefs=G722,PCMA/
X-PRE-PROCESS cmd=set data=outbound_codec_prefs=G722,PCMA/
So where can I disable the L16 codec, or why is a transcoding
necessary?
Your playing a tone, we need to encode that tone into the codec of the
channel. You could make it stop transcoding by not providing ringback
but we are still doing some transcoding for the tone detection in
openzap that you won't see via log messages. Why is this transcoding
a problem?
Mike
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Re: [Freeswitch-users] Realm Value
I need someone with this issue to provide me ssh access to their box so I can fix this problem for everyone. No one has done so yet. Please find me on irc if you can provide access. Mike On Feb 23, 2009, at 10:16 AM, Brian West wrote: You have something on your system thats causing the audio detect to see you have odbc installed.. easiest way to get around this is to just install the devel headers. http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011762.html /b On Feb 23, 2009, at 8:42 AM, Ali Al-Rubaie wrote: I could compile and install FS 1.0.2 successsfully so, do I need to install ODBC-devel package for 1.0.3 version? Thanks, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH
Thanks Giovanni. Were you planning to check in the sample skype.conf.xml into the default FreeSWITCH conf folder? If so, just be aware the default config causes freeswitch to hang right after a load mod_skypiax (if you do not have skype running or specify a nonexistant skype user). regards, Carlos On Thu, Feb 19, 2009 at 7:32 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote: On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot carlos.tal...@gmail.com wrote: One question I have, is ringback suppose to work with mod_skypiax? Whenever I dial a number I get a few seconds of dead air before the call is answered. I've tried adding ringback and transfer_ringback into the dialplan just before the bridge command but no go. Am I missing something? Thanks. Carlos, ringback now works without tricks, and Skypiax is in trunk. Both remote ringing and early media are treated as remote ringing right now (eg: no early media, just ringing). I'll add early media support in the near future. Thanks a lot for testing and exercising skypiax, and please let me know any hint, suggestion, feature request, etc Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot carlos.tal...@gmail.com wrote: Giovannia, great work on mod_skypiax. I've been testing it under Windows and it sounds great including PSTN calls. I plan to include it as part of the Windows MSI build. One question I have, is ringback suppose to work with mod_skypiax? Whenever I dial a number I get a few seconds of dead air before the call is answered. I've tried adding ringback and transfer_ringback into the dialplan just before the bridge command but no go. Am I missing something? Thanks. regards, Carlos ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_erlang_event compile problem
On Mon, Feb 23, 2009 at 11:09:28AM +0100, Leon de Rooij wrote: Everything is running on an Ubuntu Hardy Xen domu with kernel 2.6.24-23-xen. Oh, this might explain some things.. Erlang is version R12B5 and was compiled from source with options -- enable-hipe, --enable-smp-support en --enable-threads. I'm running this too. FS is trunk version 12197. Fine too. I did copy the configuration file to ~freeswitch/conf/autoload_configs Also, I just checked the 'empd -names', after both FS and an erl shell have been started: r...@erlyfs:~# epmd -names epmd: up and running on port 4369 with data: name ldr at port 57114 name freeswitch at port 8031 So that should be fine.. Yes, that's correct. I also tried loading mod_erlang_event from modules.conf, and starting FS as root, but - not surprisingly - that didn't make any difference. I've been looking in wireshark, what exactly is going over the line, and the strange thing is, that erl opens a TCP connection, a SYN packet is sent to FS, after which FS immediately returns an RST/ACK packet and thus closes the connection.. I still don't see anything in the FS CLI. Is there anything I can do to get more verbose output from FS - esp info about why the connection was closed ? It looks like ei_accept_tmo is the one resetting the connection, not my code. I can't even get an error out of it when, for example, I telnet to port 8031, it just closes the connection instantly with no error to the console. Is it possible that something is screwy with the loopback device in a xen guest? Can you get normal erlang nodes on that host to net_adm:ping each other? Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] OpenZAP codec Question: Why l...@8000 codec for incoming calls
Hi Mike, thx. Today we had some failing test fax sessions (g711/PCMA) and my first thought was that it could be caused by FS during transcoding. So I looked into FS logfile and found those hints about transcoding. But ringback shouldn't be the problem. Fax path was from FAX device (source) through a voip cpe, through a SBC through a SS7 PSTN device into PSTN. Then from PSTN through AVAYA PBX trough sangoma A104d into freeswitch. From there RTP goes to a Cisco ATA to be converted to TDM and consumed by a FAX device (Target). So we have a lot of points to look at ... regrads Helmut Your playing a tone, we need to encode that tone into the codec of the channel. You could make it stop transcoding by not providing ringback but we are still doing some transcoding for the tone detection in openzap that you won't see via log messages. Why is this transcoding a problem? Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Mit freundlichen Grüßen Helmut Kuper Finanzdienstleistungen und Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.ku...@ewetel.de ___ EWE TEL GmbH Cloppenburger Straße 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Geschäftsführung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] OpenZAP codec Question: Why l...@8000 codec for incoming calls
Hi Mike, thx. Today we had some failing test fax sessions (g711/PCMA) and my first thought was that it could be caused by FS during transcoding. So I looked into FS logfile and found those hints about transcoding. But ringback shouldn't be the problem. Fax path was from FAX device (source) through a voip cpe, through a SBC through a SS7 PSTN device into PSTN. Then from PSTN through AVAYA PBX trough sangoma A104d into freeswitch. From there RTP goes to a Cisco ATA to be converted to TDM and consumed by a FAX device (Target). So we have a lot of points to look at ... regrads Helmut Your playing a tone, we need to encode that tone into the codec of the channel. You could make it stop transcoding by not providing ringback but we are still doing some transcoding for the tone detection in openzap that you won't see via log messages. Why is this transcoding a problem? Mike ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Deployment information and use cases
Mesquita, Relatively speaking, I feel like we are near the end of our project roll out. Perhaps the case would be stronger once everything is completed. At that time, I will be very glad to share the story on the wiki -- and hopefully elsewhere! Ben On 2/21/2009 at 8:56 AM, João Mesquita jmesqu...@gmail.com wrote: Ben, thank you for your story. I would very much like to add this to the wiki if you don't mind and everyone else agrees. What do you think guys? Use cases are _ALWAYS_ a good thing for new users. Mesquita On Feb 19, 2009, at 6:23 PM, Ben Holtsclaw wrote: Raul, I am in the process of rolling out a FreeSWITCH IP PBX solution similar to what you describe. When I was trying to procure funds for a FreeSWITCH solution, I looked for the same information you're after, but came up with little. I'll briefly describe what we're trying to accomplish, and the tools I'm using to do it. This is probably more information than what you are looking for, but maybe it will also benefit someone else. We had several schools with aging or dying PBX's or KSU's. Each site had something different system, and was supported by a different VAR. Of course, the VAR's charged some outlandish fee to make onsite repair visits. Some number of Centrex lines supplied each school's dial tone. All in all, we had a very outdated and financially draining mess. Our district's long term goal had been to move to a more unified phone system. That made sense for many reasons, the chief of which was cost. We already had a strong fiber WAN in place. Why not use that for trunking and eliminate the monthly cost of the Centrex lines? That's the path we started down. Being a public entity, we had to be sure to explore all possible avenues. We looked at everything from traditional PBX's with IP add-on modules for trunking to a full blown Cisco CallManager solution. With third party proprietary systems, we were just never able to find the sweet spot between required feature set and cost. Would Cisco have been a workable solution? Absolutely. Could our small, rural, K12 public school district afford that? Not in a million years. I looked at several software packages -- some open source, some not -- but always came back to FreeSWITCH. The scalability and active development community were major factors for us. Server Hardware. Each of our five sites has a dedicated FreeSWITCH server. For hardware, we went with Dell PowerEdge 1950's with dual quad core Xeon 2.33 GHz processors, and 4GB of RAM. I have mirrored disks set up with enough space to accommodate users' voicemail. Each server will average only about 60 voicemail boxes, and we're storing sound as MP3. Disk space shouldn't be an issue. We have always been a Novell shop, so SLES is naturally our Linux distribution of choice. We chose to go with server hardware at each site so that in the event of a WAN outage, we would still at least have intra-building and emergency communication (see below). Telephony Hardware. Each of our servers includes Sangoma hardware. We actually looked at doing IP trunking to a carrier from our network core, but decided to use telco provided PRI's instead. Presently, we have two PRI's that connect to a FreeSWITCH server at the center of our network via a Sangoma A102 dual port telephony card. All calls to and from the PSTN traverse this primary server. Servers at each remote site include one of Sangoma's A200 analog cards. Emergency calls to 911 route out over this analog card through one of at least two POTS lines that remain connected at each site. Not only does this provide some redundancy in the event of a WAN outage, but it ensures proper caller location is delivered to the 911 dispatcher. Granted, there are some other solutions for the latter, but this seemed to be the most cost effective solution for us. Telephone Desksets. We chose to go with Aastra for the telephones. The standard phone that we will place in each classroom and office is the 9143i. This is an attractive phone with an adequate feature set at a price we can afford. The person that is primarily responsible for answering the phone at each site will have an Aastra 57i and some number of 560M expansion modules. We have purchased roughly 300 Aastra desksets. Logical Layout. As new sites come online, their primary phone number is being moved from the Centrex to our PRI group. All inbound calls hit our primary server, and then FreeSWITCH bridges to the appropriate secondary server based on the DID it received. On the reverse, each servers dial plan is set up to route outbound calls (save 911) to the primary server where FreeSWITCH bridges with Openzap. Site to site calls, accomplished via four digit dialing, do not hit the primary server. Outbound calls to the PSTN deliver the site's DID as the calling number. In other words, if a user from site two calls my cell phone, I see site two's published telephone number on my caller ID. Our dial plans are set up so that receptionists at
Re: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup
On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz r.pankr...@fh-wolfenbuettel.de wrote: Hello, when hanging up a call with portaudio automatically the next call that is incoming or held is accepted. Is it possible to configure PA that way, that after hanging up (doesn't matter whether caller or callee) no call is activated automatically? I want to choose if I accept the next call or not. Thanks in advance René Just following up - did this get resolved? -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Random problems with cepstral text to speech
Hello, if Cepstral 4.x is the way to go does anybody know where to get the demo version? BRs, Claudio I think you'll have to contact Cepstral on this one. I've tried to find older revisions on their site and I can't find any way to get any voices prior to 5.1. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Deployment information and use cases
Hello, I run FreeSWITCH as a PBX solution for several companies, all sharing a single server in a vritual pbx deployment. Dialplans and user directories are all separate and handled per domains. Currently, there is about 250 phones set to use it, about 200 more will be migrated soon from Asterisk (I'm still using it as a PSTN PRI gateway). Everything is designed per domain, so it's easy to add more servers, add more sites into a company's dialplan, lcr etc. I really love FS as it saved me a lot of trouble I had with Asterisk. Ognjen (sekil) On Wed, Feb 18, 2009 at 1:20 AM, Raul Fragoso r...@etellicom.com wrote: Hello FreeSWITCHERS, My company is currently creating a suite of applications which uses FreeSWITCH as the back-end for an IP-PBX solution. We currently have a prospect to have our first customer installation - a governmental department. That is a tender to have an IP-PBX installation to connect their four office branches, each one with about 300 users - which I am sure FreeSWITCH is able to handle. Since this is an official tender, it's part of their protocol to ask about real sites using the product. Having said that, would you mind sharing some information about your experience with FreeSWITCH deployments ? No need to give many details, but a short summary with company name (if possible), when it was deployed, server equipment, number of users, number of concurrent calls, what kind of functions and services are used and overall capacity of the system. I would really appreciate if you can share that information. And if you guys agree (and explicitly manifest your agreement), I can compile the information in the FreeSWITCH wiki under a Use Cases page so it can serve as a common reference as well. Kind regards, Raul Fragoso ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Help debuging core dump
Hi Guys I'm having problems with seg faults about every 10 mins with call loads 200. I've processed the core dump (http://pastebin.freeswitch.org/7436) but I'm unsure what I should be looking for. I don't see the point where the crash occurred. Can someone point me to where I should be looking? FreeSWITCH Version 1.0.trunk (12246) Regards, ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help debuging core dump
It looks like a file rewind operation. does the lua script use the input callback to rewind a file? It maybe be a race in some other thread can you paste a thread apply all bt from the same core to look at the other threads. On Mon, Feb 23, 2009 at 12:58 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi Guys I'm having problems with seg faults about every 10 mins with call loads 200. I've processed the core dump (http://pastebin.freeswitch.org/7436) but I'm unsure what I should be looking for. I don't see the point where the crash occurred. Can someone point me to where I should be looking? FreeSWITCH Version 1.0.trunk (12246) Regards, ___ Freeswitch-users mailing list Freeswitch-users@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 FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP dump to DB
Basically, you are trying to build what Empirix has with their Hammer tool. You can create an application that is basically a mix of tshark and a database feeder. You sniff with tshark and going to basically pipe it to another application that will read the pcap file, parse it, and load it into the db for you. There are plenty of modules out there that will read pcap for you. On Fri, Feb 20, 2009 at 11:15 AM, kokoska rokoska kokoska.roko...@post.czwrote: jonathan augenstine napsal(a): You can tcpdump and then use wireshark to graph the calls. When the dump is displayed in wireshark, select 'Statistics' - VoIP Calls. You will see a display of all VoIP calls. Select the one you want graphed, or select them all and you will see REINVITE and REFER interaction as well as RTP streams. Thank you very much, jonathan, for your interest! I use ngrep+wireshark many times a day, but I'm affraid it is not suitable for that amount of data. Even with few hundreds MiBs of pcap file wireshark becoms very slow and I can't imagine how to load 50-100 GiB file with milions of calls and try to search for one of them :-) And, even worse, I should rotate the file and, don't end with call divided to multiple files... Best regards, kokoska.rokoska ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP dump to DB
Joseph Bajin napsal(a): Basically, you are trying to build what Empirix has with their Hammer tool. Thank you very much, Joseph, for your interest! I have never heard about Empirix (I'll look at it), but what I'm trying to build is something like SER/Kamailio/OpenSIPS sip_trace module. You can create an application that is basically a mix of tshark and a database feeder. You sniff with tshark and going to basically pipe it to another application that will read the pcap file, parse it, and load it into the db for you. There are plenty of modules out there that will read pcap for you. Thank you once more, Joseph, for suggestion! I think about it - it will be challenge for me to write robust and still fast enough (thousands messages per second) SIP parser + DB feeder :-) Best regards, kokoska.rokoska ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] FREESwitch on Windows Server 2003
Hello: I have successfully loaded the Windows implementation (SVN 11602 - 02/02/09) from your site and it runs fine. I configured a Linksys SPA 2102 and have acquired dial tone and the '999X' tests work. I have not been able to establish connection with either FreeWorldDialup or Broadvoice as of yet. Which files do I need to edit and what are the proper entries to enable connection to FreeWorldDialup and Broadvoice? Example files and where they reside in the file structure would be very much appreciated. Thank you All the Best, Steve Steve Walker President SONASEARCH, INC 425/883-1984 NOTICE: The information contained in this document is intended by Sonasearch, Inc. or one of its subsidiaries for the use of the named individuals or entities to which it is addressed and may contain information that is privileged or otherwise confidential. It is not intended for transmission to, or receipt by, any individual or entity other than the named addressee (or a person authorized to deliver it to the named addressee) except as otherwise expressly permitted in this document. If you have received this document in error, please destroy it without copying or forwarding it, and notify the sender of the error by calling Sonasearch at (425) 883-1984. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_erlang_event compile problem
Leon, I think I found the problem. I shouldn't have been defaulting to binding to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the module to actually bind to 0.0.0.0 correctly and made it the default in the config file. Erlang nodes by default bind to 0.0.0.0, so I decided to make mod_erlang_event follow suit. Please give that a shot and see if it fixes things. Andrew ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FREESwitch on Windows Server 2003
On Mon, Feb 23, 2009 at 4:47 PM, Stephen Walker swal...@sonasearch.comwrote: Which files do I need to edit and what are the proper entries to enable connection to FreeWorldDialup and Broadvoice? Example files and where they reside in the file structure would be very much appreciated. You'll need to place a gateway configuration for Broadvoice in conf/sip_profiles/external similar to this example: http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Broadvoice The same applies to FWD. http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Free_World_Dialup_.28FWD.29 Once the gateways are configured you'll need to modify the default dial plan to recognize these gateways: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Dialing_out_via_Gatewayfor dialing out and http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gatewayfor incoming. Most of this is actually covered here: http://wiki.freeswitch.org/wiki/Installation_Guide#Windows_quick_start regards, Carlos ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP dump to DB
If you write it correctly it will work just fine. That is how most of all the other correlation engines work. Your setup is not going to be bigger than some of the large telecoms that use these systems today. On 2/23/09, kokoska.rokoska kokoska.roko...@post.cz wrote: Joseph Bajin napsal(a): Basically, you are trying to build what Empirix has with their Hammer tool. Thank you very much, Joseph, for your interest! I have never heard about Empirix (I'll look at it), but what I'm trying to build is something like SER/Kamailio/OpenSIPS sip_trace module. You can create an application that is basically a mix of tshark and a database feeder. You sniff with tshark and going to basically pipe it to another application that will read the pcap file, parse it, and load it into the db for you. There are plenty of modules out there that will read pcap for you. Thank you once more, Joseph, for suggestion! I think about it - it will be challenge for me to write robust and still fast enough (thousands messages per second) SIP parser + DB feeder :-) Best regards, kokoska.rokoska ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sent from my mobile device --Joe ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP dump to DB
Joseph Bajin napsal(a): If you write it correctly it will work just fine. Yes, this is challenge I have talked about :-) That is how most of all the other correlation engines work. I don't have enough informations but from what I heard from friendly competitors they are usualy log (SIP|ISUP) messages after they are parsed by their routing servers and not run separate tshark+parser+logger. Or they duplicate (just) SIP messages to separate machine and parse and log them there (SERlike server + sip_trace). Your setup is not going to be bigger than some of the large telecoms that use these systems today. I hope so :-) Thanks once more, Joseph, for your info! Best regards, kokoska.rokoska ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup
No, unfortunately the problem still persists. Portaudio still automatically accepts/takes the next call. René On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz r.pankr...@fh-wolfenbuettel.de wrote: Hello, when hanging up a call with portaudio automatically the next call that is incoming or held is accepted. Is it possible to configure PA that way, that after hanging up (doesn't matter whether caller or callee) no call is activated automatically? I want to choose if I accept the next call or not. Thanks in advance René Just following up - did this get resolved? -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
