- FreeSWITCH-users - FreeSWITCH Mailing Lists

live messenger RevEng
by Madovsky 11 Jul '11

11 Jul '11

for the geeks who d like to create a mod_live_messenger http://msnwcrec.arrozcru.org/vc_1_1.pdf http://www.codeproject.com/KB/macros/wlmplugin.aspx http://msnwcrec.arrozcru.org/ http://imfreedom.org/wiki/MSN

1 0

No voice channels to mobile.
by Henk Oegema 10 Jul '11

10 Jul '11

I'm facing the following problem: External calls come in via an ITSP (Localphone) and then routed to internal extension 1000. That works OK. If there's no answer, then the call is routed outbound via ITSP (Powervoip) to my mobile +316xxxxxxxxx) In this case there is NO voice communication at all. (a direct call from 1000 to a mobile is OK) <extension name="1000"> <condition field="destination_number" expression="^1000$"> .......................................................................................... .......................................................................................... <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="{originate_timeout=20}sofia/internal/1000%$${domain}"/> <----------audio OK <action application="bridge" data="{originate_timeout=20}sofia/gateway/powervoip/003163086xxxx"/> <------------NO audio both ways <action application="voicemail" data="default $${domain} 1000"/> </condition> </extension> How do I approach this problem? Rgds, Henk

2 2

Speech Recognition
by Roman Gelfand 10 Jul '11

10 Jul '11

I am looking to script a speech recognition ivr system. Has anyone used cmusphinx to accomplish? If not has anyone use a good open source speech recognition engine with freeswitch? Thanks in advance

4 5

segmentation fault with mod_spandsp
by Chris Cureau 10 Jul '11

10 Jul '11

Hi there! This morning, I did a fresh git checkout and attempted to build freeswitch with mod_freetdm and mod_flite defined in the modules.conf. I am now getting a segfault when compiling: making all mod_spandsp Making all in src /bin/bash: line 1: 22453 Segmentation fault ./make_modem_filter -m V.17 -r > v17_v32bis_rx_floating_rrc.h make[7]: *** [v17_v32bis_rx_floating_rrc.h] Error 139 make[6]: *** [all] Error 2 make[5]: *** [all-recursive] Error 1 make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 running make_modem_filter manually gets the same result. Twiddling with the commandline options doesn't help either. OS: Debian 6.0.1a amd64, also a fresh install from last night. Any idea what is happening? Thanks in advance!

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Re: [Freeswitch-users] One way calling (David Ponzone)
by Mitch Johnson 10 Jul '11

10 Jul '11

Yes, I did a reloadxml, and I even restarted FreeSWITCH. Thanks From: David Ponzone <david.ponzone(a)ipeva.fr> Date: July 10, 2011 2:02:13 PM EDT To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] One way calling (David Ponzone) Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> So, sorry for the obvious question, but did you reloadxml ? David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 10/07/2011 à 17:18, mitch Johnson a écrit : > David, > > Yes it is, it's the last entry in the default dialplan. > > Thankyou, > > Mitch > > ---------- Forwarded message ---------- > From: mitch Johnson <mitch.johnson7(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Sat, 9 Jul 2011 22:58:52 -0400 > Subject: [Freeswitch-users] One way calling > I have a phone hanging off a cisco callmanager, the DN is 8000. I have phones on the FS, DN 1000. When I dial from the Cisco phone to the Freeswitch phone (8000 to 1000) the phone rings. However, when I dial from Freeswitch to Cisco Callmanager, the call drops immediately, nothing rings out. > > From the FS_CLI, I see this output: > > 2011-07-09 22:43:59.209255 [NOTICE] switch_channel.c:812 New Channel sofia/internal/1000(a)172.16.200.60 [2657e237-f43d-4574-a7a3-7ef1fab749b8] > 2011-07-09 22:43:59.215279 [INFO] mod_dialplan_xml.c:331 Processing Test user <1000>->8000 in context public > 2011-07-09 22:43:59.219276 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/1000(a)172.16.200.60 to XML[8000@default] > 2011-07-09 22:43:59.221278 [INFO] mod_dialplan_xml.c:331 Processing Test user <1000>->8000 in context default > 2011-07-09 22:43:59.311287 [NOTICE] mod_dptools.c:929 Channel [sofia/internal/1000(a)172.16.200.60] has been answered > 2011-07-09 22:43:59.625253 [INFO] switch_rtp.c:2972 Auto Changing port from 68.225.42.67:21349 to 192.168.60.142:4000 > 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:189 sofia/internal/1000(a)172.16.200.60 has executed the last dialplan instruction, hanging up. > 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/1000(a)172.16.200.60 [CS_EXECUTE] [NORMAL_CLEARING] > 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1306 Session 7 (sofia/internal/1000(a)172.16.200.60) Ended > 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1000(a)172.16.200.60 [CS_DESTROY] > > > I've noticed that it looks as if it's trying to send the call to the Internet, all these calls are internal. My trunk to the Cisco Callmanager is below: > > <extension name="Dial to CM8"> > <condition field="destination_number" expression="^(800\d)$"> > <action application="bridge" data="sofia/internal/$1(a)172.16.200.100"/> > </condition> > </extension> > > > > Below is a successful call from the Callmanager to the Freeswitch: > > 2011-07-09 22:45:56.543784 [NOTICE] switch_channel.c:812 New Channel sofia/internal/8000(a)172.16.200.100 [d06d7795-0c35-4bfe-85bf-9f01498bfafc] > 2011-07-09 22:45:56.810823 [NOTICE] sofia.c:5285 Channel [sofia/internal/8000(a)172.16.200.100] has been answered > 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 <8000>->1000 in context public > 2011-07-09 22:45:56.812617 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/8000(a)172.16.200.100 to XML[1000@default] > 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 <8000>->1000 in context default > 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *1 execute_extension::dx XML features > 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/8000.2011-07-09-22-45-56.wav > 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *3 execute_extension::cf XML features > 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *4 execute_extension::att_xfer XML features > 2011-07-09 22:45:56.828985 [NOTICE] switch_channel.c:812 New Channel sofia/internal/sip:1000@192.168.60.142:60936 [846acd6a-3188-4569-9b81-2da9b3e88a87] > 2011-07-09 22:45:57.010788 [INFO] sofia.c:729 sofia/internal/sip:1000@192.168.60.142:60936 Update Callee ID to "Outbound Call" <1000> > 2011-07-09 22:45:57.010788 [NOTICE] sofia.c:4779 Ring-Ready sofia/internal/sip:1000@192.168.60.142:60936! > 2011-07-09 22:46:03.091789 [NOTICE] sofia.c:5285 Channel [sofia/internal/sip:1000@192.168.60.142:60936] has been answered > 2011-07-09 22:46:03.924748 [INFO] switch_rtp.c:2972 Auto Changing port from 68.225.42.67:22147 to 192.168.60.142:4002 > 2011-07-09 22:46:04.920744 [NOTICE] sofia.c:537 Hangup sofia/internal/sip:1000@192.168.60.142:60936 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2011-07-09 22:46:04.924746 [NOTICE] switch_ivr_bridge.c:1328 Hangup sofia/internal/8000(a)172.16.200.100 [CS_EXECUTE] [NORMAL_CLEARING] > 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 9 (sofia/internal/sip:1000@192.168.60.142:60936) Ended > 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/sip:1000@192.168.60.142:60936 [CS_DESTROY] > 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 8 (sofia/internal/8000(a)172.16.200.100) Ended > 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/8000(a)172.16.200.100 [CS_DESTROY] > > > I have done more extensive debugs and nothing really jumps out at me. > > Any help would be greatly appreciated, > > Thanks, > > Mitch > > > ---------- Forwarded message ---------- > From: David Ponzone <david.ponzone(a)ipeva.fr> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Sun, 10 Jul 2011 13:01:55 +0200 > Subject: Re: [Freeswitch-users] One way calling > Mitch > >> >> <extension name="Dial to CM8"> >> <condition field="destination_number" expression="^(800\d)$"> >> <action application="bridge" data="sofia/internal/$1(a)172.16.200.100"/> >> </condition> >> </extension> >> > > Is this extension in the default dialplan ? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)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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Re: [Freeswitch-users] One way calling (David Ponzone)
by mitch Johnson 10 Jul '11

10 Jul '11

David, Yes it is, it's the last entry in the default dialplan. Thankyou, Mitch ---------- Forwarded message ---------- > From: mitch Johnson <mitch.johnson7(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Sat, 9 Jul 2011 22:58:52 -0400 > Subject: [Freeswitch-users] One way calling > I have a phone hanging off a cisco callmanager, the DN is 8000. I have > phones on the FS, DN 1000. When I dial from the Cisco phone to the > Freeswitch phone (8000 to 1000) the phone rings. However, when I dial from > Freeswitch to Cisco Callmanager, the call drops immediately, nothing rings > out. > > From the FS_CLI, I see this output: > > 2011-07-09 22:43:59.209255 [NOTICE] switch_channel.c:812 New Channel > sofia/internal/1000(a)172.16.200.60 [2657e237-f43d-4574-a7a3-7ef1fab749b8] > 2011-07-09 22:43:59.215279 [INFO] mod_dialplan_xml.c:331 Processing Test > user <1000>->8000 in context public > 2011-07-09 22:43:59.219276 [NOTICE] switch_ivr.c:1606 Transfer > sofia/internal/1000(a)172.16.200.60 to XML[8000@default] > 2011-07-09 22:43:59.221278 [INFO] mod_dialplan_xml.c:331 Processing Test > user <1000>->8000 in context default > 2011-07-09 22:43:59.311287 [NOTICE] mod_dptools.c:929 Channel > [sofia/internal/1000(a)172.16.200.60] has been answered > 2011-07-09 22:43:59.625253 [INFO] switch_rtp.c:2972 Auto Changing port from > 68.225.42.67:21349 to 192.168.60.142:4000 > 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:189 > sofia/internal/1000(a)172.16.200.60 has executed the last dialplan > instruction, hanging up. > 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:191 Hangup > sofia/internal/1000(a)172.16.200.60 [CS_EXECUTE] [NORMAL_CLEARING] > 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1306 Session 7 > (sofia/internal/1000(a)172.16.200.60) Ended > 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1308 Close > Channel sofia/internal/1000(a)172.16.200.60 [CS_DESTROY] > > > I've noticed that it looks as if it's trying to send the call to the > Internet, all these calls are internal. My trunk to the Cisco Callmanager > is below: > > <extension name="Dial to CM8"> > <condition field="destination_number" expression="^(800\d)$"> > <action application="bridge" data="sofia/internal/$ > 1(a)172.16.200.100"/> > </condition> > </extension> > > > > Below is a successful call from the Callmanager to the Freeswitch: > > 2011-07-09 22:45:56.543784 [NOTICE] switch_channel.c:812 New Channel > sofia/internal/8000(a)172.16.200.100 [d06d7795-0c35-4bfe-85bf-9f01498bfafc] > 2011-07-09 22:45:56.810823 [NOTICE] sofia.c:5285 Channel [sofia/internal/ > 8000(a)172.16.200.100] has been answered > 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 > <8000>->1000 in context public > 2011-07-09 22:45:56.812617 [NOTICE] switch_ivr.c:1606 Transfer > sofia/internal/8000(a)172.16.200.100 to XML[1000@default] > 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 > <8000>->1000 in context default > 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *1 > execute_extension::dx XML features > 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *2 > record_session::/usr/local/freeswitch/recordings/8000.2011-07-09-22-45-56.wav > 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *3 > execute_extension::cf XML features > 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *4 > execute_extension::att_xfer XML features > 2011-07-09 22:45:56.828985 [NOTICE] switch_channel.c:812 New Channel > sofia/internal/sip:1000@192.168.60.142:60936[846acd6a-3188-4569-9b81-2da9b3e88a87] > 2011-07-09 22:45:57.010788 [INFO] sofia.c:729 sofia/internal/ > sip:1000@192.168.60.142:60936 Update Callee ID to "Outbound Call" <1000> > 2011-07-09 22:45:57.010788 [NOTICE] sofia.c:4779 Ring-Ready sofia/internal/ > sip:1000@192.168.60.142:60936! > 2011-07-09 22:46:03.091789 [NOTICE] sofia.c:5285 Channel [sofia/internal/ > sip:1000@192.168.60.142:60936] has been answered > 2011-07-09 22:46:03.924748 [INFO] switch_rtp.c:2972 Auto Changing port from > 68.225.42.67:22147 to 192.168.60.142:4002 > 2011-07-09 22:46:04.920744 [NOTICE] sofia.c:537 Hangup sofia/internal/ > sip:1000@192.168.60.142:60936 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2011-07-09 22:46:04.924746 [NOTICE] switch_ivr_bridge.c:1328 Hangup > sofia/internal/8000(a)172.16.200.100 [CS_EXECUTE] [NORMAL_CLEARING] > 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 9 > (sofia/internal/sip:1000@192.168.60.142:60936) Ended > 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close > Channel sofia/internal/sip:1000@192.168.60.142:60936 [CS_DESTROY] > 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 8 > (sofia/internal/8000(a)172.16.200.100) Ended > 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close > Channel sofia/internal/8000(a)172.16.200.100 [CS_DESTROY] > > > I have done more extensive debugs and nothing really jumps out at me. > > Any help would be greatly appreciated, > > Thanks, > > Mitch > > > ---------- Forwarded message ---------- > From: David Ponzone <david.ponzone(a)ipeva.fr> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Sun, 10 Jul 2011 13:01:55 +0200 > Subject: Re: [Freeswitch-users] One way calling > Mitch > > > <extension name="Dial to CM8"> > <condition field="destination_number" expression="^(800\d)$"> > <action application="bridge" data="sofia/internal/$ > 1(a)172.16.200.100"/> > </condition> > </extension> > > > Is this extension in the default dialplan ? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

2 1

One way calling
by mitch Johnson 10 Jul '11

10 Jul '11

I have a phone hanging off a cisco callmanager, the DN is 8000. I have phones on the FS, DN 1000. When I dial from the Cisco phone to the Freeswitch phone (8000 to 1000) the phone rings. However, when I dial from Freeswitch to Cisco Callmanager, the call drops immediately, nothing rings out. >From the FS_CLI, I see this output: 2011-07-09 22:43:59.209255 [NOTICE] switch_channel.c:812 New Channel sofia/internal/1000(a)172.16.200.60 [2657e237-f43d-4574-a7a3-7ef1fab749b8] 2011-07-09 22:43:59.215279 [INFO] mod_dialplan_xml.c:331 Processing Test user <1000>->8000 in context public 2011-07-09 22:43:59.219276 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/1000(a)172.16.200.60 to XML[8000@default] 2011-07-09 22:43:59.221278 [INFO] mod_dialplan_xml.c:331 Processing Test user <1000>->8000 in context default 2011-07-09 22:43:59.311287 [NOTICE] mod_dptools.c:929 Channel [sofia/internal/1000(a)172.16.200.60] has been answered 2011-07-09 22:43:59.625253 [INFO] switch_rtp.c:2972 Auto Changing port from 68.225.42.67:21349 to 192.168.60.142:4000 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:189 sofia/internal/1000(a)172.16.200.60 has executed the last dialplan instruction, hanging up. 2011-07-09 22:44:00.365253 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/1000(a)172.16.200.60 [CS_EXECUTE] [NORMAL_CLEARING] 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1306 Session 7 (sofia/internal/1000(a)172.16.200.60) Ended 2011-07-09 22:44:00.365253 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1000(a)172.16.200.60 [CS_DESTROY] I've noticed that it looks as if it's trying to send the call to the Internet, all these calls are internal. My trunk to the Cisco Callmanager is below: <extension name="Dial to CM8"> <condition field="destination_number" expression="^(800\d)$"> <action application="bridge" data="sofia/internal/$1(a)172.16.200.100 "/> </condition> </extension> Below is a successful call from the Callmanager to the Freeswitch: 2011-07-09 22:45:56.543784 [NOTICE] switch_channel.c:812 New Channel sofia/internal/8000(a)172.16.200.100 [d06d7795-0c35-4bfe-85bf-9f01498bfafc] 2011-07-09 22:45:56.810823 [NOTICE] sofia.c:5285 Channel [sofia/internal/ 8000(a)172.16.200.100] has been answered 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 <8000>->1000 in context public 2011-07-09 22:45:56.812617 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/8000(a)172.16.200.100 to XML[1000@default] 2011-07-09 22:45:56.812617 [INFO] mod_dialplan_xml.c:331 Processing 8000 <8000>->1000 in context default 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *1 execute_extension::dx XML features 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/8000.2011-07-09-22-45-56.wav 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *3 execute_extension::cf XML features 2011-07-09 22:45:56.817598 [INFO] switch_ivr_async.c:3013 Bound B-Leg: *4 execute_extension::att_xfer XML features 2011-07-09 22:45:56.828985 [NOTICE] switch_channel.c:812 New Channel sofia/internal/sip:1000@192.168.60.142:60936[846acd6a-3188-4569-9b81-2da9b3e88a87] 2011-07-09 22:45:57.010788 [INFO] sofia.c:729 sofia/internal/ sip:1000@192.168.60.142:60936 Update Callee ID to "Outbound Call" <1000> 2011-07-09 22:45:57.010788 [NOTICE] sofia.c:4779 Ring-Ready sofia/internal/ sip:1000@192.168.60.142:60936! 2011-07-09 22:46:03.091789 [NOTICE] sofia.c:5285 Channel [sofia/internal/ sip:1000@192.168.60.142:60936] has been answered 2011-07-09 22:46:03.924748 [INFO] switch_rtp.c:2972 Auto Changing port from 68.225.42.67:22147 to 192.168.60.142:4002 2011-07-09 22:46:04.920744 [NOTICE] sofia.c:537 Hangup sofia/internal/ sip:1000@192.168.60.142:60936 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2011-07-09 22:46:04.924746 [NOTICE] switch_ivr_bridge.c:1328 Hangup sofia/internal/8000(a)172.16.200.100 [CS_EXECUTE] [NORMAL_CLEARING] 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 9 (sofia/internal/sip:1000@192.168.60.142:60936) Ended 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/sip:1000@192.168.60.142:60936 [CS_DESTROY] 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1306 Session 8 (sofia/internal/8000(a)172.16.200.100) Ended 2011-07-09 22:46:04.929254 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/8000(a)172.16.200.100 [CS_DESTROY] I have done more extensive debugs and nothing really jumps out at me. Any help would be greatly appreciated, Thanks, Mitch

2 1

FreeSwitch/voip on EC2 or other cloud/virtual services
by Sanjay Arora 09 Jul '11

09 Jul '11

Hello all I am looking to find out from users about voip on cloud, especially FS on Amazon EC2. Google gave very few results. There were very few positive posts/articles, among them Eric Wickstrom's post on this list around 2009<http://www.spinics.net/lists/freeswitch/msg03870.html>but many seemed to be having clock/echo related problems. I would like to ask if any of you are successfully using Amazon as a voip hosting provider & what are the issues involved/stumbling blocks. I am looking to install a sme multi-tenant production system and will be selling voip minutes to a small focus target segment. I am mostly a marketing man & a small-time hobby sys-admin, so I'd also like to know how difficult it is. On a side note, I would like to solicit your opinion of Sipwise free & pro edition <http://www.sipwise.com/products/comparison/> especially with respect to deployment on Amazon cloud. With best regards & thanks. Sanjay.

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NOTIFY event not sent to user
by Tim St. Pierre 09 Jul '11

09 Jul '11

Hello, I'm trying to send notify events to Aastra phones for screen pops and that sort of thing. I'm triggering the event using ESL from a PHP script. If I subscribe to the NOTIFY events on fs_cli, I can see my event coming in quite nicely, but nothing gets sent to the phone. I am doing a packet capture between the phone and Freeswitch, and although I can see all the regular call setup and MWI packets, I don't get my notify packets coming through. Here's the event as it is seen on fs_cli: RECV EVENT Core-UUID: c8db9457-8e51-e011-8f26-001517ac7a54 FreeSWITCH-Hostname: stefan.151front.communicatefreely.net FreeSWITCH-IPv4: 66.207.200.21 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2011-07-06 15:08:28 Event-Date-GMT: Wed, 06 Jul 2011 19:08:28 GMT Event-Date-Timestamp: 1309979308360046 Event-Calling-File: mod_event_socket.c Event-Calling-Function: read_packet Event-Calling-Line-Number: 1131 Command: sendevent NOTIFY Event-Name: NOTIFY profile: internal event-string: aastra-xml user: 5101(a)communicatefreely.net host: stefan.151front.communicatefreely.net content-type: application/xml Content-Length: 223 Content-Length: 223 <AastraIPPhoneStatus Beep="yes" triggerDestroyOnExit="no"> <Session>1</Session> <Message Index="test" Type="alert" Timeout="15"> Hello, anyone there? </Message> </AastraIPPhoneStatus> freeswitch@default> Just for background, if I do sofia status profile internal user 5101(a)communicatefreely.net I get all the details about the phone, and the username and hostname's match what's in the packet above. Any suggestions?

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how-to catch errors in session streamFile
by Jason Moran 08 Jul '11

08 Jul '11

Once in a while I pass a bad file name or some other bad thing happens with session.streamFile. If I have the console up I can see an error such as [ERR] mod_sndfile.c:194 Error Opening File [the_file_path_and_name] [System error : No such file or directory.] However, session.streamFile returns false whether it gets an error or successfully plays the file. Is there any other way to capture success or capture that an error happened? -Jason Moran

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