[Freeswitch-users] Sofia SIP external profile gateway names
Hi, I am facing a problem with Sofia SIP external profile. Basically i have 10+ accounts, say a101 - a110, to a single service provider, say xyz.com. For each account a have created a gateway in external profile's directory i.e. /usr/local/freeswitch/conf/sip_profiles/external/a101.xml up to a110.xml. At first the problem was none of the accounts were registering since, FS was trying to send registration requests to host a101 instead of xyz.com, but later when i set xyz.com as proxy and register proxy address, it successfully registered all accounts. Now i am facing a similar problem in dialplan, for example if i try to dialout via gateway a101, call immediately fails with NORMAL_TEMPORARY_FAILURE. When i trun on sip tracing i don't see any INVITE message sent to provider xyz.com. 2009-10-22 06:51:55.325393 [NOTICE] switch_channel.c:613 New Channel sofia/external/00923344224...@a101 [b7663caf-cd29-4213-9059-ae880d49b0ca] 2009-10-22 06:51:55.516382 [NOTICE] sofia.c:4039 Hangup sofia/external/00923344224...@a101 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] API CALL [originate(sofia/external/00923344224...@a101 )] output: -ERR NORMAL_TEMPORARY_FAILURE 2009-10-22 06:51:55.527381 [NOTICE] switch_core_session.c:1132 Session 2 (sofia/external/00923344224...@a101) Ended 2009-10-22 06:51:55.527381 [NOTICE] switch_core_session.c:1134 Close Channel sofia/external/00923344224...@a101 [CS_DESTROY] I think its trying to look up a101 instead of xyz.com to send INVITE. Kindly help. Thank you. -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ 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] Call custom variable in condition
Hi Michael, The feature is already documented here: http://wiki.freeswitch.org/wiki/Dialplan_XML#Clarification http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions Perhaps the reason *why* it's the way it is can be expanded a bit ? regards, Leon On Oct 21, 2009, at 7:02 PM, Michael Collins wrote: On Wed, Oct 21, 2009 at 9:39 AM, Anthony Minessale anthony.miness...@gmail.com wrote: It not only makes sense it's well documented on the wiki page. The set line is not happening right when it's encountered, the set line is copied into the channel and executed later after the whole dialplan is parsed. The dialplan is a pre-processor not a runtime engine. Here is a new feature in pre-1.0.5 (svn trunk) Some applications like set can now be executed within the dialplan but you should use it sparingly. action application=set data=testing=true inline=true/ I'm getting ready to document this feature. For the sake of edification, why is it best to use this sparingly, other than wide- spread use making dialplans all cluttered? -MC The inline=true makes it execute inside the dialplan and it's never copied into your resulting extension because it's executed immediately. ___ 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] UUID of the newly originated call?
I'm using 'FreeSWITCH Version 1.0.trunk (15106)'. Anthony Minessale-2 wrote: which revision of FS are you using? On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. nagale...@gmail.com wrote: I've tried with origination_uuid. First, I tried with SIP and my program executes successfully as what I expected. This program initiates a new call when a call comes and let the new call to eavesdrop the landed call. When, I experimented with PRI(openzap), I'm facing the following error. And, I am unable to make any calls(even from CLI). It is reporting the same error for the subsequent calls. Error:2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9] Full log is in http://pastebin.freeswitch.org/10780 My script is here, http://pastebin.freeswitch.org/10781 What is this error for and how to avoid this? Is there any other way to get the uuid of the originated call except explicitly defining(origination_uuid).?! -- Regards, Nagalenoj H. ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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 -- View this message in context: http://www.nabble.com/Re%3A-UUID-of-the-newly-originated-call--tp25987024p26006565.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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-users Digest, Vol 40, Issue 179
cond would be helpful here? I updated the wiki on this one just now
with a bit more detail. It is a api call. so, you'd use it like:
${cond(eval ? trueval : falseval)}
so to get a value of ERR if the var my myvar is 15 you could:
${cond(${myvar} 15 ? ERR : OK)}
If both sides of the comparison operator are numeric then it does
numeric comparison otherwise it does lexical string comparison.
On Wed, Oct 21, 2009 at 11:41 PM, Michael Collins m...@freeswitch.org wrote:
On Wed, Oct 21, 2009 at 9:08 PM, Ahmed Munir ahmedmunir...@gmail.com
wrote:
Hi,
Thanks for reply, it really helped me. One more thing to ask, how can we
make decision against ,, =, = in condition header? Like we use == for
action and != for anti-action.
Kindly highlight it.
You can only do greater than and less than in the date/time matching. See
the date/time example in the default.xml dialplan file.
You can also use regular expressions if you're in a pinch. For example, if
you need to match numbers = 1100 and = 1500 you could just use this regex:
^(1[1234]\d\d|1500)$
The real question, though, is this: what types of values do you need to
match for GT or LT? Date/time? Money? Other? That will determine if you need
to use a script or just the dialplan.
-MC
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Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
TC
TCI have enabled crash-protection and when i do SIP = H323 call it
doesn't
TCgenerate coredumps... it is just this thread that is crashing ... pls
check
TCthe log downbelow:
core dump in case enabled crash-protection may be unusable, i have a case
then
my module crash silently, after this crash-protection is killing sip leg
and after
this i get core dump where backtrace show me segfault in libc6, i spent one
day to
understand this situation, and then i disable crash-protection i see there
is actualy
it crashes. disable crash-protection and show backtrace of crash, i think
result will
be different.
TC2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC==FSH323Connection::decodeCapability
TC
TC
TC
TCWell, I'm not sure if the backtrace is from 1.0.4 ... i will disable
TCcrass-protection and will send new logs to you.
TC
TC
TCAlso, if you like i can give you access to the machine itself...
TC
TCT.
TC
Hi, here is the FS log without crash-protection:
http://pastebin.freeswitch.org/10796 and here is the backtrace:
http://pastebin.freeswitch.org/10797
my dialplan looks ok, so i guess it is up to the module.
extension name=ENYTHING_ELSE
condition field=destination_number
expression=^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
!--action application=set data=bypass_media=false/--
action application=set data=proxy_media=true/
!--action application=bridge data=opal/h323:0...@${ncx_ip}/--
action application=bridge data=h323/0...@${ncx_ip}/
/condition
/extension
please advice,
T.
C уважением With Best Regards
Георгиевский Юрий.Georgiewskiy Yuriy
+7 4872 711666+7 4872 711666
факс +7 4872 711143 fax +7 4872 711143
Компания ООО Ай Ти Сервис IT Service Ltd
http://nkoort.ru http://nkoort.ru
JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru
YG129-RIPEYG129-RIPE
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[Freeswitch-users] srtp g729 bypass
Hi I have set up a freeswitch with TLS and SRTP support. I'm sending encrypted calls to Freswitch and the Freeswitch forwards the calls to an asterisk unencrypted. I have issue by using G729 in this scenario. My UA supports g279, asterisk supports g729 transcoding and I understood that freeswitch supports the g729 in bypass mode. Should it work? What I have to change in config? So the scenario, I would like to do: UA|==TLS+SRTP(G729)=|FREESWITCH|ClearSIP+RTP(G729)|Asterisk Thank you in advance. Szasz Szabolcs ___ 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] Fwd: mod_opal - call charged before H.225 connect
TCHi, here is the FS log without crash-protection: TChttp://pastebin.freeswitch.org/10796 and here is the backtrace: TChttp://pastebin.freeswitch.org/10797 i fix this crash already, please download latest version from same link as previous, recompile and try again. outgoing works, I can place an end-to-end call ... except the RTP is realy delayed ... after approx 30 sec of conversation the audio is delayed more than 10 seconds but i have 2 way audio for outgoing calls:) Do you need some logs ? Inbound cals still the same... i suppose you didn't have a chance working on that as well ... T. ___ 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] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-22 15:59 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TC TC TC TCHi, here is the FS log without crash-protection: TC TChttp://pastebin.freeswitch.org/10796 and here is the backtrace: TC TChttp://pastebin.freeswitch.org/10797 TC TC i fix this crash already, please download latest version from same link TC as previous, recompile and try again. TC TC TCoutgoing works, I can place an end-to-end call ... except the RTP is realy TCdelayed ... after approx 30 sec of conversation the audio is delayed more TCthan 10 seconds but i have 2 way audio for outgoing calls:) TC TCDo you need some logs ? try disable medai-proxy, there is issue with rtp now then medai-proxy or transcoding enabled. TCInbound cals still the same... i suppose you didn't have a chance working on TCthat as well ... sorry i don't remember what the same, show extension and logs of inbound call. at this time at this time i have working transit h323-sip in both direction, ivr is work too, there is some issues present but basicaly it work. C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE___ 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] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TCOn Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga tculj...@gmail.com wrote: TC TC TC TCHi, here is the FS log without crash-protection: TC TChttp://pastebin.freeswitch.org/10796 and here is the backtrace: TC TChttp://pastebin.freeswitch.org/10797 TC TC i fix this crash already, please download latest version from same link TC as previous, recompile and try again. TC TC TC outgoing works, I can place an end-to-end call ... except the RTP is realy TC delayed ... after approx 30 sec of conversation the audio is delayed more TC than 10 seconds but i have 2 way audio for outgoing calls:) TC TC TCone more thing ... it is H323 endpoint = SIP phone audio that is delayed. TCSIP phone = H323 endpoint is ok! hm, i have such issue but in reverce direction now. TC Do you need some logs ? TC TC TC Inbound cals still the same... i suppose you didn't have a chance working TC on that as well ... TC TC T. TC TC C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE___ 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] Fwd: mod_opal - call charged before H.225 connect
crash protection has been completely removed from FreeSWITCH, I certianly hope you are working on this against SVN trunk? Also you have been given an svn area and a jira category for this so you should move all the info from this thread to jira http://jira.freeswitch.org It's much easier to collaberate this kind of development when you have the code in SVN. 2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TCOn Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga tculj...@gmail.com wrote: TC TC TC TCHi, here is the FS log without crash-protection: TC TChttp://pastebin.freeswitch.org/10796 and here is the backtrace: TC TChttp://pastebin.freeswitch.org/10797 TC TC i fix this crash already, please download latest version from same link TC as previous, recompile and try again. TC TC TC outgoing works, I can place an end-to-end call ... except the RTP is realy TC delayed ... after approx 30 sec of conversation the audio is delayed more TC than 10 seconds but i have 2 way audio for outgoing calls:) TC TC TCone more thing ... it is H323 endpoint = SIP phone audio that is delayed. TCSIP phone = H323 endpoint is ok! hm, i have such issue but in reverce direction now. TC Do you need some logs ? TC TC TC Inbound cals still the same... i suppose you didn't have a chance working TC on that as well ... TC TC T. TC TC C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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] Fwd: mod_opal - call charged before H.225 connect
Have you started moving the code into our SVN and using our ticketing / issue tracker to help you manage issues? /b On Oct 22, 2009, at 9:13 AM, Georgiewskiy Yuriy wrote: On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre ...: TCOn Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga tculj...@gmail.com wrote: TC TC TC TCHi, here is the FS log without crash-protection: TC TChttp://pastebin.freeswitch.org/10796 and here is the backtrace: TC TChttp://pastebin.freeswitch.org/10797 TC TC i fix this crash already, please download latest version from same link TC as previous, recompile and try again. TC TC TC outgoing works, I can place an end-to-end call ... except the RTP is realy TC delayed ... after approx 30 sec of conversation the audio is delayed more TC than 10 seconds but i have 2 way audio for outgoing calls:) TC TC TCone more thing ... it is H323 endpoint = SIP phone audio that is delayed. TCSIP phone = H323 endpoint is ok! hm, i have such issue but in reverce direction now. TC Do you need some logs ? TC TC TC Inbound cals still the same... i suppose you didn't have a chance working TC on that as well ... TC TC T. TC TC C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129- RIPE___ 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] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-22 09:30 -0500, Brian West wrote freeswitch-us...@lists.freeswit...: hm, you not tell me what account created, and i don't try to do this. BWHave you started moving the code into our SVN and using our BWticketing / issue tracker to help you manage issues? BW BW/b BW BWOn Oct 22, 2009, at 9:13 AM, Georgiewskiy Yuriy wrote: BW BW On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre BW ...: BW BW TCOn Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga tculj...@gmail.com BW wrote: BW TC BW TC BW TC TCHi, here is the FS log without crash-protection: BW TC TChttp://pastebin.freeswitch.org/10796 and here is the BW backtrace: BW TC TChttp://pastebin.freeswitch.org/10797 BW TC BW TC i fix this crash already, please download latest version from BW same link BW TC as previous, recompile and try again. BW TC BW TC BW TC outgoing works, I can place an end-to-end call ... except the BW RTP is realy BW TC delayed ... after approx 30 sec of conversation the audio is BW delayed more BW TC than 10 seconds but i have 2 way audio for outgoing calls:) BW TC BW TC BW TCone more thing ... it is H323 endpoint = SIP phone audio that is BW delayed. BW TCSIP phone = H323 endpoint is ok! BW BW hm, i have such issue but in reverce direction now. BW BW TC Do you need some logs ? BW TC BW TC BW TC Inbound cals still the same... i suppose you didn't have a BW chance working BW TC on that as well ... BW TC BW TC T. BW TC BW TC BW BW C уважением With Best Regards BW Георгиевский Юрий.Georgiewskiy Yuriy BW +7 4872 711666+7 4872 711666 BW факс +7 4872 711143 fax +7 4872 711143 BW Компания ООО Ай Ти Сервис IT Service Ltd BW http://nkoort.ru http://nkoort.ru BW JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru BW YG129-RIPEYG129- BW RIPE___ BW FreeSWITCH-users mailing list BW FreeSWITCH-users@lists.freeswitch.org BW http://lists.freeswitch.org/mailman/listinfo/freeswitch-users BW UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- BW users BW http://www.freeswitch.org BW BW BW___ BWFreeSWITCH-users mailing list BWFreeSWITCH-users@lists.freeswitch.org BWhttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users BWUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users BWhttp://www.freeswitch.org BW C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE___ 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] Fwd: mod_opal - call charged before H.225 connect
AS per the email you and I exchanged we created the account and the mod_h323 folder in endpoints /b On Oct 22, 2009, at 9:34 AM, Georgiewskiy Yuriy wrote: hm, you not tell me what account created, and i don't try to do this. ___ 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] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-22 09:27 -0500, Anthony Minessale wrote freeswitch-us...@lists.f...: AMcrash protection has been completely removed from FreeSWITCH, I certianly AMhope you are working on this against SVN trunk? i don't have trunk at this time, my current work is based on 1.0.4 version. AMAlso you have been given an AMsvn area and a jira category for this so you should move all the info from AMthis thread to jira http://jira.freeswitch.org AM AMIt's much easier to collaberate this kind of development when you have the AMcode in SVN. AM AM AM2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru AM AM On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote AM freeswitch-us...@lists.fre...: AM AM TCOn Thu, Oct 22, 2009 at 3:59 PM, Tihomir Culjaga tculj...@gmail.com AM wrote: AM TC AM TC AM TC TCHi, here is the FS log without crash-protection: AM TC TChttp://pastebin.freeswitch.org/10796 and here is the backtrace: AM TC TChttp://pastebin.freeswitch.org/10797 AM TC AM TC i fix this crash already, please download latest version from same AM link AM TC as previous, recompile and try again. AM TC AM TC AM TC outgoing works, I can place an end-to-end call ... except the RTP is AM realy AM TC delayed ... after approx 30 sec of conversation the audio is delayed AM more AM TC than 10 seconds but i have 2 way audio for outgoing calls:) AM TC AM TC AM TCone more thing ... it is H323 endpoint = SIP phone audio that is AM delayed. AM TCSIP phone = H323 endpoint is ok! AM AM hm, i have such issue but in reverce direction now. AM AM TC Do you need some logs ? AM TC AM TC AM TC Inbound cals still the same... i suppose you didn't have a chance AM working AM TC on that as well ... AM TC AM TC T. AM TC AM TC AM AM C уважением With Best Regards AM Георгиевский Юрий.Georgiewskiy Yuriy AM +7 4872 711666+7 4872 711666 AM факс +7 4872 711143 fax +7 4872 711143 AM Компания ООО Ай Ти Сервис IT Service Ltd AM http://nkoort.ru http://nkoort.ru AM JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru AM YG129-RIPEYG129-RIPE AM AM ___ AM FreeSWITCH-users mailing list AM FreeSWITCH-users@lists.freeswitch.org AM http://lists.freeswitch.org/mailman/listinfo/freeswitch-users AM UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users AM http://www.freeswitch.org AM AM AM AM AM C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE___ 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] Porting Freeswitch to ARM?
Thanks Mike for the link. I'll investigate more whether running FS on ARM-based devices is a good idea. For those interested, another thread on the subject: http://www.nabble.com/Freeswitch-vs.-Asterisk-speed-on-ARM-td25086585.html -- View this message in context: http://www.nabble.com/Porting-Freeswitch-to-ARM--tp25531239p26011319.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] Fwd: mod_opal - call charged before H.225 connect
TC
TCDo you need some logs ?
try disable medai-proxy, there is issue with rtp now then medai-proxy or
transcoding enabled.
Outbound calls:
disabled rtp proxy and it is still the same issue ... audio delay H323 =
SIP endpoint.
Inbound calls:
This is the extension i use to register my Avaya SIP phone to FS.
include
user id=1001
params
param name=password value=$${default_password}/
param name=vm-password value=1001/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1001/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1001/
variable name=effective_caller_id_number value=1001/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
/variables
/user
/include
This is my h323.conf.xml
configuration name=h323.conf description=H323 Endpoints
settings
param name=trace-level value=4/
param name=context value=default/
param name=dialplan value=XML/
param name=codec-prefs value=PCMU,PCMA,GSM,G729,G726/
param name=gk-address value=/!-- empty to disable, * to
search LAN --
param name=gk-identifer value=/ !-- optional name of gk --
param name=gk-interface value=/ !-- optional listener interface
name --
/settings
listeners
listener name=default
param name=h323-ip value=10.4.62.7/
param name=h323-port value=1720/
/listener
/listeners
/configuration
I'm using default context and an inbound call looks for a registered user in
default context where 1001 user is registered to.
here is the log for an outgoing call:
http://pastebin.freeswitch.org/10799and here is a tshark output:
http://pastebin.freeswitch.org/10800
there are 2 thing that are not working here:
1. no audio at all!
2. hangup from SIP User side doesn't release the H323 leg
two points for your reference in the logs:
1. Here, SIP User disconnected the SIP leg, but nothing was triggered in
mod_h323 ... as the callback function on_hangup (in mod_h323.cpp) was
never triggered!
freeswi...@subzero
freeswi...@subzero
freeswi...@subzero recv 371 bytes from udp/[10.4.62.89]:5060 at
14:39:36.714521:
BYE sip:mod_so...@10.4.62.7:5060 SIP/2.0
From: sip:1...@10.4.62.89 sip%3a1...@10.4.62.89
;transport=udp;tag=-22166b474ae08abf-7_T10.4.62.89
To: sip:1001282...@10.4.62.7 sip%3a1001282...@10.4.62.7
;tag=Qpc53NZ2cZc1N
Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
CSeq: 127 BYE
Via: SIP/2.0/UDP
10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
Content-Length: 0
Max-Forwards: 70
Supported: replaces
2009-10-22 16:39:36.714604 [NOTICE] sofia.c:322 Hangup sofia/internal/
sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [CS_CONSUME_MEDIA]
[NORMAL_CLEARING]
2009-10-22 16:39:36.714604 [DEBUG] switch_channel.c:1683 Send signal
sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [KILL]
2009-10-22 16:39:36.714604 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [BREAK]
send 520 bytes to udp/[10.4.62.89]:5060 at 14:39:36.715258:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
From: sip:1...@10.4.62.89 sip%3a1...@10.4.62.89
;transport=udp;tag=-22166b474ae08abf-7_T10.4.62.89
To: sip:1001282...@10.4.62.7 sip%3a1001282...@10.4.62.7
;tag=Qpc53NZ2cZc1N
Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
CSeq: 127 BYE
User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:503
(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) State
CONSUME_MEDIA going to sleep
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) Running State
Change CS_HANGUP
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) State HANGUP
2009-10-22 16:39:36.721097 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 hanging up, cause:
NORMAL_CLEARING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 Standard HANGUP,
cause: NORMAL_CLEARING
2009-10-22 16:39:36.721097
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-22 09:27 -0500, Anthony Minessale wrote freeswitch-us...@lists.f...: AMcrash protection has been completely removed from FreeSWITCH, I certianly AMhope you are working on this against SVN trunk? i don't have trunk at this time, my current work is based on 1.0.4 version. Yuriy, it is better if we move this through a jira ticket, this way it is a mess. So if you agree, we can open a ticket where we can follow up all issues with mod_h323. Also, the same applies to FS trunk... first i wanted to see if i was doing something wrong when i tried your module. Now, when you fixed outgoing calls it is time to go on trunk as when we finish this 1.0.4 will be outdated and obsolete. so, to continue on this topic i suggest: 1. open a jira ticket 2. move to fs-trunk 3. upload the current src of mod_h323 to the FSSVN do you agree ? Tihomir. ___ 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] Problem when load CDR Data on the database in Real Time
Hello Anyone, Now, I'm attempting to load CDR Data on the database in Real Time. by following the instruction on this link http://wiki.freeswitch.org/wiki/CDR However, when trying to send create_table.rb to the database, I'm still struggling with connecting to mySQL database Note: my Operation System is WindowXP 2002. the ruby version that I used is 1.9.1 p129. Below is the error message show in the Command Prompt: - C:\cdrruby create_table.rb C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters/mysql.rb:157 :in `query': Mysql::Error: Commands out of sync; you can't run this command now (Sequel::DatabaseDisconnectError) from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters /mysql.rb:157:in `_execute' from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters /mysql.rb:140:in `block in execute' from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/connecti on_pool.rb:112:in `hold' from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database .rb:482:in `synchronize' from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/adapters /mysql.rb:140:in `execute' from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database .rb:313:in `execute_dui' from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database .rb:306:in `execute_ddl' from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database /schema_methods.rb:188:in `create_table_from_generator' from C:/Ruby19/lib/ruby/gems/1.9.1/gems/sequel-3.5.0/lib/sequel/database /schema_methods.rb:73:in `create_table' from create_table.rb:6:in `main' main:306: [BUG] Segmentation fault ruby 1.9.1p129 (2009-05-12 revision 23412) [i386-mswin32] -- control frame -- c:0001 p: s:0002 b:0002 l:0010c4 d:0010c4 TOPmain:306 --- -- Ruby level backtrace information- [NOTE] You may encounter a bug of Ruby interpreter. Bug reports are welcome. For details: http://www.ruby-lang.org/bugreport.html This application has requested the Runtime to terminate it in an unusual way. Please contact the application's support team for more information. C:\cdr - Has anyone ever had this problem? Pleaes advise. PB. _ Windows 7: Simplify your PC. Learn more. http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen1:102009___ 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] Domain Question
I have a noobish question about setting up FS. I have it installed and running. I setup a soft client on the machine fs is on and point it to the ip address of the FS instance and it registers with no issues. I then setup an entry in my etc/hosts files mydomain.localhost and changed the domain in the soft client. The registration now fails. with the soft phone giving a 503 error service unavailable And I am watching the cli and I see no errors come through with the domain but I see it register with the IP :( I change it back to the ip address and it works. Any direction you guys could point me in would be great. ___ 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] Qustion about INFO messages after Connect/Answer
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Mike,
here it is:
Dialplan:
extension name=Local_Extension
condition field=destination_number expression=^(10[01][0-9])$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=set data=transfer_ringback=$${hold_music}/
action application=set data=hangup_after_bridge=true/
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
/condition
/extension
Debug-Log:
recv 1521 bytes from udp/[85.16.245.206]:1024 at 15:41:02.405834:
INVITE sip:1...@85.16.246.12:5061;user=phone SIP/2.0
Via: SIP/2.0/UDP 85.16.245.206:1024;branch=z9hG4bK-i48n75d62qts;rport
From: 1001 an PBX1 sip:1...@85.16.246.12:5061;tag=7xpim4o1go
To: sip:1...@85.16.246.12:5061;user=phone
Call-ID: 3c31304b7a80-no9xsnjj0bol
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:1...@85.16.245.206:1024;line=eg3wp69a;reg-id=1
X-Serialnumber: 0004134002CB
P-Key-Flags: resolution=31x13, keys=4
User-Agent: snom820/8.2.16
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 732
v=0
o=root 411395140 411395140 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 49852 RTP/SAVP 0 8 9 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:Um06txYC7vwXdZDohXJ15te5hZ/iWmPd4voRbdby
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=audio 49852 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
send 331 bytes to udp/[85.16.245.206]:1024 at 15:41:02.406500:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
85.16.245.206:1024;branch=z9hG4bK-i48n75d62qts;rport=1024
From: 1001 an PBX1 sip:1...@85.16.246.12:5061;tag=7xpim4o1go
To: sip:1...@85.16.246.12:5061;user=phone
Call-ID: 3c31304b7a80-no9xsnjj0bol
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15180M
Content-Length: 0
2009-10-22 17:41:02.406509 [DEBUG] sofia.c:4907 IP 85.16.245.206
Approved by acl clients[]. Access Granted.
2009-10-22 17:41:02.406509 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/1...@85.16.246.12:5061 [4d941750-bf21-11de-9c3f-adfc1789590a]
2009-10-22 17:41:02.407509 [DEBUG] sofia.c:3493 Channel
sofia/internal/1...@85.16.246.12:5061 entering state [received][100]
2009-10-22 17:41:02.407509 [DEBUG] sofia.c:3500 Remote SDP:
v=0
o=root 411395140 411395140 IN IP4 85.16.245.206
s=call
c=IN IP4 85.16.245.206
t=0 0
m=audio 49852 RTP/SAVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:Um06txYC7vwXdZDohXJ15te5hZ/iWmPd4voRbdby
a=ptime:20
m=audio 49852 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
2009-10-22 17:41:02.407509 [DEBUG] sofia.c:3621
(sofia/internal/1...@85.16.246.12:5061) State Change CS_NEW - CS_INIT
2009-10-22 17:41:02.407509 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1...@85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:306
(sofia/internal/1...@85.16.246.12:5061) Running State Change CS_INIT
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/1...@85.16.246.12:5061) State INIT
2009-10-22 17:41:02.407509 [DEBUG] mod_sofia.c:83
sofia/internal/1...@85.16.246.12:5061 SOFIA INIT
2009-10-22 17:41:02.407509 [DEBUG] mod_sofia.c:111
(sofia/internal/1...@85.16.246.12:5061) State Change CS_INIT - CS_ROUTING
2009-10-22 17:41:02.407509 [DEBUG] switch_core_session.c:977 Send signal
sofia/internal/1...@85.16.246.12:5061 [BREAK]
2009-10-22 17:41:02.407509 [DEBUG] switch_core_state_machine.c:330
Re: [Freeswitch-users] Proxy media mode with T.38 re-invite
An update for Tony, Brian, Mike, and everyone on the list... I was able to get some phone time with the team yesterday. Tony worked on my machine, found the issue, and had it committed within 30 minutes. I've been testing T.38 all morning between the fax machines in the office with few issues. THANKS AGAIN GUYS! On Wed, Oct 21, 2009 at 2:54 PM, Anthony Minessale anthony.miness...@gmail.com wrote: can you try trunk and let me know right away, if it's still not working i may need ssh access and call you on the phone. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ 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] Proxy media mode with T.38 re-invite
On Thu, Oct 22, 2009 at 5:44 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: An update for Tony, Brian, Mike, and everyone on the list... I was able to get some phone time with the team yesterday. Tony worked on my machine, found the issue, and had it committed within 30 minutes. I've been testing T.38 all morning between the fax machines in the office with few issues. and what these few issues are? :P THANKS AGAIN GUYS! ___ 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] Domain Question
Im going to guess its because mydomain.localhost doesn't resolve outside the machine itself so the softphone never ends up knowing wtf to do. /b On Oct 22, 2009, at 10:42 AM, freeswitch noob wrote: I have a noobish question about setting up FS. I have it installed and running. I setup a soft client on the machine fs is on and point it to the ip address of the FS instance and it registers with no issues. I then setup an entry in my etc/hosts files mydomain.localhost and changed the domain in the soft client. The registration now fails. with the soft phone giving a 503 error service unavailable And I am watching the cli and I see no errors come through with the domain but I see it register with the IP :( I change it back to the ip address and it works. Any direction you guys could point me in would be great. ___ 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] Qustion about INFO messages after Connect/Answer
I can't get what exactly you re talking about. Can you clarify ? Also
please include the packets of interest only not the full trace if its
not relevant to the bug.
/b
On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Mike,
here it is:
Dialplan:
extension name=Local_Extension
condition field=destination_number expression=^(10[01][0-9])
$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=set data=transfer_ringback=$$
{hold_music}/
action application=set data=hangup_after_bridge=true/
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
/condition
/extension
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Re: [Freeswitch-users] Address Rupa: Database for Audio Data
Hi Rupa, Thanks again for your advice. I have been searching for the method to record in the freeswitch documentation but I'm still not sure which command and method to make the record for every call automatically. Which command or method do you use? And to make the recording start and stop automatically every time when the calls is started and end, where should I insert this command? Did you use the Mod commands 'uuid_record'? If so, where to place this commands? Please show me some clues? Thank you very much! PB Date: Tue, 13 Oct 2009 09:57:05 -0600 From: r...@rupa.comajongjit Buntaokit To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Database for Audio Data What I do is record all calls and store the call with the UUID as the filename. Then when the call is hung up a CDR entry is sent to my web server. This CDR contains callerid and other info I might want to query by. The service on the web server inserts appropriate record(s) into the database. The recordings are available to the webserver. When one clicks on the listen link, the web server serves up the recording by UUID in the recording directory. I have a process that periodically removes old recordings from that dir. I don't purge the CDRs, though that is certainly possible. On Tue, Oct 13, 2009 at 7:58 AM, Pajongjit Buntaokit pippyduck1...@hotmail.com wrote: Hi, Does anyone know whether FreeSWITCH has a function to automatically record every call as an audio file in a server or forward them to be stored in a database with additional parameters such as caller ID, date, starting time and ending time? So that these recorded audio data can be queried and retrieved with the caller ID, date and time. Any suggestion or guidance, please advise. Thank you very much! Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ 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 -- -Rupa ___ 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 _ Windows 7: It works the way you want. Learn more. http://www.microsoft.com/Windows/windows-7/default.aspx?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_evergreen2:102009___ 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] Qustion about INFO messages after Connect/Answer
I'm sure that problem is gone in svn trunk.
On Thu, Oct 22, 2009 at 11:25 AM, Brian West br...@freeswitch.org wrote:
I can't get what exactly you re talking about. Can you clarify ? Also
please include the packets of interest only not the full trace if its
not relevant to the bug.
/b
On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Mike,
here it is:
Dialplan:
extension name=Local_Extension
condition field=destination_number expression=^(10[01][0-9])
$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=set data=transfer_ringback=$$
{hold_music}/
action application=set data=hangup_after_bridge=true/
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
/condition
/extension
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Re: [Freeswitch-users] UUID of the newly originated call?
please update to svn trunk with make current and try again. On Thu, Oct 22, 2009 at 4:08 AM, Nagalenoj nagale...@gmail.com wrote: I'm using 'FreeSWITCH Version 1.0.trunk (15106)'. Anthony Minessale-2 wrote: which revision of FS are you using? On Wed, Oct 21, 2009 at 1:06 AM, Nagalenoj H. nagale...@gmail.com wrote: I've tried with origination_uuid. First, I tried with SIP and my program executes successfully as what I expected. This program initiates a new call when a call comes and let the new call to eavesdrop the landed call. When, I experimented with PRI(openzap), I'm facing the following error. And, I am unable to make any calls(even from CLI). It is reporting the same error for the subsequent calls. Error:2009-10-21 11:08:31.661252 [ERR] mod_openzap.c:1760 Session for channel 1:1 not found [UUID: 8364eb30-be03-11de-aa07-07e76b7651a9] Full log is in http://pastebin.freeswitch.org/10780 My script is here, http://pastebin.freeswitch.org/10781 What is this error for and how to avoid this? Is there any other way to get the uuid of the originated call except explicitly defining(origination_uuid).?! -- Regards, Nagalenoj H. ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.commsn%253aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com paypal%3aanthony.miness...@gmail.compaypal%253aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org sip%3a...@conference.freeswitch.orgsip%253a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org googletalk%3aconf%2b...@conference.freeswitch.orggoogletalk%253aconf%252b...@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 -- View this message in context: http://www.nabble.com/Re%3A-UUID-of-the-newly-originated-call--tp25987024p26006565.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire 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] Heartbeat question
What is heartbeat and what are the uses cases?
Sorry i didn't find much information on wiki.
Thanks.
On Sat, Oct 10, 2009 at 12:01 AM, Diego Viola diego.vi...@gmail.com wrote:
Here's my heartbeat script now.
#!/usr/bin/env ruby
require 'rubygems'
require 'fsr'
require fsr/listener/inbound
def custom_channel_heartbeat_handler(event)
puts Got a SESSION_HEARTBEAT at #{Time.now.strftime('%H:%M:%S')}
end
FSL::Inbound.add_event_hook(:SESSION_HEARTBEAT) {|event|
custom_channel_heartbeat_handler(event) }
FSR.start_ies!(FSL::Inbound, :host = localhost, :port = 8021)
Thanks again.
Diego
On Fri, Oct 9, 2009 at 9:30 PM, Diego Viola diego.vi...@gmail.com wrote:
Here is on two seconds ;)
Got a SESSION_HEARTBEAT at 17:17:13
Got a SESSION_HEARTBEAT at 17:17:15
Got a SESSION_HEARTBEAT at 17:17:17
Got a SESSION_HEARTBEAT at 17:17:19
Got a SESSION_HEARTBEAT at 17:17:21
Got a SESSION_HEARTBEAT at 17:17:23
Got a SESSION_HEARTBEAT at 17:17:25
Got a SESSION_HEARTBEAT at 17:17:27
Got a SESSION_HEARTBEAT at 17:17:29
Got a SESSION_HEARTBEAT at 17:17:31
Got a SESSION_HEARTBEAT at 17:17:33
Got a SESSION_HEARTBEAT at 17:17:35
Got a SESSION_HEARTBEAT at 17:17:37
Got a SESSION_HEARTBEAT at 17:17:39
Got a SESSION_HEARTBEAT at 17:17:41
Got a SESSION_HEARTBEAT at 17:17:43
Got a SESSION_HEARTBEAT at 17:17:45
Got a SESSION_HEARTBEAT at 17:17:47
Got a SESSION_HEARTBEAT at 17:17:49
Got a SESSION_HEARTBEAT at 17:17:51
Got a SESSION_HEARTBEAT at 17:17:53
Got a SESSION_HEARTBEAT at 17:17:55
Got a SESSION_HEARTBEAT at 17:17:57
Got a SESSION_HEARTBEAT at 17:17:59
Got a SESSION_HEARTBEAT at 17:18:01
Got a SESSION_HEARTBEAT at 17:18:03
Got a SESSION_HEARTBEAT at 17:18:05
Got a SESSION_HEARTBEAT at 17:18:07
Got a SESSION_HEARTBEAT at 17:18:09
Got a SESSION_HEARTBEAT at 17:18:11
Got a SESSION_HEARTBEAT at 17:18:13
Got a SESSION_HEARTBEAT at 17:18:15
Got a SESSION_HEARTBEAT at 17:18:17
On Fri, Oct 9, 2009 at 9:27 PM, Diego Viola diego.vi...@gmail.comwrote:
Thanks Anthony, this solved it. You rock :)
My program now outputs:
Got a SESSION_HEARTBEAT at 17:14:59
Got a SESSION_HEARTBEAT at 17:15:00
Got a SESSION_HEARTBEAT at 17:15:02
Got a SESSION_HEARTBEAT at 17:15:03
Got a SESSION_HEARTBEAT at 17:15:04
Got a SESSION_HEARTBEAT at 17:15:05
Got a SESSION_HEARTBEAT at 17:15:06
Got a SESSION_HEARTBEAT at 17:15:07
Got a SESSION_HEARTBEAT at 17:15:08
Got a SESSION_HEARTBEAT at 17:15:09
Got a SESSION_HEARTBEAT at 17:15:10
Got a SESSION_HEARTBEAT at 17:15:11
Got a SESSION_HEARTBEAT at 17:15:12
Got a SESSION_HEARTBEAT at 17:15:13
Got a SESSION_HEARTBEAT at 17:15:14
Got a SESSION_HEARTBEAT at 17:15:15
Got a SESSION_HEARTBEAT at 17:15:16
Got a SESSION_HEARTBEAT at 17:15:17
Got a SESSION_HEARTBEAT at 17:15:18
Got a SESSION_HEARTBEAT at 17:15:19
Got a SESSION_HEARTBEAT at 17:15:20
Got a SESSION_HEARTBEAT at 17:15:21
Got a SESSION_HEARTBEAT at 17:15:22
Got a SESSION_HEARTBEAT at 17:15:23
Got a SESSION_HEARTBEAT at 17:15:24
Got a SESSION_HEARTBEAT at 17:15:25
Got a SESSION_HEARTBEAT at 17:15:26
Got a SESSION_HEARTBEAT at 17:15:27
Got a SESSION_HEARTBEAT at 17:15:28
Got a SESSION_HEARTBEAT at 17:15:29
Got a SESSION_HEARTBEAT at 17:15:30
On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
Update to trunk and try it with fs_cli it for sure will let you do every
1 second
in fs_cli type
/events plain all
if you make that call you will see one every 1 second
On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola diego.vi...@gmail.comwrote:
Nope, I was just wondering why it didn't work at 1 second exactly...
On Fri, Oct 9, 2009 at 3:36 AM, William Suffill
william.suff...@gmail.com wrote:
Why do you need it every second? If you want real time channel counts
you would be able to track each create/destroy even instead of
relying on the heartbeat summary.
-- W
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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker r...@rupa.com wrote:
cond would be helpful here? I updated the wiki on this one just now
with a bit more detail. It is a api call. so, you'd use it like:
${cond(eval ? trueval : falseval)}
so to get a value of ERR if the var my myvar is 15 you could:
${cond(${myvar} 15 ? ERR : OK)}
If both sides of the comparison operator are numeric then it does
numeric comparison otherwise it does lexical string comparison.
Rupa,
Yes, you can do the set/cond API trick but you can only do it in the action
or anti-action tags, not in the condition tags. I'm sure you know that but I
want all those reading this thread to make the connection.
-MC
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Re: [Freeswitch-users] Proxy media mode with T.38 re-invite
On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga tculj...@gmail.com wrote: and what these few issues are? :P One fax machine here in the office (still testing others) correctly sends all fax pages. A minute or so after the fax is marked successful on both sides it hangs up, redials, and resends the last page... It never did it while connected to the PSTN but then again my other fax machine isn't doing it either. I'm going to test with more fax machines to see if it's an issue with that specific machine. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ 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] Proxy media mode with T.38 re-invite
Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had nothing but intermittent problems with Super G3 FAXes over T.38, unless v.34 is strictly turned off on the machine. Gabe Kristian Kielhofner wrote: On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga tculj...@gmail.com wrote: One fax machine here in the office (still testing others) correctly sends all fax pages. A minute or so after the fax is marked successful on both sides it hangs up, redials, and resends the last page... It never did it while connected to the PSTN but then again my other fax machine isn't doing it either. I'm going to test with more fax machines to see if it's an issue with that specific machine. ___ 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] Heartbeat question
On Thu, Oct 22, 2009 at 10:47 AM, Saeed Ahmad saeedahmad1...@gmail.comwrote: What is heartbeat and what are the uses cases? Sorry i didn't find much information on wiki. Thanks. A session heartbeat is just an event that is sent to your script and gives you updated information about the call in progress. It is *VERY* different from the system heartbeat, which is sent out to all event socket subscribers, not just a particular session. -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] Connecting to FS CLI...just hangs..
It just hangsand I CTRL-C out of it. [r...@ss]# ./fs_cli -H 127.0.0.1 ^C [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error] Freeswitch process is running: [r...@ss bin]# ps -ef|grep free root 8889 31039 0 12:36 pts/200:00:00 ./freeswitch root 8952 31039 0 12:42 pts/200:00:00 grep free ___ 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] Proxy media mode with T.38 re-invite
indeed, this looks like a dialect problem between your fax machine and your T.38 device. Anyhow, T.38 doesn't work well with SG3... I Always have to disable v.34 in order to have a reliable fax service. Also, cisco uses to suppress CM so that SG3 timeouts on ANSam the communication fallbacks to ordinary G3. Kristian, just for fun, what are you using to send the fax ? T. On Thu, Oct 22, 2009 at 8:16 PM, Gabriel Kuri gk...@ieee.org wrote: Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had nothing but intermittent problems with Super G3 FAXes over T.38, unless v.34 is strictly turned off on the machine. Gabe Kristian Kielhofner wrote: On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga tculj...@gmail.com wrote: One fax machine here in the office (still testing others) correctly sends all fax pages. A minute or so after the fax is marked successful on both sides it hangs up, redials, and resends the last page... It never did it while connected to the PSTN but then again my other fax machine isn't doing it either. I'm going to test with more fax machines to see if it's an issue with that specific machine. ___ 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] Connecting to FS CLI...just hangs..
Hangs for how long? Are you sure you are not just waiting on a timeout? JM On Thu, Oct 22, 2009 at 4:42 PM, Ujjval Karihaloo ujj...@simplesignal.comwrote: It just hangs….and I CTRL-C out of it. [r...@ss]# ./fs_cli -H 127.0.0.1 ^C [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error] Freeswitch process is running: [r...@ss bin]# ps -ef|grep free root 8889 31039 0 12:36 pts/200:00:00 ./freeswitch root 8952 31039 0 12:42 pts/200:00:00 grep free ___ 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] Proxy media mode with T.38 re-invite
Gabe, I don't think any of them are plus the T.38 SDP tells me the bitrate is 14400, certainly not V.34 speed. Are you saying the machine even trying to negotiate V.34 poses a problem? On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri gk...@ieee.org wrote: Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had nothing but intermittent problems with Super G3 FAXes over T.38, unless v.34 is strictly turned off on the machine. Gabe -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ 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] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
finally i fix this rtp bug, check new wersion please.
TC
TC TC
TC TCDo you need some logs ?
TC
TC try disable medai-proxy, there is issue with rtp now then medai-proxy or
TC transcoding enabled.
TC
TC
TCOutbound calls:
TC
TCdisabled rtp proxy and it is still the same issue ... audio delay H323 =
TCSIP endpoint.
TC
TC
TC
TC
TC
TC
TCInbound calls:
TC
TCThis is the extension i use to register my Avaya SIP phone to FS.
TC
TC
TCinclude
TC user id=1001
TCparams
TC param name=password value=$${default_password}/
TC param name=vm-password value=1001/
TC/params
TCvariables
TC variable name=toll_allow value=domestic,international,local/
TC variable name=accountcode value=1001/
TC variable name=user_context value=default/
TC variable name=effective_caller_id_name value=Extension 1001/
TC variable name=effective_caller_id_number value=1001/
TC variable name=outbound_caller_id_name
TCvalue=$${outbound_caller_name}/
TC variable name=outbound_caller_id_number
TCvalue=$${outbound_caller_id}/
TC variable name=callgroup value=techsupport/
TC/variables
TC /user
TC/include
TC
TC
TCThis is my h323.conf.xml
TC
TC
TCconfiguration name=h323.conf description=H323 Endpoints
TC settings
TCparam name=trace-level value=4/
TCparam name=context value=default/
TCparam name=dialplan value=XML/
TCparam name=codec-prefs value=PCMU,PCMA,GSM,G729,G726/
TCparam name=gk-address value=/!-- empty to disable, * to
TCsearch LAN --
TCparam name=gk-identifer value=/ !-- optional name of gk --
TCparam name=gk-interface value=/ !-- optional listener interface
TCname --
TC /settings
TC listeners
TClistener name=default
TC param name=h323-ip value=10.4.62.7/
TC param name=h323-port value=1720/
TC/listener
TC /listeners
TC/configuration
TC
TCI'm using default context and an inbound call looks for a registered user in
TCdefault context where 1001 user is registered to.
TC
TC
TC
TChere is the log for an outgoing call:
TChttp://pastebin.freeswitch.org/10799and here is a tshark output:
TChttp://pastebin.freeswitch.org/10800
TC
TC
TCthere are 2 thing that are not working here:
TC
TC
TC1. no audio at all!
TC2. hangup from SIP User side doesn't release the H323 leg
TC
TC
TC
TC
TC
TC
TC
TC
TC
TC
TC
TCtwo points for your reference in the logs:
TC
TC
TC1. Here, SIP User disconnected the SIP leg, but nothing was triggered in
TCmod_h323 ... as the callback function on_hangup (in mod_h323.cpp) was
TCnever triggered!
TC
TCfreeswi...@subzero
TCfreeswi...@subzero
TCfreeswi...@subzero recv 371 bytes from udp/[10.4.62.89]:5060 at
TC14:39:36.714521:
TC
TC BYE sip:mod_so...@10.4.62.7:5060 SIP/2.0
TC From: sip:1...@10.4.62.89 sip%3a1...@10.4.62.89
TC;transport=udp;tag=-22166b474ae08abf-7_T10.4.62.89
TC To: sip:1001282...@10.4.62.7 sip%3a1001282...@10.4.62.7
TC;tag=Qpc53NZ2cZc1N
TC Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
TC CSeq: 127 BYE
TC Via: SIP/2.0/UDP
TC10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
TC Content-Length: 0
TC Max-Forwards: 70
TC Supported: replaces
TC
TC
TC2009-10-22 16:39:36.714604 [NOTICE] sofia.c:322 Hangup sofia/internal/
TCsip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [CS_CONSUME_MEDIA]
TC[NORMAL_CLEARING]
TC2009-10-22 16:39:36.714604 [DEBUG] switch_channel.c:1683 Send signal
TCsofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [KILL]
TC2009-10-22 16:39:36.714604 [DEBUG] switch_core_session.c:932 Send signal
TCsofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [BREAK]
TCsend 520 bytes to udp/[10.4.62.89]:5060 at 14:39:36.715258:
TC
TC SIP/2.0 200 OK
TC Via: SIP/2.0/UDP
TC10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
TC From: sip:1...@10.4.62.89 sip%3a1...@10.4.62.89
TC;transport=udp;tag=-22166b474ae08abf-7_T10.4.62.89
TC To: sip:1001282...@10.4.62.7 sip%3a1001282...@10.4.62.7
TC;tag=Qpc53NZ2cZc1N
TC Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
TC CSeq: 127 BYE
TC User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
TC Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
TCNOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
TC Supported: timer, precondition, path, replaces
TC Content-Length: 0
TC
TC
TC2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:503
TC(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) State
TCCONSUME_MEDIA going to sleep
TC2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:398
TC(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) Running State
TCChange CS_HANGUP
TC2009-10-22 16:39:36.721097 [DEBUG]
Re: [Freeswitch-users] Address Rupa: Database for Audio Data
I use the dialplan app session_record: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session I call that in the appropriate parts before bridging to call. For incoming it is just before the bridge to user/username. For outgoing it is before I bridge to the outgoing provider. On Thu, Oct 22, 2009 at 11:30 AM, Pajongjit Buntaokit pippyduck1...@hotmail.com wrote: Hi Rupa, Thanks again for your advice. I have been searching for the method to record in the freeswitch documentation but I'm still not sure which command and method to make the record for every call automatically. Which command or method do you use? And to make the recording start and stop automatically every time when the calls is started and end, where should I insert this command? Did you use the Mod commands 'uuid_record'? If so, where to place this commands? Please show me some clues? Thank you very much! PB Date: Tue, 13 Oct 2009 09:57:05 -0600 From: r...@rupa.comajongjit Buntaokit To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Database for Audio Data What I do is record all calls and store the call with the UUID as the filename. Then when the call is hung up a CDR entry is sent to my web server. This CDR contains callerid and other info I might want to query by. The service on the web server inserts appropriate record(s) into the database. The recordings are available to the webserver. When one clicks on the listen link, the web server serves up the recording by UUID in the recording directory. I have a process that periodically removes old recordings from that dir. I don't purge the CDRs, though that is certainly possible. On Tue, Oct 13, 2009 at 7:58 AM, Pajongjit Buntaokit pippyduck1...@hotmail.com wrote: Hi, Does anyone know whether FreeSWITCH has a function to automatically record every call as an audio file in a server or forward them to be stored in a database with additional parameters such as caller ID, date, starting time and ending time? So that these recorded audio data can be queried and retrieved with the caller ID, date and time. Any suggestion or guidance, please advise. Thank you very much! Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ 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 -- -Rupa ___ 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 Windows 7: It works the way you want. Learn more. ___ 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 -- -Rupa ___ 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] Proxy media mode with T.38 re-invite
AFAIK T.38 v2 supports a max speed of 14.4 anyhow, so that's the max speed you'll ever see in the SDP. T.38 v3 supposedly supports v.34 speeds, however no one that I've seen has implemented it yet - not sure it's even an official standard? Yes, in my experience, v.34 capable FAXes do not properly negotiate down to 14.4 speeds, we've always had to disable v.34 (or whatever the option is called on your particular make/model to force it down to 14.4 or slower speeds). We use the Cisco/Linksys ATAs - I've heard there's better ATAs out there that properly negotiate the FAX down to 14.4 on the T.30 side of the connection, but never had a chance to play with them. I'd certainly be interested to hear anyone's experience with T.38 reliability and various combinations of FAX machines and make/model of ATAs. Gabe Kristian Kielhofner wrote: Gabe, I don't think any of them are plus the T.38 SDP tells me the bitrate is 14400, certainly not V.34 speed. Are you saying the machine even trying to negotiate V.34 poses a problem? On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri gk...@ieee.org wrote: Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had nothing but intermittent problems with Super G3 FAXes over T.38, unless v.34 is strictly turned off on the machine. Gabe ___ 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] 3 way conferencing question
Hello gents,
I know that it is possible to make a 3-way conferencing using att_xfer and
by pressing 0.
But I'm more interested in the way of doing 3-way conferencing using mod
conference and bind_app. Saying in other words, I want to do the same thing
what 3-way conferencing using att_xfer but with mod conference (eventually
it will be more then 3 people there)
extension name=mystuff_enum
condition field=destination_number expression=^(.*)$
action application=answer/
action application=bind_meta_app data=7 ab s
execute_extension::conference XML features/
action application=set data=hangup_after_bridge=true/
action application=bridge
data=sofia/external/${destination_numb...@$${dialoutserver}/
/condition
/extension
in features.xml :
extension name=conference
condition field=destination_number expression=^conference$
action application=set data=continue_on_fail=true/
action application=read data=3 10
/usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-welcome.wav
thirdperson 3 #/
!-- I guess some logic here which will put original caller, person
whom we dialed and thirdperson(numbers which we pressed) into a conference
room with unique id --
/condition
/extension
I will be glad to hear any ideas.
--
View this message in context:
http://www.nabble.com/3-way-conferencing-question-tp26015903p26015903.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: finally i fix this rtp bug, check new wersion please. if course i can do that, but tomorrow morning ... i'm not in the office anymore. BTW: can we please move the tickets to jira? it is gonna be easier to track. Tomorrow i will test on 1.0.4 but please lets move to trunk. T. ___ 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] Core Dump question!
freeswi...@ss_freeswitch sofia_gateway_data Segmentation fault (core dumped) Just ran the gateway command above w/o any parameters,,, and it core dumped.. I am sure mistakes like that happen...but I not sure if it should core dump and shutdown. ___ 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] Core Dump question!
What SVN rev of FS? What operating system? If you're not on the latest then do a make current and get to the latest SVN and see if you can replicate the issue. -MC On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo ujj...@simplesignal.comwrote: freeswi...@ss_freeswitch sofia_gateway_data Segmentation fault (core dumped) Just ran the gateway command above w/o any parameters,,, and it core dumped.. I am sure mistakes like that happen…but I not sure if it should core dump and shutdown….. ___ 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] Connecting to FS CLI...just hangs..
If I run .freeswitch , get back to the ROOT prompt and then from same window type in fs_cli...it fails...hangs forever However, if I open another new ssh session and fs_cli from the new session, it works. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of João Mesquita Sent: Thursday, October 22, 2009 1:11 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Connecting to FS CLI...just hangs.. Hangs for how long? Are you sure you are not just waiting on a timeout? JM On Thu, Oct 22, 2009 at 4:42 PM, Ujjval Karihaloo ujj...@simplesignal.commailto:ujj...@simplesignal.com wrote: It just hangsand I CTRL-C out of it. [r...@ss]# ./fs_cli -H 127.0.0.1 ^C [ERROR] libs/esl/fs_cli.c:652 main() Error Connecting [Connection Error] Freeswitch process is running: [r...@ss bin]# ps -ef|grep free root 8889 31039 0 12:36 pts/200:00:00 ./freeswitch root 8952 31039 0 12:42 pts/200:00:00 grep free ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto: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] FS Registration Contact
Hi All, I have FS registered to an ITSP. The contact is showing as follows.. Contact: sip:gw+i...@1.1.1.1:5080;transport=udp Itsp is the name of the SIP gatway and its IP is changed to 1.1.1.1 I want the Phone number (FromUser)to show in the contact header in the REGISTER msg going to the ITSP. How can I do that? Thx, Ujjval. ___ 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] Fwd: mod_opal - call charged before H.225 connect
On 2009-10-22 21:44 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TC2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru TC TC On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote TC freeswitch-us...@lists.fre...: TC TC finally i fix this rtp bug, check new wersion please. TC TC TCif course i can do that, but tomorrow morning ... i'm not in the office TCanymore. TCBTW: can we please move the tickets to jira? TC TC TCit is gonna be easier to track. TC TCTomorrow i will test on 1.0.4 but please lets move to trunk. i make it a bit later, to move tickets to jira and source to svn i need some time to undertand how this system is works, especially jira. C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE___ 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] Connecting to FS CLI...just hangs..
If it's not windows probably be safer to just do freeswitch -nc for no console. Give it a little to start up then fs_cli should be fine. netstat -anp can also be used to see that the ports have binded correctly. -- W ___ 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] FS Registration Contact
Ok I got this one...just put ext-in-contact setting and then define the extension to be same as FromUser in my provider.xml in /conf/sip-profile/external/ !--/// extension for inbound calls: *optional* same as username, if blank ///-- param name=extension value=xx/ !--extra sip params to send in the contact-- param name=extension-in-contact value=true/ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval Karihaloo Sent: Thursday, October 22, 2009 2:00 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] FS Registration Contact Hi All, I have FS registered to an ITSP. The contact is showing as follows.. Contact: sip:gw+i...@1.1.1.1:5080;transport=udp Itsp is the name of the SIP gatway and its IP is changed to 1.1.1.1 I want the Phone number (FromUser)to show in the contact header in the REGISTER msg going to the ITSP. How can I do that? Thx, Ujjval. ___ 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] Core Dump question!
I did test this on trunk and it seems to work right: freeswi...@default sofia_gateway_data -ERR Parameter missing Mike On Oct 22, 2009, at 3:58 PM, Michael Collins wrote: What SVN rev of FS? What operating system? If you're not on the latest then do a make current and get to the latest SVN and see if you can replicate the issue. -MC On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo ujj...@simplesignal.com wrote: freeswi...@ss_freeswitch sofia_gateway_data Segmentation fault (core dumped) Just ran the gateway command above w/o any parameters,,, and it core dumped.. I am sure mistakes like that happen…but I not sure if it should core dump and shutdown….. ___ 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] Proxy media mode with T.38 re-invite
We have probably 30-40 fax machines running T.38 between Linksys SPA-3102 and Cisco gateways. We have a pretty good success rate and we have been doing this for probably 2-3 years. We have a couple of them going thru an * 1.4 box and it seems to work ok. One of my projects is to try and get them all converted to go thru an FS box, I just haven't had time. Our biggest issues have always been error correction. Once we disable that on the fax machine, it seems to work pretty good. As a general rule, we have found that older cheaper machines work better. New fancy machines are unreliable and that could certainly be related to v.34 issues, but we haven't bothered to get that into it to find out. Peder -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Gabriel Kuri Sent: Thursday, October 22, 2009 2:29 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Proxy media mode with T.38 re-invite AFAIK T.38 v2 supports a max speed of 14.4 anyhow, so that's the max speed you'll ever see in the SDP. T.38 v3 supposedly supports v.34 speeds, however no one that I've seen has implemented it yet - not sure it's even an official standard? Yes, in my experience, v.34 capable FAXes do not properly negotiate down to 14.4 speeds, we've always had to disable v.34 (or whatever the option is called on your particular make/model to force it down to 14.4 or slower speeds). We use the Cisco/Linksys ATAs - I've heard there's better ATAs out there that properly negotiate the FAX down to 14.4 on the T.30 side of the connection, but never had a chance to play with them. I'd certainly be interested to hear anyone's experience with T.38 reliability and various combinations of FAX machines and make/model of ATAs. Gabe Kristian Kielhofner wrote: Gabe, I don't think any of them are plus the T.38 SDP tells me the bitrate is 14400, certainly not V.34 speed. Are you saying the machine even trying to negotiate V.34 poses a problem? On Thu, Oct 22, 2009 at 2:16 PM, Gabriel Kuri gk...@ieee.org wrote: Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had nothing but intermittent problems with Super G3 FAXes over T.38, unless v.34 is strictly turned off on the machine. Gabe ___ 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] Core Dump question!
I do have the core dump, should I open a ticket. I am running latest Freeswitch 1.0.4 and had done a make current just before it happened. Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [cid:image001.jpg@01CA5342.1F8A8880]http://www.simplesignal.com/ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Thursday, October 22, 2009 2:53 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Core Dump question! I did test this on trunk and it seems to work right: freeswi...@default sofia_gateway_data -ERR Parameter missing Mike On Oct 22, 2009, at 3:58 PM, Michael Collins wrote: What SVN rev of FS? What operating system? If you're not on the latest then do a make current and get to the latest SVN and see if you can replicate the issue. -MC On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo ujj...@simplesignal.commailto:ujj...@simplesignal.com wrote: freeswi...@ss_freeswitch sofia_gateway_data Segmentation fault (core dumped) Just ran the gateway command above w/o any parameters,,, and it core dumped.. I am sure mistakes like that happen...but I not sure if it should core dump and shutdown. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto: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 inline: image001.jpg___ 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] Core Dump question!
Yes, if this is latest SVN (after a make current) then open a jira. -MC On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo ujj...@simplesignal.comwrote: I do have the core dump, should I open a ticket. I am running latest Freeswitch 1.0.4 and had done a make current just before it happened. ___ 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] Core Dump question!
How do I tell if it's the latest...I downloaded is yesterday..and installed it from freeswitch.org Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [cid:image001.jpg@01CA5359.87CB41C0]http://www.simplesignal.com/ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, October 22, 2009 6:26 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Core Dump question! Yes, if this is latest SVN (after a make current) then open a jira. -MC On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo ujj...@simplesignal.commailto:ujj...@simplesignal.com wrote: I do have the core dump, should I open a ticket. I am running latest Freeswitch 1.0.4 and had done a make current just before it happened. inline: image001.jpg___ 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] Setting up Conference with Moderator
Hi
I have the Basic Conferencing working. Here is my Dial Plan.
I want to be able to setup a Moderator PIN different from other participants,
when I add the moderator flag it logs me in directly w/o asking for a PIN..
action application=conference
data=conference.c...@wideband+flags{moderator}+159753mailto:conference.c...@wideband+flags%7bmoderator%7d+159753/
DialPlan is below for the normal user, and it asks for the PIN with below
settings.
Ujj Inbound from SS - start
--
extension name=simplesignal !-- your provider or any name you'd like
to call it --
condition field=destination_number expression=2142349127 !-- your
DID for this gateway--
action application=conference
data=conference.c...@wideband+159753mailto:conference.c...@wideband+159753/
/condition
/extension
!--
Ujj Inbound from SS - end
And I am using the existing conference.conf.xml file in the auto_loads
directory.
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88 Inverness Circle East
Suite K105
Englewood, CO 80112
[cid:image001.jpg@01CA535A.E43E9870]http://www.simplesignal.com/
inline: image001.jpg___
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Re: [Freeswitch-users] Core Dump question!
Type version on the CLI. On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo ujj...@simplesignal.comwrote: How do I tell if it’s the latest…I downloaded is yesterday..and installed it from freeswitch.org Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [image: bvoip] http://www.simplesignal.com/ *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Thursday, October 22, 2009 6:26 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Core Dump question! Yes, if this is latest SVN (after a make current) then open a jira. -MC On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo ujj...@simplesignal.com wrote: I do have the core dump, should I open a ticket. I am running latest Freeswitch 1.0.4 and had done a make current just before it happened. ___ 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 image001.jpg___ 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] Core Dump question!
I think the (exported) means you don't have the latest svn, but probably the officially released build 1.0.4 that can be downloaded from the FS page. I think you should see something like (the latest trunk is 15203): freeswi...@internal version FreeSWITCH Version 1.0.trunk (15126) I guess you need to checkout the latest FS trunk On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo ujj...@simplesignal.comwrote: freeswi...@internal version FreeSWITCH Version 1.0.4 (exported) freeswi...@internal Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [image: bvoip] http://www.simplesignal.com/ *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Diego Viola *Sent:* Thursday, October 22, 2009 9:04 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Core Dump question! Type version on the CLI. On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo ujj...@simplesignal.com wrote: How do I tell if it’s the latest…I downloaded is yesterday..and installed it from freeswitch.org Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [image: bvoip] http://www.simplesignal.com/ *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Thursday, October 22, 2009 6:26 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Core Dump question! Yes, if this is latest SVN (after a make current) then open a jira. -MC On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo ujj...@simplesignal.com wrote: I do have the core dump, should I open a ticket. I am running latest Freeswitch 1.0.4 and had done a make current just before it happened. ___ 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 image001.jpg___ 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] Core Dump question!
He's running 1.0.4, he needs to checkout from SVN trunk and build FS from scratch. Diego On Fri, Oct 23, 2009 at 4:00 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: I think the (exported) means you don't have the latest svn, but probably the officially released build 1.0.4 that can be downloaded from the FS page. I think you should see something like (the latest trunk is 15203): freeswi...@internal version FreeSWITCH Version 1.0.trunk (15126) I guess you need to checkout the latest FS trunk On Fri, Oct 23, 2009 at 2:25 PM, Ujjval Karihaloo ujj...@simplesignal.com wrote: freeswi...@internal version FreeSWITCH Version 1.0.4 (exported) freeswi...@internal Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [image: bvoip] http://www.simplesignal.com/ *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Diego Viola *Sent:* Thursday, October 22, 2009 9:04 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Core Dump question! Type version on the CLI. On Fri, Oct 23, 2009 at 2:52 AM, Ujjval Karihaloo ujj...@simplesignal.com wrote: How do I tell if it’s the latest…I downloaded is yesterday..and installed it from freeswitch.org Ujjval Karihaloo VP Voice Engineering IP Phone: +13032428610 E-Fax: +17202391690 SimpleSignal Inc. 88 Inverness Circle East Suite K105 Englewood, CO 80112 [image: bvoip] http://www.simplesignal.com/ *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Thursday, October 22, 2009 6:26 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Core Dump question! Yes, if this is latest SVN (after a make current) then open a jira. -MC On Thu, Oct 22, 2009 at 5:04 PM, Ujjval Karihaloo ujj...@simplesignal.com wrote: I do have the core dump, should I open a ticket. I am running latest Freeswitch 1.0.4 and had done a make current just before it happened. ___ 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 image001.jpg___ 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
