January 2008 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Regex Mismatch
by Aadilkhan Maniyar 11 Jan '08

11 Jan '08

Hi, I have installed FreeSwitch on a OpenSuse box. As per the instructions given in the "Getting started" document, I am trying to make a call between tow Xlite softphones registered with the numbers 1000 and 1001 to FreeSwitch. The registration happens successfully but when I try to make a call between these two there is an error (FS responds with a 404). According to the logs there is a Regex Mismatch in one of the files. I am unable to understand where the problem is. I have attached the log with this email. I have not modified any of the files in the conf directory. Would be grateful if anyone could help me out. Thanks & Regards, Aadil

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Re: [Freeswitch-dev] Windows build problem?
by adrian.p.smith＠bt.com 11 Jan '08

11 Jan '08

Mike, I think I've found out the problem. Seems that IE had somehow cached a dodgy version of the file. I removed it from the cache directory and all is now well. Regards, Adrian -----Original Message----- From: Smith,AP,Adrian,DLN R Sent: 10 January 2008 10:42 To: 'freeswitch-dev(a)lists.freeswitch.org' Subject: RE: [Freeswitch-dev] Windows build problem? Hi Mike, Windows XP SP2 (professional). Interestingly, I tried it on a co-workers PC with the same result and also a PC running Windows Server 2003. On my home laptop running XP home, it seems to work OK. Will investigate further and let you know. Regards, Adrian -----Original Message----- From: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 09 January 2008 17:59 To: freeswitch-dev(a)lists.freeswitch.org Subject: Re: [Freeswitch-dev] Windows build problem? I have never seen this error before, what version of windows is this on? Mike On Jan 9, 2008, at 7:27 AM, <adrian.p.smith(a)bt.com> <adrian.p.smith(a)bt.com > wrote: > When building in Visual Studio 2005 I encountered the problem that the > downloads of pthreads and howl didn't succeed. They got as far as > extracting a file called "[Content]" and then stopping. > > I added an extra section at the end of util.vbs/UnCompress like this > to > resolve it: > > > If FSO.FileExists(DestFolder & "[Content]")Then > Set MyFile = fso.CreateTextFile(UtilsDir & batname, > True) > MyFile.WriteLine("@" & quote & UtilsDir & "7za.exe" & > quote & " x " & quote & DestFolder & "[Content]" & quote & " -y -o" & > quote & DestFolder & quote ) > MyFile.Close > Set oExec = WshShell.Exec(UtilsDir & batname) > Do > WScript.Echo OExec.StdOut.ReadLine() > Loop While Not OExec.StdOut.atEndOfStream > WScript.Sleep(500) > FSO.DeleteFile DestFolder & "[Content]",true > End If > > Make sense? > > Regards, > > Adrian Smith > BT Group > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org

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[Freeswitch-dev] Query on no_media variable
by sanjeev mudholkar 11 Jan '08

11 Jan '08

Hi, I have been trying to use the variable "no_media" to enable the media flow directly between the registered endpoints. Irrespective of the value of "no_media" variable the media always flows through the freeswitch. Here is my setup Two SIP end points(1001 and 1002) registered to Freeswitch. Dial plan: <extension name="test"> <condition field="destination_number" expression="^(1002)$"> <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/$${domain}/1002%freeswitch_ip"/> </condition> </extension> My requirement is to set the media path(direct or through freeswitch) based on the destination and not for all SIP calls. (Note: I had a quick look at audio_bridge_function in mod_dptools.c. I do not see the no_media variable used. Please ignore this note if this is not the right function) Thanks in advance -Sanjeev

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[Freeswitch-dev] Call Forward @ Freeswitch
by Irshad Ahamed 10 Jan '08

10 Jan '08

Do we have support for Call Forward (Unconditional/Busy/No Answer) @ Freeswitch? If so, how should I enable it? BR, Irshad.

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[Freeswitch-dev] [Freeswitch-users] mod_java - prompt clipping issue
by jonathan augenstine 10 Jan '08

10 Jan '08

I have a mod_java application that plays a prompt and then based on a database access response, sometimes plays a second prompt. What I actually do is play a prompt and then call getDigits(). What I notice happening is that the first prompt plays fine, but the second prompt has the first second clipped. You miss the first few words of the prompt. Any ideas on what might explain this behavior? Jonathan Augenstine jaugenstine(a)gmail.com

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[Freeswitch-dev] Windows build problem?
by adrian.p.smith＠bt.com 10 Jan '08

10 Jan '08

When building in Visual Studio 2005 I encountered the problem that the downloads of pthreads and howl didn't succeed. They got as far as extracting a file called "[Content]" and then stopping. I added an extra section at the end of util.vbs/UnCompress like this to resolve it: If FSO.FileExists(DestFolder & "[Content]")Then Set MyFile = fso.CreateTextFile(UtilsDir & batname, True) MyFile.WriteLine("@" & quote & UtilsDir & "7za.exe" & quote & " x " & quote & DestFolder & "[Content]" & quote & " -y -o" & quote & DestFolder & quote ) MyFile.Close Set oExec = WshShell.Exec(UtilsDir & batname) Do WScript.Echo OExec.StdOut.ReadLine() Loop While Not OExec.StdOut.atEndOfStream WScript.Sleep(500) FSO.DeleteFile DestFolder & "[Content]",true End If Make sense? Regards, Adrian Smith BT Group

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Re: [Freeswitch-dev] [Freeswitch-users] Not able to make an outgoing Call through FXO....
by Irshad Ahamed 09 Jan '08

09 Jan '08

With the same set-up, Outgoing calls are working fine with Asterisk. Are there any tools to detect the presence of dial tone??? If it's any parameter issue, can we adjust those parameters??? NOTE: I copied tones.conf available from the source code to /etc/openzap/ BR, Irshad. On Jan 5, 2008 12:00 AM, Anthony Minessale <anthmct(a)yahoo.com> wrote: > I think it's waiting for a dialtone before it sends the digits > and then giving up because it can't fine one. > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Irshad Ahamed <irshad.fswitch(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org; > freeswitch-dev(a)lists.freeswitch.org > Sent: Friday, January 4, 2008 4:24:07 AM > Subject: Re: [Freeswitch-users] Not able to make an outgoing Call through > FXO.... > > Hi Anthony, > > Thanks for the info. By adding the span-ID = 2 , I was able to receive an > incoming call successfully. > The dialplan/default.xml is as below: > <extension name="incoming"> > <condition field="destination_number" expression="^(1)$"> > <action application="bridge" data="openzap/1/1"/> >  > </condition> > </extension> > > <extension name="outgoing-fxo"> > <condition field="destination_number" expression="^(43)$"> > <action application="bridge" data="openzap/2/1/43"/> > </condition> > </extension> > > The Openzap.conf.xml looks like below: > > <configuration name="openzap.conf" description="OpenZAP Configuration"> > <settings> > <param name="debug" value="9"/> > </settings> > <analog_spans> > > <param name="tonegroup" value="us"/> > <param name="digit-timeout" value="2000"/> > <param name="max-digits" value="11"/> > <param name="dialplan" value="XML"/> > <param name="context" value="default"/> >  >  >  >  > > > <param name="tonegroup" value="us"/> > <param name="digit-timeout" value="2000"/> > <param name="max-digits" value="1"/> > <param name="dialplan" value="XML"/> > <param name="context" value="default"/> >  >  >  >  > > </analog_spans> > </configuration> > > But still I am unable to make an Outgoing Call. Please have a look at the > Freeswitch Log below: > > 2008-01-04 15:45:03 [DEBUG] zap_analog.c:664 process_event() EVENT > [OFFHOOK][1:1] STATE [DOWN] > 2008-01-04 15:45:03 [DEBUG] zap_analog.c:726 process_event() Changing > state on 1:1 from DOWN to DIALTONE > 2008-01-04 15:45:03 [DEBUG] zap_analog.c:184 zap_analog_channel_run() > ANALOG CHANNEL thread starting. > 2008-01-04 15:45:03 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > Executing state handler on 1:1 for DIALTONE > 2008-01-04 15:45:07 [DEBUG] zap_analog.c:490 zap_analog_channel_run() DTMF > 4 > 2008-01-04 15:45:07 [DEBUG] zap_analog.c:492 zap_analog_channel_run() > Changing state on 1:1 from DIALTONE to COLLECT > 2008-01-04 15:45:07 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS sig > [COLLECTED_DIGIT] > 2008-01-04 15:45:07 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > Executing state handler on 1:1 for COLLECT > 2008-01-04 15:45:07 [DEBUG] zap_analog.c:490 zap_analog_channel_run() DTMF > 2 > 2008-01-04 15:45:07 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS sig > [COLLECTED_DIGIT] > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:507 zap_analog_channel_run() > Number obtained [42] > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:508 zap_analog_channel_run() > Changing state on 1:1 from COLLECT to IDLE > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > Executing state handler on 1:1 for IDLE > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS sig > [START] > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:231 tech_init() Set codec PCMU > 20ms > 2008-01-04 15:45:09 [NOTICE] switch_channel.c:392 > switch_channel_set_name() New Chan OpenZAP/42 > [8d46eb7f-4305-49e8-a4a5-eb97af3b696b] > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:892 zap_channel_from_event() > OpenZAP/42 State Change CS_NEW -> CS_INIT > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > switch_core_session_signal_state_change() Kill OpenZAP/42 [BREAK] > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (OpenZAP/42) State INIT > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:256 channel_on_init() OpenZAP/42 > State Change CS_INIT -> CS_RING > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > switch_core_session_signal_state_change() Kill OpenZAP/42 [BREAK] > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:400 > switch_core_session_run() (OpenZAP/42) State RING > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:276 channel_on_ring() OpenZAP/42 > CHANNEL RING > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:64 > switch_core_standard_on_ring() Standard RING OpenZAP/42 > 2008-01-04 15:45:09 [INFO] mod_dialplan_xml.c:217 dialplan_hunt() > Processing OpenZAP->42! > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:106 parse_exten() test > conditions ${numbering_plan}() =~ // > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:106 parse_exten() test > conditions destination_number(42) =~ /^(1)$/ > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:108 parse_exten() Regex > mismatch > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:106 parse_exten() test > conditions destination_number(42) =~ /^(43)$/ > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:108 parse_exten() Regex > mismatch > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:106 parse_exten() test > conditions destination_number(42) =~ /^(42)$/ > 2008-01-04 15:45:09 [DEBUG] mod_dialplan_xml.c:273 dialplan_hunt() > OpenZAP/42 State Change CS_RING -> CS_EXECUTE > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > switch_core_session_signal_state_change() Kill OpenZAP/42 [BREAK] > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:406 > switch_core_session_run() (OpenZAP/42) State EXECUTE > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:294 channel_on_execute() > OpenZAP/42 CHANNEL EXECUTE > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:121 > switch_core_standard_on_execute() Standard EXECUTE > 2008-01-04 15:45:09 [NOTICE] switch_core_state_machine.c:134 > switch_core_standard_on_execute() Execute > set(use_profile=${cond(${is_lan_addr(${network_addr})} == yes ? nat : > default)}) > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:158 > switch_core_standard_on_execute() Expanded String set(use_profile=default) > 2008-01-04 15:45:09 [DEBUG] mod_dptools.c:419 set_function() SET > [use_profile]=[default] > 2008-01-04 15:45:09 [NOTICE] switch_core_state_machine.c:134 > switch_core_standard_on_execute() Execute set_user(default@${domain}) > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:158 > switch_core_standard_on_execute() Expanded String set_user( > default(a)192.168.2.51) > 2008-01-04 15:45:09 [NOTICE] switch_core_state_machine.c:134 > switch_core_standard_on_execute() Execute info() > 2008-01-04 15:45:09 [CRIT] mod_dptools.c:630 info_function() CHANNEL_DATA: > > Channel-State: CS_EXECUTE > Channel-State-Number: 4 > Channel-Name: OpenZAP/42 > Unique-ID: 8d46eb7f-4305-49e8-a4a5-eb97af3b696b > Channel-Read-Codec-Name: PCMU > Channel-Read-Codec-Rate: 8000 > Channel-Write-Codec-Name: PCMU > Channel-Write-Codec-Rate: 8000 > Caller-Username: OpenZAP > Caller-Dialplan: XML > Caller-Caller-ID-Name: OpenZAP > Caller-Caller-ID-Number: 1 > Caller-Destination-Number: 42 > Caller-Unique-ID: 8d46eb7f-4305-49e8-a4a5-eb97af3b696b > Caller-Source: mod_openzap > Caller-Context: default > Caller-Channel-Name: OpenZAP/42 > Caller-Channel-Created-Time: 1199441709953187 > Caller-Channel-Answered-Time: 0 > Caller-Channel-Hangup-Time: 0 > Caller-Channel-Transfer-Time: 0 > Caller-Screen-Bit: yes > Caller-Privacy-Hide-Name: no > Caller-Privacy-Hide-Number: no > variable_use_profile: default > variable_numbering_plan: US > variable_user_name: default > variable_domain_name: 192.168.2.51 > > > 2008-01-04 15:45:09 [NOTICE] switch_core_state_machine.c:134 > switch_core_standard_on_execute() Execute answer() > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:663 > channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:339 > switch_core_session_receive_message() Kill OpenZAP/42 [BREAK] > 2008-01-04 15:45:09 [NOTICE] mod_dptools.c:361 answer_function() Channel > [OpenZAP/42] has been answered > 2008-01-04 15:45:09 [NOTICE] switch_core_state_machine.c:134 > switch_core_standard_on_execute() Execute bridge(openzap/2/1/42)2008-01-04 > 15:45:09 [DEBUG] mod_openzap.c:231 tech_init() Set codec PCMU 20ms > 2008-01-04 15:45:09 [NOTICE] switch_channel.c:392 > switch_channel_set_name() New Chan OPENZAP/42 > [bef803ed-e932-4a22-84e7-9131070fe771] > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:802 channel_outgoing_channel() > OPENZAP/42 State Change CS_NEW -> CS_INIT > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > switch_core_session_signal_state_change() Kill OPENZAP/42 [BREAK] > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > Executing state handler on 1:1 for UP > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS sig > [UP] > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:55 analog_fxo_outgoing_call() > Changing state on 2:1 from DOWN to DIALING > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:184 zap_analog_channel_run() > ANALOG CHANNEL thread starting. > 2008-01-04 15:45:09 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > Executing state handler on 2:1 for DIALING > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (OPENZAP/42) State INIT > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:256 channel_on_init() OPENZAP/42 > State Change CS_INIT -> CS_RING > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > switch_core_session_signal_state_change() Kill OPENZAP/42 [BREAK] > 2008-01-04 15:45:09 [DEBUG] switch_ivr_originate.c:625 > switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel > 20ms > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:400 > switch_core_session_run() (OPENZAP/42) State RING > 2008-01-04 15:45:09 [DEBUG] mod_openzap.c:276 channel_on_ring() OPENZAP/42 > CHANNEL RING > 2008-01-04 15:45:09 [DEBUG] switch_ivr_originate.c:48 originate_on_ring() > OPENZAP/42 State Change CS_RING -> CS_HOLD > 2008-01-04 15:45:09 [DEBUG] switch_core_session.c:625 > switch_core_session_signal_state_change() Kill OPENZAP/42 [BREAK] > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:415 > switch_core_session_run() (OPENZAP/42) State HOLD > 2008-01-04 15:45:09 [DEBUG] switch_core_state_machine.c:204 > switch_core_standard_on_hold() Standard HOLD > > > freeswitch(a)sipxpbx.oneconv.com> > > freeswitch(a)sipxpbx.oneconv.com> > > freeswitch(a)sipxpbx.oneconv.com> 2008-01-04 15:45:39 [DEBUG] > zap_analog.c:244 zap_analog_channel_run() Changing state on 2:1 from DIALING > to BUSY > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > Executing state handler on 2:1 for BUSY > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:467 zap_analog_channel_run() > Changing state on 2:1 from BUSY to DOWN > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > Executing state handler on 2:1 for DOWN > 2008-01-04 15:45:40 [DEBUG] mod_openzap.c:916 on_fxo_signal() got FXO sig > 2:1 [STOP] > 2008-01-04 15:45:40 [NOTICE] mod_openzap.c:934 on_fxo_signal() Hangup > OPENZAP/42 [CS_HOLD] [NORMAL_CIRCUIT_CONGESTION] > 2008-01-04 15:45:40 [DEBUG] switch_channel.c:1071 > switch_channel_perform_hangup() Kill OPENZAP/42 [KILL] > 2008-01-04 15:45:40 [DEBUG] switch_core_session.c:625 > switch_core_session_signal_state_change() Kill OPENZAP/42 [BREAK] > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:648 zap_analog_channel_run() > ANALOG CHANNEL thread ended. > 2008-01-04 15:45:40 [DEBUG] switch_core_state_machine.c:394 > switch_core_session_run() (OPENZAP/42) State HANGUP > 2008-01-04 15:45:40 [WARNING] mod_openzap.c:319 channel_on_hangup() VETO > Changing state on 2:1 from DOWN to HANGUP > 2008-01-04 15:45:40 [DEBUG] mod_openzap.c:359 channel_on_hangup() > OPENZAP/42 CHANNEL HANGUP > 2008-01-04 15:45:40 [DEBUG] switch_core_state_machine.c:45 > switch_core_standard_on_hangup() Standard HANGUP OPENZAP/42, cause: > NORMAL_CIRCUIT_CONGESTION > 2008-01-04 15:45:40 [DEBUG] switch_core_session.c:683 > switch_core_session_thread() Session 4 (OPENZAP/42) Locked, Waiting on > external entities > 2008-01-04 15:45:40 [DEBUG] switch_ivr_originate.c:873 > switch_ivr_originate() Originate Resulted in Error Cause: 34 > [NORMAL_CIRCUIT_CONGESTION] > 2008-01-04 15:45:40 [INFO] mod_dptools.c:1331 audio_bridge_function() > Originate Failed. Cause: NORMAL_CIRCUIT_CONGESTION > 2008-01-04 15:45:40 [NOTICE] mod_dptools.c:1358 audio_bridge_function() > Hangup OpenZAP/42 [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION] > 2008-01-04 15:45:40 [DEBUG] switch_channel.c:1071 > switch_channel_perform_hangup() Kill OpenZAP/42 [KILL] > 2008-01-04 15:45:40 [DEBUG] switch_core_session.c:625 > switch_core_session_signal_state_change() Kill OpenZAP/42 [BREAK] > 2008-01-04 15:45:40 [DEBUG] switch_core_state_machine.c:394 > switch_core_session_run() (OpenZAP/42) State HANGUP > 2008-01-04 15:45:40 [DEBUG] mod_openzap.c:329 channel_on_hangup() Changing > state on 1:1 from UP to HANGUP > 2008-01-04 15:45:40 [DEBUG] mod_openzap.c:359 channel_on_hangup() > OpenZAP/42 CHANNEL HANGUP > 2008-01-04 15:45:40 [DEBUG] switch_core_state_machine.c:45 > switch_core_standard_on_hangup() Standard HANGUP OpenZAP/42, cause: > NORMAL_CIRCUIT_CONGESTION > 2008-01-04 15:45:40 [DEBUG] switch_core_session.c:683 > switch_core_session_thread() Session 3 (OpenZAP/42) Locked, Waiting on > external entities > 2008-01-04 15:45:40 [INFO] switch_core_session.c:689 > switch_core_session_thread() Session 3 (OpenZAP/42) Ended > 2008-01-04 15:45:40 [NOTICE] switch_core_session.c:691 > switch_core_session_thread() Close Channel OpenZAP/42 [CS_HANGUP] > 2008-01-04 15:45:40 [INFO] switch_core_session.c:689 > switch_core_session_thread() Session 4 (OPENZAP/42) Ended > 2008-01-04 15:45:40 [NOTICE] switch_core_session.c:691 > switch_core_session_thread() Close Channel OPENZAP/42 [CS_HANGUP] > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > Executing state handler on 1:1 for HANGUP > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:290 zap_analog_channel_run() > Changing state on 1:1 from HANGUP to BUSY > 2008-01-04 15:45:40 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > Executing state handler on 1:1 for BUSY > 2008-01-04 15:45:53 [DEBUG] zap_analog.c:664 process_event() EVENT > [ONHOOK][1:1] STATE [BUSY] > 2008-01-04 15:45:53 [DEBUG] zap_analog.c:690 process_event() Changing > state on 1:1 from BUSY to DOWN > 2008-01-04 15:45:53 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > Executing state handler on 1:1 for DOWN > 2008-01-04 15:45:53 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS sig > [STOP] > 2008-01-04 15:45:53 [DEBUG] zap_analog.c:648 zap_analog_channel_run() > ANALOG CHANNEL thread ended. > > Thanks, > Irshad. > > > > > > On Jan 3, 2008 7:43 PM, Anthony Minessale < anthmct(a)yahoo.com> wrote: > > > What does openzap.conf.xml look like? > > Perhaps you are using the default which only has 1 span defined. > > Try cloning the whole block so there > > is a span 1 and 2 like this: > > > > <configuration name=" openzap.conf" description="OpenZAP Configuration"> > > <settings> > > <param name="debug" value="0"/> > > </settings> > > <analog_spans> > > > > <param name="tonegroup" value="us"/> > > <param name="digit-timeout" value="2000"/> > > <param name="max-digits" value="11"/> > > <param name="dialplan" value="XML"/> > > <param name="context" value="default"/> > >  > >  > >  > >  > > > > > > > > <param name="tonegroup" value="us"/> > > <param name="digit-timeout" value="2000"/> > > <param name="max-digits" value="11"/> > > <param name="dialplan" value="XML"/> > > <param name="context" value="default"/> > >  > >  > >  > >  > > > > > > > > </analog_spans> > > </configuration> > > > > > > > > > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > iax:guest@conference.freeswitch.org/888 > > googletalk:conf+888@conference.freeswitch.org > > pstn:213-799-1400 > > > > > > ----- Original Message ---- > > From: Irshad Ahamed <irshad.fswitch(a)gmail.com > > > To: freeswitch-dev(a)lists.freeswitch.org; > > freeswitch-users(a)lists.freeswitch.org > > Sent: Wednesday, January 2, 2008 12:40:14 AM > > Subject: [Freeswitch-users] Not able to make an outgoing Call through > > FXO.... > > > > Hi, > > > > Wishing U all a very Happy n Prosperous New Year!!!! > > > > I am trying to make an outgoing call using FXO connected to an external > > telephone line. > > > > Here is the set-up: > > > > 1. FXO connected to external telephone line > > 2. FXS connected to analog phone. > > > > > > The Openzap.conf @ /etc/openzap is as follows: > > [span zt] > > name => OpenZAP > > number => 1 > > fxs-channel => 1 > > > > [span zt] > > name => OpenZAP > > number => 1 > > fxo-channel => 4 > > > > The default.xml @ /usr/local/freeswitch/conf/dialplan is configured for > > inward and outgoing calls on FXO as follows: > > > > <extension name="outgoing-fxo"> > > <condition field="destination_number" expression="^(43)$"> > > <action application="bridge" data="openzap/2/1/43"/> > > </condition> > > </extension> > > > > <extension name="incoming-fxo"> > > <condition field="destination_number" expression="^(35)$"> > > <action application="bridge" data="openzap/1/1"/> > > </condition> > > </extension> > > > > While Start-up Freeswitch following logs were seen: > > > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1770 load_config() found config for > > span > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1797 load_config() created span 1 > > of type zt > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 1 > > [name]=[OpenZAP] > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 1 > > [number]=[1] > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 1 > > [fxs-channel]=[1] > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1850 load_config() setting trunk > > type to 'FXS' start(KEWL) > > 2008-01-02 11:51:22 [INFO] zap_zt.c:168 zt_open_range() configuring > > device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:31 > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1770 load_config() found config for > > span > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1797 load_config() created span 2 > > of type zt > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 2 > > [name]=[OpenZAP] > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 2 > > [number]=[1] > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 2 > > [fxo-channel]=[4] > > 2008-01-02 11:51:22 [DEBUG] zap_io.c:1839 load_config() setting trunk > > type to 'FXO' start(KEWL) > > 2008-01-02 11:51:22 [INFO] zap_zt.c:168 zt_open_range() configuring > > device /dev/zap/channel channel 4 as OpenZAP device 2:1 fd:32 > > 2008-01-02 11:51:22 [INFO] zap_io.c:1881 load_config() Configured 2 > > channel(s) > > 2008-01-02 11:51:22 [CONSOLE] switch_loadable_module.c:748 > > switch_loadable_module_load_file() Successfully Loaded [mod_openzap] > > > > and while I made a call following logs were seen: > > > > 2008-01-02 12:04:16 [NOTICE] switch_core_state_machine.c:134 > > switch_core_standard_on_execute() Execute bridge(openzap/2/1/43)2008-01-02 > > 12:04:16 [DEBUG] mod_openzap.c:231 tech_init() Set codec PCMU 20ms > > 2008-01-02 12:04:16 [NOTICE] switch_channel.c:392 > > switch_channel_set_name() New Chan OPENZAP/43 > > [e3630387-25d6-4031-bf81-239c3e64e3fa] > > 2008-01-02 12:04:16 [DEBUG] mod_openzap.c:802 channel_outgoing_channel() > > OPENZAP/43 State Change CS_NEW -> CS_INIT > > 2008-01-02 12:04:16 [DEBUG] switch_core_session.c:625 > > switch_core_session_signal_state_change() Kill OPENZAP/43 [BREAK] > > 2008-01-02 12:04:16 [ERR] zap_io.c:884 zap_channel_outgoing_call() > > outgoing_call method not implemented! > > 2008-01-02 12:04:16 [NOTICE] mod_openzap.c:818 > > channel_outgoing_channel() Close Channel OPENZAP/43 [CS_INIT] > > 2008-01-02 12:04:16 [ERR] switch_ivr_originate.c:525 > > switch_ivr_originate() Cannot Create Outgoing Channel! cause: > > DESTINATION_OUT_OF_ORDER > > 2008-01-02 12:04:16 [DEBUG] switch_ivr_originate.c:873 > > switch_ivr_originate() Originate Resulted in Error Cause: 27 > > [DESTINATION_OUT_OF_ORDER] > > 2008-01-02 12:04:16 [INFO] mod_dptools.c:1331 audio_bridge_function() > > Originate Failed. Cause: DESTINATION_OUT_OF_ORDER > > 2008-01-02 12:04:16 [NOTICE] mod_dptools.c:1358 audio_bridge_function() > > Hangup OpenZAP/43 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] > > 2008-01-02 12:04:16 [DEBUG] switch_channel.c:1071 > > switch_channel_perform_hangup() Kill OpenZAP/43 [KILL] > > 2008-01-02 12:04:16 [DEBUG] switch_core_session.c:625 > > switch_core_session_signal_state_change() Kill OpenZAP/43 [BREAK] > > 2008-01-02 12:04:16 [DEBUG] switch_core_state_machine.c:394 > > switch_core_session_run() (OpenZAP/43) State HANGUP > > 2008-01-02 12:04:16 [DEBUG] mod_openzap.c:329 channel_on_hangup() > > Changing state on 1:1 from IDLE to HANGUP > > 2008-01-02 12:04:16 [DEBUG] mod_openzap.c:359 channel_on_hangup() > > OpenZAP/43 CHANNEL HANGUP > > 2008-01-02 12:04:16 [DEBUG] switch_core_state_machine.c:45 > > switch_core_standard_on_hangup() Standard HANGUP OpenZAP/43, cause: > > DESTINATION_OUT_OF_ORDER > > 2008-01-02 12:04:16 [DEBUG] switch_core_session.c:683 > > switch_core_session_thread() Session 1 (OpenZAP/43) Locked, Waiting on > > external entities > > 2008-01-02 12:04:16 [INFO] switch_core_session.c:689 > > switch_core_session_thread() Session 1 (OpenZAP/43) Ended > > 2008-01-02 12:04:16 [NOTICE] switch_core_session.c:691 > > switch_core_session_thread() Close Channel OpenZAP/43 [CS_HANGUP] > > 2008-01-02 12:04:16 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 1:1 for HANGUP > > 2008-01-02 12:04:17 [DEBUG] zap_analog.c:290 zap_analog_channel_run() > > Changing state on 1:1 from HANGUP to BUSY > > 2008-01-02 12:04:17 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 1:1 for BUSY > > 2008-01-02 12:04:20 [DEBUG] zap_analog.c:664 process_event() EVENT > > [ONHOOK][1:1] STATE [BUSY] > > 2008-01-02 12:04:20 [DEBUG] zap_analog.c:690 process_event() Changing > > state on 1:1 from BUSY to DOWN > > 2008-01-02 12:04:20 [DEBUG] zap_analog.c:351 zap_analog_channel_run() > > Executing state handler on 1:1 for DOWN > > 2008-01-02 12:04:20 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS > > sig [STOP] > > 2008-01-02 12:04:20 [DEBUG] zap_analog.c:648 zap_analog_channel_run() > > ANALOG CHANNEL thread ended. > > > > Could anyone help me out in getting FXO working ??? > > > > Thanks in advance... > > > > BR, > > Irshad. > > > > > > > > ------------------------------ > > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try > > it now.<http://us.rd.yahoo.com/evt=51733/*http://mobile.yahoo.com/;_ylt=Ahu06i62sR8…> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > ------------------------------ > Looking for last minute shopping deals? Find them fast with Yahoo! Search.<http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/c…> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Re: [Freeswitch-dev] [Freeswitch-users] Issue with Call Hold @ Freeswitch FXS....
by Irshad Ahamed 09 Jan '08

09 Jan '08

I tested with the latest svn revision but the same issue exists still. The SVN revision which I tested upon is ... Path: . URL: http://svn.freeswitch.org/svn/freeswitch/trunk Repository Root: http://svn.freeswitch.org/svn Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2 Revision: 7148 Node Kind: directory Schedule: normal Last Changed Author: anthm Last Changed Rev: 7148 Last Changed Date: 2008-01-09 04:00:52 +0530 (Wed, 09 Jan 2008) Properties Last Updated: 2008-01-08 11:14:19 +0530 (Tue, 08 Jan 2008) BR, Irshad. On Jan 9, 2008 5:05 AM, Anthony Minessale <anthmct(a)yahoo.com> wrote: > Please update the code to the latest revision where it > should be working right. > > In the future could you please direct issues to > http://jira.freeswitch.org so we can properly track > them. > > thanks =) > > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Irshad Ahamed <irshad.fswitch(a)gmail.com> > To: freeswitch-dev(a)lists.freeswitch.org; > freeswitch-users(a)lists.freeswitch.org > Sent: Tuesday, January 8, 2008 11:53:35 AM > Subject: Re: [Freeswitch-users] [Freeswitch-dev] Issue with Call Hold @ > Freeswitch FXS.... > > Your interpretation is right. This is what I have observed while testing > here. To be more precise, when you hangup the FXS completely, the first call > doesnt get disconnected. > > BR, > Irshad. > > On Jan 8, 2008 7:38 PM, Anthony Minessale <anthmct(a)yahoo.com> wrote: > > > Can you clarify more so I am sure I understand? > > > > You use fxs to call an extension that leads to a sip phone. > > You press FLASH on fxs to get a dialtone and make another call. > > You hangup the fxs completely (ONHOOK) and one of the 2 calls do not > > end? > > > > The current intended behavior is when you FLASH fxs you can make > > another call and switch back and forth with subsequent FLASH. But if > > the fxs is ONHOOK all the calls should end. > > > > We still need some more code to do other behaviors from fxs like > > transfer etc. > > > > > > > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > iax:guest@conference.freeswitch.org/888 > > googletalk:conf+888@conference.freeswitch.org > > pstn:213-799-1400 > > > > > > ----- Original Message ---- > > From: " freeswitch-dev(a)lists.freeswitch.org" < > > freeswitch-dev(a)lists.freeswitch.org> > > To: freeswitch-dev(a)lists.freeswitch.org; > > freeswitch-users(a)lists.freeswitch.org > > Sent: Monday, January 7, 2008 11:36:09 PM > > Subject: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS.... > > > > Set-up is as below: > > 1. Freeswitch is configured as UA with Asterisk TDM card (Openzap) 1 FXS > > + 1 FXO registered with SIPX. > > 2. SNOM300 registered with SIPX > > > > SNOM300: > > Kernel Version:snom300 linux 3.25 > > Application-Version:snom300-SIP 6.5.2 > > Rootfs-Version:snom300 jffs2 v3.36 > > Firmware-URL: > > Production > > Information:Mac:00041325329C;Version:Standard;Hardware:snom300 (MB > > V3.2_A11);Date:26/06/06;Copyright(C) snom technology AG > > 3. GRANDSTREAM registered with SIPX. > > > > GRANDSTREAM: > > Product Model: GXP2000 > > Software : Program-- 1.1.4.18 > > Version : Bootloader-- 1.1.4.6 > > > > > > Procedure followed: > > 1. Make a successful call from Freeswitch FXS to SNOM300. Put the call > > on hold @ Freeswitch FXS and make new call to GRANDSTREAM. > > 2. Accept the call @ GRANDSTREAM and disconnect the connected call @ > > FREESWITCH FXS. > > 3. The second call with GRANDSTREAM is being disconnected but the first > > call with SNOM300 is still alive at SNOM300. > > > > NOTE: Please find the Freeswitch logs and Ethereal caps @ > > http://jira.freeswitch.org/browse/MODENDP-74 > > > > BR, > > Irshad. > > > > > > ------------------------------ > > Looking for last minute shopping deals? Find them fast with Yahoo! > > Search.<http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/c…> > > > > _______________________________________________ > > Freeswitch-dev mailing list > > Freeswitch-dev(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > > ------------------------------ > Looking for last minute shopping deals? Find them fast with Yahoo! Search.<http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/c…> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Re: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS....
by Anthony Minessale 08 Jan '08

08 Jan '08

by the way thanks, I see you actually did file a bug so nevermind that req. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Irshad Ahamed <irshad.fswitch(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org; freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, January 8, 2008 11:53:35 AM Subject: Re: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS.... Your interpretation is right. This is what I have observed while testing here. To be more precise, when you hangup the FXS completely, the first call doesnt get disconnected. BR, Irshad. On Jan 8, 2008 7:38 PM, Anthony Minessale <anthmct(a)yahoo.com> wrote: Can you clarify more so I am sure I understand? You use fxs to call an extension that leads to a sip phone. You press FLASH on fxs to get a dialtone and make another call. You hangup the fxs completely (ONHOOK) and one of the 2 calls do not end? The current intended behavior is when you FLASH fxs you can make another call and switch back and forth with subsequent FLASH. But if the fxs is ONHOOK all the calls should end. We still need some more code to do other behaviors from fxs like transfer etc. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: " freeswitch-dev(a)lists.freeswitch.org" <freeswitch-dev(a)lists.freeswitch.org> To: freeswitch-dev(a)lists.freeswitch.org; freeswitch-users(a)lists.freeswitch.org Sent: Monday, January 7, 2008 11:36:09 PM Subject: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS.... Set-up is as below: 1. Freeswitch is configured as UA with Asterisk TDM card (Openzap) 1 FXS + 1 FXO registered with SIPX. 2. SNOM300 registered with SIPX SNOM300: Kernel Version:snom300 linux 3.25 Application-Version:snom300-SIP 6.5.2 Rootfs-Version:snom300 jffs2 v3.36 Firmware-URL: Production Information:Mac:00041325329C;Version:Standard;Hardware:snom300 (MB V3.2_A11);Date:26/06/06;Copyright(C) snom technology AG 3. GRANDSTREAM registered with SIPX. GRANDSTREAM: Product Model: GXP2000 Software : Program-- 1.1.4.18 Version : Bootloader-- 1.1.4.6 Procedure followed: 1. Make a successful call from Freeswitch FXS to SNOM300. Put the call on hold @ Freeswitch FXS and make new call to GRANDSTREAM. 2. Accept the call @ GRANDSTREAM and disconnect the connected call @ FREESWITCH FXS. 3. The second call with GRANDSTREAM is being disconnected but the first call with SNOM300 is still alive at SNOM300. NOTE: Please find the Freeswitch logs and Ethereal caps @ http://jira.freeswitch.org/browse/MODENDP-74 BR, Irshad. Looking for last minute shopping deals? Find them fast with Yahoo! Search. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Re: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS....
by Anthony Minessale 08 Jan '08

08 Jan '08

Can you clarify more so I am sure I understand? You use fxs to call an extension that leads to a sip phone. You press FLASH on fxs to get a dialtone and make another call. You hangup the fxs completely (ONHOOK) and one of the 2 calls do not end? The current intended behavior is when you FLASH fxs you can make another call and switch back and forth with subsequent FLASH. But if the fxs is ONHOOK all the calls should end. We still need some more code to do other behaviors from fxs like transfer etc. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: "freeswitch-dev(a)lists.freeswitch.org" <freeswitch-dev(a)lists.freeswitch.org> To: freeswitch-dev(a)lists.freeswitch.org; freeswitch-users(a)lists.freeswitch.org Sent: Monday, January 7, 2008 11:36:09 PM Subject: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS.... Set-up is as below: 1. Freeswitch is configured as UA with Asterisk TDM card (Openzap) 1 FXS + 1 FXO registered with SIPX. 2. SNOM300 registered with SIPX SNOM300: Kernel Version:snom300 linux 3.25 Application-Version:snom300-SIP 6.5.2 Rootfs-Version:snom300 jffs2 v3.36 Firmware-URL: Production Information:Mac:00041325329C;Version:Standard;Hardware:snom300 (MB V3.2_A11);Date:26/06/06;Copyright(C) snom technology AG 3. GRANDSTREAM registered with SIPX. GRANDSTREAM: Product Model: GXP2000 Software : Program-- 1.1.4.18 Version : Bootloader-- 1.1.4.6 Procedure followed: 1. Make a successful call from Freeswitch FXS to SNOM300. Put the call on hold @ Freeswitch FXS and make new call to GRANDSTREAM. 2. Accept the call @ GRANDSTREAM and disconnect the connected call @ FREESWITCH FXS. 3. The second call with GRANDSTREAM is being disconnected but the first call with SNOM300 is still alive at SNOM300. NOTE: Please find the Freeswitch logs and Ethereal caps @ http://jira.freeswitch.org/browse/MODENDP-74 BR, Irshad. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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