Re: [Freeswitch-users] Timers/DTMFs During a Call
Hello Michael! Thank you for your reply! I know about sched_api and I use Python as a scripting language. Also sched_api can be used inside python, using session.execute. However the problem is that the sched_api timer starts right after you initiate the SETUP of the second call leg. What I need is to call something, after a call CONNECT instead. One workaround would be if I can check what time it took to connect the call, but I do not know/see how to do this. I do not see a CONNECT callback function either. Best Regards, Delian Tashev P.S. Dear enlightened people, thank you for providing help to the community by replying to the list e-mails. From: Michael Collins [mailto:m...@freeswitch.org] Sent: Wednesday, August 26, 2009 11:28 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call Do you actually need Python for the IVR, or is it that you're comfortable using a scripting lang for an IVR? I like using XML for IVRs, but using scripting langs does give you a bit more power flexibility at the cost of some resources. For the record, you can do this in the dialplan using XML and sched_api without touching a scripting language. Checkout the sched_api channel variable on the wiki - it may give you the functionality you need. -MC On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM delians...@gmail.com wrote: Hello Everybody! 1. Scenario. I am writing an IVR in Python that gets a destination from the calling party (party A) and then connects to the destination (party B). When the call is CONNECTED, I want to: - Receive DTMFs - Have a timer that can call a certain function in my script. The script will have to play a message to party A. - Have a timer that can call a certain function in my script. The script will have to drop the call. Please notice that I want to do the things after the two parties are connected, and not after I send the Invite to party B. 2. Problem. I will be happy to receive help on: - Which methods should I look for to implement this. 3. Details Here is how I connect the call currently: session.execute(bridge,sofia/internal/ + destination_number + @domain.com) I have tried to create a timer callback function my_method() using: ivr_timer =threading.Timer(30,my_method) This never called the function my_method(). Maybe I am wrong in using threading.Timer and the bridge application? Maybe I need to create a new thread and a new timer using the API of freeswitch, plus to use the session.setInputCallback, plus use a conference rather than a bridge? Can you please provide any suggestions or examples? Thank you! Best Regards, Delian Tashev ___ 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] Timers/DTMFs During a Call
Looks like I was wrong about using the native Python timers. Here is how they can be used in your script: # Imports - add these new imports import time import threading # class definitions - add this new class class Timer(threading.Thread): def __init__(self, seconds): self.runTime = seconds threading.Thread.__init__(self) def run(self): time.sleep(self.runTime) console_log(debug, TIMER ) # entry point - add two rows in the entry point function that is called from freeswitch def handler(session, args): . t = Timer(10) t.start() . So what is now remaining is to get when the call CONNECTS and how to get DTMFs during the call. From: delianSPAM [mailto:delians...@gmail.com] Sent: Thursday, August 27, 2009 9:35 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call Hello Michael! Thank you for your reply! I know about sched_api and I use Python as a scripting language. Also sched_api can be used inside python, using session.execute. However the problem is that the sched_api timer starts right after you initiate the SETUP of the second call leg. What I need is to call something, after a call CONNECT instead. One workaround would be if I can check what time it took to connect the call, but I do not know/see how to do this. I do not see a CONNECT callback function either. Best Regards, Delian Tashev P.S. Dear enlightened people, thank you for providing help to the community by replying to the list e-mails. From: Michael Collins [mailto:m...@freeswitch.org] Sent: Wednesday, August 26, 2009 11:28 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call Do you actually need Python for the IVR, or is it that you're comfortable using a scripting lang for an IVR? I like using XML for IVRs, but using scripting langs does give you a bit more power flexibility at the cost of some resources. For the record, you can do this in the dialplan using XML and sched_api without touching a scripting language. Checkout the sched_api channel variable on the wiki - it may give you the functionality you need. -MC On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM delians...@gmail.com wrote: Hello Everybody! 1. Scenario. I am writing an IVR in Python that gets a destination from the calling party (party A) and then connects to the destination (party B). When the call is CONNECTED, I want to: - Receive DTMFs - Have a timer that can call a certain function in my script. The script will have to play a message to party A. - Have a timer that can call a certain function in my script. The script will have to drop the call. Please notice that I want to do the things after the two parties are connected, and not after I send the Invite to party B. 2. Problem. I will be happy to receive help on: - Which methods should I look for to implement this. 3. Details Here is how I connect the call currently: session.execute(bridge,sofia/internal/ + destination_number + @domain.com) I have tried to create a timer callback function my_method() using: ivr_timer =threading.Timer(30,my_method) This never called the function my_method(). Maybe I am wrong in using threading.Timer and the bridge application? Maybe I need to create a new thread and a new timer using the API of freeswitch, plus to use the session.setInputCallback, plus use a conference rather than a bridge? Can you please provide any suggestions or examples? Thank you! Best Regards, Delian Tashev ___ 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] Timers/DTMFs During a Call
I will try to execute and parse from python: freeswi...@internal show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,des t,application,application_data,dialplan,context,read_codec,read_rate,write_c odec,write_rate 53a62ebd-156c-4684-b616-740d7a5b609b,inbound,2009-04-23 11:14:09,1240510449,sofia/internal/1...@...,CS_EXECUTE,Mikey,1000,10.15.0.21 3,,playback,local_stream://moh,XML,default,PCMU,8000,PCMU,8000 Hoping that this will get the state of the call. If I call this check frequently I will catch the call connect I trust. From: delianSPAM [mailto:delians...@gmail.com] Sent: Thursday, August 27, 2009 9:53 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call Looks like I was wrong about using the native Python timers. Here is how they can be used in your script: # Imports - add these new imports import time import threading # class definitions - add this new class class Timer(threading.Thread): def __init__(self, seconds): self.runTime = seconds threading.Thread.__init__(self) def run(self): time.sleep(self.runTime) console_log(debug, TIMER ) # entry point - add two rows in the entry point function that is called from freeswitch def handler(session, args): . t = Timer(10) t.start() . So what is now remaining is to get when the call CONNECTS and how to get DTMFs during the call. From: delianSPAM [mailto:delians...@gmail.com] Sent: Thursday, August 27, 2009 9:35 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call Hello Michael! Thank you for your reply! I know about sched_api and I use Python as a scripting language. Also sched_api can be used inside python, using session.execute. However the problem is that the sched_api timer starts right after you initiate the SETUP of the second call leg. What I need is to call something, after a call CONNECT instead. One workaround would be if I can check what time it took to connect the call, but I do not know/see how to do this. I do not see a CONNECT callback function either. Best Regards, Delian Tashev P.S. Dear enlightened people, thank you for providing help to the community by replying to the list e-mails. From: Michael Collins [mailto:m...@freeswitch.org] Sent: Wednesday, August 26, 2009 11:28 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call Do you actually need Python for the IVR, or is it that you're comfortable using a scripting lang for an IVR? I like using XML for IVRs, but using scripting langs does give you a bit more power flexibility at the cost of some resources. For the record, you can do this in the dialplan using XML and sched_api without touching a scripting language. Checkout the sched_api channel variable on the wiki - it may give you the functionality you need. -MC On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM delians...@gmail.com wrote: Hello Everybody! 1. Scenario. I am writing an IVR in Python that gets a destination from the calling party (party A) and then connects to the destination (party B). When the call is CONNECTED, I want to: - Receive DTMFs - Have a timer that can call a certain function in my script. The script will have to play a message to party A. - Have a timer that can call a certain function in my script. The script will have to drop the call. Please notice that I want to do the things after the two parties are connected, and not after I send the Invite to party B. 2. Problem. I will be happy to receive help on: - Which methods should I look for to implement this. 3. Details Here is how I connect the call currently: session.execute(bridge,sofia/internal/ + destination_number + @domain.com) I have tried to create a timer callback function my_method() using: ivr_timer =threading.Timer(30,my_method) This never called the function my_method(). Maybe I am wrong in using threading.Timer and the bridge application? Maybe I need to create a new thread and a new timer using the API of freeswitch, plus to use the session.setInputCallback, plus use a conference rather than a bridge? Can you please provide any suggestions or examples? Thank you! Best Regards, Delian Tashev ___ 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
Re: [Freeswitch-users] Define Condition in FreeSwitch
On Wed, Aug 26, 2009 at 8:57 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
Hi,
I'm newbie. How can we translate the asterisk's condition in freeswitch
as listed below;
1. NoOp (Remote Conference Call)
2. GotoIf ($[${LEN(${DIALSTR})}=0]?3:4)
3. Hangup()
4. NoOp(Finish if-CONFERENCE-430)
Kindly reply soon.
Before I answer this question I just want you to know that there's probably
a more elegant way of doing whatever it is you're trying to do. This
dialplan snippet is pretty short. My first question would be: how does a
call get to this point? Also, what is the big picture, that is, what's the
application you're creating? Remember the golden rule: Anything that you do
in Asterisk is easier to do in FreeSWITCH, but you need to learn the ropes a
bit.
The answer to your question is, of course, It depends. :P Give us the
background on what you're doing so that we can give you an educated answer.
You could use the condition tags with actions and anti-actions. You could
also call a Lua/JavaScript/Perl/Python/etc. script to handle the logic but
that's probably overkill.
Tell us more and we'll tell you more. ;)
Thanks!
-Michael Collins
___
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] Timers/DTMFs During a Call
On Wed, Aug 26, 2009 at 11:35 PM, delianSPAM delians...@gmail.com wrote: Hello Michael! Thank you for your reply! I know about sched_api and I use Python as a scripting language. Also “sched_api” can be used inside python, using session.execute. However the problem is that the “sched_api” timer starts right after you initiate the SETUP of the second call leg. What I need is to call something, after a call CONNECT instead. One workaround would be if I can check what time it took to connect the call, but I do not know/see how to do this. I do not see a CONNECT callback function either. Perhaps you could set this channel variable to the sched_api call? http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Timers/DTMFs During a Call
To get the state use: session.getVariable(state) I hope that I will solve the Timers puzzle soon. I am still looking on getting the DTMFs during a bridged call. Best Regards, Delian Tashev From: delianSPAM [mailto:delians...@gmail.com] Sent: Thursday, August 27, 2009 10:01 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call I will try to execute and parse from python: freeswi...@internal show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,des t,application,application_data,dialplan,context,read_codec,read_rate,write_c odec,write_rate 53a62ebd-156c-4684-b616-740d7a5b609b,inbound,2009-04-23 11:14:09,1240510449,sofia/internal/1...@...,CS_EXECUTE,Mikey,1000,10.15.0.21 3,,playback,local_stream://moh,XML,default,PCMU,8000,PCMU,8000 Hoping that this will get the state of the call. If I call this check frequently I will catch the call connect I trust. From: delianSPAM [mailto:delians...@gmail.com] Sent: Thursday, August 27, 2009 9:53 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call Looks like I was wrong about using the native Python timers. Here is how they can be used in your script: # Imports - add these new imports import time import threading # class definitions - add this new class class Timer(threading.Thread): def __init__(self, seconds): self.runTime = seconds threading.Thread.__init__(self) def run(self): time.sleep(self.runTime) console_log(debug, TIMER ) # entry point - add two rows in the entry point function that is called from freeswitch def handler(session, args): . t = Timer(10) t.start() . So what is now remaining is to get when the call CONNECTS and how to get DTMFs during the call. From: delianSPAM [mailto:delians...@gmail.com] Sent: Thursday, August 27, 2009 9:35 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call Hello Michael! Thank you for your reply! I know about sched_api and I use Python as a scripting language. Also sched_api can be used inside python, using session.execute. However the problem is that the sched_api timer starts right after you initiate the SETUP of the second call leg. What I need is to call something, after a call CONNECT instead. One workaround would be if I can check what time it took to connect the call, but I do not know/see how to do this. I do not see a CONNECT callback function either. Best Regards, Delian Tashev P.S. Dear enlightened people, thank you for providing help to the community by replying to the list e-mails. From: Michael Collins [mailto:m...@freeswitch.org] Sent: Wednesday, August 26, 2009 11:28 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call Do you actually need Python for the IVR, or is it that you're comfortable using a scripting lang for an IVR? I like using XML for IVRs, but using scripting langs does give you a bit more power flexibility at the cost of some resources. For the record, you can do this in the dialplan using XML and sched_api without touching a scripting language. Checkout the sched_api channel variable on the wiki - it may give you the functionality you need. -MC On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM delians...@gmail.com wrote: Hello Everybody! 1. Scenario. I am writing an IVR in Python that gets a destination from the calling party (party A) and then connects to the destination (party B). When the call is CONNECTED, I want to: - Receive DTMFs - Have a timer that can call a certain function in my script. The script will have to play a message to party A. - Have a timer that can call a certain function in my script. The script will have to drop the call. Please notice that I want to do the things after the two parties are connected, and not after I send the Invite to party B. 2. Problem. I will be happy to receive help on: - Which methods should I look for to implement this. 3. Details Here is how I connect the call currently: session.execute(bridge,sofia/internal/ + destination_number + @domain.com) I have tried to create a timer callback function my_method() using: ivr_timer =threading.Timer(30,my_method) This never called the function my_method(). Maybe I am wrong in using threading.Timer and the bridge application? Maybe I need to create a new thread and a new timer using the API of freeswitch, plus to use the session.setInputCallback, plus use a conference rather than a bridge? Can you please provide any suggestions or examples? Thank you! Best Regards, Delian Tashev
[Freeswitch-users] need help with mod_xml_odbc
Hi, I tried to make install mod_xml_odbc and load it in freeswitch, but I am getting: 2009-08-28 00:46:55.848087 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_xml_odbc.so **/usr/local/freeswitch/mod/mod_xml_odbc.so: invalid ELF header** I had to manually copy the .so file from the mod_xml_odbc dir to /usr/local/freeswitch/mod Does anyone know what is wrong? Thanks, JB ___ 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] upstream Registrar / Mirror proxy
Hello, We would like to use FreeSwitch to test interoperability with our sip registrar platform - as our registrar doesnt handle NAT Traversal we would like to use FS as a mirror proxy . If a call goes through, FS should transparently handle nat traversal functionalities while forwarding the REGISTER/INVITE requests to our registrar server. Below some detailed requirements: Upper Registration is the capability of a SBC to proxy Registrations towards an upstream Registrar. While it is necessary that all SIP requests traverse the SBC for NAT continuity, Registrations should be allowed to be relayed towards our upstream Registrar while the SBC retains a copy of the AOR and masquerade on behalf of the UA that send the registration request. Please let us know the best way to configure FS to achieve this type of configuration. Thank you ___ 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] need help with mod_xml_odbc
You can not just copy the .so file from the directory it is just a pointer
file to the real .so ... You should always use the make install target or
make {module_name}-install target to get it properly installed
From: Juan Backson juanback...@gmail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Thu, 27 Aug 2009 16:56:09 +0800
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] need help with mod_xml_odbc
Hi,
I tried to make install mod_xml_odbc and load it in freeswitch, but I am
getting:
2009-08-28 00:46:55.848087 [CRIT] switch_loadable_module.c:871 Error Loading
module /usr/local/freeswitch/mod/mod_xml_odbc.so
**/usr/local/freeswitch/mod/mod_xml_odbc.so: invalid ELF header**
I had to manually copy the .so file from the mod_xml_odbc dir to
/usr/local/freeswitch/mod
Does anyone know what is wrong?
Thanks,
JB
___
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] upstream Registrar / Mirror proxy
A proxy won't do what the original poster is asking for. Upper registration is a special type of function performed by SBCs and not defined by any RFC yet but there are drafts out there. This question comes up quite often in various mailing lists and has been asked this list before. The answer is no freeswitch can't be configured to do this as it stands today. The only opensource SBC I know of that attempts to do upper registration at the moment is OpenSBC. Otherwise there's vendor SBCs like ACME packet, Nextone (now Genband - they call it Mirror Proxy mode) etc which do it. Regards, Steve On Thu, Aug 27, 2009 at 9:02 PM, Ken Rice kr...@freeswitch.org wrote: FreeSWITCH is a B2BUA and NOT a proxy and will not proxy any requests (REGISTER or INVITE) If you want a Proxy you should look toward OpenSIPS -- *From: *sylver_b sylve...@yahoo.com *Reply-To: *freeswitch-users@lists.freeswitch.org *Date: *Wed, 26 Aug 2009 17:14:55 -0700 (PDT) *To: *freeswitch-users@lists.freeswitch.org *Subject: *[Freeswitch-users] upstream Registrar / Mirror proxy Hello, We would like to use FreeSwitch to test interoperability with our sip registrar platform - as our registrar doesnt handle NAT Traversal we would like to use FS as a mirror proxy . If a call goes through, FS should transparently handle nat traversal functionalities while forwarding the REGISTER/INVITE requests to our registrar server. Below some detailed requirements: Upper Registration is the capability of a SBC to proxy Registrations towards an upstream Registrar. While it is necessary that all SIP requests traverse the SBC for NAT continuity, Registrations should be allowed to be relayed towards our upstream Registrar while the SBC retains a copy of the AOR and masquerade on behalf of the UA that send the registration request. Please let us know the best way to configure FS to achieve this type of configuration. Thank you -- ___ 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
[Freeswitch-users] Can an Instant Message (IM) be sent to a cell phone with FS?
Can an Instant Message (IM) be sent to a cell phone with FS? Any of these: AOL Instant Messenger (AIM) Yahoo (Y!) Blackberry Messenger Google Talk Windows Live Messenger Thanks, Merle ___ 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] Can a chat message be sent to a cell phone with FS?
Can a chat message be sent to a cell phone with FS? Thanks, Merle ___ 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] Get the Session State
Hello Everybody! How do you get the call state in Python? I have tried: . session.answer() state_result=str(session.getVariable(state)) console_log(debug,state_result) . But it returns: None Thank you! Best Regards, Delian Tashev ___ 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] need help with mod_xml_odbc
Hi, ok, the module can be loaded, but it now complains about odbc. I can't find anything missing in my odbc.ini. Could someone please point me to the right direction? 2009-08-28 02:22:35.670284 [ERR] switch_odbc.c:188 STATE: IM002 CODE 201 ERROR: [unixODBC]Missing server name, port, or database name in call to CC_connect. 2009-08-28 02:22:35.670312 [CRIT] mod_xml_odbc.c:577 Cannot Open ODBC Database! 2009-08-28 02:22:35.670319 [ERR] mod_xml_odbc.c:612 Unable to load xml_odbc config file 2009-08-28 02:22:35.670326 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_xml_odbc.so **Module load routine returned an error** I already have xml_odbc.conf cat /etc/odbc.ini ; begin odbc.ini [ODBC Data Sources] test = PostgreSQL ODBC Driver [test] Driver = /usr/local/lib/libodbcpsql.so Description = PostgreSQL Data Source DSN = test Servername = 192.168.1.133 Server = 192.168.1.133 Port = 5432 ;Socket = 4096 Protocol = 6.4 # 7.2 or other values UserName = root Password = JdqB-S Database = freeswitch ReadOnly = no ServerType = Postgres FetchBufferSize = 99 ServerOptions = ConnectOptions = ;Options = 3 Trace = 0 TraceFile = /var/log/PostgreSQL_test_trace.log Debug = 0 DebugFile = /var/log/PostgreSQL_test_debug.log ___ 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] Question about presence
sorry, but i do not know i which category i have to set this problem. could you help me with that? ___ 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] need help with mod_xml_odbc
Hi,
Finally, I got xml_odbc running, but it does not really work well for me. I
am getting:
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:325 Stopped rendering
template, called xml_odbc_render_template more than [32] times, probably
looping.
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:408 Something went horribly
wrong while generating an XML template!
My config is:
configuration name=xml_odbc.conf description=XML ODBC Configuration
settings
param name=binding value=directory/
param name=odbc-dsn value=class5_odbc:root:JdqB-S/
param name=debug value=true/
param name=keep-files-around value=true/
/settings
templates
template name=default
document type=freeswitch/xml
section name=directory
xml-odbc-do name=query on-empty-result-break-to=not-found
value=SELECT enabled, sip_password FROM agent WHERE sip_user = ${user}/
domain name=${domain}
user id=${user}
params
xml-odbc-do name=query value=SELECT sip_password FROM
agent WHERE sip_user = ${user}
param name=password, value=${sip_password}/
/xml-odbc-do
/params
/user
/domain
/section
/document
/template
/templates
/configuration
Since these two queries get data from the same table, I tried to merge them,
but could not get it to work.
Anyone has any idea?
Thanks,
JB
___
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 as SBC and kamailio - no route
Bojnour,
I'll send a trace ASAP.
What I see is that SIP header does not get updated - regex is not true then it
does not go to the peer. (I assume that is coming from kamailio config)
I'm really interested to see the updates of the project.
Thank you for the good tutorial.
Hristo
Оригинално писмо
От: rod
Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Четвъртък, 2009, Август 27 09:41:05 EEST
Hi Hristo,
I'm the author of this setup and wiki page. I did a lot of modifications
on this setup (alternative routing if failure essentially) but don't
have too much time to update the wiki.
May you please send me an ngrep trace when you call 1000:
ngrep -d any -nn -i '1000' port 5060 -W byline
I will check what's happening.
Do you have an entry for 1000 in your mysql database ?
regards,
rod
Hristo Benev a écrit :
It seems that the problem is on kamailio configuration.
Will ask on their list.
But to test i try to connect to my asterisk server and i receive 407 proxy
authentication required.
I have it setup as friend in asterisk, but still ???
Any ideas?
Thanks,
Оригинално писмо
От: Hristo Benev
Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Сряда, 2009, Август 26 22:02:13 EEST
I think that the problem is here:
-
2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing
1001-1000 in context ROUTING
Dialplan: sofia/internal/1...@209.71.254.33 parsing [ROUTING-PEER_01]
continue=false
Dialplan: sofia/internal/1...@209.71.254.33 Regex (FAIL) [PEER_01]
${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No
Route, Aborting
--
Actually Regex FAIL
I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be
PEER_01 for success?
Here is my default.xml:
--
Оригинално писмо
От: Brian West
Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no
route
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Сряда, 2009, Август 26 19:47:37 EEST
We do not blindly follow 302's as that is a dangerous thing to do.
You have to process all 302's in the dialplan.
Set this on your sofia profile
You can set these variables sip_redirect_profile,
sip_redirect_context,
sip_redirect_dialplan,
When a redirect happens you get these variables - sip_redirect_contact_%d,
sip_redirected_to,
sip_redirect_contact_user_%d,
sip_redirect_contact_host_%d,
sip_redirect_contact_params_%d,
sip_redirect_dialstring_%d,
sip_redirect_dialstring,
sip_redirected_byThen its up to you to process the redirect in your
dialplan, If you don't set the
sip_redirect_context then it'll default to redirected context and XML as
the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI
followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have
following problem when I dial 1000 Kamalio reports 302, but freeswitch does
not routeWhere to look for problems?
___
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
___
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] delay buildup in conference
very offtopic, but, if this is not to personally, but how do you get to spend literally 8-12 hours a day on a conference? :-) best -- Raimund Sacherer - RunSolutions Open Source It Consulting - Email: r...@runsolutions.com tel: 625 40 32 08 Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares On Aug 26, 2009, at 10:45 PM, Anthony Minessale wrote: There is no bug, it's all dependent on your network conditions. I spend literally 8-12 hours a day on a conference and there is no delay. The important param is param name=rtp-autoflush-during-bridge value=true/ in the sofia profile in question. if you have delay with that in place, then it's probably not FS On Wed, Aug 26, 2009 at 3:30 PM, Public Dump p...@suspiria.net wrote: Hi, I am on a quite recent version (i assume): FreeSWITCH Version 1.0.trunk (14461) Should the bug be fixed in this revision ? What config settings would a have to check to limit delay (even at the cost of reduction in quality). Thanks Von: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] Im Auftrag von Anthony Minessale Gesendet: Mittwoch, 26. August 2009 20:57 An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] delay buildup in conference which revision are you on? The defaults on the latest code and examples should be configured to minimize delay. Some of the older revisions built up some delay issues from udp buffering when timers were not synced. On Wed, Aug 26, 2009 at 12:28 PM, Public Dump p...@suspiria.net wrote: When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay ? I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module. Any advice? Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 -- 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 ___ 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] need help with mod_xml_odbc
Hi Juan,
Perhaps it loops because you didn't include the not-found template ?
Actually, I see there's a bug in the example xml_odbc.conf.xml file
where it's defined with an underscore instead of a dash, will change
that tonight..
The not-found template needs to be specified as a template in the
configuration. I think I'll define that template statically in the
module itself later.
Because it's the 'fall-through' template when it can't find a
template, you get a loop.
So, something like this should probably work for you:
templates
template name=default
xml-odbc-do name=query on-empty-result-break-to=not-found
value=SELECT sip_password FROM agent WHERE sip_user = '${user}';/
document type=freeswitch/xml
section name=directory
domain name=${domain}
user id=${user}
params
param name=password, value=${sip_password}/
/params
/user
/domain
/section
/document
/template
template name=not-found
document type=freeswitch/xml
section name=result
result status=not found/
/section
/document
/template
/templates
(I didn't include the enabled field in your select statement, as you
don't use it later, perhaps you need it in the where clause ?)
Also, note that this way the template will also be used at startup
when FS tries to get a list of all ACL's - I believe for something
else as well, have to check it - but those lookups probably don't give
a ${user} so will render the not-found anyway..
One last thing, you didn't have ${user} enclosed in quotes in your
query, so if no ${user} was given with the lookup to the module, then
your query becomes invalid, which probably breaks things as well.
Let me know if it works..
regards,
Leon
On Aug 27, 2009, at 1:40 PM, Juan Backson wrote:
Hi,
Finally, I got xml_odbc running, but it does not really work well
for me. I am getting:
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:325 Stopped
rendering template, called xml_odbc_render_template more than [32]
times, probably looping.
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:408 Something went
horribly wrong while generating an XML template!
My config is:
configuration name=xml_odbc.conf description=XML ODBC
Configuration
settings
param name=binding value=directory/
param name=odbc-dsn value=class5_odbc:root:JdqB-S/
param name=debug value=true/
param name=keep-files-around value=true/
/settings
templates
template name=default
document type=freeswitch/xml
section name=directory
xml-odbc-do name=query on-empty-result-break-to=not-found
value=SELECT enabled, sip_password FROM agent WHERE sip_user = $
{user}/
domain name=${domain}
user id=${user}
params
xml-odbc-do name=query value=SELECT sip_password
FROM agent WHERE sip_user = ${user}
param name=password, value=${sip_password}/
/xml-odbc-do
/params
/user
/domain
/section
/document
/template
/templates
/configuration
Since these two queries get data from the same table, I tried to
merge them, but could not get it to work.
Anyone has any idea?
Thanks,
JB
___
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] Questions about att_xfer
Just for information - the idea :
A---calls--- B ---att_xfer--- C, B hangs C up and goes back to A
( I`m sorry C is not answering :) )
dialplan:
---
extension name=local_number
condition field=${toll_allow} expression=local/
condition field=destination_number expression=^(\d{3})$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=bind_meta_app data=1 b s
execute_extension::dx XML features/
action application=bind_meta_app data=2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav/
action application=bind_meta_app data=3 b s
execute_extension::cf XML features/
action application=bind_meta_app data=4 b s
execute_extension::attented_xfer XML features/
action application=set data=transfer_ringback=$${hold_music}/
action application=set data=call_timeout=10/
action application=set data=hangup_after_bridge=true/
action application=set
data=continue_on_fail=USER_NOT_REGISTERED,SUBSCRIBER_ABSENT/
action application=info/
action application=set
data=bringback=${user_data(${dialed_extensi...@${domain_name} var
cringback )}/
action application=set data=ringback=${${bringback}}/
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
action application=info/
...
/condition
/extension
features.xml
-
extension name=attented_xfer
condition field=${toll_allow} expression=local/
condition field=destination_number expression=^attented_xfer$
action application=set data=continue_on_fail=true/
action application=read data=3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 3 #/
action application=set
data=bringback=${user_data(${attxfer_callth...@${domain_name} var
cringback )}/
action application=set data=ringback=${${bringback}}/
action application=info/
action application=set data=origination_cancel_key=#/
action application=att_xfer
data=user/${attxfer_callth...@${domain_name}/
/condition
/extension
After testing the patch the results are:
1) B answers the call from A and executes the feature code *4
(attented_xfer), sends the desired number ( in this case the number of
C). A goes to MusicOnHold, B is waiting for C to pick up - B sends # -
sends SIP CANCEL to C and SIP BUY to A - so all calls are dropped.
2) B answers the call from A and executes the feature code *4
(attented_xfer), but without entering the number of C - so the read
aplication timeouts after 30 sec - after that att_xfer is executed
with empty string (nothing is entered) - B and A are bridged together.
On the console without entering the extension of C:
2009-08-27 16:00:58.929756 [NOTICE] switch_core_session.c:1576 Execute
set(origination_cancel_key=#)
EXECUTE sofia/internal/sip:1...@10.17.4.107:5060 set(origination_cancel_key=#)
2009-08-27 16:00:58.929756 [DEBUG] mod_dptools.c:748
sofia/internal/sip:us...@10.17.4.107:5060 SET
[origination_cancel_key]=[#]
2009-08-27 16:00:58.929756 [NOTICE] switch_core_session.c:1576 Execute
att_xfer(user/${attxfer_callth...@${domain_name})
EXECUTE sofia/internal/sip:1...@10.17.4.107:5060 att_xfer(user/@1.1.1.1)
2009-08-27 16:00:58.929756 [WARNING] mod_dptools.c:2373 Can't find
user [...@1.1.1.1]
2009-08-27 16:00:58.929756 [ERR] switch_ivr_originate.c:1527 Cannot
create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-08-27 16:00:58.929756 [DEBUG] switch_ivr_originate.c:2167
Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_play_say.c:1402 done playing file
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:231
sofia/internal/us...@1.1.1.1 receive message [BRIDGE]
2009-08-27 16:00:58.949842 [DEBUG] switch_core_session.c:630 Send
signal sofia/internal/us...@1.1.1.1 [BREAK]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:233 Send signal
sofia/internal/sip:us...@10.17.4.107:5060 [BREAK]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:231
sofia/internal/sip:us...@10.17.4.107:5060 receive message [BRIDGE]
2009-08-27 16:00:58.949842 [DEBUG] switch_core_session.c:630 Send
signal sofia/internal/sip:us...@10.17.4.107:5060 [BREAK]
2009-08-27 16:00:58.954052 [DEBUG] switch_ivr_bridge.c:233 Send signal
sofia/internal/us...@1.1.1.1 [BREAK]
2009-08-27 16:01:22.881501 [DEBUG] sofia.c:3302 Channel
sofia/internal/us...@1.1.1.1 entering state [calling][0]
2009-08-27 16:01:22.886122 [DEBUG] sofia.c:3302 Channel
sofia/internal/us...@1.1.1.1 entering state [ready][200]
I'll try to summarize my idea - could it be possible when the # (in
the att_xfer) is executed to behave as if you're trying to make
attended transfer to non existing subscriber( SUBSCRIBER_ABSENT ), so
B can be bridged back with A.
Thank you in advance,
Anatoliy Kounitskiy
On Wed, Aug 26, 2009 at 11:11 PM, Michael Collinsm...@freeswitch.org
[Freeswitch-users] memory leak
Hello *, a memory leak showed up in our loadtests. It's (still) the same setup as in the http://jira.freeswitch.org/browse/MODSOFIA-22 bugfix. One thing I'd like to add is that fsctl shutdown restart was unable to shutdown freeswitch. The last line printed is switch_core_memory.c:567 Stopping memory pool queue. attached file is a the collected and graphed output of some ps waux command. sz should be in physical pages (2k?). I rerun the test and this time it coredumped trying to malloc() space for some playback. Anything else you need (full backtraces?) to dig into it? Cheers Beni. attachment: mem-sz.png___ 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] need help with mod_xml_odbc
Made a typo in the param, you have to leave out the comma..
regards,
Leon
On Aug 27, 2009, at 2:57 PM, Leon de Rooij wrote:
Hi Juan,
Perhaps it loops because you didn't include the not-found template ?
Actually, I see there's a bug in the example xml_odbc.conf.xml file
where it's defined with an underscore instead of a dash, will change
that tonight..
The not-found template needs to be specified as a template in the
configuration. I think I'll define that template statically in the
module itself later.
Because it's the 'fall-through' template when it can't find a
template, you get a loop.
So, something like this should probably work for you:
templates
template name=default
xml-odbc-do name=query on-empty-result-break-to=not-found
value=SELECT sip_password FROM agent WHERE sip_user = '${user}';/
document type=freeswitch/xml
section name=directory
domain name=${domain}
user id=${user}
params
param name=password, value=${sip_password}/
/params
/user
/domain
/section
/document
/template
template name=not-found
document type=freeswitch/xml
section name=result
result status=not found/
/section
/document
/template
/templates
(I didn't include the enabled field in your select statement, as you
don't use it later, perhaps you need it in the where clause ?)
Also, note that this way the template will also be used at startup
when FS tries to get a list of all ACL's - I believe for something
else as well, have to check it - but those lookups probably don't give
a ${user} so will render the not-found anyway..
One last thing, you didn't have ${user} enclosed in quotes in your
query, so if no ${user} was given with the lookup to the module, then
your query becomes invalid, which probably breaks things as well.
Let me know if it works..
regards,
Leon
On Aug 27, 2009, at 1:40 PM, Juan Backson wrote:
Hi,
Finally, I got xml_odbc running, but it does not really work well
for me. I am getting:
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:325 Stopped
rendering template, called xml_odbc_render_template more than [32]
times, probably looping.
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:408 Something went
horribly wrong while generating an XML template!
My config is:
configuration name=xml_odbc.conf description=XML ODBC
Configuration
settings
param name=binding value=directory/
param name=odbc-dsn value=class5_odbc:root:JdqB-S/
param name=debug value=true/
param name=keep-files-around value=true/
/settings
templates
template name=default
document type=freeswitch/xml
section name=directory
xml-odbc-do name=query on-empty-result-break-to=not-found
value=SELECT enabled, sip_password FROM agent WHERE sip_user = $
{user}/
domain name=${domain}
user id=${user}
params
xml-odbc-do name=query value=SELECT sip_password
FROM agent WHERE sip_user = ${user}
param name=password, value=${sip_password}/
/xml-odbc-do
/params
/user
/domain
/section
/document
/template
/templates
/configuration
Since these two queries get data from the same table, I tried to
merge them, but could not get it to work.
Anyone has any idea?
Thanks,
JB
___
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
[Freeswitch-users] mod_limit and memcache
Hello, I read something that talks about using memcache for mod_limit before. Is it something that is available now? If I have multiple instances of freeswitch that need to share the same limit status, it there any existing solution? If no existing solution is available, what is the best way to go about modifying mod_limit to accomplish limiting for multiple freeswitch servers together? Thanks, Woody ___ 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] need help with mod_xml_odbc
Hi Leon,
Thanks for your help.
I have changed it according to your comment but I am still getting the
looping error.
Would you please take a look see what else I did wrong?
Also, sip_user is an integer field, so I can't really use ''. Is there
anyway to get around that?
configuration name=xml_odbc.conf description=XML ODBC Configuration
settings
param name=binding value=directory/
param name=odbc-dsn value=class5_odbc:root:abcd/
param name=debug value=true/
param name=keep-files-around value=true/
/settings
templates
template name=default
document type=freeswitch/xml
section name=directory
xml-odbc-do name=query on-empty-result-break-to=not-found
value=SELECT enabled, sip_password FROM agent WHERE sip_user = ${user} and
enabled='t'/
domain name=${domain}
user id=${user}
params
xml-odbc-do name=query value=SELECT sip_password FROM
agent WHERE sip_user = ${user}
param name=password value=${sip_password}/
/xml-odbc-do
/params
/user
/domain
/section
/document
/template
template
document type=freeswitch/xml
section name=result
result status=not found/
/section
/document
/template
/templates
/configuration
___
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] fscore mutex locking question
Hello *,
while looking at the code i came across a region of code which is
unclear to me regarding locking issues.
One example is switch_ivr_broadcast in switch_ivr_async.c. This should
be the function called by
uuid_broadcast() and others.
in line 2341 it tries to queue an event to the bleg if it has to...
-
. if ((flags SMF_ECHO_BLEG) (other_uuid =
switch_channel_get_variable(channel, SWITCH_SIGNAL_BOND_VARIABLE))
. .(other_session = switch_core_session_locate(other_uuid))) {
---
switch_core_session_locate() does a readonly trylock on the channel
mutex returning NULL if it's unable to
aquire the lock, which brings up the following question:
If some other thread is currently holding a writelock on the channel,
the broadcast is not queued and not retried at a later time at all?
I guess it's pretty easy to cause some unexpected behaviour using some
endless loop calling uuid_setvar or some other race condition where
the channel-mutex is write-locked while calling uuid_broadcast (eg.
uuid_media?).
Could this lead to a problem in real live scenarios, or are there
other countermeasures despite chances are one in a million that you
hit that small time frame?
thx in advance
Beni.
___
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] delay buildup in conference
Our whole development team works from 6 different states so we use a 24x7 conference as part of our virtual office. There is a public conference for freeswitch users as well. http://conference.freeswitch.org sip:8...@conference.freeswitch.orgsip%3a...@conference.freeswitch.org On Thu, Aug 27, 2009 at 7:10 AM, Raimund Sacherer r...@runsolutions.comwrote: very offtopic, but, if this is not to personally, but how do you get to spend literally 8-12 hours a day on a conference? :-) best -- Raimund Sacherer - RunSolutions Open Source It Consulting - Email: r...@runsolutions.com tel: 625 40 32 08 Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares On Aug 26, 2009, at 10:45 PM, Anthony Minessale wrote: There is no bug, it's all dependent on your network conditions. I spend literally 8-12 hours a day on a conference and there is no delay. The important param is param name=rtp-autoflush-during-bridge value=true/ in the sofia profile in question. if you have delay with that in place, then it's probably not FS On Wed, Aug 26, 2009 at 3:30 PM, Public Dump p...@suspiria.net wrote: Hi, I am on a quite recent version (i assume): FreeSWITCH Version 1.0.trunk (14461) Should the bug be fixed in this revision ? What config settings would a have to check to limit delay (even at the cost of reduction in quality). Thanks *Von:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *Im Auftrag von *Anthony Minessale *Gesendet:* Mittwoch, 26. August 2009 20:57 *An:* freeswitch-users@lists.freeswitch.org *Betreff:* Re: [Freeswitch-users] delay buildup in conference which revision are you on? The defaults on the latest code and examples should be configured to minimize delay. Some of the older revisions built up some delay issues from udp buffering when timers were not synced. On Wed, Aug 26, 2009 at 12:28 PM, Public Dump p...@suspiria.net wrote: When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay ? I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module. Any advice? Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- 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 -- 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 ___ 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
Re: [Freeswitch-users] memory leak
600k is not a leak? FS can use as much as a gig of ram or more depending on what you are doing. you may want to install a fresh copy of FS, removing all your old files etc and make sure they build clean. We also have not had much luck running on ubuntu which is more of a desktop centric OS. I recommend you try your application on 64 bit CentOS which is the platform all of our paid customers use. On Thu, Aug 27, 2009 at 8:15 AM, Benedikt Fraunhofer fraunhofer.lists.freeswitch-...@traced.net wrote: Hello *, a memory leak showed up in our loadtests. It's (still) the same setup as in the http://jira.freeswitch.org/browse/MODSOFIA-22 bugfix. One thing I'd like to add is that fsctl shutdown restart was unable to shutdown freeswitch. The last line printed is switch_core_memory.c:567 Stopping memory pool queue. attached file is a the collected and graphed output of some ps waux command. sz should be in physical pages (2k?). I rerun the test and this time it coredumped trying to malloc() space for some playback. Anything else you need (full backtraces?) to dig into it? Cheers Beni. ___ 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] mod_limit and memcache
limit and memcache haven't been introduced to each other yet -- it is on my (semi-long) list of things to do. If you want it you can: 1) do it yourself and submit the patches 2) open a jira and hope someone does it 3) open a jira + bounty and someone will probably do it It will get done eventually, just hasn't been a itch for ME to scratch yet. To do it: 1) I need to make it possible to call inc/dec methods of mod_memcache to support an expiration time. 2) mod_limit.c - use the hash limit as a guide Initial pitfalls: hash limits concurrent access/modification of the hash and by implication limit_hash_item_t (hash data) by using a mutex. We can't mutex across FS instances. So perhaps split up limit_has_item_t and spread it across multiple keys. So instead of one key marked as realm_id, we could have realm_id_total_usage realm_id_rate_usage and realm_id_last_check. This does mean that rate_usage and total_usage can inc/dec independent of each other, but I think the logic will still be ok *IF* we remember to decrement earlier incremented items in the event a later item is failed. (so, say we increment rate but fail on total we need to remember to decrement rate so that we have no net effect on the counters) Alternatively, we could use CAS support and pull the limit_hash_item_t item from memcache, twiddle it and then try to put it back only if the check info is the same (no one else has changed the entry). If the entry has changed, pull the new version, do the limit logic, and try again. Loop that a few times until you succeed or give up. Problem is that CAS needs to be explicitly turned on in memcache (some distros compile with it off), is relatively new in memcache (hint: may have issues) and has some performance/memory downsides though by how much I'm not sure. Thoughts? On Thu, Aug 27, 2009 at 8:49 AM, Woody Dicksonwoodydick...@gmail.com wrote: Hello, I read something that talks about using memcache for mod_limit before. Is it something that is available now? If I have multiple instances of freeswitch that need to share the same limit status, it there any existing solution? If no existing solution is available, what is the best way to go about modifying mod_limit to accomplish limiting for multiple freeswitch servers together? Thanks, Woody ___ 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
[Freeswitch-users] Re : upstream Registrar / Mirror proxy
Thanks for the feedback Steve: we are using Genband/Nextone MSX , but we're looking for a cheaper alternative to avoid using our vports for this purpose. We'll check OpenSBC .. thanks De : Steve Kurzeja steve.kurz...@gmail.com À : freeswitch-users@lists.freeswitch.org Envoyé le : Jeudi, 27 Août 2009, 10h21mn 47s Objet : Re: [Freeswitch-users] upstream Registrar / Mirror proxy A proxy won't do what the original poster is asking for. Upper registration is a special type of function performed by SBCs and not defined by any RFC yet but there are drafts out there. This question comes up quite often in various mailing lists and has been asked this list before. The answer is no freeswitch can't be configured to do this as it stands today. The only opensource SBC I know of that attempts to do upper registration at the moment is OpenSBC. Otherwise there's vendor SBCs like ACME packet, Nextone (now Genband - they call it Mirror Proxy mode) etc which do it. Regards, Steve On Thu, Aug 27, 2009 at 9:02 PM, Ken Rice kr...@freeswitch.org wrote: FreeSWITCH is a B2BUA and NOT a proxy and will not proxy any requests (REGISTER or INVITE) If you want a Proxy you should look toward OpenSIPS From: sylver_b sylve...@yahoo.com Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 26 Aug 2009 17:14:55 -0700 (PDT) To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] upstream Registrar / Mirror proxy Hello, We would like to use FreeSwitch to test interoperability with our sip registrar platform - as our registrar doesnt handle NAT Traversal we would like to use FS as a mirror proxy . If a call goes through, FS should transparently handle nat traversal functionalities while forwarding the REGISTER/INVITE requests to our registrar server. Below some detailed requirements: Upper Registration is the capability of a SBC to proxy Registrations towards an upstream Registrar. While it is necessary that all SIP requests traverse the SBC for NAT continuity, Registrations should be allowed to be relayed towards our upstream Registrar while the SBC retains a copy of the AOR and masquerade on behalf of the UA that send the registration request. Please let us know the best way to configure FS to achieve this type of configuration. Thank you ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] need help with mod_xml_odbc
Hi Juan,
With debug=true you should be able to see what template it's trying to
render in a loop, can you tell which one that is ? (I'm guessing it
says 32 times it wants to render not-found)
In the xml you pasted in your mail, you didn't specify the name of the
not-found template, just template, that could be the problem..
To get around the problem of generating a wrong query when the ${user}
is not given, you could simply encapsulate everything with a check-
event-header, like this:
template name=default
xml-odbc-do name=check-event-header if-name=user
xml-odbc-do name=query on-empty-result-break-to=not-found
value=...etc
..rest of your template goes here..
/xml-odbc-do
/xml-odbc-do
/template
But that would mean that an empty xml is returned whenever no $
{user} is provided. I didn't test how fs will react to that when it's
looking up all the users for generating the acl list..
A way around that would be to only render your template when ${user}
is set and otherwise render the not-found template, like this:
template name=default
xml-odbc-do name=check-event-header if-name=user
xml-odbc-do name=break-to value=directory-user/
/xml-odbc-do
xml-odbc-do name=break-to value=not-found/
/template
template name=directory-user
xml-odbc-do name=query on-empty-result-break-to=not-found
value=...etc
..rest of your template goes here..
/xml-odbc-do
/template
template name=not-found
etc...
/template
---
Which comes close to how it's defined in the xml_odbc.conf.xml
example, which solves it the other way around:
...
template name=default
xml-odbc-do name=break-to value=${section}/
/template
template name=directory --
xml-odbc-do name=check-event-header if-name=purpose
!-- catches purpose gateways and network-list (any more?) --
xml-odbc-do name=break-to value=directory-${purpose}/
/xml-odbc-do
xml-odbc-do name=break-to value=directory-user/
/template
...
- First it checks what section is requested, which is nice but not
really necessary in your case, since you only bind to directory.
- Then it checks whether a purpose header is set (which was only the
case for looking up acl's and gateways and NOT for regular user
lookups last time I checked out the svn)
- If a purpose header IS set, then it tries to render directory-$
{purpose}, which isn't specified, so it falls back to not-found
which is what you want in this case
- And as a last resort - this is where regular user lookups end up, it
renders directory-user where you can put the template as specified
above..
Hope that helps ?
regards,
Leon
On Aug 27, 2009, at 3:53 PM, Juan Backson wrote:
Hi Leon,
Thanks for your help.
I have changed it according to your comment but I am still getting
the looping error.
Would you please take a look see what else I did wrong?
Also, sip_user is an integer field, so I can't really use ''. Is
there anyway to get around that?
configuration name=xml_odbc.conf description=XML ODBC
Configuration
settings
param name=binding value=directory/
param name=odbc-dsn value=class5_odbc:root:abcd/
param name=debug value=true/
param name=keep-files-around value=true/
/settings
templates
template name=default
document type=freeswitch/xml
section name=directory
xml-odbc-do name=query on-empty-result-break-to=not-found
value=SELECT enabled, sip_password FROM agent WHERE sip_user = $
{user} and enabled='t'/
domain name=${domain}
user id=${user}
params
xml-odbc-do name=query value=SELECT sip_password
FROM agent WHERE sip_user = ${user}
param name=password value=${sip_password}/
/xml-odbc-do
/params
/user
/domain
/section
/document
/template
template
document type=freeswitch/xml
section name=result
result status=not found/
/section
/document
/template
/templates
/configuration
___
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] Odd sonus warning
Hi, I have some DIDs from Bandwidth.com and when they call in I see this in the console: 2009-08-27 11:01:47 [WARNING] sofia_glue.c:2701 sofia_glue_negotiate_sdp() Hello, I see you have a Sonus! FYI, Sonus cannot follow the RFC on the proper way to send DTMF. Sadly, my creator had to spend several hours figuring this out so I thought you'd like to know that! Don't worry, DTMF will work but you may want to ask them to fix it.. Anything I should worry about? -- Greg ___ 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] Odd sonus warning
Anything I should worry about? Nothing to worry about. Freeswitch will deal with the issue, but there are a few caveats. I believe this is the most up-to date resource: http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus ___ 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] Odd sonus warning
Does your DTMF work? /b On Aug 27, 2009, at 10:06 AM, Greg Thoen wrote: Anything I should worry about? ___ 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 as SBC and kamailio - no route
I assume you asked for port 5062 since I do not have any traffic on 5060 (I
have one IP and my internal sip port is 5090 and external 5080).
If you need additional info I'll provide it.
Here is trace:
ngrep -d any -nn -i '1000' port 5062 -W byline
interface: any
filter: (ip or ip6) and ( port 5062 )
match: 1000
#
U 10.10.10.10:5090 - 127.0.0.1:5062
INVITE sip:1...@127.0.0.1:5062 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
Max-Forwards: 69.
From: Extension 1001 sip:1...@10.10.10.10;tag=978g69jZaFpBD.
To: sip:1...@127.0.0.1:5062.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: sip:mod_so...@10.10.10.10:5090.
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.
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 429.
X-ROUTE: LOOKUP.
Remote-Party-ID: Extension 1001
sip:1...@10.10.10.10;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
s=FreeSWITCH.
c=IN IP4 10.10.10.10.
t=0 0.
m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:115 G7221/32000.
a=fmtp:115 bitrate=48000.
a=rtpmap:107 G7221/16000.
a=fmtp:107 bitrate=32000.
a=rtpmap:9 G722/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.
#
U 127.0.0.1:5062 - 10.10.10.10:5090
SIP/2.0 302 PEER_01.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
From: Extension 1001 sip:1...@10.10.10.10;tag=978g69jZaFpBD.
To: sip:1...@127.0.0.1:5062;tag=458fb4012080e656b6742c09466dabcd.31c8.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: sip:fra...@peer_01.
Server: Kamailio (1.5.2-notls (i386/linux)).
Content-Length: 0.
.
#
U 10.10.10.10:5090 - 127.0.0.1:5062
ACK sip:1...@127.0.0.1:5062 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
Max-Forwards: 69.
From: Extension 1001 sip:1...@10.10.10.10;tag=978g69jZaFpBD.
To: sip:1...@127.0.0.1:5062;tag=458fb4012080e656b6742c09466dabcd.31c8.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 ACK.
Content-Length: 0.
.
Оригинално писмо
От: Hristo Benev
Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Четвъртък, 2009, Август 27 14:58:53 EEST
Bojnour,
I'll send a trace ASAP.
What I see is that SIP header does not get updated - regex is not true then
it does not go to the peer. (I assume that is coming from kamailio config)
I'm really interested to see the updates of the project.
Thank you for the good tutorial.
Hristo
Оригинално писмо
От: rod
Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Четвъртък, 2009, Август 27 09:41:05 EEST
Hi Hristo,
I'm the author of this setup and wiki page. I did a lot of modifications
on this setup (alternative routing if failure essentially) but don't
have too much time to update the wiki.
May you please send me an ngrep trace when you call 1000:
ngrep -d any -nn -i '1000' port 5060 -W byline
I will check what's happening.
Do you have an entry for 1000 in your mysql database ?
regards,
rod
Hristo Benev a écrit :
It seems that the problem is on kamailio configuration.
Will ask on their list.
But to test i try to connect to my asterisk server and i receive 407
proxy authentication required.
I have it setup as friend in asterisk, but still ???
Any ideas?
Thanks,
Оригинално писмо
От: Hristo Benev
Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no
route
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Сряда, 2009, Август 26 22:02:13 EEST
I think that the problem is here:
-
2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing
1001-1000 in context ROUTING
Dialplan: sofia/internal/1...@209.71.254.33 parsing [ROUTING-PEER_01]
continue=false
Dialplan: sofia/internal/1...@209.71.254.33 Regex (FAIL) [PEER_01]
${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No
Route, Aborting
--
Actually Regex FAIL
I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be
PEER_01 for success?
Here is my default.xml:
Re: [Freeswitch-users] Call exits after 120 seconds with hangup cause
Is the snom firmware up to the latest? yes, the firmware is the latest. I believe session timers should work properly with snom? i don't know, but i think so. we tested with session timers off and with setting the session timer to 0. both does not change anything. we played with using a stun-server and had a small success. if we are connected with a stund-server, the hangup will not come after 120 seconds. we can also call the 5900 and listen to the moh. this works fine. here we have a totally different problem. if another voip phone is connected to the server using stun (for example the 1000), both sides can not talk to each other. one won't hear, if the other side is talking. very strange (for us), but we have no idea, what we can do about that. ___ 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 exits after 120 seconds with hangup cause
Are the phones behind the same nat as FS? /b On Aug 27, 2009, at 10:39 AM, Dennis wrote: we played with using a stun-server and had a small success. if we are connected with a stund-server, the hangup will not come after 120 seconds. we can also call the 5900 and listen to the moh. this works fine. here we have a totally different problem. if another voip phone is connected to the server using stun (for example the 1000), both sides can not talk to each other. one won't hear, if the other side is talking. ___ 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] Odd sonus warning
On Thu, 2009-08-27 at 11:06 -0400, Greg Thoen wrote: Hi, I have some DIDs from Bandwidth.com and when they call in I see this in the console: 2009-08-27 11:01:47 [WARNING] sofia_glue.c:2701 sofia_glue_negotiate_sdp() Hello, I see you have a Sonus! FYI, Sonus cannot follow the RFC on the proper way to send DTMF. Sadly, my creator had to spend several hours figuring this out so I thought you'd like to know that! Don't worry, DTMF will work but you may want to ask them to fix it.. Anything I should worry about? Not unless you're a Sonus shareholder - see http://messages.finance.yahoo.com/Stocks_(A_to_Z)/Stocks_S/threadview?m=tebn=16942tid=827769mid=-1tof=15rt=1frt=2 for an amusing read ;-) --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ 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] Who is callie?
On Thu, Aug 27, 2009 at 9:04 AM, Carlos S. Antunes c...@nowthor.com wrote: Hi! I searched the wiki but couldn't find the answer. Is callie a real woman one may ask to record additional sounds? Callie is one of the voices from GM Voices. She is definitely available for custom work. Visit www.gmvoices.com for more info. Tell them that the FreeSWITCH project sent you. :) -MC Thanks! Carlos ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] memory leak
Hello Anthony, 2009/8/27 Anthony Minessale anthony.miness...@gmail.com: 600k is not a leak? FS can use as much as a gig of ram or more depending on what you are doing. it's 600*1024 * 4* 1024 /1024/1024 = ~ 2.4 Gig that's what i tried to express with the physical pages in my first post. you may want to install a fresh copy of FS, removing all your old files etc and make sure they build clean. i make current to get the SOFIA-22 bugfix. We also have not had much luck running on ubuntu which is more of a desktop centric OS. really? even the server edition? (not the netbook remix :) Is 64bit so grave different if i don't need to address more than 4 Gig of ram per process? I recommend you try your application on 64 bit CentOS which is the platform all of our paid customers use. hmm. our operations-people wont be happy with that. But as i carefully followed the other mailing-list-threads it came to my mind that i could be the culprit: somehow the original example.xml came back into the directory tree and included a lot of things, most notably the hash-inserts for last-dailed-numbers and stuff. I stripped it down to what it was before and gave it another try. thx again and (hopefully) sorry Beni. ___ 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] Get the Session State
When you say call state, do you mean something like ringing or answered? -MC On Thu, Aug 27, 2009 at 3:23 AM, delianSPAM delians...@gmail.com wrote: Hello Everybody! How do you get the call state in Python? I have tried: … session.answer() state_result=str(session.getVariable(state)) console_log(debug,state_result) … But it returns: “None” Thank you! Best Regards, Delian Tashev ___ 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] How to make a call back
On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan lakindi...@gmail.com wrote: When I give the following from the command line it calls to 1010 extension and once answered, it calls to 1000 and bridge the connection. originate user/1010 bridge(user/1000) But I want to do this in perl. So I have given as follows $session-originate($session,user/1010 bridge user/1000); But it is not working. It says user/1010 bridge user/1000 is invalid user. How to do this in perl. pls help. Are you calling this perl script from the CLI? If so you won't have the $session object because a channel does not exist for a simple API call. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Question about presence
Open on endpoint modules. We will relocate if needed. jmesquita On Thu, Aug 27, 2009 at 8:06 AM, Dennis oderm...@googlemail.com wrote: sorry, but i do not know i which category i have to set this problem. could you help me with that? ___ 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] Who is callie?
Hi! I searched the wiki but couldn't find the answer. Is callie a real woman one may ask to record additional sounds? Thanks! Carlos ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call exits after 120 seconds with hangup cause
Are the phones behind the same nat as FS? no, both phones are behind the same nat, fs is behind another nat (and the internet is inbetween). both phones are in our office and we are sitting behind a nat. we connect through our nat over the internet into the other nat, where fs is behind. ___ 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] fscore mutex locking question
The only time the session is write locked is when its about to be
free'd, to make sure nothing holds a valid pointer.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 27-Aug-09, at 10:08 AM, Benedikt Fraunhofer wrote:
Hello *,
while looking at the code i came across a region of code which is
unclear to me regarding locking issues.
One example is switch_ivr_broadcast in switch_ivr_async.c. This should
be the function called by
uuid_broadcast() and others.
in line 2341 it tries to queue an event to the bleg if it has to...
-
. if ((flags SMF_ECHO_BLEG) (other_uuid =
switch_channel_get_variable(channel, SWITCH_SIGNAL_BOND_VARIABLE))
. .(other_session = switch_core_session_locate(other_uuid))) {
---
switch_core_session_locate() does a readonly trylock on the channel
mutex returning NULL if it's unable to
aquire the lock, which brings up the following question:
If some other thread is currently holding a writelock on the channel,
the broadcast is not queued and not retried at a later time at all?
I guess it's pretty easy to cause some unexpected behaviour using some
endless loop calling uuid_setvar or some other race condition where
the channel-mutex is write-locked while calling uuid_broadcast (eg.
uuid_media?).
Could this lead to a problem in real live scenarios, or are there
other countermeasures despite chances are one in a million that you
hit that small time frame?
thx in advance
Beni.
___
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] Call exits after 120 seconds with hangup cause
Can you paste in the line that actually hangs up your call it's BLUE and says something like 2009-08-27 11:39:04.903202 [NOTICE] sofia.c:327 Hangup sofia/internal/ 1...@dev.bkw.org [CS_EXECUTE] [NORMAL_CLEARING] find the line like this for each leg of you call and put them in your reply On Thu, Aug 27, 2009 at 11:01 AM, Dennis oderm...@googlemail.com wrote: Are the phones behind the same nat as FS? no, both phones are behind the same nat, fs is behind another nat (and the internet is inbetween). both phones are in our office and we are sitting behind a nat. we connect through our nat over the internet into the other nat, where fs is behind. ___ 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] Question about presence
are you setting presence_id=u...@domain variable on the outbound leg? This is done for you in the DP via the user/ channel in the defaults but if you are not using this you have to set it manually. 2009/8/27 João Mesquita jmesqu...@freeswitch.org Open on endpoint modules. We will relocate if needed. jmesquita On Thu, Aug 27, 2009 at 8:06 AM, Dennis oderm...@googlemail.com wrote: sorry, but i do not know i which category i have to set this problem. could you help me with that? ___ 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 -- 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] Questions about att_xfer
please retest with latest SVN trunk On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy anato...@kounitskiy.com wrote: Just for information - the idea : A---calls--- B ---att_xfer--- C, B hangs C up and goes back to A ( I`m sorry C is not answering :) ) -- 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] Question about presence
are you setting presence_id=u...@domain variable on the outbound leg?
This is done for you in the DP via the user/ channel in the defaults but if
you are not using this
you have to set it manually.
in directory default we have the following:
params
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}/
/params
everthing works fine with the led lights - the only problem is the
described above.
___
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] FXO and analogue phones
Merul,
My apologies, I had meant to follow up much earlier. Would you mind going
ahead and capturing full debug log output of a call in as well as a call
out? Please use pastebin.freeswitch.org to post the logs and then reply here
with the link to the pb post.
Thanks,
MC
On Sun, Aug 23, 2009 at 10:27 AM, Merul Patel me...@mac.com wrote:
I have a Freeswitch setup working on an Alix embedded platform in
conjunction with a USB FXO device from Sangoma. My goal is to be able to
either answer incoming calls on a softphone or on a POTS handset elsewhere
in the building, and to also be able to make outgoing calls from either. For
clarity, the analogue line has two physical extensions, one connected to the
POTS and the other to the FXO.
I can make and receive calls fine, but have problems when the call is
answered on the POTS handset.
Here is the dialplan I initially used in
/opt/freeswitch/conf/dialplans/public/01_incoming.xml:
include
extension name=public_did
condition field=${strftime(%w)} expression=^(\d)$
!-- There seems to be a delay of 7 seconds from when FS starts
dealing with the call and from when it starts ringing --
action application=sleep data=23000/
action application=set data=domain_name=$${domain}/
action application=transfer data=1001 XML default/
/condition
/extension
/include
It's pretty basic, and if the softphone is not registered or does not
answer then the call goes to voicemail. However the call will always go to
voicemail, and the voicemail application will begin to execute after the
call has been answered on the POTS handset.
I've been trying to make the dialplan more useful, by having it ring the
softphone immediately, and only transfer the call to the voicemail
application if the line is still ringing. I'm in the UK, hence my choice of
frequencies in the tone_detect application:
include
extension name=public_did
condition field=${strftime(%w)} expression=^(\d)$
!-- There seems to be a delay of 7 seconds from when FS starts
dealing with the call and from when it starts ringing --
action application=set data=call_timeout=23/
action application=set data=continue_on_fail=true/
action application=set data=hangup_after_bridge=true/
action application=bridge data=sofia/internal/1001%$${domain}/
action application=sleep data=23000/
action application=tone_detect data=ring 400,450 r +5000 set
RING=true/
action application=transfer data=public_answer_and_email/
/condition
/extension
extension name=public_answer_and_email
condition field=RING expression=true
action application=answer/
action application=voicemail data=default $${domain} 1001/
/condition
/extension
/include
Unfortunately, this is not working, and the logs are not yielding anything
the is helpful to me. Is my use of the tone_detect application and the basic
dialplan correct?
Merul
___
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] Call exits after 120 seconds with hangup cause
this is the line (without stun - so we only have one leg) and we called the 5900 to moh: 2009-08-27 19:18:02.348232 [NOTICE] sofia.c:3863 Hangup sofia/internal/1...@212.18.215.102 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] we called the 5900 and waited 2 minutes... or did you mean something different? ___ 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] fifo question
On Mon, Aug 24, 2009 at 7:25 AM, Juan Backson juanback...@gmail.com wrote: Hi, Does anyone know the purpose of fifo_orbit_announce? When does fifo_orbit_announce get played? I believe this is what gets played to a caller when he/she has been in queue too long and it times out. I'm assuming this gets played and then the caller is transferred to the fifo_orbit_extension. (See also fifo_orbit_context and fifo_orbit_dialplan.) -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call exits after 120 seconds with hangup cause
do the following: 1) restart FS 2) type sofia profile internal siptrace on 3) type console loglevel debug 4) reproduce the issue 5) upload the resulting output to http://pastebin.freeswitch.org (*hint* the pass is in the dialog box) 6) let me know the pastebin number On Thu, Aug 27, 2009 at 12:25 PM, Dennis oderm...@googlemail.com wrote: this is the line (without stun - so we only have one leg) and we called the 5900 to moh: 2009-08-27 19:18:02.348232 [NOTICE] sofia.c:3863 Hangup sofia/internal/1...@212.18.215.102 [CS_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] we called the 5900 and waited 2 minutes... or did you mean something different? ___ 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] mod_voicemail email template variables
all variables referenced in the template should expand when sending the email. On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger nick.lember...@lkfd.netwrote: Is there a way to use dialplan variables in the email that gets sent with the voicemail attachement. I tried using some but nothing seems to show up, I'm guessing it's a different channel or something... Any ideas? Thanks, -Nick ___ 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] sofia profile external register gwname via XML-Curl?
I got it, gateways have to be preloaded (rescanned) before they can be registered. Best regards peter Peter P GMX schrieb: Hello, I am using XML-Curl to handle the configuration of freeswitch When I try to register a gateway via event-socket with sofia profile external register gw-name I receive back invalid gateway. After reload mod_sofia the gateway is there. Question: Does this command work with xml-curl or only with local files?? At least I see no xml-curl request when grepping network traffic. Best regards Peter ___ 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] Calls from registered gateway try to lookup Directory
I have found a strange thing in my FS installation, FS is registered via a Gateway to an external provider (QSC) in the external context. But when a call is coming in, FS does not seem to go to any context, but tries to lookup the user, as I receive the following message 2009-08-27 18:44:23.287782 [WARNING] sofia_reg.c:1773 Can't find user [026xx...@my.domain@my.domain] You must define a domain called 'my.domain' in your directory and add a user with the id=026xx...@my.domain attribute and you must configure your device to use the proper domain in it's authentication credentials. I learnt that a call from an external gateway should go to the public context. But (in CLI debug mode) there are no other messages, except the 3 lines above. What am I doing wrong? Best regards Peter Here the invite message. INVITE sip:gw+gw_xxx...@xx.xxx.xx.xxx:5080;transport=udp SIP/2.0. Via:SIP/2.0/UDP 62.206.3.xxx;branch=z9hG4bK-BroadWorks.as1-xx.xxx.xx.xxxV5080-0-778271239-1616003581-1251392025611-. From:0XXsip:0xxx...@62.206.3.xxx;user=phone;tag=1616003581-1251392025611-. To:Mesip:026xx...@my.domain. Call-ID:bw185345611270809356816...@62.206.3.xxx. CSeq:778271239 INVITE. Contact:sip:62.206.3.xxx:5060. Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE. Accept:multipart/mixed,application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp. Supported:. Max-Forwards:20. Proxy-Authorization:DIGEST cnonce=fyvqi2pf,qop=auth,uri=sip:gw+gw_026xx...@xx.xxx.xx.xxx:5080;transport=udp,realm=my.domain,username=026xx...@my.domain,nonce=21bbe70c-932a-11de-b94d-bbade892ded3,algorithm=MD5,response=6d39a2546a4aa9a1fc39e2dc07c1e934,nc=0001. Content-Type:application/sdp. Content-Length:344. . v=0. o=BroadWorks 1271473 1 IN IP4 87.234.9.178. s=-. c=IN IP4 87.234.9.178. t=0 0. m=audio 18534 RTP/AVP 8 0 2 99 18 110. a=rtpmap:99 G726-24/8000. a=rtpmap:110 X-NSE/8000. a=fmtp:110 192-194,200-202. a=X-sqn:0. a=X-cap: 1 audio RTP/AVP 110. a=X-cpar: a=rtpmap:110 X-NSE/8000. a=X-cpar: a=fmtp:110 192-194,200-202. a=X-cap: 2 image udptl t38. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_voicemail email template variables
Thanks for the fast reply! I just tried 10 random variables from http://wiki.freeswitch.org/wiki/Channel_Variables and I only see the whitespace where the variable should be. I've only been able to get the ones that are set in mod_voicemail.c circa line 1600 to work. -Nick On 8/27/2009 at 12:44 PM, in message 191c3a030908271044k63973088xeec12c578d02e...@mail.gmail.com, Anthony Minessale anthony.miness...@gmail.com wrote: all variables referenced in the template should expand when sending the email. On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger nick.lember...@lkfd.netwrote: Is there a way to use dialplan variables in the email that gets sent with the voicemail attachement. I tried using some but nothing seems to show up, I'm guessing it's a different channel or something... Any ideas? Thanks, -Nick ___ 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] Call exits after 120 seconds with hangup cause
the pastebin number is 10129 ___ 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] newbie alert. Help with dialplan
I am a long-time asterisk user (2005), so I come from an unfortunate position of having to unlearn an awful lot of stuff in order to make freeswitch do the things I want ;( I *think* i've got my head around using mod_curl_xml (?) to read all the config stuff from my webserver. I *think* I've got my head around setting up the sip clients etc etc ... However, where I am really struggling is the dialplan. For the life of me I simply cannot seem to grasp the fs way - that's no disrespect to the fs way, but perhaps the failure of this old brain to change ! have pastebinned an example of show 1234...@inboundq at http://www.pastebin.ca/1544890. If possible, would someone be able to show me how to convert this dialplan to the fs way ? If I could be given a little foot-up I'm sure that I would be able to convert the rest of the dialplan ! Thanks in advance, and please go easy ! Julian ___ 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] Questions about att_xfer
/me points to the first question on the FAQ ;) /b On Aug 27, 2009, at 1:18 PM, Anatoliy Kounitskiy wrote: Thank you all for the great job :) Now it works as I wanted!! Tomorrow I'll will try to update the wiki :) ___ 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] Calls from registered gateway try to lookup Directory
And yes, external profile is on Port 5080 and all request go to 5080. Best regards Peter Peter P GMX schrieb: I have found a strange thing in my FS installation, FS is registered via a Gateway to an external provider (QSC) in the external context. But when a call is coming in, FS does not seem to go to any context, but tries to lookup the user, as I receive the following message 2009-08-27 18:44:23.287782 [WARNING] sofia_reg.c:1773 Can't find user [026xx...@my.domain@my.domain] You must define a domain called 'my.domain' in your directory and add a user with the id=026xx...@my.domain attribute and you must configure your device to use the proper domain in it's authentication credentials. I learnt that a call from an external gateway should go to the public context. But (in CLI debug mode) there are no other messages, except the 3 lines above. What am I doing wrong? Best regards Peter Here the invite message. INVITE sip:gw+gw_xxx...@xx.xxx.xx.xxx:5080;transport=udp SIP/2.0. Via:SIP/2.0/UDP 62.206.3.xxx;branch=z9hG4bK-BroadWorks.as1-xx.xxx.xx.xxxV5080-0-778271239-1616003581-1251392025611-. From:0XXsip:0xxx...@62.206.3.xxx;user=phone;tag=1616003581-1251392025611-. To:Mesip:026xx...@my.domain. Call-ID:bw185345611270809356816...@62.206.3.xxx. CSeq:778271239 INVITE. Contact:sip:62.206.3.xxx:5060. Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE. Accept:multipart/mixed,application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp. Supported:. Max-Forwards:20. Proxy-Authorization:DIGEST cnonce=fyvqi2pf,qop=auth,uri=sip:gw+gw_026xx...@xx.xxx.xx.xxx:5080;transport=udp,realm=my.domain,username=026xx...@my.domain,nonce=21bbe70c-932a-11de-b94d-bbade892ded3,algorithm=MD5,response=6d39a2546a4aa9a1fc39e2dc07c1e934,nc=0001. Content-Type:application/sdp. Content-Length:344. . v=0. o=BroadWorks 1271473 1 IN IP4 87.234.9.178. s=-. c=IN IP4 87.234.9.178. t=0 0. m=audio 18534 RTP/AVP 8 0 2 99 18 110. a=rtpmap:99 G726-24/8000. a=rtpmap:110 X-NSE/8000. a=fmtp:110 192-194,200-202. a=X-sqn:0. a=X-cap: 1 audio RTP/AVP 110. a=X-cpar: a=rtpmap:110 X-NSE/8000. a=X-cpar: a=fmtp:110 192-194,200-202. a=X-cap: 2 image udptl t38. ___ 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] Questions about att_xfer
Thank you all for the great job :)
Now it works as I wanted!!
Tomorrow I'll will try to update the wiki :)
But with few words ( I tried to strip the extensions to minimum for
more easy read for users).
In the dialplan I have created the extension showed below:
extension name=local_number
condition field=destination_number expression=^(\d{3})$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=bind_meta_app data=1 b s
execute_extension::attented_xfer XML features/
action application=set data=transfer_ringback=$${hold_music}/
action application=set data=call_timeout=10/
action application=set data=hangup_after_bridge=true/
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
/condition
/extension
In the features.xml file I have extension attented_xfer, that is using
the att_xfer application :
extension name=attented_xfer
condition field=destination_number expression=^attented_xfer$
action application=set data=continue_on_fail=true/
action application=read data=3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 3 #/
action application=set data=origination_cancel_key=#/
action application=att_xfer
data=user/${attxfer_callth...@${domain_name}/
/condition
/extension
Also I have set the the origination cancel key will be # - so if B
wants to cancel the call to C and to go back to A - just uses the #
key :)
Again - thank you very much :) :)
Anatoliy Kounitskiy
On Thu, Aug 27, 2009 at 7:48 PM, Anthony
Minessaleanthony.miness...@gmail.com wrote:
please retest with latest SVN trunk
On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy
anato...@kounitskiy.com wrote:
Just for information - the idea :
A---calls--- B ---att_xfer--- C , B hangs C up and goes back to A
( I`m sorry C is not answering :) )
--
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
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@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
--
Anatoliy Kounitskiy
-
E-mail: anato...@kounitskiy.com
Mobile: +359898913540
___
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] Questions about att_xfer
Thanks! I'll be in touch to help you if you need any wiki editing
assistance.
-MC
On Thu, Aug 27, 2009 at 11:18 AM, Anatoliy Kounitskiy
anato...@kounitskiy.com wrote:
Thank you all for the great job :)
Now it works as I wanted!!
Tomorrow I'll will try to update the wiki :)
But with few words ( I tried to strip the extensions to minimum for
more easy read for users).
In the dialplan I have created the extension showed below:
extension name=local_number
condition field=destination_number expression=^(\d{3})$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=bind_meta_app data=1 b s
execute_extension::attented_xfer XML features/
action application=set data=transfer_ringback=$${hold_music}/
action application=set data=call_timeout=10/
action application=set data=hangup_after_bridge=true/
action application=bridge
data=user/${dialed_extensi...@${domain_name}/
/condition
/extension
In the features.xml file I have extension attented_xfer, that is using
the att_xfer application :
extension name=attented_xfer
condition field=destination_number expression=^attented_xfer$
action application=set data=continue_on_fail=true/
action application=read data=3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 3 #/
action application=set data=origination_cancel_key=#/
action application=att_xfer
data=user/${attxfer_callth...@${domain_name}/
/condition
/extension
Also I have set the the origination cancel key will be # - so if B
wants to cancel the call to C and to go back to A - just uses the #
key :)
Again - thank you very much :) :)
Anatoliy Kounitskiy
On Thu, Aug 27, 2009 at 7:48 PM, Anthony
Minessaleanthony.miness...@gmail.com wrote:
please retest with latest SVN trunk
On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy
anato...@kounitskiy.com wrote:
Just for information - the idea :
A---calls--- B ---att_xfer--- C, B hangs C up and goes back to A
( I`m sorry C is not answering :) )
--
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
--
Anatoliy Kounitskiy
-
E-mail: anato...@kounitskiy.com
Mobile: +359898913540
___
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] Call exits after 120 seconds with hangup cause
I think the problem is the session-timeout is too long and your nat mapping is being deleted. try setting it to a smaller value like 20 or 40 sec or try setting those phones to register more frequently (every 10 or 20 sec) On Thu, Aug 27, 2009 at 1:11 PM, Dennis oderm...@googlemail.com wrote: the pastebin number is 10129 ___ 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] Call exits after 120 seconds with hangup cause
I think the problem is the session-timeout is too long and your nat mapping is being deleted. we changed the param session-timeout (in internal) to 20 and 40, without success. we changed the minimum-session-expires to 20, although we knew, the allowed minimum is 90 and 90 was shown in the console - no success. try setting it to a smaller value like 20 or 40 sec or try setting those phones to register more frequently (every 10 or 20 sec) we changed all possible options to 10 and 20, again without success :( ___ 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 exits after 120 seconds with hangup cause
If you look at your trace the call is sending a re-invite over and over and over again with no reply you need to examine your network topology and find out why the packets FS is sending to your phone never make it. also try disabling session-timers on the snom On Thu, Aug 27, 2009 at 2:09 PM, Dennis oderm...@googlemail.com wrote: I think the problem is the session-timeout is too long and your nat mapping is being deleted. we changed the param session-timeout (in internal) to 20 and 40, without success. we changed the minimum-session-expires to 20, although we knew, the allowed minimum is 90 and 90 was shown in the console - no success. try setting it to a smaller value like 20 or 40 sec or try setting those phones to register more frequently (every 10 or 20 sec) we changed all possible options to 10 and 20, again without success :( ___ 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] Can a chat message be sent to a cell phone with FS?
In a short while (for any value of short) will be available for testing mod-celliax, an interface to the cellular phones networks for voice calls and SMSs. -giovanni On 8/27/09, Merle J. Ebbert se02005-...@yahoo.com wrote: Can a chat message be sent to a cell phone with FS? Thanks, Merle ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_voicemail email template variables
you should be able to for instance put
action application=set data=test_var=this is a test/
right before the voicemail app is called
then put
${test_var} in your template.
making sure to issue reloadxml or restart FS
On Thu, Aug 27, 2009 at 1:06 PM, Nick Lemberger nick.lember...@lkfd.netwrote:
Thanks for the fast reply!
I just tried 10 random variables from
http://wiki.freeswitch.org/wiki/Channel_Variables and I only see the
whitespace where the variable should be. I've only been able to get the
ones that are set in mod_voicemail.c circa line 1600 to work.
-Nick
On 8/27/2009 at 12:44 PM, in message
191c3a030908271044k63973088xeec12c578d02e...@mail.gmail.com, Anthony
Minessale anthony.miness...@gmail.com wrote:
all variables referenced in the template should expand when sending the
email.
On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger
nick.lember...@lkfd.netwrote:
Is there a way to use dialplan variables in the email that gets sent
with
the voicemail attachement. I tried using some but nothing seems to show
up,
I'm guessing it's a different channel or something...
Any ideas?
Thanks,
-Nick
___
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
--
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] Call exits after 120 seconds with hangup cause
If you look at your trace the call is sending a re-invite over and over and over again with no reply you need to examine your network topology and find out why the packets FS is sending to your phone never make it. also try disabling session-timers on the snom are you talking about our problem, that we get a hangup after 120 seconds? i just ask to avoid, that we are talking about different things. if we connect without stun and call the 5900, we can hear the music, so i assume, that we are receiving packets from fs!? but after about 120 seconds, we receive the hangup. before everything was fine and as expected. we also tried disabling session-timers in the snom - same problem. thant's the problem, we tried soo many things, but simply nothing changes the problem or makes something different (like sending the hangup later or earlier). ___ 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 exits after 120 seconds with hangup cause
yes i am talking about that same problem. I am not talking about audio I am talking about sip messages. look at the trace you made from my earlier request You can see the same invite being sent 10 times and never got any answer. so you need to investigate it. On Thu, Aug 27, 2009 at 2:42 PM, Dennis oderm...@googlemail.com wrote: If you look at your trace the call is sending a re-invite over and over and over again with no reply you need to examine your network topology and find out why the packets FS is sending to your phone never make it. also try disabling session-timers on the snom are you talking about our problem, that we get a hangup after 120 seconds? i just ask to avoid, that we are talking about different things. if we connect without stun and call the 5900, we can hear the music, so i assume, that we are receiving packets from fs!? but after about 120 seconds, we receive the hangup. before everything was fine and as expected. we also tried disabling session-timers in the snom - same problem. thant's the problem, we tried soo many things, but simply nothing changes the problem or makes something different (like sending the hangup later or earlier). ___ 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
[Freeswitch-users] Can I stream a file to a parked call.
Hi there,
I know there are other ways of doing this. I am just trying get to
know have fun with the FreeSWITCH API.
I am using originate and park a call:
fsApi.Execute(originate,
string.Format([origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
park, blegSession.Uuid, OutgoingCallerID, NumberToDial));
That works great, the phone rings. I want to play something to this
called party when they pick up.
while (!blegSession.answered())
{
Log.WriteLine(LogLevel.Alert,
Inside::CallReturns:!Session.answered::Loop);
blegSession.sleep(500, 1);
}
string promptFile = prompts/whisper.wav;
blegSession.StreamFile(promptFile, 0);
This works - but the audio is choppy and slow.
Is there something I need to do to that parked call before streaming that file?
Thanks for any input.
Phil
___
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] Who is callie?
Mike, Sure! I am planning on doing a session soon, maybe in a couple of weeks or so. My only question is whether GM Voices license will allow this kind of thing. Do you or anyone else know? Carlos Michael Jerris wrote: Also, note, we have a few sound files we would like to add to our default set. If you are doing a session anyways and are willing please let me know if we can throw a few other prompts in your list to be recorded. Mike On Aug 27, 2009, at 12:18 PM, Michael Collins wrote: On Thu, Aug 27, 2009 at 9:04 AM, Carlos S. Antunes c...@nowthor.com mailto:c...@nowthor.com wrote: Hi! I searched the wiki but couldn't find the answer. Is callie a real woman one may ask to record additional sounds? Callie is one of the voices from GM Voices. She is definitely available for custom work. Visit www.gmvoices.com http://www.gmvoices.com/ for more info. Tell them that the FreeSWITCH project sent you. :) -MC Thanks! Carlos ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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 http://www.freeswitch.org/ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto:FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Can I stream a file to a parked call.
try wedging {ignore_early_media=true} before the first [ in your dial string
and eliminate the code waiting for answer.
On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones pjinthe...@gmail.com wrote:
Hi there,
I know there are other ways of doing this. I am just trying get to
know have fun with the FreeSWITCH API.
I am using originate and park a call:
fsApi.Execute(originate,
string.Format([origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
park, blegSession.Uuid, OutgoingCallerID, NumberToDial));
That works great, the phone rings. I want to play something to this
called party when they pick up.
while (!blegSession.answered())
{
Log.WriteLine(LogLevel.Alert,
Inside::CallReturns:!Session.answered::Loop);
blegSession.sleep(500, 1);
}
string promptFile = prompts/whisper.wav;
blegSession.StreamFile(promptFile, 0);
This works - but the audio is choppy and slow.
Is there something I need to do to that parked call before streaming that
file?
Thanks for any input.
Phil
___
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] Can a chat message be sent to a cell phone with FS?
On Thu, Aug 27, 2009 at 12:20 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote: In a short while (for any value of short) will be available for testing mod-celliax, an interface to the cellular phones networks for voice calls and SMSs. -giovanni We eagerly anticipate it's arrival! :) -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] Mixing mod_curl_xml dynamic dialplans and static ones
I have mod_curl_xml working, but I also have several static dialplans
in dialplan/public/ and I can't seem to get it to search those if my
php generated xml page does not return a result.
I know in the static public.xml there is this
extension name=unloop
condition field=${unroll_loops} expression=^true$/
condition field=${sip_looped_call} expression=^true$
action application=deflect data=${destination_number}/
/condition
/extension
Do I need my php page to generate something like that so it will
continue looking in the static pages if it does not find a match in
the xml_curl dynamic dialplan?
I don't know if I have given enough data for someone to point me in
the right direction...
This is in the console when I have mod_xml_curl enabled:
2009-08-27 16:39:52 [INFO] mod_dialplan_xml.c:252 dialplan_hunt()
Processing -15854199896 in context public
Dialplan: sofia/internal/585...@208.34.86.39 parsing [public-
curl_test] continue=false
Dialplan: sofia/internal/585...@208.34.86.39 Regex (FAIL)
[curl_test] destination_number(15854199896) =~ /^(18775844111)$/
break=on-false
2009-08-27 16:39:52 [INFO] switch_core_state_machine.c:136
switch_core_standard_on_routing() No Route, Aborting
--
Greg
___
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] Mixing mod_curl_xml dynamic dialplans and static ones
Your php should return a not found, also you can stack static and non-static
XML dialplans on your technology profile settings
For example, on your internal.xml (in the default sip profiles) the dialplan
setting can be set to ³XML,XML:/path/to/some/static.xml² . Once it processes
thru the xml_curl responses it will hit the static.xml
From: Greg Thoen gr...@cgicommunications.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Thu, 27 Aug 2009 17:13:42 -0400
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Mixing mod_curl_xml dynamic dialplans and static
ones
I have mod_curl_xml working, but I also have several static dialplans in
dialplan/public/ and I can't seem to get it to search those if my php
generated xml page does not return a result.
I know in the static public.xml there is this
extension name=unloop
condition field=${unroll_loops} expression=^true$/
condition field=${sip_looped_call} expression=^true$
action application=deflect data=${destination_number}/
/condition
/extension
Do I need my php page to generate something like that so it will continue
looking in the static pages if it does not find a match in the xml_curl
dynamic dialplan?
I don't know if I have given enough data for someone to point me in the
right direction...
This is in the console when I have mod_xml_curl enabled:
2009-08-27 16:39:52 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing
-15854199896 in context public
Dialplan: sofia/internal/585...@208.34.86.39 parsing [public-curl_test]
continue=false
Dialplan: sofia/internal/585...@208.34.86.39 Regex (FAIL) [curl_test]
destination_number(15854199896) =~ /^(18775844111)$/ break=on-false
2009-08-27 16:39:52 [INFO] switch_core_state_machine.c:136
switch_core_standard_on_routing() No Route, Aborting
--
Greg
___
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] Mixing mod_curl_xml dynamic dialplans and static ones
On Thu, Aug 27, 2009 at 2:32 PM, Ken Rice kr...@freeswitch.org wrote: Your php should return a not found, also you can stack static and non-static XML dialplans on your technology profile settings For example, on your internal.xml (in the default sip profiles) the dialplan setting can be set to “XML,XML:/path/to/some/static.xml” . Once it processes thru the xml_curl responses it will hit the static.xml Ken, That's a very cool feature. I wasn't aware of it. I will see about adding this knowledge to the wiki if it isn't there already. -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] Using a Virtual Extension
Hello, I have setup multiple FreeSWITCH servers for a conferencing application that uses "virtual" extensions as links between servers. Basically, I create a connection between the servers using an extension that does not have a live person on the channel -- so that when calls are connected on a remote server, the call joins the same conference that is linked between the servers and can participate on the call. In my code, I identify the same extension when linking between the servers for ALL conference calls. For example: Server1: Conference Call ABC starts --> Conference Name = ABC Server1: A link is created to Server2, by placing a call using extension , so there is then a conference between Server1 and Server2 without any live person on the call yet. Server1: Dials a live person that is placed in conference ABC. Server2: Dials a live person that is placed in conference ABC (Members in the conference can now hear all participants in Conference ABC on both servers). Then, Server1: Conference Call XYZ starts --> Conference Name = XYZ Server1: A link is created to Server2, using extension again. Server1: Dials a live person that is placed in conference XYZ. Server2: Dials a live person that placed in conference XYZ (Members in the conference can now hear all participants in Conference XYZ on both servers). So far this works great! However, my question is whether or not using the same extension is a bad idea? does it matter how many times I setup the link between the servers using the same extension for every conference? - 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] memory leak
Anthony can you ( or anyone else alao ). Please elaborate on what makes centos 5.3 o much better for Freeswitch. Is there some specific library vesiion on centos that makes a massive difference ? Reason I ask ... I personally only have a preference for debian, but others may have policy mandated Os's For their companies, and it would be great to have some info about this. I'd imagine this actually boils down to a requirement for libfoo version X ... And if we ran those library versions on other OS's we'd be fine ? Jay I recommend you try your application on 64 bit CentOS which is the platform all of our paid customers use. hmm. our operations-people wont be happy with that. ___ 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] Using a Virtual Extension
So far this works great! However, my question is whether or not using the same extension is a bad idea? does it matter how many times I setup the link between the servers using the same extension for every conference? - T Tina, Glad to hear your multi-machine conference thing is working. Most likely I'd say that you are okay to use the same extension number. It's really just an administrative thing. As long as people know what number to dial it should be just fine. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] memory leak
Mostly based on that is what we developed it with and continue to use and have had the best luck and least problems with. I am sure you are right about it being certain version of the core toolchain I will guess it's the combo of libc and kernel and the fact that CentOS has a purist attitude towards patching such code. Some people use stable debian with no issues, other have reported problems resolved by switching to CentOS so presumably you could roll your own debian that was similar but we don't have the resources to try to find out what makes what happen etc. If you can configure debian to work, it's fine, if we get some inexplicable problem on debian we will start to wonder though. it's happened more than once with people doing nothing other than changing OS =D On Thu, Aug 27, 2009 at 4:55 PM, Jay Binks jaybi...@gmail.com wrote: Anthony can you ( or anyone else alao ). Please elaborate on what makes centos 5.3 o much better for Freeswitch. Is there some specific library vesiion on centos that makes a massive difference ? Reason I ask ... I personally only have a preference for debian, but others may have policy mandated Os's For their companies, and it would be great to have some info about this. I'd imagine this actually boils down to a requirement for libfoo version X ... And if we ran those library versions on other OS's we'd be fine ? Jay I recommend you try your application on 64 bit CentOS which is the platform all of our paid customers use. hmm. our operations-people wont be happy with that. ___ 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] memory leak
Jay Binks jaybi...@gmail.com wrote: Reason I ask ... I personally only have a preference for debian, but others may have policy mandated Os's For their companies, and it would be great to have some info about this. The only problem I've had with FreeSWITCH under Debian Squeeze and Sid involves TLS-related segmentation faults that appear to be related to something in the version of libssl supplied with Debian. The same problem can't be reproduced on Fedora, for example, but it does occur under Debian Lenny as well as Debian Squeeze and Sid (i.e., testing and unstable, respectively). besides this, all of the issues that I have encountered turned out to be (usually short-lived) bugs in FreeSWITCH or one of the libraries included in the source tree - they're mostly FreeSWITCH issues. I should point out that the FreeSWITCH developers are very good at avoiding the introduction of bugs into their code and that known bugs get fixed. It appears to be an unchanging fact about programming that with a large and complex project, even given highly knowledgeable, experienced, committed and talented developers (as we have with FreeSWITCH), sometimes, bugs do slip through. ___ 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] How to make a call back
No. In the dial plan I said, application=perl data=The perl script. I also checked $session-execute(bridge,user/1010). This is working fine. But originate is not working as I expected. On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins m...@freeswitch.org wrote: On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan lakindi...@gmail.com wrote: When I give the following from the command line it calls to 1010 extension and once answered, it calls to 1000 and bridge the connection. originate user/1010 bridge(user/1000) But I want to do this in perl. So I have given as follows $session-originate($session,user/1010 bridge user/1000); But it is not working. It says user/1010 bridge user/1000 is invalid user. How to do this in perl. pls help. Are you calling this perl script from the CLI? If so you won't have the $session object because a channel does not exist for a simple API call. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Authorizations when using DNS SRV bug?
Brian, You've been vindicated. Callcentric is now advertising zero weighted SRV records! :) I've re-enabled SRV lookups for the Callcentric profile and will monitor to see if I get any errors. Carlos Brian West wrote: Or as I have argued today they should fix their SRV records to be zero weighted. /b On Aug 20, 2009, at 1:36 PM, Michael Jerris wrote: You can bypass the srv records if you like by passing a :port with the hostname where you use it in freeswitch. ___ 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
