[Freeswitch-users] Sofia SIP external profile gateway names
Hi, I am facing a problem with Sofia SIP external profile. Basically i have 10+ accounts, say a101 - a110, to a single service provider, say xyz.com. For each account a have created a gateway in external profile's directory i.e. /usr/local/freeswitch/conf/sip_profiles/external/a101.xml up to a110.xml. At first the problem was none of the accounts were registering since, FS was trying to send registration requests to host a101 instead of xyz.com, but later when i set xyz.com as proxy and register proxy address, it successfully registered all accounts. Now i am facing a similar problem in dialplan, for example if i try to dialout via gateway a101, call immediately fails with NORMAL_TEMPORARY_FAILURE. When i trun on sip tracing i don't see any INVITE message sent to provider xyz.com. 2009-10-22 06:51:55.325393 [NOTICE] switch_channel.c:613 New Channel sofia/external/00923344224...@a101 [b7663caf-cd29-4213-9059-ae880d49b0ca] 2009-10-22 06:51:55.516382 [NOTICE] sofia.c:4039 Hangup sofia/external/00923344224...@a101 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] API CALL [originate(sofia/external/00923344224...@a101 )] output: -ERR NORMAL_TEMPORARY_FAILURE 2009-10-22 06:51:55.527381 [NOTICE] switch_core_session.c:1132 Session 2 (sofia/external/00923344224...@a101) Ended 2009-10-22 06:51:55.527381 [NOTICE] switch_core_session.c:1134 Close Channel sofia/external/00923344224...@a101 [CS_DESTROY] I think its trying to look up a101 instead of xyz.com to send INVITE. Kindly help. Thank you. -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Some help with my post-paid billing project
://www.freeswitch.org -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Registering a large number of SIP users
Hi, I am creating a load test setup for FreeSWITCH using Sofia SIP. I have two machines both configured with mod_xml_curl, one machine (lets call it SIP Server) has 100 SIP accounts. Now i want to register second machine (lets call it SIP Client) to all these 100 SIP accounts on first machine. How can i do that? One approach that i can think of is that to create a new profile (or use existing external profile) on SIP Client and add all accounts to it as gateways having, *param name=register value=true* define in their configuration. Is my approach correct? Are there any better ways to do this? Thank you. -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Registering a large number of SIP users
Yes i have used this tool before, but sip registration is just first part of my load test, i will be make SIP call load testing too. Also, i want to use FreeSWITCH against FreeSWITCH to test its capability both as SIP Server and SIP Client. Thank you. On Tue, Oct 13, 2009 at 6:21 PM, Ryanny Lin ryan...@gmail.com wrote: You may try this tool, sipp, to execute a load test. http://sipp.sourceforge.net/ 2009/10/13 Muhammad Shahzad shaherya...@googlemail.com Hi, I am creating a load test setup for FreeSWITCH using Sofia SIP. I have two machines both configured with mod_xml_curl, one machine (lets call it SIP Server) has 100 SIP accounts. Now i want to register second machine (lets call it SIP Client) to all these 100 SIP accounts on first machine. How can i do that? One approach that i can think of is that to create a new profile (or use existing external profile) on SIP Client and add all accounts to it as gateways having, *param name=register value=true* define in their configuration. Is my approach correct? Are there any better ways to do this? Thank you. -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sincerely regards, Wen-Jen ___ 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 -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup, Agenda For This Week
Great. Just added it. Is there any user limit on this? Thank you. On Wed, Oct 7, 2009 at 6:15 PM, Chris Chen chris.chen2...@gmail.com wrote: Hi Muhammad, the simple and reliable solution for you where SIP is being blocked is add conf+...@conference.freeswitch.orgconf%2b...@conference.freeswitch.orgto your Goolgetalk buddy list, and you can call from there to join the conference, simple and straightforward. Chris On Wed, Oct 7, 2009 at 12:24 AM, Muhammad Shahzad shaherya...@googlemail.com wrote: I got a lot of problem last week for making conference call. I was at home (conference call starts at 2200hours PKST, my time) and unable to make SIP call since the government has blocked it. So my only choice was Skype, but unfortunately DTMF wasn't working, i get connected on Skypiax5 for about 7-8 times, but DTMF problem never let me get in to conference. Is there any solution for this? I really need to discuss a lot of things about FS documentation in conference, like what FS community can expect from it and what not? how i have planned it? how much progress has done? what are the problems me and my team are facing (which has slow us down considerably)? etc. etc. I have a solution for it but that needs testing. The plan is to use one of my FS servers to connect my jingle calls from GTalk to conference server over SIP. How can i test this setup with conference server, any ideas? If any one else also interested in getting connected to weekly conference call through this setup then i can also extend this setup as needed. Thank you. On Tue, Oct 6, 2009 at 1:30 AM, Brian West br...@freeswitch.org wrote: It always supported 48kHz CELT but the conference itself was running at 32kHz so everyone 48k had to be down sampled. Now you all get to be up sampled. w00t! /b On Oct 5, 2009, at 2:03 PM, Michael Collins wrote: * Starting with the upcoming meeting (Oct 9) the conference will support 48kHz CELT codec. ___ 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 -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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 -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH Weekly Conference Call Followup, Agenda For This Week
I got a lot of problem last week for making conference call. I was at home (conference call starts at 2200hours PKST, my time) and unable to make SIP call since the government has blocked it. So my only choice was Skype, but unfortunately DTMF wasn't working, i get connected on Skypiax5 for about 7-8 times, but DTMF problem never let me get in to conference. Is there any solution for this? I really need to discuss a lot of things about FS documentation in conference, like what FS community can expect from it and what not? how i have planned it? how much progress has done? what are the problems me and my team are facing (which has slow us down considerably)? etc. etc. I have a solution for it but that needs testing. The plan is to use one of my FS servers to connect my jingle calls from GTalk to conference server over SIP. How can i test this setup with conference server, any ideas? If any one else also interested in getting connected to weekly conference call through this setup then i can also extend this setup as needed. Thank you. On Tue, Oct 6, 2009 at 1:30 AM, Brian West br...@freeswitch.org wrote: It always supported 48kHz CELT but the conference itself was running at 32kHz so everyone 48k had to be down sampled. Now you all get to be up sampled. w00t! /b On Oct 5, 2009, at 2:03 PM, Michael Collins wrote: * Starting with the upcoming meeting (Oct 9) the conference will support 48kHz CELT codec. ___ 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 -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!
Yes, i had same problem, then i changed stun server to one of our own servers. You may try some of public stun servers listed on below link, http://www.voip-info.org/wiki/view/STUN Thank you. On Fri, Oct 2, 2009 at 2:58 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Anyone have this issue? On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Hi! I have just started to use dingaling again, and noticed I constantly get a stun error. 2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed! stun.fwdnet.net:3478 [Remote Address Error!] I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers and keep getting this error with dingaling. I have no problems with inbound sip calls, so I don't think its the actual stun server. Has anyone else seen this? I am using: FreeSWITCH Version 1.0.trunk (14952) 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 -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH Start up Script - Ubuntu Server
Glad to hear about init script for archlinux, my favourite distro for
development. :-)
Let me try it.
Thank you.
On Mon, Sep 28, 2009 at 6:46 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Yep.. it works for me. You will probably have to modify these lines
to match the user/group that FS normally run user on your system:
FS_USER=freeswitch
FS_GROUP=freeswitch
On Mon, Sep 28, 2009 at 10:37 AM, Aloysius Thevarajah Lloyd
lloyd.aloys...@gmail.com wrote:
I try earlier today this script ... but it is not working. Did you try ?
On Sun, Sep 27, 2009 at 8:12 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
What about this one for Debian...
http://wiki.freeswitch.org/wiki/Freeswitch_init
On Mon, Sep 28, 2009 at 9:15 AM, Aloysius Thevarajah Lloyd
lloyd.aloys...@gmail.com wrote:
Yes. I have seen the scripts. But I could not find a suitable one for
Ubuntu.
Thank you.
LLoyd
2009/9/27 João Mesquita jmesqu...@freeswitch.org
Only 3 init scripts available on trunk today (${SVNROOT}/build) are
for
archlinux, redhat or suse.
We would love to have more for other distros.
Regards,
jmesquita
On Sun, Sep 27, 2009 at 10:42 AM, Aloysius Thevarajah Lloyd
lloyd.aloys...@gmail.com wrote:
Hi All,
I am trying to setup FreeSwitch on a Ubuntu Server.
Where can I find the start up(boot time) script for FreeSwitch on a
Ubuntu Server?
Thank you .
Lloyd
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--
|
|
| FATAL ERROR ---
O X |
|___|
|You have moved the mouse.
|
| Windows must be restarted for the changes to take effect. |
|OK
|
/
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
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Re: [Freeswitch-users] Using Sofia-SIP with ICE NAT Traversal Mechanism
No mod_dingaling does not use LibNICE. However, i have plans to integrate NICE with Sofia in mod_msn project, which is at the moment moving with very slow pace due to some trouble in reverse engineering MSNP-18 protocol (used in Windows Live Messenger 2009). Thank you. On Wed, Sep 23, 2009 at 6:37 PM, afshin afzali a.afzali2...@gmail.comwrote: Hi, I know that FreeSWITCH uses libdingaling to talk to Jingle call parties. Also I know that Jingle Protocol uses ICE protocol to traverse NAT devices. So : does libdingaling use an open library such as libnice for ICE? Is it possible to use the ICE implementation in Sofia-SIP endpoint? If not, how could I integrate an open ICE library in Sofia-SIP? Regards, -- afshin ___ 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 -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Using Sofia-SIP with ICE NAT Traversal Mechanism
Yup, that's a good idea but not in my project list right now. Thank you. On Wed, Sep 23, 2009 at 7:17 PM, afshin afzali a.afzali2...@gmail.comwrote: Don't you think is better to integrate LibNICE to FreeSWITCH's RTP stack than a module such as mod_msn / mod_dingaling ? -- afshin On Wed, Sep 23, 2009 at 4:21 PM, Muhammad Shahzad shaherya...@googlemail.com wrote: No mod_dingaling does not use LibNICE. However, i have plans to integrate NICE with Sofia in mod_msn project, which is at the moment moving with very slow pace due to some trouble in reverse engineering MSNP-18 protocol (used in Windows Live Messenger 2009). Thank you. On Wed, Sep 23, 2009 at 6:37 PM, afshin afzali a.afzali2...@gmail.comwrote: Hi, I know that FreeSWITCH uses libdingaling to talk to Jingle call parties. Also I know that Jingle Protocol uses ICE protocol to traverse NAT devices. So : does libdingaling use an open library such as libnice for ICE? Is it possible to use the ICE implementation in Sofia-SIP endpoint? If not, how could I integrate an open ICE library in Sofia-SIP? Regards, -- afshin ___ 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 -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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 -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Using Sofia-SIP with ICE NAT Traversal Mechanism
Personally i am not a fan of GLib as well and always prefer STL over it due to so many good reasons. But on the other hand libnice is the only library that has Microsoft extensions to ICE protocol, which are required for mod_msn to work. So far on mod_msn, i am able to send and receive voice call requests to / from WLM 2009, but upon answer two way voice is not working. I am able to hear voice from WML 2009 but WML 2009 client can't hear anything. So nowadays i am reviewing libNICE to fix this problem. If this does not work (and so far it does not seems to work) then i would like to use FS ICE library, provided that you guys allow me to extend it to support Microsoft extensions..! Thank you. On Wed, Sep 23, 2009 at 8:40 PM, Michael Jerris m...@jerris.com wrote: We already have ice support in freeswitch, granted it is the slightly twisted ice from the old jingle, but this should not be difficult to fix. Knowing what I know about libnice architechture I can say almost without doubt that it will never fit well into freeeswitch. Is the basis of this question and you loooking for an ice library on the sofia list just to support ice in sip? If so, for both sip and msn the path of least resistance and probably the only way that would work would be to address this within our existing ice implementation. Mike On Sep 23, 2009, at 10:14 AM, Brian West br...@freeswitch.org wrote: I'm not comfortable adding libnice into FreeSWITCH as it depends on glib and that would add bloat in my opinion... is there no other license compatible option? /b On Sep 23, 2009, at 8:42 AM, Muhammad Shahzad wrote: Yup, that's a good idea but not in my project list right now. Thank you. On Wed, Sep 23, 2009 at 7:17 PM, afshin afzali a.afzali2...@gmail.com a.afzali2...@gmail.com wrote: Don't you think is better to integrate LibNICE to FreeSWITCH's RTP stack than a module such as mod_msn / mod_dingaling ? -- afshin On Wed, Sep 23, 2009 at 4:21 PM, Muhammad Shahzad shaherya...@googlemail.com shaherya...@googlemail.com wrote: No mod_dingaling does not use LibNICE. However, i have plans to integrate NICE with Sofia in mod_msn project, which is at the moment moving with very slow pace due to some trouble in reverse engineering MSNP-18 protocol (used in Windows Live Messenger 2009). Thank you. On Wed, Sep 23, 2009 at 6:37 PM, afshin afzali a.afzali2...@gmail.com a.afzali2...@gmail.com wrote: Hi, I know that FreeSWITCH uses libdingaling to talk to Jingle call parties. Also I know that Jingle Protocol uses ICE protocol to traverse NAT devices. So : does libdingaling use an open library such as libnice for ICE? Is it possible to use the ICE implementation in Sofia-SIP endpoint? If not, how could I integrate an open ICE library in Sofia-SIP? Regards, -- afshin ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org/http://www.freeswitch.org -- | | | FATAL ERROR --- O X | |___| |You have moved the mouse. | | Windows must be restarted for the changes to take effect. | |OK | / Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.comshari_78...@hotmail.com Email: shaherya...@googlemail.comshaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 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 FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org/http://www.freeswitch.org
Re: [Freeswitch-users] Call Tracing
In that case you should turn on sip trace for profile where your callcentric
peer is configured. By default FS comes with two profiles namely internal
and external. If you haven't created any new profile and configured your
users and peers in these two profiles then you should try turning on sip
trace for external profile too (or just external profile alone).
*sofia profile external siptrace on*
Please check your peer configuration and turn on sip trace on appropriate
profile.
Thank you.
On Sun, Sep 20, 2009 at 5:49 PM, Klaus Teller klaus.tel...@gmx.net wrote:
Thanks. I tried that and what it shows me is the trace between my peer and
the SIP provider (i.e. les.net). The call is actually coming from
callcentric and i don't see that in the trace. Is it supposed to show this?
Klaus.
Original-Nachricht
Datum: Sun, 20 Sep 2009 17:11:50 +0600
Von: Muhammad Shahzad shaherya...@googlemail.com
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Call Tracing
there are a few variable that you can set in
/usr/local/freeswitch/conf/vars.xml.
* X-PRE-PROCESS cmd=set data=call_debug=false/
X-PRE-PROCESS cmd=set data=console_loglevel=info/
*
You can change it to something like (and then restart FS),
* X-PRE-PROCESS cmd=set data=call_debug=true/
X-PRE-PROCESS cmd=set data=console_loglevel=debug/
*
Usually it will give you enough information about call processing,
however
just in case you are looking for SIP trace of a call only then you can
enable it on per-profile basis at run-time,
for example,
*sofia profile internal siptrace on*
this will enable SIP trace for all calls to / from sofia internal profile
(which also includes directory users).
You can run following command on FS console to get information on what
profile etc. are available as well as their status.
*sofia status*
For more info consult Wiki page at,
http://wiki.freeswitch.org/wiki/Sofia
Thank you.
On Sun, Sep 20, 2009 at 4:44 PM, Klaus Teller klaus.tel...@gmx.net
wrote:
Hi T.,
I just tried that but i don't see anything different on the console. My
test call is going via callcentric and les.net, but besides the final
hop
which i normally see in the channel name, there is nothing else.
Any idea what i might be doing wrong here?
Thanks,
Klaus.
Original-Nachricht
Datum: Sun, 20 Sep 2009 10:33:01 +0200
Von: Tihomir Culjaga tculj...@gmail.com
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Call Tracing
switch.conf.xml (btw: in console you can enable/disable logging on
the
fly
-
F8/F7)
param name=loglevel value=debug/
your relevant sip profile:
param name=sip-trace value=yes/
T.
On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller klaus.tel...@gmx.net
wrote:
Hi,
Say i have an inbound VoIP/SIP call that hits my FS box. Is it
possible
to
to extract information about the intermediate hops that the call or
the
signaling went through? If so, what information can i get?
Thanks,
Gregoire.
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---
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Cell: +92 334
Re: [Freeswitch-users] Any FreeSWITCH training courses out there?
With help from Pakistan Software Export Board (PSEB), we formed Asterisk Pakistan community forum in early 2008. This forum is still active and we arranged many workshops during last 18 months in all major cities of Pakistan. It was a great success and we effectively introduced Asterisk in so many government and private sectors. FreeSWITCH is very new in Pakistan and a very few people have heard its name here right now. So, we (me and some of my friends from Pakistan Open Source Software Foundation) are trying to develop some skilled personals for FreeSWITCH, before we approach Ministry of Information Technology to launch a campaign similar to Asterisk Pakistan Forum for FreeSWITCH. So, that if our proposal gets approval we would have enough resources to execute workshops all over Pakistan for FS training. All people in this mailing list (especially Pakistanis) who are interested in this, may contact me off list for participation and coordination in these efforts. The goal is to secure greatest share for Pakistan in this newly emerging technology and its benefits. Thank you. On Mon, Sep 21, 2009 at 9:53 AM, Mitul Limbani mi...@enterux.com wrote: Gavin, Sorry for the earlier mail, I can see that you mentioned Asterisk to Freeswitch course, we have pretty much under gone the same cycle and have put that as the part of our training course, it's named: FreeSWITCH for AstMasters Please do get in touch off the list, also if anyone else is interested in this course do get in touch with me. Thanks Regards, Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 21-Sep-2009, at 1:17 AM, Gavin Henry gavin.he...@gmail.com wrote: Hi all, Is there anyone out there doing beginner courses or conversion courses from an Asterisk mindset? Cheers. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] User Creation with DB in Freeswitch
Yes, XML CURL is really cool. I once bombed FS with 12,000 wrong registrations (bad username or password) in less then 50 seconds (49496 ms to be exact) and it processed all of them and gave correct responses using XML CURL. I am willing to do this test again soon, with correct registration data this time, to see how many registration Sofia SIP module configured with XML CURL module can handle at a time. Thank you. On Mon, Sep 21, 2009 at 3:42 PM, Brian West br...@freeswitch.org wrote: You can't put the users directly into a db with FreeSWITCH you'll have to serve up the XML document via XML CURL or write your own module to do so via the module interfaces provided. /b On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: Yes use odbc in fs Thanks Regards,Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] User Creation with DB in Freeswitch
I searched my sent emails and found the results, copying it below (after removing some sensitive info), 1,000 Calls == Total 1000 REGISTER calls sent in 890 ms at rate of 1123/sec Total 1000 responses receieved in 4516 ms at rate of 221/sec: Detailed responses received: - 403 responses: 1000 (Forbidden) -- TOTAL responses: 1000 (rate=221/sec) Maximum outstanding job: 894 Peak memory size: 15MB 5,000 Calls == Total 5000 REGISTER calls sent in 28539 ms at rate of 175/sec Total 5000 responses receieved in 36398 ms at rate of 137/sec: Detailed responses received: - 403 responses: 5000 (Forbidden) -- TOTAL responses: 5000 (rate=137/sec) Maximum outstanding job: 1001 Peak memory size: 63MB 10,000 Calls == Total 1 REGISTER calls sent in 60741 ms at rate of 164/sec Total 9289 responses receieved in 62740 ms at rate of 148/sec: Detailed responses received: - 403 responses: 9289 (Forbidden) -- TOTAL responses: 9289 (rate=148/sec) Maximum outstanding job: 1047 Peak memory size: 78MB 12,000 Calls == Total 12000 REGISTER calls sent in 49496 ms at rate of 242/sec Total 12314 responses receieved in 60582 ms at rate of 203/sec: Detailed responses received: - 403 responses:12314 (Forbidden) -- TOTAL responses:12314 (rate=203/sec) Maximum outstanding job: 1018 Peak memory size: 143MB So, FS doesn't crash even on 12,000 bad registrations (600 regs per second). I did tweak its configurations a little however no change was made to source code to make this happen. :-) Thank you. On Mon, Sep 21, 2009 at 4:07 PM, Muhammad Shahzad shaherya...@googlemail.com wrote: Yes, XML CURL is really cool. I once bombed FS with 12,000 wrong registrations (bad username or password) in less then 50 seconds (49496 ms to be exact) and it processed all of them and gave correct responses using XML CURL. I am willing to do this test again soon, with correct registration data this time, to see how many registration Sofia SIP module configured with XML CURL module can handle at a time. Thank you. On Mon, Sep 21, 2009 at 3:42 PM, Brian West br...@freeswitch.org wrote: You can't put the users directly into a db with FreeSWITCH you'll have to serve up the XML document via XML CURL or write your own module to do so via the module interfaces provided. /b On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: Yes use odbc in fs Thanks Regards,Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call Tracing
there are a few variable that you can set in /usr/local/freeswitch/conf/vars.xml. * X-PRE-PROCESS cmd=set data=call_debug=false/ X-PRE-PROCESS cmd=set data=console_loglevel=info/ * You can change it to something like (and then restart FS), * X-PRE-PROCESS cmd=set data=call_debug=true/ X-PRE-PROCESS cmd=set data=console_loglevel=debug/ * Usually it will give you enough information about call processing, however just in case you are looking for SIP trace of a call only then you can enable it on per-profile basis at run-time, for example, *sofia profile internal siptrace on* this will enable SIP trace for all calls to / from sofia internal profile (which also includes directory users). You can run following command on FS console to get information on what profile etc. are available as well as their status. *sofia status* For more info consult Wiki page at, http://wiki.freeswitch.org/wiki/Sofia Thank you. On Sun, Sep 20, 2009 at 4:44 PM, Klaus Teller klaus.tel...@gmx.net wrote: Hi T., I just tried that but i don't see anything different on the console. My test call is going via callcentric and les.net, but besides the final hop which i normally see in the channel name, there is nothing else. Any idea what i might be doing wrong here? Thanks, Klaus. Original-Nachricht Datum: Sun, 20 Sep 2009 10:33:01 +0200 Von: Tihomir Culjaga tculj...@gmail.com An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] Call Tracing switch.conf.xml (btw: in console you can enable/disable logging on the fly - F8/F7) param name=loglevel value=debug/ your relevant sip profile: param name=sip-trace value=yes/ T. On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller klaus.tel...@gmx.net wrote: Hi, Say i have an inbound VoIP/SIP call that hits my FS box. Is it possible to to extract information about the intermediate hops that the call or the signaling went through? If so, what information can i get? Thanks, Gregoire. -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser ___ 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 -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] FreeSWITCH Documentation
Hi, I have observed that one of the major hurdle while writing patches and / or bug fixes is lack of doxygen documentation for FS source code. For example it took me 5+ days to understand mod_dingaling code and its hooks into FS source code to write up soft reload patch, while it could have taken less then 3 days to do so if source code documentation was available. So, since right now i have some human resources including myself available, I would like to document all source code (or at least core FS code i.e. everything that has switch_ prefix) using doxygen. I know its a huge task and will take a while to complete but at least lets get it started. If anyone else wants to participate as well in this task, then we can team up to complete it quickly. Let me know if you guys are interested. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
I am also available for FS configuration on various Linux distributions and Wiki / documentation. Thank you. On Wed, Sep 16, 2009 at 11:44 PM, Diego Toro dft...@yahoo.com wrote: Hi, count on me for testing and answering questions on Windows and spanish support. Diego http://lacarretade.blogspot.com/ --- On *Wed, 9/16/09, Michael Collins m...@freeswitch.org* wrote: From: Michael Collins m...@freeswitch.org Subject: Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects To: freeswitch-users@lists.freeswitch.org Date: Wednesday, September 16, 2009, 9:56 AM On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mahé t.m...@telemaque.frhttp://us.mc335.mail.yahoo.com/mc/compose?to=t.m...@telemaque.fr wrote: Hi, Count on me for answering questions on IRC when I'm in, and for subprojects I'm in too as you know ;) Merci! Okay, what's your IRC nick and when are you generally on line? Also, I'm pretty sure that you're fluent in French which is good because we need more multilingual people out there. Last question: what are your areas of expertise? I'd like to keep a list of people and what they're good at so we know whom to ask first when questions come up. Thanks again! -MC -Inline Attachment Follows- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orghttp://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-us...@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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_dingaling: dl_login command syntax
actually, mod_dingaling is not reading configuration from xml_curl unless we reload mod_dingaling, which obviously fails if dingaling profile is in call etc. So, i am writing a patch right now to enable this functionality, almost finished just to perfect some memory management things. Thank you. On Fri, Sep 11, 2009 at 10:27 PM, Michael Jerris m...@jerris.com wrote: What errors do you get? Mike On Sep 4, 2009, at 8:35 AM, Muhammad Shahzad wrote: Hi, i am have FS SVN revision 14760, i am trying to use mod_xml_curl against mod_dingaling. When i call xml_curl url in browser i get mod_dingaling configuration correctly, also when i do reload mod_dingaling it fetches its configuration from xml_curl correctly. BUT when i try to use dl_login command to login a jingle profile it does not work. I have tried both syntax, Syntax 1: === dl_login profile=abcd Where abcd is a valid jingle profile fetch-able from xml_curl. Syntax 2: === dl_login name=abcd;login= x...@gmail.com/talk;pass=YYY;dialplan=XML;context=public;rtp-ip=auto;sasl=plain;tls=true;exten=1001 All these values are correct and work if i reload mod_dingaling but they don't work with dl_login, and give following output. USAGE: Existing Profile: dl_login profile=profile_name Dynamic Profile: dl_login var1=val1;var2=val2;varN=valN I don't think xml_curl has any role in this syntax. Can you please correct me if i am doing something wrong in here or is it a bug in mod_dingaling. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_dingaling: dl_login command syntax
sure, i have a full QA department who will take case of all possible cases. Then it can be tested by our community. Thank you. On Fri, Sep 11, 2009 at 11:51 PM, Brian West br...@freeswitch.org wrote: Also for tests make sure you fuzz test it also .. giving it invalid data shouldn't crash ... so try that when you're done too. /b On Sep 11, 2009, at 12:18 PM, Muhammad Shahzad wrote: actually, mod_dingaling is not reading configuration from xml_curl unless we reload mod_dingaling, which obviously fails if dingaling profile is in call etc. So, i am writing a patch right now to enable this functionality, almost finished just to perfect some memory management things. 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_dingaling: dl_login command syntax
great, can you share it with me? Thank you. On Sat, Sep 12, 2009 at 2:16 AM, Brian West br...@freeswitch.org wrote: Kewl I have a fuzz test I do also thats automated that throws all kinds of crazy stuff at all the api's. /b On Sep 11, 2009, at 2:52 PM, Muhammad Shahzad wrote: sure, i have a full QA department who will take case of all possible cases. Then it can be tested by our community. 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] XML Dial Plan vs Language Modules
Hi, I couple of my team members are working on translating a very long Asterisk Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link below, http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables The dial plan variables are not getting initialized as expected. I was just wondering if we move this variable get and set stuff to any language module say mod_perl, will that make any difference performance wise? I mean we will be invoking a Perl interpreter for each incoming call, won't that be expensive in terms of RAM and CPU usage and thus reducing number of calls this FS deployment can handle? I have guys with programming skills in Perl, PHP, Python, Java and LUA languages. Which language do you recommend for this, again in terms of speed and performance? Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] XML Dial Plan vs Language Modules
Thank you so much. Of course we are not doing a blind translation, but at the very basic we will need to get and set certain variable at different stage of call processing. Another question in same context, Can we do post-hangup call processing? I mean like in Asterisk, we have extension h which is called after hangup. Can you guide a bit how to do it in FS? Does FS has any such special extensions? Thank you. On Fri, Sep 4, 2009 at 12:06 PM, Michael Collins m...@freeswitch.org wrote: On Thu, Sep 3, 2009 at 10:59 PM, Muhammad Shahzad shaherya...@googlemail.com wrote: Hi, I couple of my team members are working on translating a very long Asterisk Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link below, Before you go through all the trouble of translating the dialplan be sure to review the application itself. In many cases just doing a dialplan translation results in less efficient use of FreeSWITCH's powerful features. Be sure that you are looking at the way FreeSWITCH handles various situations and take advantage of its power and ease of use. http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables The dial plan variables are not getting initialized as expected. I was just wondering if we move this variable get and set stuff to any language module say mod_perl, will that make any difference performance wise? I mean we will be invoking a Perl interpreter for each incoming call, won't that be expensive in terms of RAM and CPU usage and thus reducing number of calls this FS deployment can handle? I have guys with programming skills in Perl, PHP, Python, Java and LUA languages. Which language do you recommend for this, again in terms of speed and performance? Lua is very portable and we've done tests with hundreds of concurrent Lua scripts running. The other languages are heavier but they'll still handle quite a few concurrent sessions. Just be sure that you don't do the bridge app right in the script, use transfer instead and have the dialplan process any bridging that you need to do. -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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] mod_dingaling: dl_login command syntax
Hi, i am have FS SVN revision 14760, i am trying to use mod_xml_curl against mod_dingaling. When i call xml_curl url in browser i get mod_dingaling configuration correctly, also when i do reload mod_dingaling it fetches its configuration from xml_curl correctly. BUT when i try to use dl_login command to login a jingle profile it does not work. I have tried both syntax, Syntax 1: === dl_login profile=abcd Where abcd is a valid jingle profile fetch-able from xml_curl. Syntax 2: === dl_login name=abcd;login= x...@gmail.com/talk;pass=YYY;dialplan=XML;context=public;rtp-ip=auto;sasl=plain;tls=true;exten=1001 All these values are correct and work if i reload mod_dingaling but they don't work with dl_login, and give following output. USAGE: Existing Profile: dl_login profile=profile_name Dynamic Profile: dl_login var1=val1;var2=val2;varN=valN I don't think xml_curl has any role in this syntax. Can you please correct me if i am doing something wrong in here or is it a bug in mod_dingaling. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS performance under windows
If you want to try FS on Windows only for feature testing etc. then its okay, however for production deployments (that includes load testing) i strongly recommend CentOS 5.x. As far as configuration migration is concerned, you don't need to change any configuration files, simply copy them to Linux installation. Thank you. On Tue, Sep 1, 2009 at 2:13 PM, Dmitry Kadantsev kadantse...@gmail.comwrote: Hi folk, First of all, thank you for FS - really strong project. I have already asked this once in other thread but didn't got any answer. So, I'll try to re-ask. We are playing currently with FS under Windows 2008 64bit. So far there are some issues but I hope we'll solve it in nearest future. After FS will be configured correctly we plan to play with performance things on FS. The question is: Does it makes any sense to try to setup FS under Win for a same performance level possible under Linux (e.g. CentOs)? Or it's just wasting of time? An additional question is: Are there any important and well know issues during migration from Win to Lin. Or it is just like copying of all configs into Linux installation? Thank you -- Best regards, Dmitry Kadantsev ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] SIP codec preference order
Hi, I have a FS gateway (SVN revision 14537) that is is receiving SIP calls from different source gateways and sending it to one single destination gateway. Now each source gateway can talk in one specific codec and FS itself is not doing any transcoding. So i enabled all possible codecs that this FS may receive from source gateways. The problem is that the source gateways who are talking in codec that are in first three preferred codecs list in sip profile are working fine, while the codecs that are at 4th or greater preference order number do not work. The call is received and accepted by destination gateway then it gets terminated almost immediately. Do note that destination gateway is not FS, its some CISCO device that is also accepting all possible codecs. Can you guys suggest why it is happening and what are the possible solutions, other then transcode of course. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP codec preference order
i just upgraded it to 14599 and its working fine now. Thank you. On Sun, Aug 23, 2009 at 2:07 AM, Brian West br...@freeswitch.org wrote: Can you provide a little bit of log detail? /b On Aug 22, 2009, at 3:53 PM, Muhammad Shahzad wrote: Can you guys suggest why it is happening and what are the possible solutions, other then transcode of course. ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Is there any freeswitch show version command
Hi, How can we check the version / svn revision of a running FS instance? I know, this is kind of a stupid question, but i sometimes run into situation where i don't know or don't have access to FS source, nor i can restart it to get its version string. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is there any freeswitch show version command
thanks. On Mon, Aug 17, 2009 at 2:35 AM, Mathieu Rene mrene_li...@avgs.ca wrote: Hi, type: version at the CLI, it'll tell you. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 16-Aug-09, at 4:33 PM, Muhammad Shahzad wrote: Hi, How can we check the version / svn revision of a running FS instance? I know, this is kind of a stupid question, but i sometimes run into situation where i don't know or don't have access to FS source, nor i can restart it to get its version string. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] if using centos you should read this
CentOS has been a trusted platfrom for me from last 3+ years. I have developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS boxes, and 27 out of 49 Asterisk box are still ruining on CentOS in production environment. I really wish and hope this great project continues. I don't know any of its developers personally but i am quite sure they will resolve their differences professionally and put this project back on track. Thank you. On Fri, Jul 31, 2009 at 12:47 AM, Brian West br...@freeswitch.org wrote: You're just trying to introduce FUD CentOS is an open source project and it will carry on in Lance's absence. I know Russ personally and I don't think its going to end like this. /b On Jul 30, 2009, at 1:39 PM, Saji Honey wrote: If you are using centOS for your Freeswitch installation, you should probable read the article on planet.centos.org and the www.centos.org the open letter to “Lance Davis” one of the founders of centOS. CONFIDENTIAL NOTICE : If you have received this email in error, please immediately notify the sender by email at the address shown above. This email may contain confidential or legally privileged information that is intended only for the use of the individual or entity named in this email. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution or reliance upon the contents of this email is strictly prohibited. Please delete from your files if you are not the intended recipient. Thank you for your compliance. ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] if using centos you should read this
Please read my email as, CentOS has been a trusted platfrom for me from last 3+ years. I have developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS boxes, and 27 out of 49 Asterisk box are still *running* on CentOS in production environment. I really wish and hope this great project continues. I don't know any of its developers personally but i am quite sure they will resolve their differences professionally and put this project back on track. This damn Google Spell made meaning of my entire post the possite. ;-( Thank you. On Fri, Jul 31, 2009 at 11:21 AM, Michael Collins m...@freeswitch.orgwrote: On Thu, Jul 30, 2009 at 9:57 PM, Muhammad Shahzad shaherya...@googlemail.com wrote: CentOS has been a trusted platfrom for me from last 3+ years. I have developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS boxes, and 27 out of 49 Asterisk box are still ruining on CentOS in production environment. I really wish and hope this great project continues. I don't know any of its developers personally but i am quite sure they will resolve their differences professionally and put this project back on track. The guys doing the work have vowed to continue the project. The only real issues are who controls the centos.org domain name and how to handle donations to the project. CentOS isn't going anywhere but forward. -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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] A possible bug in FS causing Linux Kernel crash
Hi, I am having random Linux Kernel crash problems while running FreeSWITCH as Skype to/from SIP gateway on one of our production servers. This machine is running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS svn revision number 13754. At time of Kernel crash i could find following crash messages which point to some source code file in FS source tree. - Kernel Begin 3 Time(s): === 3 Time(s): [c0404eff] syscall_call+0x7/0xb 3 Time(s): [c043ed22] sys_delete_module+0x192/0x1b8 3 Time(s): [c0449011] audit_syscall_entry+0x14b/0x17d 3 Time(s): [c049f4fe] remove_proc_entry+0x139/0x18c 3 Time(s): [f8d96281] alsa_sound_exit+0xa/0x30 [snd] 3 Time(s): [f8d96304] snd_info_done+0x46/0x49 [snd] 3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() (Not tainted) 1 Time(s): snd-malloc: Memory leak? pages not freed = 1 -- Kernel End - While the problem seems to arise from ALSA kernel module but it blames FS file fs/proc/generic.c:732 for this. The only FS module that is using ALSA is mod_skypiax but as far as i remember that module is using FS internal routines to allocate and de-allocate sound driver services for Skype client. Please suggest a solution. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash
Thanks. Let me try it and let you know the results. Thank you. On Sun, Jul 26, 2009 at 5:40 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote: Performance problems and other issues (eg crashes on ALSA drivers) has been reported for Skypiax on CentOS, albeit various users got good success on same CentOS. The section down below, Extreme Performances on Linux solves all problems for the user that got issues on CentOS. http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#.22Extreme.22_Performances_on_Linux On Sun, Jul 26, 2009 at 2:37 PM, Giovanni Maruzzelligmar...@celliax.org wrote: Ciao Muhammad, I've got many problems with ALSA drivers, including various kind of crashes. To make a looong story short, use the alsa_drivers version 1.0.20, they have not yet crashed on me. Also, if you want to test it, you can compile the customized snd-dummy driver you find in the svn code, it is a try to have much more efficiency bot in softirqs and context switches, allows for 64 Skype instances (128 subdevices), etc. it is to be compiled with alsa_drivers 1.0.20 too. Is my feeling (I mean, almost sure) they got spin_locking wrong in previous versions, and it crashes the kernel when you really use it (Skype clients have a demented usage of alsa). BTW, I'm in the process of revamp the code, fix the bugs and apply patches. Please, have a look at the new wiki page with lots of new content, I'll send a mail to the ML tomorrow :-) Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Sun, Jul 26, 2009 at 2:19 PM, Muhammad Shahzadshaherya...@googlemail.com wrote: Hi, I am having random Linux Kernel crash problems while running FreeSWITCH as Skype to/from SIP gateway on one of our production servers. This machine is running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS svn revision number 13754. At time of Kernel crash i could find following crash messages which point to some source code file in FS source tree. - Kernel Begin 3 Time(s): === 3 Time(s): [c0404eff] syscall_call+0x7/0xb 3 Time(s): [c043ed22] sys_delete_module+0x192/0x1b8 3 Time(s): [c0449011] audit_syscall_entry+0x14b/0x17d 3 Time(s): [c049f4fe] remove_proc_entry+0x139/0x18c 3 Time(s): [f8d96281] alsa_sound_exit+0xa/0x30 [snd] 3 Time(s): [f8d96304] snd_info_done+0x46/0x49 [snd] 3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() (Not tainted) 1 Time(s): snd-malloc: Memory leak? pages not freed = 1 -- Kernel End - While the problem seems to arise from ALSA kernel module but it blames FS file fs/proc/generic.c:732 for this. The only FS module that is using ALSA is mod_skypiax but as far as i remember that module is using FS internal routines to allocate and de-allocate sound driver services for Skype client. Please suggest a solution. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_managed users?
Sorry for replying late. I have tried mod_managed again on same machine (Lenovo 3000 N200), same OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249. It compiles correctly this time but gives following error upon make install, = making install mod_managed make[5]: *** No rule to make target `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'. Stop. make[4]: *** [install] Error 1 make[3]: *** [mod_managed-install] Error 1 make[2]: *** [install-recursive] Error 1 = Here is compilation log when executing make, if it could of any help. = making all mod_managed Compiling freeswitch_managed.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp Compiling freeswitch_wrap.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp Demo.cs(58,14): warning CS0169: The private method `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used Compilation succeeded - 1 warning(s) Compiling mod_managed.cpp... /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo mod_managed.cpp libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -fPIC -DPIC -o .libs/mod_managed.o mod_managed.cpp: In function ‘void InitManagedSession(ManagedSession*, char* (*)(void*, switch_input_type_t), void (*)())’: mod_managed.cpp:97: warning: deprecated conversion from string constant to ‘char*’ libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o mod_managed.o /dev/null 21 Creating mod_managed.la... cat: .libs/mod_managed.log: No such file or directory = Thank you. On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad shaherya...@googlemail.com wrote: I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro dft...@yahoo.com wrote: Hey, I am here :) I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull. I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. I use c# application and sqlserver 2005, using FS and mod_managed. Diego --- On *Thu, 7/16/09, Michael Giagnocavo m...@giagnocavo.net* wrote: From: Michael Giagnocavo m...@giagnocavo.net Subject: [Freeswitch-users] mod_managed users? To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org Date: Thursday, July 16, 2009, 4:43 PM Hey, if there are any mod_managed users on this list, I’d love it if you were able to let me know. I’d like to get feedback, positive or negative, on what worked, what didn’t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael -Inline Attachment Follows- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orghttp://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-us
Re: [Freeswitch-users] mod_managed users?
I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro dft...@yahoo.com wrote: Hey, I am here :) I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull. I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. I use c# application and sqlserver 2005, using FS and mod_managed. Diego --- On *Thu, 7/16/09, Michael Giagnocavo m...@giagnocavo.net* wrote: From: Michael Giagnocavo m...@giagnocavo.net Subject: [Freeswitch-users] mod_managed users? To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org Date: Thursday, July 16, 2009, 4:43 PM Hey, if there are any mod_managed users on this list, I’d love it if you were able to let me know. I’d like to get feedback, positive or negative, on what worked, what didn’t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael -Inline Attachment Follows- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orghttp://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-us...@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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] SIP Trace Option at Runtime
Hi, Is there any CLI command to enable / disable SIP packet trace at runtime. I do know an option in SIP profile which enables / disable SIP trace but it to apply it i have reload mod_sofia, which at many times fail due to a running call. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skypiax Parameters Informations Request
I think you can use it has long as remote end-point supports it.
Thank you.
On Fri, Jul 10, 2009 at 3:48 PM, Meftah Tayeb tayeb.mef...@gmail.comwrote:
hello Muhammad ,
thank you
what about hig cality audio codec to use?
speex is good?
thanks
Muhammad Shahzad wrote:
Destination parameter actually specifies the extension on which this Skype
user is reachable within FreeSWITCH dialplan for incoming calls.
If this parameter is specified in per_interface_settings xml tag then it
will override the value of this parameter in global_settings xml tag,
otherwise value of this parameter from global_settings xml tag will be used.
Here is an example (see below), the user test.01 is reachable on dialplan
extension 2000 (since it has its own destination defined in
per_interface_settings xml tag), whereas test.02 is reachable on dialplan
extension 5000 (since it does not have destination parameter defined and
thus it will use value for this parameter in global_settings xml tag).
global_settings
param name=debug value=8/
param name=codec-master value=us/
param name=dialplan value=XML/
param name=context value=default/
param name=codec-prefs value=gsm,ulaw/
param name=codec-rates value=8000,16000/
param name=hold-music value=$${moh_uri}/
param name=destination value=5000/
/global_settings
per_interface_settings
interface id=1 name=test.01
param name=hold-music value=$${moh_uri}/
param name=dialplan value=XML/
param name=context value=default/
param name=X11-display value=:101/
param name=tcp_cli_port value=15556/
param name=tcp_srv_port value=15557/
param name=skype_user value=test.01/
param name=destination value=2000/
/interface
interface id=2 name=test.02
param name=hold-music value=$${moh_uri}/
param name=dialplan value=XML/
param name=context value=default/
param name=X11-display value=:102/
param name=tcp_cli_port value=15558/
param name=tcp_srv_port value=15559/
param name=skype_user value=test.02/
/interface
/per_interface_settings
Now the codec, Skype has its own proprietory code for Skype to Skype calls.
The codec we specify in Skypiax configuration file is actually used for
Skype to/from non-Skype calls. Consider following dial plan example (with
skypiax configuration given above),
extension name=skype_incoming-01
condition field=destination_number expression=^2000$
action application=bridge data=sofia/internal/1000/
/condition
/extension
If a remote Skype user dials test.01 from his/her Skype client, then
FreeSWITCH will route this call to SIP user 1000 and codecs specified in
Skypiax configuration will be offered to destination SIP endpoint (SIP user
1000 in this case).
Thank you.
On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb tayeb.mef...@gmail.comwrote:
hello,
i have the folowing parameter in Skypiax.conf.xml:
configuration name=skypiax.conf description=Skypiax Configuration
global_settings
param name=destination value=1000/
each call that will to by routed to this destination??
per_interface_settings
interface id=1 name=skypiax1
param name=destination value=1000/
Each Call will to by routed to this destination?
each codecs that is pocible to use it with Skypiax? all? speex?
this codecs is used beetwan skypiax and the remote peer?
thanks
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--
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
--
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http://www.eset.com
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http://www.eset.com
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Freeswitch-users
Re: [Freeswitch-users] Skypiax Parameters Informations Request
Destination parameter actually specifies the extension on which this Skype
user is reachable within FreeSWITCH dialplan for incoming calls.
If this parameter is specified in per_interface_settings xml tag then it
will override the value of this parameter in global_settings xml tag,
otherwise value of this parameter from global_settings xml tag will be used.
Here is an example (see below), the user test.01 is reachable on dialplan
extension 2000 (since it has its own destination defined in
per_interface_settings xml tag), whereas test.02 is reachable on dialplan
extension 5000 (since it does not have destination parameter defined and
thus it will use value for this parameter in global_settings xml tag).
global_settings
param name=debug value=8/
param name=codec-master value=us/
param name=dialplan value=XML/
param name=context value=default/
param name=codec-prefs value=gsm,ulaw/
param name=codec-rates value=8000,16000/
param name=hold-music value=$${moh_uri}/
param name=destination value=5000/
/global_settings
per_interface_settings
interface id=1 name=test.01
param name=hold-music value=$${moh_uri}/
param name=dialplan value=XML/
param name=context value=default/
param name=X11-display value=:101/
param name=tcp_cli_port value=15556/
param name=tcp_srv_port value=15557/
param name=skype_user value=test.01/
param name=destination value=2000/
/interface
interface id=2 name=test.02
param name=hold-music value=$${moh_uri}/
param name=dialplan value=XML/
param name=context value=default/
param name=X11-display value=:102/
param name=tcp_cli_port value=15558/
param name=tcp_srv_port value=15559/
param name=skype_user value=test.02/
/interface
/per_interface_settings
Now the codec, Skype has its own proprietory code for Skype to Skype calls.
The codec we specify in Skypiax configuration file is actually used for
Skype to/from non-Skype calls. Consider following dial plan example (with
skypiax configuration given above),
extension name=skype_incoming-01
condition field=destination_number expression=^2000$
action application=bridge data=sofia/internal/1000/
/condition
/extension
If a remote Skype user dials test.01 from his/her Skype client, then
FreeSWITCH will route this call to SIP user 1000 and codecs specified in
Skypiax configuration will be offered to destination SIP endpoint (SIP user
1000 in this case).
Thank you.
On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb tayeb.mef...@gmail.comwrote:
hello,
i have the folowing parameter in Skypiax.conf.xml:
configuration name=skypiax.conf description=Skypiax Configuration
global_settings
param name=destination value=1000/
each call that will to by routed to this destination??
per_interface_settings
interface id=1 name=skypiax1
param name=destination value=1000/
Each Call will to by routed to this destination?
each codecs that is pocible to use it with Skypiax? all? speex?
this codecs is used beetwan skypiax and the remote peer?
thanks
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--
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
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[Freeswitch-users] Interactive Connectivity Establishment (ICE) support in FS
Hi, Do we have ICE support in FreeSWITCH. If so, any module as example that is using it? If not then i would like to write one for my mod_msn module, do we have any FS API that i would need to implement in this case? Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Baby Update!
Congratulations to Ray and Samantha. Lets see what new features and bug fixes we will get in their new version..! ;-) Thank you. On 7/2/09, Brian West br...@freeswitch.org wrote: FreeSWITCHers, Kaiden Anthony Chandler will arrive sometime Friday July 3rd 2009!!! So to help out with any last minute expenses and help ease things up for Ray and Samantha and remove some of the worry I'm going to donate $100 dollars myself to the cause... never know diapers and various other expenses that come up. Be sure to select the personal option on paypal so they don't take any money from the transaction. Paypal: intralan...@gmail.com And congratulations to Ray and Samantha on their first Boy! Thanks everyone you're a great community! /b ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Talk to freeswitch from Adobe AIR application through XMLRPC
Good job man! This is really useful. Thank you. On Mon, Jun 29, 2009 at 8:23 AM, Prabhuram Mohan mprabhu...@gmail.comwrote: Hi All, Abode AIR based on flash player can be used to connect to freeswitch and issue commands through XMLRPC. I read the internet to do this plumbing and also got help through generous fellow developers from AIR freeswitch community. Now that it is successfully done, here is the complete code for the benefit of people who are following suite. Comments are welcome!! Code available here - http://neoalchemist.tumblr.com/post/132134683 Thanks Prabhu ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] How to enable compact SIP headers in mod_sofia
Hi, Is it possible to enable compact SIP headers in mod_sofia configuration? If yes, then how to do so? Kindly give an example. Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to enable compact SIP headers in mod_sofia
Ok, thanks, i will take care of it in my code where necessary. Thank you. On Thu, Jun 18, 2009 at 12:54 AM, Brian West br...@freeswitch.org wrote: Its not possible right now but you could if you enable the config option and apply the tag... its something I have thought about adding but wasn't high on my list. NTATAG_SIPFLAGS(MSG_FLG_COMPACT) http://sofia-sip.sourceforge.net/refdocs/nta/nta__tag_8h.html#346ad7ff8e886335f8fec40c65983ca6 /b On Jun 17, 2009, at 1:43 PM, Muhammad Shahzad wrote: Hi, Is it possible to enable compact SIP headers in mod_sofia configuration? If yes, then how to do so? Kindly give an example. 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] MPL Confusion
Thanks, I will look at it in more details as you suggested. I try to be online to discuss mod_msn and mod_yahoo on FS IRC channel this after noon Danish time. Thank you. On Sun, Jun 14, 2009 at 10:39 PM, Brian West br...@freeswitch.org wrote: For clarification ... Read section 3.2 and 3.3 of the MPL 1.1 The simplest way I can describe it is how it was described to me What's yours is yours and what's mine is mine!. /b On Jun 11, 2009, at 11:45 PM, Muhammad Shahzad wrote: I have some confusion about FreeSWITCH's Mozilla Public License 1.1. I do understand that me or any one can change provided code according to our customization needs and we are not bound to share our changes as long as we are not distributing it, right? ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] MPL Confusion
Hi, I have some confusion about FreeSWITCH's Mozilla Public License 1.1. I do understand that me or any one can change provided code according to our customization needs and we are not bound to share our changes as long as we are not distributing it, right? Now, i have been doing RD on MSN and Yahoo voice chat services, I have now completed by research and now would like to write up FS modules to communicate with these servers. But as you all know both MSN and Yahoo provide SIP based VOIP services, however they are not using standard SIP stack and have their own versions of customized SIP stack. So, in order to write an endpoint for these servers, instead of writing everything from scretch, i can using existing mod_sofia endpoint and customize it to make it compatible with MSN and Yahoo SIP stack. So here are my questions, 1. Is it possible under MPL, that i make a copy of mod_sofia as say mod_msn and develop it to work with MSN, similarly mod_yahoo for Yahoo voice chat service? 2. If yes, how can i mention my role in these modules development, i.e. as developer or as contributor? Also i wish to include my work, once completed, in FreeSWITCH, can you provide me the guidelines and / or eligibility criteria to do so, any link on FS site etc.? Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems
Sorry, i didn't visited the Jira link you mentioned. Now i know the issue and I have replied it there. Thank you. On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote: Hi all, there are problems for mod_skypiax in recent centos, with more than a handful of concurrent Skype calls. Probably the problem is ALSA-related. Until it is solved, for production please use Ubuntu 8.04 (see below), some other Linux distro (and please write here your experience), or Windows. I modified the wiki page to reflect this ( http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) If someone with CentOS knowledge can chime in I'll be grateful :-). Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for all infos, and feel free to contact me directly. -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems
Thanks. I didn't make any special arrangements for FS or Skypiax to work on CentOS 5.3. I only enabled CentOS Plus yum repository and then install PAE kernel with following commands, root ~# yum update root ~# yum install kernel-PAE i installed PAE kernel just because i wanted to increase System RAM to 8GB before i deploy it for production use, so i can double or even triple Skypiax channels whenever i need so, without system or FS shutdown. As far as a skypiax configuration is concerned, i did modified mod_skypiax.c to add a couple of commands to dynamically add and remove Skypiax interfaces in a running FS process. However, this code does not replaces or changes any previous code. Other then that there is no significant change in configuration steps. Though i did use mod_xml_curl to dynamically update skypiax interface configuration in FS. Thank you. On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote: Ciao Muhammad! What a good news! Centos is the most stable and performing platform for FS, so I would really love to test and document on the wiki how to have a stable centos mod_skypiax installation. I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE ), and begin to test. In the mean time, do you have any hint, special procedure, etc you have done for having skypiax working well? Please, please, please let be in contact! :-) Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 8:33 AM, Muhammad Shahzadshaherya...@googlemail.com wrote: Sorry, i didn't visited the Jira link you mentioned. Now i know the issue and I have replied it there. Thank you. On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli gmar...@celliax.org wrote: Hi all, there are problems for mod_skypiax in recent centos, with more than a handful of concurrent Skype calls. Probably the problem is ALSA-related. Until it is solved, for production please use Ubuntu 8.04 (see below), some other Linux distro (and please write here your experience), or Windows. I modified the wiki page to reflect this ( http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) If someone with CentOS knowledge can chime in I'll be grateful :-). Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for all infos, and feel free to contact me directly. -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems
I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable kernel. I have heard 64bit ALSA drivers have bad sound issues, but never used it personally. As for source code of my modifications, i made those change to develop a customized commercial solution for large European firm, so i would need their permissions to provide you the required official patch. Let me write them an offical request for this. Thank you. On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote: Ciao Muhammad, first thanks a lot for sharing your experience and help us in making a better software! From the name of the kernel, seems that you are using centos5.2 is this correct? I just tried centos5.3 (64bit) with centosplus kernel, but no luck. I'm now installing a centos5.2 (64), I will test it with centosplus kernel and with its normal kernel. BTW, I would like *really* a lot to have and integrate your addition to the code (also if it needs some labor from me, no problem). Would you like to send it to me, so I will integrate in the main trunk and you don't have no more to maintain it? (so you can develop other cool features for mod_skypiax ;-) )? -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 11:16 AM, Muhammad Shahzadshaherya...@googlemail.com wrote: Thanks. I didn't make any special arrangements for FS or Skypiax to work on CentOS 5.3. I only enabled CentOS Plus yum repository and then install PAE kernel with following commands, root ~# yum update root ~# yum install kernel-PAE i installed PAE kernel just because i wanted to increase System RAM to 8GB before i deploy it for production use, so i can double or even triple Skypiax channels whenever i need so, without system or FS shutdown. As far as a skypiax configuration is concerned, i did modified mod_skypiax.c to add a couple of commands to dynamically add and remove Skypiax interfaces in a running FS process. However, this code does not replaces or changes any previous code. Other then that there is no significant change in configuration steps. Though i did use mod_xml_curl to dynamically update skypiax interface configuration in FS. Thank you. On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli gmar...@celliax.org wrote: Ciao Muhammad! What a good news! Centos is the most stable and performing platform for FS, so I would really love to test and document on the wiki how to have a stable centos mod_skypiax installation. I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE ), and begin to test. In the mean time, do you have any hint, special procedure, etc you have done for having skypiax working well? Please, please, please let be in contact! :-) Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 8:33 AM, Muhammad Shahzadshaherya...@googlemail.com wrote: Sorry, i didn't visited the Jira link you mentioned. Now i know the issue and I have replied it there. Thank you. On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli gmar...@celliax.org wrote: Hi all, there are problems for mod_skypiax in recent centos, with more than a handful of concurrent Skype calls. Probably the problem is ALSA-related. Until it is solved, for production please use Ubuntu 8.04 (see below), some other Linux distro (and please write here your experience), or Windows. I modified the wiki page to reflect this ( http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) If someone with CentOS knowledge can chime in I'll be grateful :-). Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for all infos, and feel free to contact me directly. -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email
Re: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems
I am glad to share the patch to enable dynamic Skypiax interfaces in FS. Please do note that however, that i started working on it on May 22, 2009. So any officaily changes made to mod_skypiax.c since then will not appear in it and will be lost if you apply this patch blindly. I request Giovanni Maruzzelli to carefully merge this patch in main stream code before committing it to FS SVN. Thank you. On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad shaherya...@googlemail.com wrote: I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable kernel. I have heard 64bit ALSA drivers have bad sound issues, but never used it personally. As for source code of my modifications, i made those change to develop a customized commercial solution for large European firm, so i would need their permissions to provide you the required official patch. Let me write them an offical request for this. Thank you. On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote: Ciao Muhammad, first thanks a lot for sharing your experience and help us in making a better software! From the name of the kernel, seems that you are using centos5.2 is this correct? I just tried centos5.3 (64bit) with centosplus kernel, but no luck. I'm now installing a centos5.2 (64), I will test it with centosplus kernel and with its normal kernel. BTW, I would like *really* a lot to have and integrate your addition to the code (also if it needs some labor from me, no problem). Would you like to send it to me, so I will integrate in the main trunk and you don't have no more to maintain it? (so you can develop other cool features for mod_skypiax ;-) )? -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 11:16 AM, Muhammad Shahzadshaherya...@googlemail.com wrote: Thanks. I didn't make any special arrangements for FS or Skypiax to work on CentOS 5.3. I only enabled CentOS Plus yum repository and then install PAE kernel with following commands, root ~# yum update root ~# yum install kernel-PAE i installed PAE kernel just because i wanted to increase System RAM to 8GB before i deploy it for production use, so i can double or even triple Skypiax channels whenever i need so, without system or FS shutdown. As far as a skypiax configuration is concerned, i did modified mod_skypiax.c to add a couple of commands to dynamically add and remove Skypiax interfaces in a running FS process. However, this code does not replaces or changes any previous code. Other then that there is no significant change in configuration steps. Though i did use mod_xml_curl to dynamically update skypiax interface configuration in FS. Thank you. On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli gmar...@celliax.org wrote: Ciao Muhammad! What a good news! Centos is the most stable and performing platform for FS, so I would really love to test and document on the wiki how to have a stable centos mod_skypiax installation. I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE ), and begin to test. In the mean time, do you have any hint, special procedure, etc you have done for having skypiax working well? Please, please, please let be in contact! :-) Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 8:33 AM, Muhammad Shahzadshaherya...@googlemail.com wrote: Sorry, i didn't visited the Jira link you mentioned. Now i know the issue and I have replied it there. Thank you. On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli gmar...@celliax.org wrote: Hi all, there are problems for mod_skypiax in recent centos, with more than a handful of concurrent Skype calls. Probably the problem is ALSA-related. Until it is solved, for production please use Ubuntu 8.04 (see below), some other Linux distro (and please write here your experience), or Windows. I modified the wiki page to reflect this ( http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) If someone with CentOS knowledge can chime in I'll be grateful :-). Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34for all infos, and feel free to contact me directly. -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039
Re: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems
You are welcome. Let me elaborate my setup here, I have two machines, one for development, this is basically my lenovo 3000 N200 laptop, it has following specs, 1. Intel 1.6 GHz with 1GB RAM. 2. CentOS 5.3 with Kernel 2.6.18-128.1.6.el5. 3. FS SVN revision Revision ID 13613. root ~# uname -a Linux localhost.localdomain 2.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:19:18 EDT 2009 i686 i686 i386 GNU/Linux root ~# cat /etc/issue CentOS release 5.3 (Final) Kernel \r on an \m I am using this machine extensively for my development projects, including Skypiax. Yesterday i gave a presentation to the board of directors of the said firm, regarding existing status of my project. They tested the setup with 2-3 concurrent SIP - SKYPIAX and then SKYPIAX to SIP calls without any problem. So, i believe this configuration works without any sound issue...! The second machine is my test machine in a remote data center. I didn't prepare this machine, however, from SSH console i can see it has following specs, 1. Intel(R) Xeon(R) CPU E5405 @ 2.00GHz with 4 GB of RAM. 2. CentOS 5.3 with kernel 2.6.18-92.1.22.el5.centos.plusPAE. 3. FS SVN revision Revision ID 13613. root ~# uname -a Linux localhost.localdomain 2.6.18-92.1.22.el5.centos.plusPAE #1 SMP Wed Dec 17 11:32:56 EST 2008 i686 i686 i386 GNU/Linux root ~# cat /etc/issue CentOS release 5.3 (Final) Kernel \r on an \m Each machine that i use always, get update with yum update command BEFORE i do anything else on it. Hope this info will be helpful for you. Can you give me step by step procedure of your testing that is producing this bad sound result? I would like to perform this test on my both machines and see if i get the same results too. Thank you. On Wed, Jun 10, 2009 at 6:33 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote: Ciao Muhammad, you're faster than light :-)! the patch will be integrated very soon, I'll let you know when I'm done it. Keep enhancements, patches, bug fixes, etc flowing! thanks again, and thanks to the firm that so quickly understood and authorized you, -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 2:29 PM, Muhammad Shahzadshaherya...@googlemail.com wrote: I am glad to share the patch to enable dynamic Skypiax interfaces in FS. Please do note that however, that i started working on it on May 22, 2009. So any officaily changes made to mod_skypiax.c since then will not appear in it and will be lost if you apply this patch blindly. I request Giovanni Maruzzelli to carefully merge this patch in main stream code before committing it to FS SVN. Thank you. On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad shaherya...@googlemail.com wrote: I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable kernel. I have heard 64bit ALSA drivers have bad sound issues, but never used it personally. As for source code of my modifications, i made those change to develop a customized commercial solution for large European firm, so i would need their permissions to provide you the required official patch. Let me write them an offical request for this. Thank you. On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli gmar...@celliax.org wrote: Ciao Muhammad, first thanks a lot for sharing your experience and help us in making a better software! From the name of the kernel, seems that you are using centos5.2 is this correct? I just tried centos5.3 (64bit) with centosplus kernel, but no luck. I'm now installing a centos5.2 (64), I will test it with centosplus kernel and with its normal kernel. BTW, I would like *really* a lot to have and integrate your addition to the code (also if it needs some labor from me, no problem). Would you like to send it to me, so I will integrate in the main trunk and you don't have no more to maintain it? (so you can develop other cool features for mod_skypiax ;-) )? -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Wed, Jun 10, 2009 at 11:16 AM, Muhammad Shahzadshaherya...@googlemail.com wrote: Thanks. I didn't make any special arrangements for FS or Skypiax to work on CentOS 5.3. I only enabled CentOS Plus yum repository and then install PAE kernel with following commands, root ~# yum update root ~# yum install kernel-PAE i installed PAE kernel just because i wanted to increase System RAM to 8GB before i deploy it for production use, so i can double or even triple Skypiax channels whenever i need so, without system or FS shutdown. As far as a skypiax
Re: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems
Hi, What kind of problem you are referring to? I am using Skypiax from latest FS trunk revision no. 13613 on CentOS 5.3, Kernel 2.6.18-92.1.22.el5.centos.plusPAE without any problem, the system seems stable and going in production very soon. However, i would like to mention here that i have customized it a bit to add a couple of new commands to allow dynamic Skypiax interface addition and deletion in a running FreeSWITCH process, But instead of changing any existing code i have merely added new code to the exiting, so this shouldn't have resolved the problem you are referring to. The overall performance of both Skypiax and FS are excellent and we are extremely thankful to you guys for developing such great software. If you guys or anyone else need any help in setting up FS or Skypiax on CentOS, do write to me. Thank you. On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote: Hi all, there are problems for mod_skypiax in recent centos, with more than a handful of concurrent Skype calls. Probably the problem is ALSA-related. Until it is solved, for production please use Ubuntu 8.04 (see below), some other Linux distro (and please write here your experience), or Windows. I modified the wiki page to reflect this ( http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk ) If someone with CentOS knowledge can chime in I'll be grateful :-). Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for all infos, and feel free to contact me directly. -giovanni Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Freeswitch taking too long to start up
Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, make current command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Freeswitch taking too long to start up
Yes, this resolves the problem. Thank you. On Tue, Jun 2, 2009 at 5:27 PM, dujinfang dujinf...@gmail.com wrote: Actually Brain mentioned that you can comment out switch_nat_init(); in switch_core.c On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: As I understand it, a new ‘feature’ was added over the weekend to resolve NAT. If you’re firewall is not allowing ICMP then FS waits until it times out. At this time there is no option to disable it. Regards -- *From:* freeswitch-users-boun...@lists.freeswitch.org [ mailto:freeswitch-users-boun...@lists.freeswitch.orgfreeswitch-users-boun...@lists.freeswitch.org ] *On Behalf Of *Muhammad Shahzad *Sent:* 02 June 2009 11:40 *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Freeswitch taking too long to start up Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, make current command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Freeswitch taking too long to start up
I had to upgrade again svn revision to use this switch, but it works. Thank you. On Wed, Jun 3, 2009 at 1:12 AM, Keith Laaks kei...@voxtelecom.co.za wrote: Hi, Try starting using the -nonat switch. Best Regards Keith *From:* Muhammad Shahzad [mailto:shaherya...@googlemail.com] *Sent:* 02 June 2009 14:39 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Freeswitch taking too long to start up Yes, this resolves the problem. Thank you. On Tue, Jun 2, 2009 at 5:27 PM, dujinfang dujinf...@gmail.com wrote: Actually Brain mentioned that you can comment out switch_nat_init(); in switch_core.c On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote: As I understand it, a new ‘feature’ was added over the weekend to resolve NAT. If you’re firewall is not allowing ICMP then FS waits until it times out. At this time there is no option to disable it. Regards -- *From:* freeswitch-users-boun...@lists.freeswitch.org [ mailto:freeswitch-users-boun...@lists.freeswitch.orgfreeswitch-users-boun...@lists.freeswitch.org ] *On Behalf Of *Muhammad Shahzad *Sent:* 02 June 2009 11:40 *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Freeswitch taking too long to start up Hi, I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using 32bit CentOS 5.3, make current command completes successfully without any errors but when i start freeswitch it take considerable time (roughly 90 - 120 seconds) to start up. During this time no message is display on console. Once successfully started, it works fine. However, this initial delay is really annoying. Is there anyway to reduce/remove this delay? Thank you. -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how to disbale : switch_core_sqldb()
I think passing -nosql as argument to freeswitch start up command will do this. For example, bash# freeswitch -hp -nosql -nc Thank you. On Mon, May 25, 2009 at 1:24 PM, mashudi mashudifl...@telkom.co.id wrote: Hi Guys, How to disable process starting of sql DB when we starting FreeSwitch ? here is the log from starting FreeSwitch : 32m2009-05-25 14:00:18 [INFO] switch_core_sqldb.c:494 switch_core_sqldb_start() Opening DB thank you in advance, mashudi * Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATEspasi[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya hubungi 147 atau ketik INFO, sms ke 345. ___ 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 -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
