Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...
Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I’ve been using FS as a gateway to our OCS server for some time. It’s used just for testing, so it’s not really used every day. I don’t know when, but after some trunk update (right now I running r13945) of FS it doesn’t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter param name=register-transport value=tcp/ set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it’s sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Originating a call from lua with rudimentary error checking
check that s is nil. /b On Jun 24, 2009, at 8:12 PM, John Wehle wrote: What's the recommended way to check if the session constructor was successful (i.e. the number could be dialed)? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Originate works but dialplan does not work?
Thanks a lot it works for me... Edmar Cruz wrote: If then, what bridge i shall call to? Like this? action application=bridge data=sofia/external/@1...@116.541.23.12/ dujinfang wrote: put your extension in dialplan/public.xml instead of sip_profiles/ external/myprofile.xml On Jun 24, 2009, at 6:02 PM, Edmar Cruz wrote: Ooops.. Sorry wrong spelling... Same issue Jason White-14 wrote: Edmar Cruz darklio...@yahoo.com wrote: Here is my dialplan on sip_profiles/external/myprofile.xml extension name=dialmyprof condition field=destination_number expression=^(\d+)$ action application=set data=gate_site_id=1/ action application=bridge The above should be $1 not @1 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24181933.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/Originate-works-but-dialplan-does-not-work--tp24181208p24199253.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] How to change database of freeswitch cdr to MySQL?
Hi, How can i change my freeswitch database instead of CSV file, I make it mysql. Can you tell me how? Thanks -- View this message in context: http://www.nabble.com/How-to-change-database-of-freeswitch-cdr-to-MySQL--tp24200644p24200644.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_dingaling no audio
Two questions for you:
1) Do you have extension 888 in your public context?
2)Can you put your internal Ip address of FS in rtp-ip instead of
$${bind_server_ip} just to make sure it get the right IP?
3) param name=disable-rtp-auto-adjust value=true/ is not really
required at least for my working setup behind the NAT router.
On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Chris, thanks for your help. Here's my client.xml
include
!-- Client Profile (Original mode) --
!-- to use this profile take the x- away from the open and close tags so
its profile and /profile --
profile type=client
param name=name value=gmail.com/
param name=login value=user...@gmail.com/talk/
param name=password value=/
param name=dialplan value=XML/
param name=context value=public/
param name=message value=Jingle all the way/
param name=rtp-ip value=$${bind_server_ip}/
!--param name=ext-sip-ip value=$${external_sip_ip}/--
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=disable-rtp-auto-adjust value=true/
param name=auto-login value=true/
!-- SASL plain or md5 --
param name=sasl value=plain/
!-- if the server where the jabber is hosted is not the same as the
one in the jid --
param name=server value=talk.google.com/
!-- Enable TLS or not --
param name=tls value=true/
!-- disable to trade async for more calls --
param name=use-rtp-timer value=true/
!-- default extension (if one cannot be determined) --
param name=exten value=888/
!-- VAD choose one --
!-- param name=vad value=in/ --
!-- param name=vad value=out/ --
param name=vad value=both/
!--param name=avatar value=/path/to/tiny.jpg/--
!--param name=candidate-acl value=rfc1918.auto/--
/profile
/include
On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen chris.chen2...@gmail.comwrote:
Please provide your client.xml detail with confidential information
crossout, I have gtalk client and server working properly behind the NAT.
I should be able to help you.
Chris
On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang jingwei.y...@gmail.comwrote:
Thanks seven. External IPs have sound echo this time with ext-rtp-ip
uncommented and disable-rtp-auto-adjust=true. However, internal IP has no
audio this time no matter what value disable-rtp-auto-adjust is...
On Thu, Jun 25, 2009 at 11:24 AM, seven dujinf...@gmail.com wrote:
uncomment ext-rtp-ip
On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote:
Hi seven, thanks for your reply. I've commented out ext-rtp-ip and put
disable-rtp-auto-adjust inside client.xml. No matter what value this
parameter has (true or false), local IP is able to hear the echo but
external ones still have no audio.
On Wed, Jun 24, 2009 at 6:01 PM, seven dujinf...@gmail.com wrote:
search wiki from sth. like disable_rtp_autoajust , I don't remember the
exact var.
On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote:
Hi Guys,
Here's my situation:
The freeswitch server and my machine are behind the same LAN. If I
commented out ext-rtp-ip from client.xml, I'm able to hear the echo (by
*originate dingaling/gmail.com/user...@gmail.com echo*).
However, external calls have no sound at all no matter whether this
param is commented out or not.
Please kindly let me know what other params to set to resolve this
issue.
Thanks,
-Jingwei
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Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...
Please open a jira and attach sip traces. /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West br...@freeswitch.org wrote: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I’ve been using FS as a gateway to our OCS server for some time. It’s used just for testing, so it’s not really used every day. I don’t know when, but after some trunk update (right now I running r13945) of FS it doesn’t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter param name=register-transport value=tcp/ set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it’s sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 no audio
3) param name=disable-rtp-auto-adjust value=true/ is not really required at least for my working setup behind the NAT router. ok, this param is originally added for another problem http://jira.freeswitch.org/browse/MODENDP-198 . But I think it might be useful for this. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...
Done, added as issue SFSIP-157. Regards, Peter Olsson Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 25 juni 2009 10:16 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I've been using FS as a gateway to our OCS server for some time. It's used just for testing, so it's not really used every day. I don't know when, but after some trunk update (right now I running r13945) of FS it doesn't send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter param name=register-transport value=tcp/ set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it's sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.orgmailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.orgmailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.comhttp://www.cluecon.com/ !DSPAM:4a4333c332936913812693! ___ Freeswitch-users mailing list Freeswitch-users@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 change database of freeswitch cdr to MySQL?
You can use FS XML Curl - FS sends XML CDRs to a web server of your choice, and there you do whatever you want with these CDRs, like store in a database. There are also pre-built solutions available, check here: http://versafon.com/versafonweb/CommercialSupport.jsp Edmar Cruz wrote: Hi, How can i change my freeswitch database instead of CSV file, I make it mysql. Can you tell me how? 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
Re: [Freeswitch-users] mod_dingaling no audio
if you are behind nat , you will not want to disable auto-adjust that is what the feature is there to help you with. On Thu, Jun 25, 2009 at 10:16 AM, Seven Du dujinf...@gmail.com wrote: 3) param name=disable-rtp-auto-adjust value=true/ is not really required at least for my working setup behind the NAT router. ok, this param is originally added for another problem http://jira.freeswitch.org/browse/MODENDP-198 . But I think it might be useful for this. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...
Fixed revision 13948. /b On Jun 25, 2009, at 10:22 AM, Peter Olsson wrote: Done, added as issue SFSIP-157. Regards, Peter Olsson Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org ] För Brian West Skickat: den 25 juni 2009 10:16 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I’ve been using FS as a gateway to our OCS server for some time. It’s used just for testing, so it’s not really used every day. I don’t know when, but after some trunk update (right now I running r13945) of FS it doesn’t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter param name=register-transport value=tcp/ set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it’s sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:4a4333c332936913812693! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 change database of freeswitch cdr to MySQL?
Are you the owner of Versafon? /b On Jun 25, 2009, at 10:24 AM, paul.degt wrote: You can use FS XML Curl - FS sends XML CDRs to a web server of your choice, and there you do whatever you want with these CDRs, like store in a database. There are also pre-built solutions available, check here: http://versafon.com/versafonweb/CommercialSupport.jsp Edmar Cruz wrote: Hi, How can i change my freeswitch database instead of CSV file, I make it mysql. Can you tell me how? 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
Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...
Can you verify that this is fixed... I think its related to the same issue... /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West br...@freeswitch.org wrote: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I’ve been using FS as a gateway to our OCS server for some time. It’s used just for testing, so it’s not really used every day. I don’t know when, but after some trunk update (right now I running r13945) of FS it doesn’t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter param name=register-transport value=tcp/ set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it’s sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 change database of freeswitch cdr to MySQL?
I am an employee, why? Brian West wrote: Are you the owner of Versafon? /b On Jun 25, 2009, at 10:24 AM, paul.degt wrote: You can use FS XML Curl - FS sends XML CDRs to a web server of your choice, and there you do whatever you want with these CDRs, like store in a database. There are also pre-built solutions available, check here: http://versafon.com/versafonweb/CommercialSupport.jsp Edmar Cruz wrote: Hi, How can i change my freeswitch database instead of CSV file, I make it mysql. Can you tell me how? 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 ___ Freeswitch-users mailing list Freeswitch-users@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] Outgoing sofia calls not using tcp anymore...
I upgraded to 13950, still the same, keeping the same loop like the console log showing: 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 Chris On Thu, Jun 25, 2009 at 12:06 PM, Brian West br...@freeswitch.org wrote: Can you verify that this is fixed... I think its related to the same issue... /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West br...@freeswitch.org wrote: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I’ve been using FS as a gateway to our OCS server for some time. It’s used just for testing, so it’s not really used every day. I don’t know when, but after some trunk update (right now I running r13945) of FS it doesn’t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter param name=register-transport value=tcp/ set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it’s sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
since you are advertising on our site regularly now perhaps you could ask your boss to sponsor the project. On Thu, Jun 25, 2009 at 11:15 AM, paul.degt paul.d...@gmail.com wrote: I am an employee, why? Brian West wrote: Are you the owner of Versafon? /b On Jun 25, 2009, at 10:24 AM, paul.degt wrote: You can use FS XML Curl - FS sends XML CDRs to a web server of your choice, and there you do whatever you want with these CDRs, like store in a database. There are also pre-built solutions available, check here: http://versafon.com/versafonweb/CommercialSupport.jsp Edmar Cruz wrote: Hi, How can i change my freeswitch database instead of CSV file, I make it mysql. Can you tell me how? 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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...
What can I say - you guys provide far much better (and quicker) support then any commersial solution :) Thanks for the help! /Peter Från: Brian West br...@freeswitch.org Skickat: den 25 juni 2009 17:53 Till: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Fixed revision 13948. /b On Jun 25, 2009, at 10:22 AM, Peter Olsson wrote: Done, added as issue SFSIP-157. Regards, Peter Olsson Från: freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Brian West Skickat: den 25 juni 2009 10:16 Till: freeswitch-users@lists.freeswitch.orgmailto:freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I’ve been using FS as a gateway to our OCS server for some time. It’s used just for testing, so it’s not really used every day. I don’t know when, but after some trunk update (right now I running r13945) of FS it doesn’t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter param name=register-transport value=tcp/ set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it’s sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.orgmailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.orgmailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.comhttp://www.cluecon.com/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.orgmailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.orgmailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.comhttp://www.cluecon.com/ !DSPAM:4a439d8f32931361515932! ___ Freeswitch-users mailing list Freeswitch-users@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] Outgoing sofia calls not using tcp anymore...
I see what is wrong you're running it all on the same IP and the 302 is back to the same IP but a different port I need to fix that logic to compare the port number also. /b On Jun 25, 2009, at 11:24 AM, Chris Chen wrote: I upgraded to 13950, still the same, keeping the same loop like the console log showing: 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 Chris On Thu, Jun 25, 2009 at 12:06 PM, Brian West br...@freeswitch.org wrote: Can you verify that this is fixed... I think its related to the same issue... /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West br...@freeswitch.org wrote: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I’ve been using FS as a gateway to our OCS server for some time. It’s used just for testing, so it’s not really used every day. I don’t know when, but after some trunk update (right now I running r13945) of FS it doesn’t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter param name=register-transport value=tcp/ set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it’s sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
I didn't see the last message from Paul as advertizement. He was explaining about the FS XML Curl. I think he may have mentioned the link to versafon even if he didn't work there. On Thu, Jun 25, 2009 at 12:27 PM, Anthony Minessale anthony.miness...@gmail.com wrote: since you are advertising on our site regularly now perhaps you could ask your boss to sponsor the project. On Thu, Jun 25, 2009 at 11:15 AM, paul.degt paul.d...@gmail.com wrote: I am an employee, why? Brian West wrote: Are you the owner of Versafon? /b On Jun 25, 2009, at 10:24 AM, paul.degt wrote: You can use FS XML Curl - FS sends XML CDRs to a web server of your choice, and there you do whatever you want with these CDRs, like store in a database. There are also pre-built solutions available, check here: http://versafon.com/versafonweb/CommercialSupport.jsp Edmar Cruz wrote: Hi, How can i change my freeswitch database instead of CSV file, I make it mysql. Can you tell me how? 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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
I talked about it but with the current state of VOIP market it's seems problematic. :-( I did not see my posts as strictly ads since we offer free software as well and also share my limited knowledge about FS. Anthony Minessale wrote: since you are advertising on our site regularly now perhaps you could ask your boss to sponsor the project. On Thu, Jun 25, 2009 at 11:15 AM, paul.degt paul.d...@gmail.com mailto:paul.d...@gmail.com wrote: I am an employee, why? Brian West wrote: Are you the owner of Versafon? /b On Jun 25, 2009, at 10:24 AM, paul.degt wrote: You can use FS XML Curl - FS sends XML CDRs to a web server of your choice, and there you do whatever you want with these CDRs, like store in a database. There are also pre-built solutions available, check here: http://versafon.com/versafonweb/CommercialSupport.jsp Edmar Cruz wrote: Hi, How can i change my freeswitch database instead of CSV file, I make it mysql. Can you tell me how? Thanks ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
I will have to ask my boss about that, most probably he will ask same in return. EdPimentl wrote: To be fair ... when mentioning Freeswitch here http://versafon.com/versafonweb/Software.jsp a common courtesy would be to provide a link to Freeswitch http://www.freeswitch.org Sincerely, -E ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
Done. :-) Brian West wrote: And possibly present the word FreeSWITCH in the proper case! ;) /b On Jun 25, 2009, at 11:48 AM, EdPimentl wrote: To be fair ... when mentioning Freeswitch here http://versafon.com/versafonweb/Software.jsp a common courtesy would be to provide a link to Freeswitch http://www.freeswitch.org Sincerely, -E Brian West br...@freeswitch.org mailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com http://www.cluecon.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
Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...
Brian, 13952 fixed the problem, now SIP over TCP to my Exchange 2007 UM is working. Thanks for your great work. Chris On Thu, Jun 25, 2009 at 12:53 PM, Brian West br...@freeswitch.org wrote: I found the problem... the fs_path refactor regression number 2 was just fixed.. It was assuming the route_uri was the contact and making it stick to the wrong place to send the invite... you should be able to update now and it work correctly. /b On Jun 25, 2009, at 11:35 AM, Anthony Minessale wrote: are you redirecting it to yourself by any chance because of some proxy in your network? On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen chris.chen2...@gmail.comwrote: I upgraded to 13950, still the same, keeping the same loop like the console log showing: 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 Chris On Thu, Jun 25, 2009 at 12:06 PM, Brian West br...@freeswitch.orgwrote: Can you verify that this is fixed... I think its related to the same issue... /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West br...@freeswitch.orgwrote: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I’ve been using FS as a gateway to our OCS server for some time. It’s used just for testing, so it’s not really used every day. I don’t know when, but after some trunk update (right now I running r13945) of FS it doesn’t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter param name=register-transport value=tcp/ set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set
Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
On 06/25/2009 01:04 PM, paul.degt wrote: I will have to ask my boss about that, most probably he will ask same in That doesn't really make sense... FreeSWITCH isn't using or benefitting from your software... but yours is from FreeSWITCH -Ray ___ Freeswitch-users mailing list Freeswitch-users@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 change database of freeswitch cdr to MySQL?
Right which is why i said it was a suggestion not a demand. If you want to help answer email in here or answer questions on a regular basis that's just as valuable. if your boss has no money to sponsor maybe he can donate you =D On Thu, Jun 25, 2009 at 11:57 AM, paul.degt paul.d...@gmail.com wrote: I talked about it but with the current state of VOIP market it's seems problematic. :-( I did not see my posts as strictly ads since we offer free software as well and also share my limited knowledge about FS. Anthony Minessale wrote: since you are advertising on our site regularly now perhaps you could ask your boss to sponsor the project. On Thu, Jun 25, 2009 at 11:15 AM, paul.degt paul.d...@gmail.com mailto:paul.d...@gmail.com wrote: I am an employee, why? Brian West wrote: Are you the owner of Versafon? /b On Jun 25, 2009, at 10:24 AM, paul.degt wrote: You can use FS XML Curl - FS sends XML CDRs to a web server of your choice, and there you do whatever you want with these CDRs, like store in a database. There are also pre-built solutions available, check here: http://versafon.com/versafonweb/CommercialSupport.jsp Edmar Cruz wrote: Hi, How can i change my freeswitch database instead of CSV file, I make it mysql. Can you tell me how? Thanks ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.commsn%253aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.compaypal%253aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.orgsip%253a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.orggoogletalk%253aconf%252b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...
are you redirecting it to yourself by any chance because of some proxy in your network? On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen chris.chen2...@gmail.comwrote: I upgraded to 13950, still the same, keeping the same loop like the console log showing: 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 Chris On Thu, Jun 25, 2009 at 12:06 PM, Brian West br...@freeswitch.org wrote: Can you verify that this is fixed... I think its related to the same issue... /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 sip%3a...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/ 1...@192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West br...@freeswitch.org wrote: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I’ve been using FS as a gateway to our OCS server for some time. It’s used just for testing, so it’s not really used every day. I don’t know when, but after some trunk update (right now I running r13945) of FS it doesn’t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter param name=register-transport value=tcp/ set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it’s sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
To be fair ... when mentioning Freeswitch here http://versafon.com/versafonweb/Software.jsp a common courtesy would be to provide a link to Freeswitchhttp://www.freeswitch.org Sincerely, -E ___ Freeswitch-users mailing list Freeswitch-users@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 change database of freeswitch cdr to MySQL?
And possibly present the word FreeSWITCH in the proper case! ;) /b On Jun 25, 2009, at 11:48 AM, EdPimentl wrote: To be fair ... when mentioning Freeswitch here http://versafon.com/versafonweb/Software.jsp a common courtesy would be to provide a link to Freeswitch Sincerely, -E Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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 change database of freeswitch cdr to MySQL?
It's a suggestion not a demand. If you follow the link it's a list of products and services related to FS. We prefer that anyone who wants to sell stuff made from FS or as an accessory to FS would consider sponsoring the project or ClueCon. We provide both commercial and free support for FreeSWITCH and the amount of free help we have time to give is directly impacted by how many people who use FreeSWITCH for commercial purposes give back to us in the form of volunteer developers, support contracts and sponsorship. Trust me, if we don't ask very few will realize it on their own. On Thu, Jun 25, 2009 at 11:35 AM, Harmeet Singh harm...@litatel.com wrote: I didn't see the last message from Paul as advertizement. He was explaining about the FS XML Curl. I think he may have mentioned the link to versafon even if he didn't work there. On Thu, Jun 25, 2009 at 12:27 PM, Anthony Minessale anthony.miness...@gmail.com wrote: since you are advertising on our site regularly now perhaps you could ask your boss to sponsor the project. On Thu, Jun 25, 2009 at 11:15 AM, paul.degt paul.d...@gmail.com wrote: I am an employee, why? Brian West wrote: Are you the owner of Versafon? /b On Jun 25, 2009, at 10:24 AM, paul.degt wrote: You can use FS XML Curl - FS sends XML CDRs to a web server of your choice, and there you do whatever you want with these CDRs, like store in a database. There are also pre-built solutions available, check here: http://versafon.com/versafonweb/CommercialSupport.jsp Edmar Cruz wrote: Hi, How can i change my freeswitch database instead of CSV file, I make it mysql. Can you tell me how? 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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
On Thu, Jun 25, 2009 at 9:48 AM, EdPimentl edpime...@gmail.com wrote: To be fair ... when mentioning Freeswitch here http://versafon.com/versafonweb/Software.jsp a common courtesy would be to provide a link to Freeswitchhttp://www.freeswitch.org Not only that but spelling FreeSWITCH correctly would be a nice touch, no? :) -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...
I found the problem... the fs_path refactor regression number 2 was just fixed.. It was assuming the route_uri was the contact and making it stick to the wrong place to send the invite... you should be able to update now and it work correctly. /b On Jun 25, 2009, at 11:35 AM, Anthony Minessale wrote: are you redirecting it to yourself by any chance because of some proxy in your network? On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen chris.chen2...@gmail.com wrote: I upgraded to 13950, still the same, keeping the same loop like the console log showing: 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 Chris On Thu, Jun 25, 2009 at 12:06 PM, Brian West br...@freeswitch.org wrote: Can you verify that this is fixed... I think its related to the same issue... /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West br...@freeswitch.org wrote: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I’ve been using FS as a gateway to our OCS server for some time. It’s used just for testing, so it’s not really used every day. I don’t know when, but after some trunk update (right now I running r13945) of FS it doesn’t send the SIP traffic using tcp anymore (OCS only accepts tcp or tls). My configuration is quite easy, I have a sofia gateway configured to OCS, this has the parameter param name=register-transport value=tcp/ set in the config (nothing in the config has changed for ages). Then in the dialplan I use this gateway to connect the calls. When doing a siptrace I can see that the headers has transport=tcp set correctly, but according to the trace it’s sent using udp instead of tcp. Has something changed so I need to configure it in another way, or is it just simply a bug? I just wanted to check this before issuing a jira case and providing more specific information and debug traces etc. /Peter ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org
Re: [Freeswitch-users] FreeSwitch at backend
On Wed, Jun 24, 2009 at 10:53 PM, Jingwei Yang jingwei.y...@gmail.comwrote: Thanks Michael, fs_eslib sounds the one for Java. I'll give it a try. Let us know how it goes. We like success stories! :) -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
FreeSWITCH Solutions will soon be offering a product gallery where companies who use FS can become certified partners and display their products. On Thu, Jun 25, 2009 at 12:08 PM, Raymond Chandler intralan...@freeswitch.org wrote: On 06/25/2009 01:04 PM, paul.degt wrote: I will have to ask my boss about that, most probably he will ask same in That doesn't really make sense... FreeSWITCH isn't using or benefitting from your software... but yours is from FreeSWITCH -Ray ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
FS users may well be benefiting - thus FS itself benefits as well indirectly. Raymond Chandler wrote: On 06/25/2009 01:04 PM, paul.degt wrote: I will have to ask my boss about that, most probably he will ask same in That doesn't really make sense... FreeSWITCH isn't using or benefitting from your software... but yours is from FreeSWITCH -Ray ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
Is it possible to unsusbcribe from specific threads on this list? Specifically I am looking for C code that removes useless banter so my brain doesn't hurt so much... -Original Message- From: paul.degt [mailto:paul.d...@gmail.com] Sent: Thursday, June 25, 2009 10:24 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL? FS users may well be benefiting - thus FS itself benefits as well indirectly. Raymond Chandler wrote: On 06/25/2009 01:04 PM, paul.degt wrote: I will have to ask my boss about that, most probably he will ask same in That doesn't really make sense... FreeSWITCH isn't using or benefitting from your software... but yours is from FreeSWITCH -Ray ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Nibblebill and multiple gateway
Did this work? Would love an update on this error/issue.
_
From: Michael Jerris [mailto:m...@jerris.com]
Sent: Wednesday, June 24, 2009 8:15 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway
try adding
action application=set data=import=nibble_rate/
before the bridge and report back results.
Mike
On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote:
Dear All,
Look like nibblebill does't work with multiple gatreway.
I try
action application=set data=nibble_account=0838833133/
action application=bridge
data={absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626
734...@203.xxx.xxx.xxx|[nibble_rate=0.5]sofia/external/6626734...@202.xxx.xx
x.xxx
nibblebill not found nibble_rate
But
action application=set data=nibble_rate=0.05/
action application=set data=nibble_account=0838833133/
action application=bridge
data={absolute_codec_string='GSM,G729'}sofia/external/6626734...@203.xxx.xx
x.xxx|sofia/external/6626734...@202.xxx.xxx.xxx
Work fine
What's difference from set application and [] ?
Best Regards.
Dome C.
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Re: [Freeswitch-users] How to change database of freeswitch cdr to MySQL?
On 06/25/2009 11:24 AM, paul.degt wrote: You can use FS XML Curl - FS sends XML CDRs to a web server of your choice, and there you do whatever you want with these CDRs, like store in a database. There are also pre-built solutions available, check here: http://versafon.com/versafonweb/CommercialSupport.jsp There are also a few xml curl options from within the project already... for free, check here: http://fisheye.freeswitch.org/browse/FreeSWITCH/scripts/contrib/intralanman/PHP/fs_curl , here: http://fisheye.freeswitch.org/browse/FreeSWITCH/scripts/contrib/swk , and here: http://fisheye.freeswitch.org/browse/FreeSWITCH/scripts/contrib/trixter/xml-curl For posting CDR's to db specifically, there are a couple of options already in tree that could easily be modified to meet your needs (whatever they may be), check here: http://fisheye.freeswitch.org/browse/FreeSWITCH/scripts/contrib/trixter/xml-cdr and here: http://fisheye.freeswitch.org/browse/FreeSWITCH/scripts/contrib/intralanman/perl/cdr Coincidentally, there's a java esl lib in tree similar to the one you might see on the afforementioned site, check here: http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/java -Ray ___ Freeswitch-users mailing list Freeswitch-users@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] Nibblebill and multiple gateway
Just test.
i use javascript
session.execute(set, import=nibble_rate);
session.execute(set, import=nibble_account);
session.execute(bridge,
{absolute_codec_string='GSM,G729'}[nibble_rate=0.5,nibble_account=0838833133]sofia/external/x...@.xxx.xxx.xx);
when call connected nibble do nothing i found heartbeat
mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT!
when call disconnect nibble update amont.
mod_nibblebill.c:478 Billing 16 secs
I think nibble still not found variable channel.
Let's me share more information
I want to use nibblebill for callingcard. (i have develop billing by
myself). i plan to use javascript connect to ODBC
when customer call my script query balance and say.
and then i loop for get destination (my customer want to dial many number).
when i got number my script query
gateway from DB. i have 3 route and order by cost.
First plan i use
session.execute(bridge,
[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/x...@provder1
|[nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/external/x...@provder2);
i modify nibblebill for match provider with my billing.
this case still fail.
now i try
if
(session.ready()){
s = new
Session({absolute_codec_string='GSM,G729'}sofia/external/x...@provider1
}
if
(s.ready()){
session.execute(set,
nibble_rate=2.5);
session.execute(set,
nibble_account=+acaller);
session.execute(set,
hangup_after_bridge=false);
session.execute(set,
provider_id=+dialprovider_id[1]);
bridge(session,s);
}
and check hangup cause before try other provider.
Please guide me it's right way or not ?
Dome C.
2009/6/26 Darren Schreiber d...@d-man.org
Did this work? Would love an update on this error/issue.
--
*From:* Michael Jerris [mailto:m...@jerris.com]
*Sent:* Wednesday, June 24, 2009 8:15 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Nibblebill and multiple gateway
try adding action application=set data=import=nibble_rate/
before the bridge and report back results.
Mike
On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote:
Dear All,
Look like nibblebill does't work with multiple gatreway.
I try
action application=set
data=nibble_account=0838833133/
action application=bridge
data={absolute_codec_string='GSM,G729'}[nibble_rate=0.3]
sofia/external/6626734...@203.xxx.xxx.xxx|[nibble_rate=0.5]
sofia/external/6626734...@202.xxx.xxx.xxx
nibblebill not found nibble_rate
But
action application=set data=nibble_rate=0.05/
action application=set
data=nibble_account=0838833133/
action application=bridge data={absolute_codec_string='GSM,
G729'}sofia/external/6626734...@203.xxx.xxx.xxxG729%27%7Dsofia/external/6626734...@203.xxx.xxx.xxx
|sofia/external/6626734...@202.xxx.xxx.xxx%7Csofia/external/6626734...@202.xxx.xxx.xxx
Work fine
What's difference from set application and [] ?
Best Regards.
Dome C.
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Re: [Freeswitch-users] Nibblebill and multiple gateway
I said to just add the set import=nibble_rate, your re-setting it for
no reason (and getting rid of the change that should have helped) by
your import=nibble_account line
Mike
On Jun 25, 2009, at 2:04 PM, Dome Charoenyost wrote:
Just test.
i use javascript
session.execute(set, import=nibble_rate);
session.execute(set, import=nibble_account);
session.execute(bridge, {absolute_codec_string='GSM,G729'}
[nibble_rate=0.5,nibble_account=0838833133]sofia/external/x...@.xxx.xxx.xx
);
when call connected nibble do nothing i found heartbeat
mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT!
when call disconnect nibble update amont.
mod_nibblebill.c:478 Billing 16 secs
I think nibble still not found variable channel.
Let's me share more information
I want to use nibblebill for callingcard. (i have develop billing by
myself). i plan to use javascript connect to ODBC
when customer call my script query balance and say.
and then i loop for get destination (my customer want to dial many
number). when i got number my script query
gateway from DB. i have 3 route and order by cost.
First plan i use
session.execute(bridge,
[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/
x...@provder1|
[nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/external/
x...@provder2);
i modify nibblebill for match provider with my billing.
this case still fail.
now i try
if (session.ready()){
s = new Session({absolute_codec_string='GSM,G729'}sofia/
external/x...@provider1
}
if (s.ready()){
session.execute(set, nibble_rate=2.5);
session.execute(set, nibble_account=+acaller);
session.execute(set, hangup_after_bridge=false);
session.execute(set, provider_id=+dialprovider_id[1]);
bridge(session,s);
}
and check hangup cause before try other provider.
Please guide me it's right way or not ?
Dome C.
2009/6/26 Darren Schreiber d...@d-man.org
Did this work? Would love an update on this error/issue.
From: Michael Jerris [mailto:m...@jerris.com]
Sent: Wednesday, June 24, 2009 8:15 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway
try adding
action application=set data=import=nibble_rate/
before the bridge and report back results.
Mike
On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote:
Dear All,
Look like nibblebill does't work with multiple gatreway.
I try
action application=set data=nibble_account=0838833133/
action application=bridge
data={absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734...@203.xxx.xxx.xxx
|[nibble_rate=0.5]sofia/external/6626734...@202.xxx.xxx.xxx
nibblebill not found nibble_rate
But
action application=set data=nibble_rate=0.05/
action application=set data=nibble_account=0838833133/
action application=bridge
data={absolute_codec_string='GSM,G729'}sofia/external/6626734...@203.xxx.xxx.xxx
|sofia/external/6626734...@202.xxx.xxx.xxx
Work fine
What's difference from set application and [] ?
Best Regards.
Dome C.
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[Freeswitch-users] Originating a call from lua with rudimentary error checking
What's the recommended way to check if the session constructor was
successful (i.e. the number could be dialed)?
check that s is nil.
Doesn't work ... s is never nil. Type shows it as userdata
even if Session failed. Specifically my test was:
local s = freeswitch.Session (
{ignore_early_media=true,origination_caller_id_name= ..
caller .. }loopback/ .. destination .. /default/XML)
stream:write (type(s))
if s == nil then
stream:write (-ERR call failed\n)
return
end
and I dialed an unreachable number.
and that s.ready() is true
Checking s.ready() results in:
[ERR] freeswitch_lua.cpp:102 session is not initalized
if Session failed.
What I'm looking for is a way to try to originate a call which doesn't
throw ERR messages if the attempt fails.
Explicitly calling session.originate seems to allow you to check if
the call was successful ... is there a particular reason it's discouraged?
I'm happy to avoid it if a better approach is available, however I'm
having trouble finding one.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495 | |
-
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Re: [Freeswitch-users] Originating a call from lua with rudimentary error checking
I think this is an oversight
update to trunk and session.ready() should work as expected.
On Thu, Jun 25, 2009 at 2:35 PM, John Wehle j...@feith.com wrote:
What's the recommended way to check if the session constructor was
successful (i.e. the number could be dialed)?
check that s is nil.
Doesn't work ... s is never nil. Type shows it as userdata
even if Session failed. Specifically my test was:
local s = freeswitch.Session (
{ignore_early_media=true,origination_caller_id_name= ..
caller .. }loopback/ .. destination .. /default/XML)
stream:write (type(s))
if s == nil then
stream:write (-ERR call failed\n)
return
end
and I dialed an unreachable number.
and that s.ready() is true
Checking s.ready() results in:
[ERR] freeswitch_lua.cpp:102 session is not initalized
if Session failed.
What I'm looking for is a way to try to originate a call which doesn't
throw ERR messages if the attempt fails.
Explicitly calling session.originate seems to allow you to check if
the call was successful ... is there a particular reason it's discouraged?
I'm happy to avoid it if a better approach is available, however I'm
having trouble finding one.
-- John
-
| Feith Systems | Voice: 1-215-646-8000 | Email: j...@feith.com |
|John Wehle| Fax: 1-215-540-5495 | |
-
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--
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Re: [Freeswitch-users] Compiling freeswitch for Dragonfly BSD
On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote: On Jun 23, 2009, at 10:15 PM, Vincent Stemen vince.freeswi...@hightek.org wrote: Here is what I did and the results: Checked out the current trunk with svn. Patched /usr/include/sys/resource.h Since Dragonfly has fixed or will be fixing this future releases I patched the system header to add RLIMIT_AS rather than patching freeswitch to use RLIMIT_VMEM. Can we make a patch ifdefing on RLIMIT_AS to make this always work without patches to system header files? Thanks for the responses Michael. I did this for attempting to compile freeswitch-1.0.3 and trunk as of a couple months ago. It would not apply to the current freeswitch trunk though. Apparently there have been changes to that area of the code. Since RLIMIT_AS is apparently a posix standard definition, I think this is fixed in Dragonfly HEAD and should not be a problem with future releases. I could go ahead and make a new patch when I get a chance if you still want me to, for compatibility with older Dragonfly releases. Compiling = Still lots of warnings of: warning: return makes pointer from integer without a cast Errors: It is apparently not checking return codes from make. It continues even when there are errors. Is this intentional?? su_alloc.c: In function `su_salloc': su_alloc.c:1518: warning: return makes pointer from integer without a cast gmake[9]: *** [su_alloc.lo] Error 1 gmake[8]: *** [all] Error 2 Making all in features LTCOMPILE features.lo ... Making all in sresolv LTCOMPILE sres.lo LTCOMPILE sres_cache.lo LTCOMPILE sres_blocking.lo LTCOMPILE sresolv.lo LTCOMPILE sres_sip.lo sres_sip.c: In function `sres_sip_new': sres_sip.c:267: warning: return makes pointer from integer without a cast gmake[8]: *** [sres_sip.lo] Error 1 Making all in ipt LTCOMPILE base64.lo LTCOMPILE token64.lo LINK libipt.la ... There are about 12 errors of this nature before ending with Making all in nua LTCOMPILE nua.lo nua.c: In function `nua_create': nua.c:141: warning: return makes pointer from integer without a cast nua.c:144: warning: return makes pointer from integer without a cast gmake[9]: *** [nua.lo] Error 1 gmake[8]: *** [all] Error 2 gmake[8]: *** No rule to make target `iptsec/libiptsec.la', needed by `libsofia-sip-ua.la'. Stop. gmake[7]: *** [all-recursive] Error 1 Making all in packages gmake[6]: *** [all-recursive] Error 1 gmake[5]: *** [all] Error 2 gmake[4]: *** [/u1/falcon/ports/freeswitch-20090623/work/ freeswitch-20090623/libs/sofia-sip/libsofia-sip-ua/libsofia-sip- ua.la] Error 2 gmake[3]: *** [mod_sofia-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + gmake install + +--+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 Can you post a bug to Jira.freeswitch.org with all these warnings, even better with patches to fix it. It says it has been successfully built. Apparently part of the same problem of not checking the return codes. Patches to fix this appreciated Heh :-) OK. If I get it working and we end up using freeswitch, I will probably take a look at seeing if I can fix some or all of these warnings and create patches. It does not say what most of the errors are except for near the last when it says No rule to make target `iptsec/libiptsec.la' It just says Error 1 or Error 2 which does not tell me what the problem is. ___ Freeswitch-users mailing list Freeswitch-users@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] Compiling freeswitch for Dragonfly BSD
On Tue, Jun 23, 2009 at 11:19:51PM -0400, Andrew Thompson wrote: On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote: Ok. I did this. Compilation still failed but there are significant improvements since the last time. Here is what I did and the results: It looks like some the games that sofia plays with errno makes Dragonfly unhappy. I also noticed that where the code checks for BSD-like systems (*BSD and OSX) in libsofia-sip-ua/su/sofia-sip/su_errno.h, DragonFly is omitted, so obviously one of the first steps would be to fix that (if applicable). If you disable mod_sofia in modules conf, do the rest of the default modules build OK? OK. I commented out endpoints/mod_sofia. It looks like that eliminated all the errors except the one I get at the end. making all mod_spidermonkey cd config; gmake -j1 export cd pr; gmake -j1 export cd include; gmake export cd md; gmake export ../../../config/./nsinstall: cannot make symbolic link /u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/js/nsprpub/dist/include/nspr/.: File exists gmake[9]: *** [export] Error 1 gmake[8]: *** [export] Error 2 gmake[7]: *** [export] Error 2 gmake[6]: *** [export] Error 2 gmake[5]: *** [/u1/falcon/ports/freeswitch-20090623/work/freeswitch-20090623/libs/js/libjs.la] Error 2 gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_spidermonkey-all] Error 1 gmake[2]: *** [all-recursive] Error 1 So, it looks like most all the problems, except for that symlink error, including the pointer cast warnings, are related to the sofia module. I notice a lot of the modules seem to be redirecting the output somewhere. Not only do they just say Error 1 or Error 2 when there is an error, they also do not show the compile commands. They just output something like Making built-sources in su or Compiling src/switch_apr.c Is there a log file somewhere that contains the actual compile commands and error output so you can find out what happened when there is a error? Or perhaps a configuration to enable it to come out on the console? For the record, DragonFly and FreeBSD have rather seriously diverged at this point, DragonFly forked from FreeBSD back in the 4.10 days or so and has changed a *lot* of things since, so I don't think it's gonna be quite as easy as you expected (but it's far from impossible either). Andrew True, architecturally Dragonfly is becoming very different. They seem to be trying to maintain fairly good API compatibility though. Enough to constantly allow them to bring across major sub-systems, such as sound and SATA drivers, etc, from FreeBSD. So far, they have been pretty good about correcting it as soon as possible whenever one of us finds an incompatibility (Such as the RLIMIT_AS issue). Usually, all I have to do is add -D__FreeBSD__ to CFLAGS and CPPFLAGS to compile packages that do not natively know about Dragonfly yet. Which is what I am doing with freeswitch. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...
Will it be a Windows build with the fix available soon? Drago From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris Chen Sent: Thursday, June 25, 2009 1:05 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore... Brian, 13952 fixed the problem, now SIP over TCP to my Exchange 2007 UM is working. Thanks for your great work. Chris On Thu, Jun 25, 2009 at 12:53 PM, Brian West br...@freeswitch.org wrote: I found the problem... the fs_path refactor regression number 2 was just fixed.. It was assuming the route_uri was the contact and making it stick to the wrong place to send the invite... you should be able to update now and it work correctly. /b On Jun 25, 2009, at 11:35 AM, Anthony Minessale wrote: are you redirecting it to yourself by any chance because of some proxy in your network? On Thu, Jun 25, 2009 at 11:24 AM, Chris Chen chris.chen2...@gmail.com wrote: I upgraded to 13950, still the same, keeping the same loop like the console log showing: 2009-06-25 12:23:31.951936 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 mailto:sip%3a...@192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.953179 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:31.993935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 mailto:sip%3a...@192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:31.995134 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.35933 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 mailto:sip%3a...@192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.37184 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.77935 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 mailto:sip%3a...@192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.79108 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.119937 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 mailto:sip%3a...@192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 2009-06-25 12:23:32.121170 [DEBUG] sofia.c:3214 Channel sofia/internal/1...@192.168.0.250 entering state [calling][0] 2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 mailto:sip%3a...@192.168.0.250 ;transport=tcp Setting proxy route to sofia/internal/1...@192.168.0.250 Chris On Thu, Jun 25, 2009 at 12:06 PM, Brian West br...@freeswitch.org wrote: Can you verify that this is fixed... I think its related to the same issue... /b On Jun 25, 2009, at 8:07 AM, Chris Chen wrote: I am having the same issue, now the SIP trunk over TCP between FS and Exchange 2007 UM just stops working, just stuck in a loop like this: 2009-06-25 00:02:55.542672 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.584468 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 mailto:sip%3a...@192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.585657 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.626474 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 mailto:sip%3a...@192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.627632 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.668479 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 mailto:sip%3a...@192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.669608 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] 2009-06-25 00:02:55.710473 [DEBUG] sofia_glue.c:1692 sip:1...@192.168.0.250 mailto:sip%3a...@192.168.0.250 ;transport=tcp Setting proxy route to sofia/external/1...@192.168.0.250 2009-06-25 00:02:55.711631 [DEBUG] sofia.c:3214 Channel sofia/external/1...@192.168.0.250 entering state [calling][0] On my Exchange 2007 side nothing was changed which used to work fine with FS Chris On Thu, Jun 25, 2009 at 4:15 AM, Brian West br...@freeswitch.org wrote: Please open a jira and attach sip traces of register and phone calls. /b On Jun 25, 2009, at 2:36 AM, Peter Olsson wrote: I've been using FS as a gateway to our OCS server for some time. It's used just for testing, so it's not really used every day. I don't know when, but after some trunk update (right now I running r13945) of FS it doesn't send the SIP traffic
Re: [Freeswitch-users] Outgoing sofia calls not using tcp anymore...
I don't think the windows build was updated to include the bug... but you can build it with MSVC Express Edition which is Free from Microsoft. /b On Jun 25, 2009, at 4:55 PM, Drago Totev wrote: Will it be a Windows build with the fix available soon? Drago Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] Compiling freeswitch for Dragonfly BSD
On Jun 25, 2009, at 5:49 PM, Vincent wrote: On Tue, Jun 23, 2009 at 11:19:51PM -0400, Andrew Thompson wrote: On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote: Ok. I did this. Compilation still failed but there are significant improvements since the last time. Here is what I did and the results: It looks like some the games that sofia plays with errno makes Dragonfly unhappy. I also noticed that where the code checks for BSD-like systems (*BSD and OSX) in libsofia-sip-ua/su/sofia-sip/su_errno.h, DragonFly is omitted, so obviously one of the first steps would be to fix that (if applicable). If you disable mod_sofia in modules conf, do the rest of the default modules build OK? OK. I commented out endpoints/mod_sofia. It looks like that eliminated all the errors except the one I get at the end. making all mod_spidermonkey cd config; gmake -j1 export cd pr; gmake -j1 export cd include; gmake export cd md; gmake export ../../../config/./nsinstall: cannot make symbolic link /u1/falcon/ ports/freeswitch-20090623/work/freeswitch-20090623/libs/js/nsprpub/ dist/include/nspr/.: File exists gmake[9]: *** [export] Error 1 gmake[8]: *** [export] Error 2 gmake[7]: *** [export] Error 2 gmake[6]: *** [export] Error 2 gmake[5]: *** [/u1/falcon/ports/freeswitch-20090623/work/ freeswitch-20090623/libs/js/libjs.la] Error 2 gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_spidermonkey-all] Error 1 gmake[2]: *** [all-recursive] Error 1 you can also comment out that module and see if you get further. So, it looks like most all the problems, except for that symlink error, including the pointer cast warnings, are related to the sofia module. I notice a lot of the modules seem to be redirecting the output somewhere. Not only do they just say Error 1 or Error 2 when there is an error, they also do not show the compile commands. They just output something like Making built-sources in su or Compiling src/switch_apr.c Is there a log file somewhere that contains the actual compile commands and error output so you can find out what happened when there is a error? Or perhaps a configuration to enable it to come out on the console? VERBOSE=1 gmake For the record, DragonFly and FreeBSD have rather seriously diverged at this point, DragonFly forked from FreeBSD back in the 4.10 days or so and has changed a *lot* of things since, so I don't think it's gonna be quite as easy as you expected (but it's far from impossible either). Andrew True, architecturally Dragonfly is becoming very different. They seem to be trying to maintain fairly good API compatibility though. Enough to constantly allow them to bring across major sub-systems, such as sound and SATA drivers, etc, from FreeBSD. So far, they have been pretty good about correcting it as soon as possible whenever one of us finds an incompatibility (Such as the RLIMIT_AS issue). Usually, all I have to do is add -D__FreeBSD__ to CFLAGS and CPPFLAGS to compile packages that do not natively know about Dragonfly yet. Which is what I am doing with freeswitch. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Orphaned calls left on FS after redirect off of FS
Dear Anthony and Brian, Firstly please accept my apologies for wasting your time. Brian's request for the SVN number prompted me to realise I was running with an out of date version of FS. When I synced up to the head of the trunk and reran my tests the scenario I described below worked perfectly with no stuck calls. Therefore the sequence Park, Ringing (ring back), Redirect using the event API has provided me with the automated redirection I was seeking. Thank you for your advice earlier this week and prompt turnaround of fixes for the problems I encountered with bridged and deflected calls. Regards Richard Lamkin richard.lam...@mettoni.com From: Brian West [mailto:br...@freeswitch.org] Sent: 24 June 2009 19:21 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Orphaned calls left on FS after redirect offof FS I have tried to reproduce this issue but haven't been able too... What SVN Rev are you on? /b On Jun 24, 2009, at 10:29 AM, Richard Lamkin wrote: I am using the API to manage calls as they arrive at FS from a trunk I have a very simple Dial plan rule that parks the incoming call. extension name=Trunk_02031701648 condition field=destination_number expression=^02031701648$ action application=park / /condition /extension Once the call is parked via the API I first send a ringing (to keep the originator happy) sendmsg uuid call-command: execute execute-app-name: respond execute-app-arg: 180 Via the API I then redirect the call on to another PSTN number back through the same gateway sendmsg UUID call-command: execute execute-app-name: redirect execute-app-arg: sip:destination@194.0.147.16 The redirection works well and the originator and destination are connected correctly. But after the call has left FS I'm still left with some call debris which I cannot clear down using sendmsg UUID call-command: execute execute-app-name: hangup execute-app-arg: cause code Using command api show channels I find the following held on FS The only way I've found to remove these calls is api hupall - uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,appl ication,application_data,dialplan,context,read_codec,read_rate,write_cod ec,write_rate 132a3362-3bb2-8e46-a11b-9bd46ab2d706,2009-06-24 15:10:15,1245852615,sofia/TrunkExternal/0203196...@194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728...@194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 c2b40d55-0b5f-ff45-9541-cdcecc451e2c,2009-06-24 15:18:00,1245853080,sofia/TrunkExternal/0203196...@194.0.147.16:5060,CS_ EXECUTE,0203196598,0203196598, 194.0.147.16,02031701648,redirect,sip:0189728...@194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 b03fa6b3-a436-db4b-add5-dfd0658b8867,2009-06-24 15:22:53,1245853373,sofia/TrunkExternal/0203196...@194.0.147.16:5060,CS_ EXECUTE,0203196599,0203196599, 194.0.147.16,02031701648,redirect,sip:0189728...@194.0.147.16,XML,Public ,PCMU,8000,PCMU,8000 57ce0f01-a84d-6e49-a66d-0d771849ebb4,2009-06-24 15:40:30,1245854430,sofia/TrunkExternal/0189728...@194.0.147.16:5060,CS_ EXECUTE,0189728400,0189728400, 194.0.147.16,02031701648,redirect,sip:0701137...@194.0.147.16,XML,Public ,PCMA,8000,PCMA,8000 4 total. --- The SIP signalling is correct with an outgoing 302 moved temporarily [with the new destination in the contact] which is then Ack'ed by the switch. From a SIP point of view the call no longer on FS. The only way I've found to remove these phantom calls is either api hupall, or restart the Sip profile. Any suggestions on how I can remove these phantom calls without recourse to api hupall. api hupall kills any incoming calls as well as the stuck calls. Regards Richard Lamkin richard.lam...@mettoni.com * Please consider the environment before printing this e-mail * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN * ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com http://www.cluecon.com/
Re: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins
Everyone that wanted to help this project to pay Arsen to write this please paypal br...@freeswitch.org when you can so I can gather it all up and send it to Arsen... Everyone that has sent money already thank you... ;) http://code.google.com/p/unimrcp/source/browse/trunk/plugins/mrcp-flite/src/mrcp_flite.c http://code.google.com/p/unimrcp/source/browse/trunk/plugins/mrcp-pocketsphinx/src/mrcp_pocketsphinx.c So the progress is moving forward Please pitch in. Thanks, Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] CDR loss possibility if FS freezes?
No the goal is to NOT crash in the first place. Are you experiencing a crash? If so http://wiki.freeswitch.org/wiki/Reporting_Bugs is how you would report it. Thanks, Brian On Jun 25, 2009, at 7:34 PM, Muhammad Danish Moosa wrote: Hi Can somebody tell if FS freezes/crashes due to any reason. Does it logs both log of each call before dying? If we run it on large scale like 2-3k calls , a simple crash can cost a lot if it dies silently. One more aspect is , after freezing it will no more send/rec packets to any endpoint ,may result in inaccurate logging on endpoint. It should somehow send BYE ? -- Muhammad Danish Moosa Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] CDR loss possibility if FS freezes?
Well this isn't specific to FS crashing. The machine losing power would have the same effect, no? -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Muhammad Danish Moosa Sent: Thursday, June 25, 2009 6:35 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] CDR loss possibility if FS freezes? Hi Can somebody tell if FS freezes/crashes due to any reason. Does it logs both log of each call before dying? If we run it on large scale like 2-3k calls , a simple crash can cost a lot if it dies silently. One more aspect is , after freezing it will no more send/rec packets to any endpoint ,may result in inaccurate logging on endpoint. It should somehow send BYE ? -- Muhammad Danish Moosa ___ Freeswitch-users mailing list Freeswitch-users@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 no audio
I guess you have the problem here,
in client.xml you have
param name=context value=public/
but you only define extension 888 in default context,
that's why nobody can reach you from public.
under /usr/local/freeswitch/conf/dialplan
define extension 888 in public.xml to the proper extension you expect, and
check the console log from fs_cli when you do gtalk calling to your gmail
client, you will find out the solution to your issue.
chris
On Thu, Jun 25, 2009 at 10:15 PM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Chris. thanks for the reply. Here're my answers.
On Thu, Jun 25, 2009 at 9:02 PM, Chris Chen chris.chen2...@gmail.comwrote:
Two questions for you:
1) Do you have extension 888 in your public context?
What public context are you saying? I only defined 888.xml in
/usr/local/freeswitch/conf/directory/default.
2)Can you put your internal Ip address of FS in rtp-ip instead of
$${bind_server_ip} just to make sure it get the right IP?
I changed it to the internal Ip, but still no echo.
3) param name=disable-rtp-auto-adjust value=true/ is not really
required at least for my working setup behind the NAT router.
Thanks, I've commented it out.
On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Chris, thanks for your help. Here's my client.xml
include
!-- Client Profile (Original mode) --
!-- to use this profile take the x- away from the open and close tags
so its profile and /profile --
profile type=client
param name=name value=gmail.com/
param name=login value=user...@gmail.com/talk/
param name=password value=/
param name=dialplan value=XML/
param name=context value=public/
param name=message value=Jingle all the way/
param name=rtp-ip value=$${bind_server_ip}/
!--param name=ext-sip-ip value=$${external_sip_ip}/--
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=disable-rtp-auto-adjust value=true/
param name=auto-login value=true/
!-- SASL plain or md5 --
param name=sasl value=plain/
!-- if the server where the jabber is hosted is not the same as the
one in the jid --
param name=server value=talk.google.com/
!-- Enable TLS or not --
param name=tls value=true/
!-- disable to trade async for more calls --
param name=use-rtp-timer value=true/
!-- default extension (if one cannot be determined) --
param name=exten value=888/
!-- VAD choose one --
!-- param name=vad value=in/ --
!-- param name=vad value=out/ --
param name=vad value=both/
!--param name=avatar value=/path/to/tiny.jpg/--
!--param name=candidate-acl value=rfc1918.auto/--
/profile
/include
On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen
chris.chen2...@gmail.comwrote:
Please provide your client.xml detail with confidential information
crossout, I have gtalk client and server working properly behind the NAT.
I should be able to help you.
Chris
On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang
jingwei.y...@gmail.comwrote:
Thanks seven. External IPs have sound echo this time with ext-rtp-ip
uncommented and disable-rtp-auto-adjust=true. However, internal IP has no
audio this time no matter what value disable-rtp-auto-adjust is...
On Thu, Jun 25, 2009 at 11:24 AM, seven dujinf...@gmail.com wrote:
uncomment ext-rtp-ip
On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote:
Hi seven, thanks for your reply. I've commented out ext-rtp-ip and
put disable-rtp-auto-adjust inside client.xml. No matter what value
this
parameter has (true or false), local IP is able to hear the echo but
external ones still have no audio.
On Wed, Jun 24, 2009 at 6:01 PM, seven dujinf...@gmail.com wrote:
search wiki from sth. like disable_rtp_autoajust , I don't remember
the exact var.
On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote:
Hi Guys,
Here's my situation:
The freeswitch server and my machine are behind the same LAN. If I
commented out ext-rtp-ip from client.xml, I'm able to hear the echo
(by
*originate dingaling/gmail.com/user...@gmail.com echo*).
However, external calls have no sound at all no matter whether this
param is commented out or not.
Please kindly let me know what other params to set to resolve this
issue.
Thanks,
-Jingwei
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Re: [Freeswitch-users] mod_dingaling no audio
Hi Chris. thanks for the reply. Here're my answers.
On Thu, Jun 25, 2009 at 9:02 PM, Chris Chen chris.chen2...@gmail.comwrote:
Two questions for you:
1) Do you have extension 888 in your public context?
What public context are you saying? I only defined 888.xml in
/usr/local/freeswitch/conf/directory/default.
2)Can you put your internal Ip address of FS in rtp-ip instead of
$${bind_server_ip} just to make sure it get the right IP?
I changed it to the internal Ip, but still no echo.
3) param name=disable-rtp-auto-adjust value=true/ is not really
required at least for my working setup behind the NAT router.
Thanks, I've commented it out.
On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Chris, thanks for your help. Here's my client.xml
include
!-- Client Profile (Original mode) --
!-- to use this profile take the x- away from the open and close tags
so its profile and /profile --
profile type=client
param name=name value=gmail.com/
param name=login value=user...@gmail.com/talk/
param name=password value=/
param name=dialplan value=XML/
param name=context value=public/
param name=message value=Jingle all the way/
param name=rtp-ip value=$${bind_server_ip}/
!--param name=ext-sip-ip value=$${external_sip_ip}/--
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=disable-rtp-auto-adjust value=true/
param name=auto-login value=true/
!-- SASL plain or md5 --
param name=sasl value=plain/
!-- if the server where the jabber is hosted is not the same as the
one in the jid --
param name=server value=talk.google.com/
!-- Enable TLS or not --
param name=tls value=true/
!-- disable to trade async for more calls --
param name=use-rtp-timer value=true/
!-- default extension (if one cannot be determined) --
param name=exten value=888/
!-- VAD choose one --
!-- param name=vad value=in/ --
!-- param name=vad value=out/ --
param name=vad value=both/
!--param name=avatar value=/path/to/tiny.jpg/--
!--param name=candidate-acl value=rfc1918.auto/--
/profile
/include
On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen chris.chen2...@gmail.comwrote:
Please provide your client.xml detail with confidential information
crossout, I have gtalk client and server working properly behind the NAT.
I should be able to help you.
Chris
On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang
jingwei.y...@gmail.comwrote:
Thanks seven. External IPs have sound echo this time with ext-rtp-ip
uncommented and disable-rtp-auto-adjust=true. However, internal IP has no
audio this time no matter what value disable-rtp-auto-adjust is...
On Thu, Jun 25, 2009 at 11:24 AM, seven dujinf...@gmail.com wrote:
uncomment ext-rtp-ip
On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote:
Hi seven, thanks for your reply. I've commented out ext-rtp-ip and
put disable-rtp-auto-adjust inside client.xml. No matter what value this
parameter has (true or false), local IP is able to hear the echo but
external ones still have no audio.
On Wed, Jun 24, 2009 at 6:01 PM, seven dujinf...@gmail.com wrote:
search wiki from sth. like disable_rtp_autoajust , I don't remember
the exact var.
On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote:
Hi Guys,
Here's my situation:
The freeswitch server and my machine are behind the same LAN. If I
commented out ext-rtp-ip from client.xml, I'm able to hear the echo (by
*originate dingaling/gmail.com/user...@gmail.com echo*).
However, external calls have no sound at all no matter whether this
param is commented out or not.
Please kindly let me know what other params to set to resolve this
issue.
Thanks,
-Jingwei
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Re: [Freeswitch-users] mod_dingaling no audio
I have been testing dingaling all day... I added switch_nat routines to poke holes in the nat if needed if you're behind upnp or nat-pmp. /b On Jun 25, 2009, at 9:30 PM, Chris Chen wrote: I guess you have the problem here, in client.xml you have param name=context value=public/ but you only define extension 888 in default context, that's why nobody can reach you from public. under /usr/local/freeswitch/conf/dialplan define extension 888 in public.xml to the proper extension you expect, and check the console log from fs_cli when you do gtalk calling to your gmail client, you will find out the solution to your issue. chris ___ Freeswitch-users mailing list Freeswitch-users@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] Nibblebill and multiple gateway
2009/6/26 Michael Jerris m...@jerris.com:
I said to just add the set import=nibble_rate, your re-setting it for no
reason (and getting rid of the change that should have helped) by your
import=nibble_account line
I test it agin.
import work. nibble can see nibble_rate , nibble_account in channel
but i can't change nibble heratbeat so nibble use default heartbeat.
Dome C.
Mike
On Jun 25, 2009, at 2:04 PM, Dome Charoenyost wrote:
Just test.
i use javascript
session.execute(set, import=nibble_rate);
session.execute(set, import=nibble_account);
session.execute(bridge,
{absolute_codec_string='GSM,G729'}[nibble_rate=0.5,nibble_account=0838833133]sofia/external/x...@.xxx.xxx.xx);
when call connected nibble do nothing i found heartbeat
mod_callbackbill.c:550 Received request via SESSION_HEARTBEAT!
when call disconnect nibble update amont.
mod_nibblebill.c:478 Billing 16 secs
I think nibble still not found variable channel.
Let's me share more information
I want to use nibblebill for callingcard. (i have develop billing by
myself). i plan to use javascript connect to ODBC
when customer call my script query balance and say.
and then i loop for get destination (my customer want to dial many number).
when i got number my script query
gateway from DB. i have 3 route and order by cost.
First plan i use
session.execute(bridge,
[nibble_rate=0.5,nibble_account=xxx,provider_id=1]sofia/external/x...@provder1|[nibble_rate=0.5,nibble_account=xxx,provider_id=2]sofia/external/x...@provder2);
i modify nibblebill for match provider with my billing.
this case still fail.
now i try
if
(session.ready()){
s = new
Session({absolute_codec_string='GSM,G729'}sofia/external/x...@provider1
}
if
(s.ready()){
session.execute(set,
nibble_rate=2.5);
session.execute(set,
nibble_account=+acaller);
session.execute(set,
hangup_after_bridge=false);
session.execute(set,
provider_id=+dialprovider_id[1]);
bridge(session,s);
}
and check hangup cause before try other provider.
Please guide me it's right way or not ?
Dome C.
2009/6/26 Darren Schreiber d...@d-man.org
Did this work? Would love an update on this error/issue.
From: Michael Jerris [mailto:m...@jerris.com]
Sent: Wednesday, June 24, 2009 8:15 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway
try adding
action application=set data=import=nibble_rate/
before the bridge and report back results.
Mike
On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote:
Dear All,
Look like nibblebill does't work with multiple gatreway.
I try
action application=set
data=nibble_account=0838833133/
action application=bridge
data={absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734...@203.xxx.xxx.xxx|[nibble_rate=0.5]sofia/external/6626734...@202.xxx.xxx.xxx
nibblebill not found nibble_rate
But
action application=set data=nibble_rate=0.05/
action application=set
data=nibble_account=0838833133/
action application=bridge
data={absolute_codec_string='GSM,G729'}sofia/external/6626734...@203.xxx.xxx.xxx|sofia/external/6626734...@202.xxx.xxx.xxx
Work fine
What's difference from set application and [] ?
Best Regards.
Dome C.
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Re: [Freeswitch-users] CDR loss possibility if FS freezes?
true dat... but again our goal is to not crash in the first place :P... nice APC can take care of the no power thing. /b On Jun 25, 2009, at 8:16 PM, Michael Giagnocavo wrote: Well this isn’t specific to FS crashing. The machine losing power would have the same effect, no? -Michael Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.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] CDR loss possibility if FS freezes?
In case of bad battery in APC, are cdr's logged prior to system failure? On Thursday, June 25, 2009, Brian West br...@freeswitch.org wrote: true dat... but again our goal is to not crash in the first place :P... nice APC can take care of the no power thing. /b On Jun 25, 2009, at 8:16 PM, Michael Giagnocavo wrote: Well this isn’t specific to FS crashing. The machine losing power would have the same effect, no? -Michael Brian westbr...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com http://www.cluecon.com/ -- Shannon ___ Freeswitch-users mailing list Freeswitch-users@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 no audio
Hi Chris, here's the one that confuses me. As far as I understand, the
extension 888 defined in public.xml is for picking up incoming calls. It
should have no influence on outgoing calls, right? If not, what is to write
to fit my case? (originate
dingaling/gmail.com/user...@gmail.combridge(dingaling/
gmail.com/user...@gmail.com), both userAAA and userBBB can be internal or
external).
Anyway, I've defined a extension 888 in public.xml. Frankly speaking, I'm
not quite sure what to include. So I make it very simple.
extension name=gtalk
condition field=destination_number expression=^(888)$
action application=voicemail data=default $${domain} 888/
/condition
/extension
Here are three relative parameters in client.xml:
param name=rtp-ip value=192.168.1.100/
param name=ext-rtp-ip value=$${external_rtp_ip}/
!--param name=disable-rtp-auto-adjust value=true/--
Still, I got no echo for internal Ip calls. Please let me know where goes
wrong.
Thanks,
-Jingwei
On Fri, Jun 26, 2009 at 10:30 AM, Chris Chen chris.chen2...@gmail.comwrote:
I guess you have the problem here,
in client.xml you have
param name=context value=public/
but you only define extension 888 in default context,
that's why nobody can reach you from public.
under /usr/local/freeswitch/conf/dialplan
define extension 888 in public.xml to the proper extension you expect, and
check the console log from fs_cli when you do gtalk calling to your gmail
client, you will find out the solution to your issue.
chris
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Re: [Freeswitch-users] CDR loss possibility if FS freezes?
You could use something like nibble bill and at most loose the last interval of the call before its billed. You're going thru a lot of what if's ... You can't account for everything and you shouldn't have all your eggs in the same basket. /b On Jun 25, 2009, at 10:32 PM, Shannon wrote: In case of bad battery in APC, are cdr's logged prior to system failure? ___ Freeswitch-users mailing list Freeswitch-users@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] CDR loss possibility if FS freezes?
Of course not. This is why many do billing in icrements like mod_nibblebill does. Radius (although not yet with our module) and diamater both work this way and solve this issue. This in combination with session timers adress this and the hangup issue during a catastophic switch or network failure. On Jun 25, 2009, at 11:32 PM, Shannon shan...@sacredhearts.us wrote: In case of bad battery in APC, are cdr's logged prior to system failure? On Thursday, June 25, 2009, Brian West br...@freeswitch.org wrote: true dat... but again our goal is to not crash in the first place :P... nice APC can take care of the no power thing. /b On Jun 25, 2009, at 8:16 PM, Michael Giagnocavo wrote: Well this isn’t specific to FS crashing. The machine losing power would have the same effect, no? -Michael Brian westbr...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com http://www.cluecon.com/ -- Shannon ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CDR loss possibility if FS freezes?
Does nibblebill update balances in real-time for each and every call? Does it do every second (micro/nano second?). How does it affect the performance vs if its done at end of call? I know that is not desirable for calling card applications. Harmeet On Thu, Jun 25, 2009 at 11:42 PM, Brian West br...@freeswitch.org wrote: You could use something like nibble bill and at most loose the last interval of the call before its billed. You're going thru a lot of what if's ... You can't account for everything and you shouldn't have all your eggs in the same basket. /b On Jun 25, 2009, at 10:32 PM, Shannon wrote: In case of bad battery in APC, are cdr's logged prior to system failure? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CDR loss possibility if FS freezes?
Well you would do it every 20-60 seconds maybe... It would be silly to do it every micro/nano second... it would cost you more in cpu and you don't gain much. /b On Jun 25, 2009, at 10:53 PM, Harmeet Singh wrote: Does nibblebill update balances in real-time for each and every call? Does it do every second (micro/nano second?). How does it affect the performance vs if its done at end of call? I know that is not desirable for calling card applications. Harmeet ___ Freeswitch-users mailing list Freeswitch-users@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] CDR loss possibility if FS freezes?
Is solar an option? ;) /b On Jun 25, 2009, at 10:48 PM, Harmeet Singh wrote: Just make sure the power is always there! ...I know that some parts of the world this is not easy to achieve. Harmeet ___ Freeswitch-users mailing list Freeswitch-users@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] CDR loss possibility if FS freezes?
clouds On Thu, Jun 25, 2009 at 22:56, Brian West br...@freeswitch.org wrote: Is solar an option? ;) /b On Jun 25, 2009, at 10:48 PM, Harmeet Singh wrote: Just make sure the power is always there! ...I know that some parts of the world this is not easy to achieve. Harmeet ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Shannon ___ Freeswitch-users mailing list Freeswitch-users@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] CDR loss possibility if FS freezes?
Can the interval be easily configures based on the destination? Like small interval for destinations with cost per minute $1.00 and large intervals for cheaper destinations? Harmeet On Thu, Jun 25, 2009 at 11:56 PM, Brian West br...@freeswitch.org wrote: Well you would do it every 20-60 seconds maybe... It would be silly to do it every micro/nano second... it would cost you more in cpu and you don't gain much. /b On Jun 25, 2009, at 10:53 PM, Harmeet Singh wrote: Does nibblebill update balances in real-time for each and every call? Does it do every second (micro/nano second?). How does it affect the performance vs if its done at end of call? I know that is not desirable for calling card applications. Harmeet ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CDR loss possibility if FS freezes?
yes you could. why not check it out and set it up ... its rather powerful. /b On Jun 25, 2009, at 11:06 PM, Harmeet Singh wrote: Can the interval be easily configures based on the destination? Like small interval for destinations with cost per minute $1.00 and large intervals for cheaper destinations? Harmeet ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CDR loss possibility if FS freezes?
I also forgot about Nights. /b On Jun 25, 2009, at 11:07 PM, SP wrote: clouds On Thu, Jun 25, 2009 at 22:56, Brian West br...@freeswitch.org wrote: Is solar an option? ;) /b ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CDR loss possibility if FS freezes?
windmills ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] multiple gateways not working?
include
extension name=multiple
condition field=destination_number expression=^(\d+)$
action application=set data=gate_site_id=1/
action application=set
data=effective_caller_id_name=${effective_caller_id_name}/
action application=set
data=effective_caller_id_number=${effective_caller_id_number}/
action application=transfer data=$ XML default/--
action application=bridge data=sofia/default/$...@116.80.80.101/
/condition
condition field=destination_number expression=^(\d+)$
action application=set data=gate_site_id=1/
action application=set
data=effective_caller_id_name=${effective_caller_id_name}/
action application=set
data=effective_caller_id_number=${effective_caller_id_number}/
action application=transfer data=$ XML default/--
action application=bridge data=sofia/default/$...@116.80.80.102/
/condition
/extension
/include
Is this correct for multiple gateways? When I try this the first gateway
works but the second gateway does not work?
What is the solution for this can u help me?
Thanks
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