[Freeswitch-users] How to stop the playback files
Dear All, I have played the list of voice files in playback like the following by using ESL perl module, $conn-execute(set,playback_delimiter=!); $conn-execute(set,playback_sleep_val=100); $conn-playback($sound_path.ivr/ivr-welcome_to_freeswitch.wav!ivr/ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr-you_may_exit_by_hanging_up.wav); In a loop I am checking the DTMF event, if that event comes I should stop the above palyback. How can I do it? Regards, Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Channel variables not being set when FS calls user
Hello, in this page http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide we can read: = Variables: Any variables defined in the domain or user will be defined as channel variables when there is a call to user or when there is an inbound calls from that user. = I can see the channel variables are set when there is an incoming call from the user, but not when FS sends a call to the user. Can someone confirm what is the expected behavior? br, takeshi ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Retrieve conference member state using cli?
Type conference at the cli and the help will be displayed... same can be called over XML RPC. /b On Oct 27, 2009, at 11:51 PM, Lon Baker wrote: Just like you can call conference list for conferences, is there a way to retrieve the profile state of a conference member using the cli or xml/rpc? Lon ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 stop the playback files
http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators Are you using sync or async socket? /b On Oct 28, 2009, at 3:29 AM, velusamy velu wrote: Dear All, I have played the list of voice files in playback like the following by using ESL perl module, $conn-execute(set,playback_delimiter=!); $conn-execute(set,playback_sleep_val=100); $conn-playback($sound_path.ivr/ivr-welcome_to_freeswitch.wav!ivr/ ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr- you_may_exit_by_hanging_up.wav); In a loop I am checking the DTMF event, if that event comes I should stop the above palyback. How can I do it? Regards, Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Channel variables not being set when FS calls user
Yes it is the expected behavior... if you wish to set them on an inbound call TO the user you'll use the set_user api to do so. Its kinda like sudo in FreeSWITCH it'll load up all the variables for that user. /b On Oct 28, 2009, at 3:33 AM, mayamatakeshi wrote: I can see the channel variables are set when there is an incoming call from the user, but not when FS sends a call to the user. Can someone confirm what is the expected behavior? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] db
Hi, can i use sqlserver instead of sqllite. and can two freeswitch servers can share same database(sqllite or sqlserver). any help would be great. Thanks Srinivasula Reddy K ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Channel variables not being set when FS calls user
Got it. Thanks. On Wed, Oct 28, 2009 at 5:57 PM, Brian West br...@freeswitch.org wrote: Yes it is the expected behavior... if you wish to set them on an inbound call TO the user you'll use the set_user api to do so. Its kinda like sudo in FreeSWITCH it'll load up all the variables for that user. /b On Oct 28, 2009, at 3:33 AM, mayamatakeshi wrote: I can see the channel variables are set when there is an incoming call from the user, but not when FS sends a call to the user. Can someone confirm what is the expected behavior? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
Handling of fastStart in CallProceeding is commented out in h323plus library, this is exploration from h323plus developers about this: Yes that should be mera. The problem is that Callproceeding does not always come from the remote it may be generated by the gatekeeper. this is a feature .. called force_callproceeding. It means MERA will send a provisional CallProceeding in order not to timeout on calls that don't respond with that message on time. If this message contains a faststart element it is certanly a bug and it has to be reported to them. MERA where sending fast start elements in the Call proceeding and connect. The call proceeding where not valid and causing the media to fail. well if there is a correct faststart element within a connect message (or alerting or facility or progress), the originator should adjust the media resources accordingly. Here what could went wrong is just the media before the next faststart element in the row. Normally (although valid) EP's do not set Fast Start in Call proceeding so the code was disabled to resolve the MERA issue. well, this is unlikely as fast start element can be included in call proceeding message. The developer's task is to determine whether a call proceeding message is to be trusted or not. Also, provisional call proceeding messages don't have faststart element included! There are equipment (Cisco PGW / HSI) that are sending call proceeding with faststart element and h245Controll (OLC + TCS/MSD) that has to be treated correctly. Unfortunately, just disabling handling of callproceeding faststart element is not a real option... if you wont read bugs file in mod_h323, there is explaned how to enable it. of course i can enable it during build time but this will not solve interop issues later we can encounter... Do you maybe have some sniffs/traces of the wrong call proceeding message ? ...anyhow this is the expected behaviour when a GK/Proxy sends a provisional Call Proceding to the terminator and later it receives the real Call Proceeding carring faststart and h245Controll element within. Entities in the signalling path shall also use the Facility message or the Progress message to convey any new information (such as Q.931 information elements, CallProceeding-UUIE fields, tunnelled non-H.323 protocols, and encapsulated H.245 messages) received in a Call Proceeding message to the other endpoint if the entity has already sent a Call Proceeding message. This will allow the entity, for example, to transmit the fastStart element to facilitate proper establishment of a Fast Connect call and/or a Progress Indicator to indicate the presence of in-band tones and announcements. When using the Facility message to carrying such information extracted from the Call Proceeding message, the reason in the Facility should be set to forwardedElements. in other words: ORIG GKTERM - Setup OLC = Call proceeding (prov)= Setup OLC = Call proceeding (OLC+TCS/MSD) = Facility (OLC+TCS/MSD)= --- normal call establishment scenario follows --- ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 stop the playback files
I am using async socket. Shall I set the regular expression for palyback_terminators? On Wed, Oct 28, 2009 at 2:26 PM, Brian West br...@freeswitch.org wrote: http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators Are you using sync or async socket? /b On Oct 28, 2009, at 3:29 AM, velusamy velu wrote: Dear All, I have played the list of voice files in playback like the following by using ESL perl module, $conn-execute(set,playback_delimiter=!); $conn-execute(set,playback_sleep_val=100); $conn-playback($sound_path.ivr/ivr-welcome_to_freeswitch.wav!ivr/ ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr- you_may_exit_by_hanging_up.wav); In a loop I am checking the DTMF event, if that event comes I should stop the above palyback. How can I do it? Regards, Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] db
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core Please don't cross post if possible. /b On Oct 28, 2009, at 4:07 AM, srinivasula reddy wrote: Hi, can i use sqlserver instead of sqllite. and can two freeswitch servers can share same database(sqllite or sqlserver). any help would be great. Thanks Srinivasula Reddy K ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 stop the playback files
Hello Velusamy! You may also be interested in using the method: http://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits It play a file, gets DTMFs and can be terminated if the maximum number of digits is reached (i.e. 1 digit). Delian Tashev -Original Message- From: Brian West [mailto:br...@freeswitch.org] Sent: Wednesday, October 28, 2009 10:57 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] How to stop the playback files http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators Are you using sync or async socket? /b On Oct 28, 2009, at 3:29 AM, velusamy velu wrote: Dear All, I have played the list of voice files in playback like the following by using ESL perl module, $conn-execute(set,playback_delimiter=!); $conn-execute(set,playback_sleep_val=100); $conn-playback($sound_path.ivr/ivr-welcome_to_freeswitch.wav!ivr/ ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr- you_may_exit_by_hanging_up.wav); In a loop I am checking the DTMF event, if that event comes I should stop the above palyback. How can I do it? Regards, Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Problem with inbound call answered but no sound
What kind of router are you behind? /b On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote: Thanks for the reply, Brian. Did something in FS change between v15183 and v15225 to make this occur? I ask because this same configuration worked OK in the earlier version. Lars ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP provider with extern rtp server
Here is a debug log from a call from an internal phone out to an external (my iPhone with nbr 91316356): http://pastebin.freeswitch.org/108578 Ivan Den 27. okt. 2009 kl. 18:34 skrev Eliot Gable: No, the IP address the media originates from does not need to be tied to the SIP IP address. Can you send a Wireshark capture taken on the FreeSWITCH server of both call legs? Or, if you can, pastebin a debug log from FreeSWITCH console with sofia loglevel set to 9 and siptrace on for any Sofia SIP profiles involved. On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold i...@myrvold.org wrote: The server is on a public IP, so there is no nat issue here. I can also see the rtp messages on wireshark starting just after the 183 Session Progress message on the server, but just in one direction, coming in to the server. So it looks like Freeswitch is stopping the rtp. Is this because the rtp originates from another ip than the sip provider ip? Ivan Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable: Make sure you let their media IPs through your firewall. Also, if you are behind a NAT, check you have things passing to the correct internal address. On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold i...@myrvold.org wrote: I have used a SIP provider for more than a year. A few days ago, he said he was moving to a new server, and asked me to reconfigure. I did, and everything seemed to work fine, until I did an outgoing call to an external telephone. I found out I had no audio, in neither direction. Incoming calls was working fine. My provider said that the rtp is not going through the sip server, as it did earlier, but now through several other IP's. Do I have to do some special configuration to handle that? Ivan ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Can not record session. Media not enabled on channel.
Correct - compiled but did not run. Works fine now. I'll see if the error shows up again and let you know if it does. Thanks, MA Anthony Minessale wrote: won't compile or won't run? maybe you should try rebuilding it. On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz maciej.aniserow...@gmail.com wrote: Sorry, trunk does not compile on win7, here are the details: rev.15247 --- Microsoft Visual C++ Debug Library --- Debug Assertion Failed! - Original Message - *From:* [hidden email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3899478i=0 *To:* [hidden email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3899478i=1 *Sent:* Monday, October 26, 2009 10:32 PM *Subject:* Re: [Freeswitch-users] Can not record session. Media not enabled on channel. On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz [hidden email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3895104i=0 wrote: Yes, I can confirm - this exact error occurs each time when I start recording before the call is answered (just after sending ORIGINATE command) - but I think that's completely understandable that media is not ready for an unanswered call. But... is there any other event that guarantees media to be ready? Update to latest SVN and try again. -MC ___ FreeSWITCH-users mailing list [hidden email]http://n2.nabble.com/user/SendEmail.jtp?type=nodenode=3895104i=1 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: Re: [Freeswitch-users] Can not record session. Media not enabled on channel.http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3899478.html Sent from the freeswitch-users mailing list archivehttp://n2.nabble.com/freeswitch-users-f2379917.htmlat Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp3857858p3906568.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] No RTP destination auto-correction
I'm seeing a strange issue with eyebeam behind NAT. If I make a call from eyebeam to FS, I can see FS receives packets from eyebeam's nat address (confirmed with tcpdump and mod_event_socket DTMF) but it doesn't change the RTP destination to the source of those packets and sends the packets to the private address advertised in the SDP. However, if I do hold/unhold and send some digits, FS logs Auto Changing port from X to Y and starts to send the packets to the correct address. This always work after a hold/unhold + digits but never happen just after the first INVITE transaction. Also, I don't see the problem if eyebeam receives a call from FS. Has anyone seen this? br, takeshi ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] No RTP destination auto-correction
are you = r15256 because an issue with this was fixed in that revision. On Wed, Oct 28, 2009 at 11:35 AM, mayamatakeshi mayamatake...@gmail.comwrote: I'm seeing a strange issue with eyebeam behind NAT. If I make a call from eyebeam to FS, I can see FS receives packets from eyebeam's nat address (confirmed with tcpdump and mod_event_socket DTMF) but it doesn't change the RTP destination to the source of those packets and sends the packets to the private address advertised in the SDP. However, if I do hold/unhold and send some digits, FS logs Auto Changing port from X to Y and starts to send the packets to the correct address. This always work after a hold/unhold + digits but never happen just after the first INVITE transaction. Also, I don't see the problem if eyebeam receives a call from FS. Has anyone seen this? br, takeshi ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No RTP destination auto-correction
have you updated to the latest SVN? /b On Oct 28, 2009, at 11:35 AM, mayamatakeshi wrote: I'm seeing a strange issue with eyebeam behind NAT. If I make a call from eyebeam to FS, I can see FS receives packets from eyebeam's nat address (confirmed with tcpdump and mod_event_socket DTMF) but it doesn't change the RTP destination to the source of those packets and sends the packets to the private address advertised in the SDP. However, if I do hold/unhold and send some digits, FS logs Auto Changing port from X to Y and starts to send the packets to the correct address. This always work after a hold/unhold + digits but never happen just after the first INVITE transaction. Also, I don't see the problem if eyebeam receives a call from FS. Has anyone seen this? br, takeshi ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] SIP UPDATE Method
Thank you very much, Anthony. I really appreciate your help and excellent work of you and your team. Regards, Dorn B. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, October 26, 2009 12:59:01 PM Subject: Re: [Freeswitch-users] SIP UPDATE Method in r15233 i put it back to the way it originally was but I may have to remove that if it causes more problems. We tried to handle update for display updating which was not working but the handler for it was still in place which may have broken some automatic behavior regarding update so I added it back to how it was originally to determine that. On Mon, Oct 26, 2009 at 2:50 PM, Anthony Minessale anthony.miness...@gmail.com wrote: but i think the minimum value you can set is 120 On Mon, Oct 26, 2009 at 2:47 PM, Anthony Minessale anthony.miness...@gmail.com wrote: I think session timers will use invite if there is no update. the session-timeout profile param should control that but you have to double the number you actually want because it sends the new invite at the halfway point. On Mon, Oct 26, 2009 at 2:29 PM, DJB djbin...@yahoo.com wrote: We were trying to see whether we can adjust call duration on Session timers. It was a question from the application developers. I am not sure what they are trying to do exactly. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, October 26, 2009 11:45:31 AM Subject: Re: [Freeswitch-users] SIP UPDATE Method what exactly are you expecting to use it for? We never really supported it anyway. On Mon, Oct 26, 2009 at 12:19 PM, DJB djbin...@yahoo.com wrote: I wonder whether you will consider to put it back on the next version 1.0.5 since 1.0.4 has it? Regards, Dorn B. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, October 26, 2009 8:08:59 AM Subject: Re: [Freeswitch-users] SIP UPDATE Method It was never there before and it caused extreme havoc once we added it so we took it away again. On Sun, Oct 25, 2009 at 12:55 AM, DJB djbin...@yahoo.com wrote: I am wondering why after I update to trunk-15225, the Allow: UPDATE method is no longer there. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Am I missing something here? Thank you. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com
[Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is available for testing!
Hello FreeSWITCHers, The latest FreeSWITCH version is now available for download on the fileshttp://files.freeswitch.org/freeswitch-1.0.5pre3.tar.gzsite. The announcement story is in the main FreeSWITCH http://www.freeswitch.org/node/212 site. Please download and test, and then test some more. We need your feedback. The sooner we get your feedback, the more quickly we can roll the official 1.0.5. Thanks! -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] No RTP destination auto-correction
I'm really sorry, guys. I was just some revisions behind and forgot to make current and test before posting. It's fine now. Thank you. On Thu, Oct 29, 2009 at 1:51 AM, Brian West br...@freeswitch.org wrote: have you updated to the latest SVN? /b On Oct 28, 2009, at 11:35 AM, mayamatakeshi wrote: I'm seeing a strange issue with eyebeam behind NAT. If I make a call from eyebeam to FS, I can see FS receives packets from eyebeam's nat address (confirmed with tcpdump and mod_event_socket DTMF) but it doesn't change the RTP destination to the source of those packets and sends the packets to the private address advertised in the SDP. However, if I do hold/unhold and send some digits, FS logs Auto Changing port from X to Y and starts to send the packets to the correct address. This always work after a hold/unhold + digits but never happen just after the first INVITE transaction. Also, I don't see the problem if eyebeam receives a call from FS. Has anyone seen this? br, takeshi ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] error in loading spidermonkey
Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04 runs on fedora 8 VM. Any clue? thanks, [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary** ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] error in loading spidermonkey
Per http://jira.freeswitch.org/browse/MDXMLINT-23 Try to configure it with --without-libcurl. -- W ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP provider with extern rtp server
On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org wrote: Here is a debug log from a call from an internal phone out to an external (my iPhone with nbr 91316356): http://pastebin.freeswitch.org/108578 Ivan Uh... you wanna try that PB number again? -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] Problem with hangin bri
Okay, obligatory questions: Which version of FreeSWITCH are you running? Which PRI library are you using? Which BRI card are you using? -MC On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawliński ja...@pawlinski.pl wrote: Hi, I have some problems with bri status. I have 3 chanel isdn modem, and zaptel compatible quad bri card. I can invoke calls from my voip phone to cell phone, and vice versa, but when i make inbound and outbound connection in nearly same time something goes wrong with chanells and after few calls all of them has hangup status. There is log about that in attachement, see is already in use waiting for it to become available. phrase. Time of this event is about 14:43:35. Unload and Load open_zap module helped, but its not an solution because of lost connections. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] meaning of created_time channel variable.
On Mon, Oct 26, 2009 at 12:20 AM, velusamy velu velu.techni...@gmail.comwrote: Dear All, What is the value of created_time channel variable? Is this epoch seconds? It's epoch microseconds. Divide by 1,000,000 to get epoch seconds... -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] Problem with inbound call answered but no sound
BroadXent ADSL 8120 è Netscreen 5XP From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, October 28, 2009 7:13 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem with inbound call answered but no sound What kind of router are you behind? /b On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote: Thanks for the reply, Brian. Did something in FS change between v15183 and v15225 to make this occur? I ask because this same configuration worked OK in the earlier version. Lars ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] error in loading spidermonkey
The libcurl is broken on your distro. You can fix this by configuring with --without-libcurl which will use our working in tree copy instead of the broken one from your distro. Mike On Oct 28, 2009, at 1:39 PM, Erwin Davis wrote: Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04 runs on fedora 8 VM. Any clue? thanks, [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/ freeswitch/mod/mod_spidermonkey.so **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary** ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] FreeSWITCH on FreeBSD
Hello! I'm researching a phone system for an organization with 8-12 members. The place has an Internet connection via Verizon FIOS. My first choice of the operating system is FreeBSD (preferably -- on amd64 platform). FreeBSD's port (net/freeswitch) seems nice and currently installs version 1.0.4.3. Although fairly proficient with Unix and FreeBSD in particular, I'm very new to the field of computer telephony, so my questions are most likely to be rather naive. I apologize... * What hardware is known to work well on FreeBSD? * What else -- other than a computer with a phone card -- will I need? * Do end-user phones have to be regular analog phones (all plugged-in to the computer), or IP-phones, such as made by Avaya (connecting via an Ethernet switch)? * How is the computer connecting to the world? Do I plug-in the incoming analog phone line(s) somewhere, or is it going to go through the Internet? * If the connection to the world is over the Internet, don't we need an account with someone else, or is Verizon known to work with FreeSWITCH-based PBX-systems directly? * Should I not simply buy a used Avaya-system like this: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=110450339950_trksid=p2759.l1259 What will I gain and what will I lose (other than personal satisfaction, of course) by going that route instead of building my own box? Thanks a lot! Yours, -mi ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Multi-tenancy context manipulation
Hello, it's my first post, and I'm quite new to Freeswitch... I'm looking for a bit of direction / advice on Multi-tenant set-ups. I have configured Freeswitch as per the fine Multi-tenant guide on the Wiki, but this leaves some aspects such as transfers and IVR still in the default context. For example, transfering a call that is passed to features.xml, or connecting to an extension thorough demo_ivr will fail if the call is left in the default context. Would you recommend a separate features.xml for each tenants context? If the features provided by features.xml and the ivr are the same for all tenants, could you advise on the best way to implement one features.xml for all tenants? Kind regards, Marty Hickman ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Network cable disconnection
Hi, I have configured FreeSwitch and played it by making calls. Every thing went fine. Then I tried to see what happens if network cable is unplugged. Freeswitch shown an error saying ping failed to the gateway and IP address is changed to local system IP. After sometime network cable plugged back in and in another 30 seconds freeswitch has shown it got the IP of gateway and changed back from local IP to gateway IP. After that when I tried to make calls, calls are not going through. Even there is no log displayed in Freeswitch. Please find below the logs... freeswi...@qa360basexp 2009-10-28 16:46:13.156250 [WARNING] sofia.c:2814 Ping failed ihub-trunk 2009-10-28 16:46:13.234375 [INFO] mod_sofia.c:3160 IP change detected [192.168.12.10]-[192.168.20.138] []-[] 2009-10-28 16:46:13.265625 [NOTICE] sofia_glue.c:3444 Reload XML [Success] 2009-10-28 16:46:13.265625 [INFO] switch_time.c:661 Timezone reloaded 0 definitions 2009-10-28 16:46:14.156250 [NOTICE] sofia.c:921 Waiting for worker thread 2009-10-28 16:46:14.156250 [NOTICE] sofia_glue.c:3505 deleted gateway ihub-trunk 2009-10-28 16:46:14.156250 [NOTICE] sofia_reg.c:2122 Added gateway 'ihub-trunk' to profile 'TrunkExternal' 2009-10-28 16:46:14.156250 [NOTICE] sofia.c:2767 Started Profile TrunkExternal [sofia_reg_TrunkExternal] 2009-10-28 16:46:14.156250 [ERR] sofia.c:805 Error Creating SIP UA for profile: TrunkExternal 2009-10-28 16:46:14.156250 [NOTICE] sofia_glue.c:3505 deleted gateway ihub-trunk 2009-10-28 16:46:53.234375 [INFO] mod_sofia.c:3160 IP change detected [192.168.20.138]-[192.168.12.10] []-[] 2009-10-28 16:46:53.281250 [NOTICE] sofia_glue.c:3444 Reload XML [Success] 2009-10-2816:46:53.281250 [INFO] switch_time.c:661 Timezone reloaded 0 definitions --After this there were no messages Calls went through if I reload freeswitch. Another surprise thing is I checked for the active ports in the system before and after cable unplugged. Port 5060 disappeared after cable is plugged out. This doesn't display even after cable is plugged back in. Thanks Regards Suneel * 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
[Freeswitch-users] IVR Intro Clipped
I notice that when I call IVR from the PSTN, the Welcome to Freeswitch... introduction is clipped at the beginning, so it sounds like come to Freeswitch. If I call 5000 internally, then I always hear the full introduction. What can I do to resolve this? My XML config looks like: extension name=ivr_demo condition field=destination_number expression=5000 action application=answer/ action application=start_dtmf/ action application=ivr data=demo_ivr/ /condition /extension Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] IVR Intro Clipped
Sleep 1000 ms... we usually bring up media too fast before the other end is ready. /b On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote: I notice that when I call IVR from the PSTN, the Welcome to Freeswitch... introduction is clipped at the beginning, so it sounds like come to Freeswitch. If I call 5000 internally, then I always hear the full introduction. What can I do to resolve this? My XML config looks like: extension name=ivr_demo condition field=destination_number expression=5000 action application=answer/ action application=start_dtmf/ action application=ivr data=demo_ivr/ /condition /extension Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] SIP provider with extern rtp server
Oh, what happened to it? Anyway, here is a new pb: http://pastebin.freeswitch.org/10867 Ivan Den 28. okt. 2009 kl. 19:12 skrev Michael Collins: On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org wrote: Here is a debug log from a call from an internal phone out to an external (my iPhone with nbr 91316356): http://pastebin.freeswitch.org/108578 Ivan Uh... you wanna try that PB number again? -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem with hangin bri
Hi I'm also working on this project, so i can answer your questions Which version of FreeSWITCH are you running? FreeSWITCH Version 1.0.trunk (15246) Which PRI library are you using? openzap Native stack openzap.conf [span zt BRI1] trunk_type = bri b-channel = 1-2 d-channel= 3 openzap.conf configuration name=openzap.conf description=OpenZAP Configuration settings param name=debug value=0/ /settings pri_spans span id=1 param name=mode value=user/ param name=dialect value=q931/ param name=dialplan value=XML/ param name=context value=default/ /span /pri_spans /configuration Which BRI card are you using? Producer: http://www.phoniceq.com/ card model: http://quadbri.phoniceq.com/ Card instalation process (instruction from producer) 1) download bristuff staff from http://junghanns.net/downloads/bristuff-0.4.0-RC3h.tar.gz or http://junghanns.net/downloads/bristuff-0.3.0-PRE-1y-z.tar.gz unpack it and go to bristuff-* 2) download patcher from http://quadbri.phoniceq.com/driver/bristuff/qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch patch it using patch -p0 qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch 3) you can check card using zttest (result should be 99.x) Producer has said, that we are first client, it wants to use this card in freeswitch we are using 1 port (S/T interface). Our NT is NT1 plus 2b1q Thanks Michael Collins pisze: Okay, obligatory questions: Which version of FreeSWITCH are you running? Which PRI library are you using? Which BRI card are you using? -MC On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawliński ja...@pawlinski.pl mailto:ja...@pawlinski.pl wrote: Hi, I have some problems with bri status. I have 3 chanel isdn modem, and zaptel compatible quad bri card. I can invoke calls from my voip phone to cell phone, and vice versa, but when i make inbound and outbound connection in nearly same time something goes wrong with chanells and after few calls all of them has hangup status. There is log about that in attachement, see is already in use waiting for it to become available. phrase. Time of this event is about 14:43:35. Unload and Load open_zap module helped, but its not an solution because of lost connections. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto:FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Mariusz Kołodziejczyk Advanced Developing Architecture S.C. tel. : +48 609 381 316 e-mail : mariusz_k...@wp.pl ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] IVR Intro Clipped
I modified my dialplan as shown, but the clipping persists. Should the sleep be placed somewhere else? extension name=ivr_demo condition field=destination_number expression=5000 action application=sleep data=1000\ action application=answer/ action application=start_dtmf/ action application=ivr data=demo_ivr/ /condition /extension Best Regards, Jerry -Original Message- From: Brian West [mailto:br...@freeswitch.org] Sent: Wednesday, October 28, 2009 1:51 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] IVR Intro Clipped Sleep 1000 ms... we usually bring up media too fast before the other end is ready. /b On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote: I notice that when I call IVR from the PSTN, the Welcome to Freeswitch... introduction is clipped at the beginning, so it sounds like come to Freeswitch. If I call 5000 internally, then I always hear the full introduction. What can I do to resolve this? My XML config looks like: extension name=ivr_demo condition field=destination_number expression=5000 action application=answer/ action application=start_dtmf/ action application=ivr data=demo_ivr/ /condition /extension Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] IVR Intro Clipped
Hello Yes. Answer first. --FC On Wed, Oct 28, Jerry Richards wrote: I modified my dialplan as shown, but the clipping persists. Should the sleep be placed somewhere else? extension name=ivr_demo condition field=destination_number expression=5000 action application=sleep data=1000\ action application=answer/ action application=start_dtmf/ action application=ivr data=demo_ivr/ /condition /extension Best Regards, Jerry -Original Message- From: Brian West [mailto:br...@freeswitch.org] Sent: Wednesday, October 28, 2009 1:51 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] IVR Intro Clipped Sleep 1000 ms... we usually bring up media too fast before the other end is ready. /b On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote: I notice that when I call IVR from the PSTN, the Welcome to Freeswitch... introduction is clipped at the beginning, so it sounds like come to Freeswitch. If I call 5000 internally, then I always hear the full introduction. What can I do to resolve this? My XML config looks like: extension name=ivr_demo condition field=destination_number expression=5000 action application=answer/ action application=start_dtmf/ action application=ivr data=demo_ivr/ /condition /extension Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] IVR Intro Clipped
Jerry, Put the sleep after the answer. That should fix it. Shelby Jerry Richards wrote: I modified my dialplan as shown, but the clipping persists. Should the sleep be placed somewhere else? extension name=ivr_demo condition field=destination_number expression=5000 action application=sleep data=1000\ action application=answer/ action application=start_dtmf/ action application=ivr data=demo_ivr/ /condition /extension Best Regards, Jerry -Original Message- From: Brian West [mailto:br...@freeswitch.org] Sent: Wednesday, October 28, 2009 1:51 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] IVR Intro Clipped Sleep 1000 ms... we usually bring up media too fast before the other end is ready. /b On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote: I notice that when I call IVR from the PSTN, the Welcome to Freeswitch... introduction is clipped at the beginning, so it sounds like come to Freeswitch. If I call 5000 internally, then I always hear the full introduction. What can I do to resolve this? My XML config looks like: extension name=ivr_demo condition field=destination_number expression=5000 action application=answer/ action application=start_dtmf/ action application=ivr data=demo_ivr/ /condition /extension Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Estimating Call Capacity
That's really nice looking software. Thanks for the pointer. Cliff On Tue, 2009-10-27 at 16:06 -0400, Gregory Boehnlein wrote: I'm fond of Vqmanager from ManagEngine. It is a passive SIP monitor. I.E. you mirror the ports that your FS or Asterisk boxes, and VQmanager sniffs the mirrors, tracking all sorts of good data. You can install it on a Centos box, and get a free trial. http://www.manageengine.com/products/vqmanager/index.html What is really cool is that it actually monitors the RTP/RTCP as well as all of the SIP headers and archives the calls, so you can look at calls from several days ago and see EXACTLY what happened on them. I have used this extensively to pinpoint bad Level 3 and X/O media gateways.. Much better than trying to sniff packets in real-time and MAYBE catch a problem.. I've also used it to find/fix several SIP issues w/ odd endpoints.. Very easy to see.. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch- users-boun...@lists.freeswitch.org] On Behalf Of Cliff Wells Sent: Tuesday, October 27, 2009 3:52 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity A little off-topic, but since call-capacity is the subject, what are people using to analyze their CDR's to discover this? I'm handling about 30k calls per day but have only a bandwidth-based guesstimate of the peak number of concurrent calls I'm handling. If there's an open source solution, I'd appreciate a pointer. Regards, Cliff On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote: Although, FYI, I just benchmarked mod_xml_curl on a separate web app server from FS with FS on a Dell R710 with their current best processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32 GB memory. The web app server is less than half the power of the R710. I maxed the web app server at 300 calls per second (both setting up and tearing down) and the R710 running FS was 65% idle. No audio was being proxied through FS, though. If I were running the web app server on an equivalent R710, they probably would have been on-par with each other in performance. Extrapolating, I expect that in such a case I should be able to get at least 650 CPS out of FS, though for production I would probably limit it to 400 CPS or less so I leave room for miscellaneous tasks. I maxed out the R710 at over 16,000 simultaneous calls (again, no audio proxying) but the only reason I couldn't do more was because I hit some sort of thread creation limit in Linux. There was about 17 GB of memory used for this many calls. This should give you some ballpark idea of what you can accomplish with FS. At some point, I will track down and resolve the thread creation issue, at which time I believe call limits will be limited either by a complex combination of available memory, the speed of the processor, the cost of thread context switching, calls per second setup rate, and call duration. -- Eliot Gable -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Monday, October 26, 2009 4:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur vinuth.madi...@gmail.com wrote: Here are a few benchmarks that I had stumbled upon. http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+ x2200+M2 Please remember NO benchmarks are endorsed by the FS community or developers, because there are just too many variables, and a simple figure is just useful for marketing hype, not for real dimensioning. You MUST do your own benchmarking, so you get an idea about how to dimension for your own use case and hardware. Thanks, Vinuth. On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org wrote: I highly doubt it... You can wait for someone to post their results but in the end you'll have to do your own load testing because not everyone's numbers will jive with your use case. Which is the reason the project never posts or endorses a set call count. /b On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: Are there any benchmarking test results available publicly? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org
Re: [Freeswitch-users] Estimating Call Capacity
Hi Shelby, Thanks! That's pretty useful. I also note that this same info is available from the CLI, although I was curious as to what some of the numbers indicated: UP 0 years, 13 days, 19 hours, 25 minutes, 48 seconds, 887 milliseconds, 975 microseconds 529509 session(s) since startup 26 session(s) 0/30 1000 session(s) max Specifically, 26 sessions 0/30... I take it this means there are 26 current sessions, but I'm unsure of what the 0/30 means. Cliff On Tue, 2009-10-27 at 15:19 -0500, Shelby Ramsey wrote: Cliff, Try using xml_rpc ... status or show channels will give you what you need. SDR Cliff Wells wrote: A little off-topic, but since call-capacity is the subject, what are people using to analyze their CDR's to discover this? I'm handling about 30k calls per day but have only a bandwidth-based guesstimate of the peak number of concurrent calls I'm handling. If there's an open source solution, I'd appreciate a pointer. Regards, Cliff On Mon, 2009-10-26 at 18:01 -0400, Eliot Gable wrote: Although, FYI, I just benchmarked mod_xml_curl on a separate web app server from FS with FS on a Dell R710 with their current best processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32 GB memory. The web app server is less than half the power of the R710. I maxed the web app server at 300 calls per second (both setting up and tearing down) and the R710 running FS was 65% idle. No audio was being proxied through FS, though. If I were running the web app server on an equivalent R710, they probably would have been on-par with each other in performance. Extrapolating, I expect that in such a case I should be able to get at least 650 CPS out of FS, though for production I would probably limit it to 400 CPS or less so I leave room for miscellaneous tasks. I maxed out the R710 at over 16,000 simultaneous calls (again, no audio proxying) but the only reason I couldn't do more was because I hit some sort of thread creation limit in Linux. There was about 17 GB of memory used for this many calls. This should give you some ballpark idea of what you can accomplish with FS. At some point, I will track down and resolve the thread creation issue, at which time I believe call limits will be limited either by a complex combination of available memory, the speed of the processor, the cost of thread context switching, calls per second setup rate, and call duration. -- Eliot Gable -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Monday, October 26, 2009 4:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur vinuth.madi...@gmail.com wrote: Here are a few benchmarks that I had stumbled upon. http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2 Please remember NO benchmarks are endorsed by the FS community or developers, because there are just too many variables, and a simple figure is just useful for marketing hype, not for real dimensioning. You MUST do your own benchmarking, so you get an idea about how to dimension for your own use case and hardware. Thanks, Vinuth. On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org wrote: I highly doubt it... You can wait for someone to post their results but in the end you'll have to do your own load testing because not everyone's numbers will jive with your use case. Which is the reason the project never posts or endorses a set call count. /b On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: Are there any benchmarking test results available publicly? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___
Re: [Freeswitch-users] Estimating Call Capacity
sessions per second now/max /b On Oct 28, 2009, at 4:58 PM, Cliff Wells wrote: current sessions, but I'm unsure of what the 0/30 means. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Problem with hangin bri
Thanks. Can you collect debug logs of this happening? See http://wiki.freeswitch.org/wiki/Reporting_Bugs for helpful tips on collecting debug information. Use pastebin to dump all the log info and reply here with the link. We don't have too many BRI users but I believe there are a few so hopefully we can help you get up and running. -MC 2009/10/28 Mariusz Kołodziejczyk mariusz_k...@wp.pl Hi I'm also working on this project, so i can answer your questions Which version of FreeSWITCH are you running? FreeSWITCH Version 1.0.trunk (15246) Which PRI library are you using? openzap Native stack openzap.conf [span zt BRI1] trunk_type = bri b-channel = 1-2 d-channel= 3 openzap.conf configuration name=openzap.conf description=OpenZAP Configuration settings param name=debug value=0/ /settings pri_spans span id=1 param name=mode value=user/ param name=dialect value=q931/ param name=dialplan value=XML/ param name=context value=default/ /span /pri_spans /configuration Which BRI card are you using? Producer: http://www.phoniceq.com/ card model: http://quadbri.phoniceq.com/ Card instalation process (instruction from producer) 1) download bristuff staff from http://junghanns.net/downloads/bristuff-0.4.0-RC3h.tar.gz or http://junghanns.net/downloads/bristuff-0.3.0-PRE-1y-z.tar.gz unpack it and go to bristuff-* 2) download patcher from http://quadbri.phoniceq.com/driver/bristuff/qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch patch it using patch -p0 qozap-bristuff-0.3.0-PRE-1y-j-enableLEDS.patch 3) you can check card using zttest (result should be 99.x) Producer has said, that we are first client, it wants to use this card in freeswitch we are using 1 port (S/T interface). Our NT is NT1 plus 2b1q Thanks Michael Collins pisze: Okay, obligatory questions: Which version of FreeSWITCH are you running? Which PRI library are you using? Which BRI card are you using? -MC On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawliński ja...@pawlinski.pl mailto:ja...@pawlinski.pl wrote: Hi, I have some problems with bri status. I have 3 chanel isdn modem, and zaptel compatible quad bri card. I can invoke calls from my voip phone to cell phone, and vice versa, but when i make inbound and outbound connection in nearly same time something goes wrong with chanells and after few calls all of them has hangup status. There is log about that in attachement, see is already in use waiting for it to become available. phrase. Time of this event is about 14:43:35. Unload and Load open_zap module helped, but its not an solution because of lost connections. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto:FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Mariusz Kołodziejczyk Advanced Developing Architecture S.C. tel. : +48 609 381 316 e-mail : mariusz_k...@wp.pl ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Multi-tenancy context manipulation
On Wed, Oct 28, 2009 at 11:17 AM, Martin Hickman ma...@makolink.com wrote: Hello, it's my first post, and I'm quite new to Freeswitch... I'm looking for a bit of direction / advice on Multi-tenant set-ups. I have configured Freeswitch as per the fine Multi-tenant guide on the Wiki, but this leaves some aspects such as transfers and IVR still in the default context. For example, transfering a call that is passed to features.xml, or connecting to an extension thorough demo_ivr will fail if the call is left in the default context. Would you recommend a separate features.xml for each tenants context? If the features provided by features.xml and the ivr are the same for all tenants, could you advise on the best way to implement one features.xml for all tenants? Kind regards, Marty Hickman How many contexts and tenants are you looking at? Also, will that change in the future, that is, will you be adding tenants? Keep that in mind because you might be looking at a good case for using mod_xml_curl... -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] mod_opal - call charged before H.225 connect
On 2009-10-06 22:47 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: Hi, now i have another questions about fs and not sip protocols, call scheme like this - user1-sip-fs-sip-user2, as i see if in sip profile enable inbound-proxy-media SDP is goes end-to-end if sip is on both call legs, if call goes throuch another protocol module, for example my mod_h323 (call scheme like user1-sip-fs-h323-someH323Ep) there is no way to negotiate codec because i have no SDP from called party for example and have no way to insert it on sip leg, or i don't see this way? C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is available for testing!
Is the plan to run 1.0.5 as a stable branch with bug fix updates or will it be the case of just follow the trunk like it is with any problems you encounter in 1.0.4? 2009/10/29 Michael Collins m...@freeswitch.org Hello FreeSWITCHers, The latest FreeSWITCH version is now available for download on the fileshttp://files.freeswitch.org/freeswitch-1.0.5pre3.tar.gzsite. The announcement story is in the main FreeSWITCH http://www.freeswitch.org/node/212 site. Please download and test, and then test some more. We need your feedback. The sooner we get your feedback, the more quickly we can roll the official 1.0.5. -- Craig Askings Network Engineer | Over the Wire Pty Ltd cr...@overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is available for testing!
Craig Askings cr...@overthewire.com.au wrote: Is the plan to run 1.0.5 as a stable branch with bug fix updates or will it be the case of just follow the trunk like it is with any problems you encounter in 1.0.4? I can't speak for the developers, but I would expect the current policy to continue: bug fixes are placed in trunk, and bug reports should be submitted against whatever the current trunk revision is at the time of lodging the bug reports. The maintainers of FreeSWITCH are very effective, in general, at keeping SVN trunk relatively stable. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Freeswitch seems to doesn't reknow dial tone after the first call using OpenZAP (analog spans)
Hi to all, this is my first message here! I'm trying to setup a freeswitch box using 1.0.4 version compiled from sources, ubuntu 8.04.03 lts with lastest updates and lastest zaptel modules/tools compiled from sources. As hardware i'm using an OpenVox A800P. My problem is the following: if i do a call trought openzap it works, but when i hang down and try a second call it doesn't work! Looking with ztmonitor and hearing recorded audio file it seems that freeswitch do the job on the first outgoing call but, on the second, it does nothing because it doesn't reknow dial tone (but it is there). If i call the freeswitch mananged number (i can answer or not, nothing change) i can do another call, but, in any case, the successive call doesn't start. Looking with ztmonitor 1 -v i see that before the first call i get the special dial tone (a continue 425 frequency to state that the line is free to dial sended by the provider) but after the first call is done, before the second, i didn't see any special dial tone ... that appears for less than a sec when freeswitch starts to compose the number. Here logs for first call -- 2009-10-28 21:38:39.616112 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-10-28 21:38:39.616112 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DIALING 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_INIT 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT 2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz) State Change CS_INIT - CS_ROUTING 2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT going to sleep 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_ROUTING 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING 2009-10-28 21:38:39.616112 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz CHANNEL ROUTING 2009-10-28 21:38:39.616112 [DEBUG] switch_ivr_originate.c:63 (OpenZAP/1:1/xyz) State Change CS_ROUTING - CS_CONSUME_MEDIA 2009-10-28 21:38:39.616112 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING going to sleep 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA 2009-10-28 21:38:39.616112 [DEBUG] switch_core_state_machine.c:503 (OpenZAP/1:1/xyz) State CONSUME_MEDIA 2009-10-28 21:38:40.475897 [DEBUG] ozmod_analog.c:655 Detected tone DIAL on 1:1 2009-10-28 21:38:40.475897 [DEBUG] mod_openzap.c:1216 got FXO sig 1:1 [TONE_DETECTED] -- Here logs for second call -- 2009-10-28 21:39:07.105916 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-10-28 21:39:07.105916 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DIALING 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_INIT 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT 2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/xyz) State Change CS_INIT - CS_ROUTING 2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:481 (OpenZAP/1:1/xyz) State INIT going to sleep 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_ROUTING 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING 2009-10-28 21:39:07.118644 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/xyz CHANNEL ROUTING 2009-10-28 21:39:07.118644 [DEBUG] switch_ivr_originate.c:63 (OpenZAP/1:1/xyz) State Change CS_ROUTING - CS_CONSUME_MEDIA 2009-10-28 21:39:07.118644 [DEBUG] switch_core_session.c:932 Send signal OpenZAP/1:1/xyz [BREAK] 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:484 (OpenZAP/1:1/xyz) State ROUTING going to sleep 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:398 (OpenZAP/1:1/xyz) Running State Change CS_CONSUME_MEDIA 2009-10-28 21:39:07.118644 [DEBUG] switch_core_state_machine.c:503 (OpenZAP/1:1/xyz) State CONSUME_MEDIA -- Following my configs! Really thanks for the help, Best Regards, Daniele --- ZAPTEL.CONF --- # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: OPVXA1200/0 OpenVox A1200P/A800P Board 1 (MASTER) fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxoks=5 # Global data loadzone= it defaultzone = it ---
Re: [Freeswitch-users] error in loading spidermonkey
fixed. Thanks, On Wed, Oct 28, 2009 at 11:51 AM, Michael Jerris m...@jerris.com wrote: The libcurl is broken on your distro. You can fix this by configuring with --without-libcurl which will use our working in tree copy instead of the broken one from your distro. Mike On Oct 28, 2009, at 1:39 PM, Erwin Davis wrote: Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04 runs on fedora 8 VM. Any clue? thanks, [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/ freeswitch/mod/mod_spidermonkey.so **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary** ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] pass arguments into javascript
Hi, new to javascript. I tried to pass two arguments into javascript, action application=javascript data=test.js $1 $2/ In test.js, I tried to use argv[1] to retrieve $1 and argv[2] to retrieve $2, however, the javascript test.js complained about argv[] as undefined variables. How to retrieve the passing arguments in a javascript same as the case above. 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] SIP provider with extern rtp server
See that 200 OK that keeps coming in over and over and over and over again? That's because they never received your ACK. If you can turn on sofia loglevel to 9 and then watch where you send the ACK, you will probably have your answer to why the other system did not receive it. If you're still not sure what's going on, post another pastebin with sofia loglevel set to 9. On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold i...@myrvold.org wrote: Oh, what happened to it? Anyway, here is a new pb: http://pastebin.freeswitch.org/10867 Ivan Den 28. okt. 2009 kl. 19:12 skrev Michael Collins: On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org wrote: Here is a debug log from a call from an internal phone out to an external (my iPhone with nbr 91316356): http://pastebin.freeswitch.org/108578 Ivan Uh... you wanna try that PB number again? -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] pass arguments into javascript
Should work fine. I use this: var calling_num = argv[0]; var called_num = argv[1]; Are you sure you actually had valid data in $1 and $2? Try to call it from the CLI: jsrun test.js testvar1 testvar2 On Wed, Oct 28, 2009 at 10:22 PM, Erwin Davis davis.er...@gmail.com wrote: Hi, new to javascript. I tried to pass two arguments into javascript, action application=javascript data=test.js $1 $2/ In test.js, I tried to use argv[1] to retrieve $1 and argv[2] to retrieve $2, however, the javascript test.js complained about argv[] as undefined variables. How to retrieve the passing arguments in a javascript same as the case above. 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 -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
