Re: [Freeswitch-users] [CRIT] mod_event_socket.c:337 Lost 8456 events!
Anthony, thank you very much for your response. The daemon that was reading the events froze, so apparently that was the source of the problem and your explanation fits perfectly. On Mon, Dec 28, 2009 at 12:47 PM, Anthony Minessale anthony.miness...@gmail.com wrote: most likely cause would be connecting a socket then not regularly reading from it causing the buffer to fill up. any event socket connection must select on the socket and do regular read attempts or all the events will accumulate on the server side until some sanity check is reached and it begins to throw them away, the fist time there is room in this buffer again (when you consume some from the socket leaving space in the queue) it will report how many have been lost since the last read. One way to cause this would be suspend fs_cli with ctl-z and bring it back to the foreground after some time. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] [CRIT] mod_event_socket.c:337 Lost 8456 events!
I just got into the fs cli and when I ran a 'show calls' I got the following message: 2009-12-24 09:58:20.058365 [CRIT] mod_event_socket.c:337 Lost 8456 events! What does this mean? does it mean the event_socket did not report 8456 events? Why could this happen? The answer to this is pretty critical to me, as I make and monitor calls through the socket. Thanks for your help! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Javascript system calls
Hi, I wanted to know what is the javascript equivalent of lua's os.execute(). I need to run a command from within a js script. Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Equivalent of canreinvite?
I'm looking for the equivalent configuration parameter or option of Asterisk's canreinvite (http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite). Is there anything like this for configuring a gateway? (there's no info about it on the wiki). Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Equivalent of canreinvite?
Thanks, but I would like to keep FS in the media path. What would be the equivalent of an Asterisk sip.conf's canreinvite=no? On Tue, Dec 15, 2009 at 1:45 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Closest thing I've found: http://wiki.freeswitch.org/wiki/Channel_Variables#bypass_media_after_bridge ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Equivalent of canreinvite?
Thanks! On Tue, Dec 15, 2009 at 3:12 PM, Frank Carmickle fr...@carmickle.com wrote: On Tue, Dec 15, Nicolas Brenner wrote: Thanks, but I would like to keep FS in the media path. What would be the equivalent of an Asterisk sip.conf's canreinvite=no? It's that way by default. Fs wants to listen for events on a channel in the default config. See bind_meta_app. --FC ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Changing User-Agent String
I had a voip provider which wouldn't accept calls from Freeswitch because of the user-agent string. I had to change it to Asterisk and then everything worked. Nico On Wed, Nov 18, 2009 at 12:21 PM, Brian West br...@freeswitch.org wrote: you do realize that is NOT the purpose of the user-agent string... changing might break things in some people's configs due to some assumptions made about the user agent on the far side for interop purposes... its your choice to change it but it servers NO purpose doing so. /b On Nov 18, 2009, at 9:17 AM, Ujjval Karihaloo wrote: Not sure I am the only one changing *User-Agent*….but I just want a way for our Customers to know the purpose of the server when they talk to it. There is FreeSwitch written into the SDP “o” line as well…which I don’t care about, I want to have something in there that identifies the purpose of the server. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 gateway registration
Is there some way to make FS register with the gateway that is rejecting the authentication? is it FS or the SIP server at fault? Why would X-Lite work and FS not? Thanks again for your time and help. On Tue, Oct 6, 2009 at 5:46 PM, Brian West br...@freeswitch.org wrote: btw My mistake it doesn't assume auth it just calculates the response hash differently on this case where qop isn't present. /b On Oct 6, 2009, at 4:22 PM, Nicolas Brenner wrote: What does the qop parameter stand for? Apparently because of that parameter, FS sends a new REGISTER including 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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 gateway registration
You are missing the point, it is only rejecting auth for FS, Asterisk and X-Lite work fine with the same config for that gateway. On Wed, Oct 7, 2009 at 10:20 AM, Brian West br...@freeswitch.org wrote: I would suspect its a PEBKAC. I mean if you could register to a gateway that rejected auth... what purpose would auth serve in the first place? /b On Oct 7, 2009, at 8:48 AM, Nicolas Brenner wrote: Is there some way to make FS register with the gateway that is rejecting the authentication? is it FS or the SIP server at fault? Why would X-Lite work and FS not? Thanks again for your time and help. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 gateway registration
That happens with both gateways though, one works and the other doesn't. Would the rport have anything to do with the registration failing? The big difference to me is that the working gateway replies a 401 Unauthorized containing: WWW-Authenticate: Digest realm=pxextmy.redvoiss.net, nonce=4acac8fe248a9075a13773274684392a65a40240, qop=auth. Whereas the non-working gateway's 401 has: WWW-Authenticate: Digest realm=216.72.10.39, nonce=4acac08249c439decb2bea539282faf755c80b0c. What does the qop parameter stand for? Apparently because of that parameter, FS sends a new REGISTER including this: Authorization: Digest username=x, realm=pxextmy.redvoiss.net, nonce=4acac8fe248a9075a13773274684392a65a40240, cnonce=h1DCSizTEi2eMQAdCe9KJA, algorithm=MD5, uri=sip: pxextmy.redvoiss.net, response=05adb2a7f9d7772e57dc846257484f5d, qop=auth, nc=0001. Instead, on the non-working gateway case, FS sends a REGISTER with this: Authorization: Digest username=y, realm=216.72.10.39, nonce=4acac08249c439decb2bea539282faf755c80b0c, algorithm=MD5, uri=sip: 216.72.10.39, response=8311db7666779df89d5223e16a611826. Notice the absence of the qop and nc parameters. I'm guessing the lack of those parameters causes the gateway (SIP server) to use another nonce and hence reject the mismatching REGISTER. BTW, registration from an X-Lite softphone works. Thanks! Nicolas On Tue, Oct 6, 2009 at 10:31 AM, Brian West br...@freeswitch.org wrote: This looks like you have an ALG messing with packets... notice it says rport 5080 but we are sending to 5060. /b On Oct 5, 2009, at 11:42 PM, Nicolas Brenner wrote: Ignore my previous email, the traces were incomplete, got much better (and complete) traces with ngrep (found a suggestion from Brian in the list archive, thanks!) The gateway that registers: - http://pastebin.freeswitch.org/10607 The one that doesn't: - http://pastebin.freeswitch.org/10608 Thanks again for your time and help! Nicolas On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner nico...@medularis.comwrote: There was no sane way of doing that, so I ended up logging the trace from the cli. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 gateway registration
There was no sane way of doing that, so I ended up logging the trace from the cli. Here's the bad registration: - http://pastebin.freeswitch.org/10605 Here's the good one: - http://pastebin.freeswitch.org/10606 I am not sure if the second one is complete because for some reason the first few packages don't appear on the console when doing 'sofia profile external restart reloadxml' and 'sofia profile external siptrace on' or viceversa. Anyway, thanks for your time, and I hope those traces help in figuring out what's going on. Nicolas PS: Is there anyway to get the same format from a pcap dump as with the siptrace feature on the cli? On Mon, Oct 5, 2009 at 12:20 PM, Michael Collins m...@freeswitch.org wrote: On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner nico...@medularis.comwrote: Mike, how exactly should I format the file? I got the pcap file, how do I convert it to text so that you can easily read it? you can open it with wireshark, follow the TCP or UDP stream, then just copy paste the text as needed... -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 gateway registration
Ignore my previous email, the traces were incomplete, got much better (and complete) traces with ngrep (found a suggestion from Brian in the list archive, thanks!) The gateway that registers: - http://pastebin.freeswitch.org/10607 The one that doesn't: - http://pastebin.freeswitch.org/10608 Thanks again for your time and help! Nicolas On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner nico...@medularis.comwrote: There was no sane way of doing that, so I ended up logging the trace from the cli. Here's the bad registration: - http://pastebin.freeswitch.org/10605 Here's the good one: - http://pastebin.freeswitch.org/10606 I am not sure if the second one is complete because for some reason the first few packages don't appear on the console when doing 'sofia profile external restart reloadxml' and 'sofia profile external siptrace on' or viceversa. Anyway, thanks for your time, and I hope those traces help in figuring out what's going on. Nicolas PS: Is there anyway to get the same format from a pcap dump as with the siptrace feature on the cli? On Mon, Oct 5, 2009 at 12:20 PM, Michael Collins m...@freeswitch.orgwrote: On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner nico...@medularis.comwrote: Mike, how exactly should I format the file? I got the pcap file, how do I convert it to text so that you can easily read it? you can open it with wireshark, follow the TCP or UDP stream, then just copy paste the text as needed... -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 gateway registration
Here it is: - http://pastebin.freeswitch.org/10582 (it is the pcap file I sent on the first email of this thread, converted to text with 'tshark -V -r') On Sun, Oct 4, 2009 at 5:40 PM, Michael Jerris m...@jerris.com wrote: can you send a link of a text sip trace please. On Oct 1, 2009, at 3:29 PM, Nicolas Brenner wrote: Any ideas about this? The SIP provider is offering H323, but I'm not quite sure about that, is mod_opal working right? Thanks! Nicolas On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner nico...@medularis.comwrote: Anthony, thanks. Below are my config files for the two gateways from the sip trace. Both files are located in conf/directory/default. - redvoiss.xml (the one that works) include user id=gateway_redvoiss gateways gateway name=redvoiss-pp param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=pxextmy.redvoiss.net/ param name=realm value=pxextmy.redvoiss.net/ param name=proxy value=pxextmy.redvoiss.net/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2010/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - orange.xml (the one that doesn't work) include user id=gateway_orange gateways gateway name=orange param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=216.72.10.39/ param name=realm value=216.72.10.39/ param name=proxy value=216.72.10.39/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2011/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - If I remove the register=true param for the non-working gateway, I don't get the registration error on the cli, but then all call attempts get rejected with a 401 Unauthorized, and I get a hangup cause of NORMAL_UNSPECIFIED. Best, Nicolas On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale anthony.miness...@gmail.com wrote: 900 level errors are sofia internal errors so probably something is wrong with your gateway config xml. if you want to send it with any critical info replaced with XXX maybe we can see the issue for you. On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner nico...@medularis.comwrote: Hello everyone, I am trying to add a gateway, but after configuring it just like the others gateways I have, it is failing to register with a message like this: 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed with status Operation has no matching challenge [904]. failure #1 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed Registration, setting retry to 10 seconds. I captured the sip traffic and noticed that when trying to register with one gateway (the one that works), I get a Trying reply immediately followed by a 401 Unauthorized which contains a WWW-Authenticate: digest with a qop=auth parameter. Then Freeswitch replies with a second REGISTER including a large Authorization: digest section with cnonce and nc=0001 parameters. The gateway which doesn't register, doesn't send the qop=auth parameter together with the 401 Unauthorized, and then Freeswitch sends a Authorization: digest section on the second REGISTER with no cnonce or nc parameters. I know very little abouth SIP, so I'm wondering what this qop=auth parameter means and how does it affect the registration process. Is there any way to do without the qop=auth parameter? Also, I tried registering with X-Lite directly to the gateway, and it worked, so it appears to be a problem in the Freeswitch/gateway combination. (Note: X-Lite sends an Authorization: digest section on the _first_ REGISTER, apparently this makes a difference) Attached is a sip trace for the registration traffic when doing sofia profile external restart reloadxml on the cli, captured with tshark -i eth0 -o rtp.heuristic_rtp: TRUE -w /tmp/capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or t38' ___ FreeSWITCH-users mailing list
Re: [Freeswitch-users] Problem with gateway registration
Mike, how exactly should I format the file? I got the pcap file, how do I convert it to text so that you can easily read it? On Sun, Oct 4, 2009 at 6:48 PM, Michael Jerris m...@jerris.com wrote: I've never been able to read these, why exactly do I need a text protocol to be decoded for me? Ends up being too much noise so I just don't bother. Mike On Oct 4, 2009, at 6:19 PM, Nicolas Brenner wrote: Here it is: - http://pastebin.freeswitch.org/10582 (it is the pcap file I sent on the first email of this thread, converted to text with 'tshark -V -r') On Sun, Oct 4, 2009 at 5:40 PM, Michael Jerris m...@jerris.com wrote: can you send a link of a text sip trace please. On Oct 1, 2009, at 3:29 PM, Nicolas Brenner wrote: Any ideas about this? The SIP provider is offering H323, but I'm not quite sure about that, is mod_opal working right? Thanks! Nicolas On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner nico...@medularis.comwrote: Anthony, thanks. Below are my config files for the two gateways from the sip trace. Both files are located in conf/directory/default. - redvoiss.xml (the one that works) include user id=gateway_redvoiss gateways gateway name=redvoiss-pp param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=pxextmy.redvoiss.net/ param name=realm value=pxextmy.redvoiss.net/ param name=proxy value=pxextmy.redvoiss.net/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2010/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - orange.xml (the one that doesn't work) include user id=gateway_orange gateways gateway name=orange param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=216.72.10.39/ param name=realm value=216.72.10.39/ param name=proxy value=216.72.10.39/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2011/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - If I remove the register=true param for the non-working gateway, I don't get the registration error on the cli, but then all call attempts get rejected with a 401 Unauthorized, and I get a hangup cause of NORMAL_UNSPECIFIED. Best, Nicolas On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale anthony.miness...@gmail.com wrote: 900 level errors are sofia internal errors so probably something is wrong with your gateway config xml. if you want to send it with any critical info replaced with XXX maybe we can see the issue for you. On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner nico...@medularis.com wrote: Hello everyone, I am trying to add a gateway, but after configuring it just like the others gateways I have, it is failing to register with a message like this: 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed with status Operation has no matching challenge [904]. failure #1 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed Registration, setting retry to 10 seconds. I captured the sip traffic and noticed that when trying to register with one gateway (the one that works), I get a Trying reply immediately followed by a 401 Unauthorized which contains a WWW-Authenticate: digest with a qop=auth parameter. Then Freeswitch replies with a second REGISTER including a large Authorization: digest section with cnonce and nc=0001 parameters. The gateway which doesn't register, doesn't send the qop=auth parameter together with the 401 Unauthorized, and then Freeswitch sends a Authorization: digest section on the second REGISTER with no cnonce or nc parameters. I know very little abouth SIP, so I'm wondering what this qop=auth parameter means and how does it affect the registration process. Is there any way to do without the qop=auth parameter? Also, I tried registering with X-Lite directly to the gateway, and it worked, so it appears to be a problem in the Freeswitch/gateway combination. (Note: X-Lite sends an Authorization: digest section
Re: [Freeswitch-users] Problem with gateway registration
Any ideas about this? The SIP provider is offering H323, but I'm not quite sure about that, is mod_opal working right? Thanks! Nicolas On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner nico...@medularis.comwrote: Anthony, thanks. Below are my config files for the two gateways from the sip trace. Both files are located in conf/directory/default. - redvoiss.xml (the one that works) include user id=gateway_redvoiss gateways gateway name=redvoiss-pp param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=pxextmy.redvoiss.net/ param name=realm value=pxextmy.redvoiss.net/ param name=proxy value=pxextmy.redvoiss.net/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2010/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - orange.xml (the one that doesn't work) include user id=gateway_orange gateways gateway name=orange param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=216.72.10.39/ param name=realm value=216.72.10.39/ param name=proxy value=216.72.10.39/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2011/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - If I remove the register=true param for the non-working gateway, I don't get the registration error on the cli, but then all call attempts get rejected with a 401 Unauthorized, and I get a hangup cause of NORMAL_UNSPECIFIED. Best, Nicolas On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale anthony.miness...@gmail.com wrote: 900 level errors are sofia internal errors so probably something is wrong with your gateway config xml. if you want to send it with any critical info replaced with XXX maybe we can see the issue for you. On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner nico...@medularis.comwrote: Hello everyone, I am trying to add a gateway, but after configuring it just like the others gateways I have, it is failing to register with a message like this: 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed with status Operation has no matching challenge [904]. failure #1 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed Registration, setting retry to 10 seconds. I captured the sip traffic and noticed that when trying to register with one gateway (the one that works), I get a Trying reply immediately followed by a 401 Unauthorized which contains a WWW-Authenticate: digest with a qop=auth parameter. Then Freeswitch replies with a second REGISTER including a large Authorization: digest section with cnonce and nc=0001 parameters. The gateway which doesn't register, doesn't send the qop=auth parameter together with the 401 Unauthorized, and then Freeswitch sends a Authorization: digest section on the second REGISTER with no cnonce or nc parameters. I know very little abouth SIP, so I'm wondering what this qop=auth parameter means and how does it affect the registration process. Is there any way to do without the qop=auth parameter? Also, I tried registering with X-Lite directly to the gateway, and it worked, so it appears to be a problem in the Freeswitch/gateway combination. (Note: X-Lite sends an Authorization: digest section on the _first_ REGISTER, apparently this makes a difference) Attached is a sip trace for the registration traffic when doing sofia profile external restart reloadxml on the cli, captured with tshark -i eth0 -o rtp.heuristic_rtp: TRUE -w /tmp/capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or t38' Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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
Re: [Freeswitch-users] Freeswitch Failover
Just finished downloading the whole torrent and seeding now. On Tue, Sep 29, 2009 at 3:04 PM, Dan White dwh...@olp.net wrote: I've downloaded 0Kb in 56 minutes. Anyone mind setting up a seeder? I'm on a pretty fast connection. While waiting, I found the following discussions: I found the following threads while searching for redundancy and failover: http://lists.freeswitch.org/pipermail/freeswitch-users/2007-September/001499.html (round robin) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-April/013069.html (load balancing, scalability) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/017349.html (a discussion of setting up multiple FS boxes in multiple datacenters) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-December/009723.html (comparing FS to a commercial product) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011770.html (WAN redundancy) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/010924.html (Comparing against commercial SBCs) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006710.html (Hardware redundancy) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-January/001934.html (Generic heartbeat failover question) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-August/017972.html (Failing over an originating call based on originate_timeout) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-August/005740.html (HA via Ultra Monkey) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-April/013097.html (gateway redundancy) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-August/005688.html (redundancy via openser) http://lists.freeswitch.org/pipermail/freeswitch-users/2007-August/001363.html (failover based on xml-curl) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-July/016904.html (multiple gateways) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-February/002146.html (dead gateway detection) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-June/003979.html (load balancing and failover via mod_xml_curl) http://lists.freeswitch.org/pipermail/freeswitch-users/2007-September/001445.html (multiple gateways) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-June/015363.html (hot failover) http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/014071.html (failover extension) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/008532.html (an interesting discussion on failing over registrations) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-October/thread.html#7734 (clustering via DNS SRV) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-November/007845.html (hot failover and 6 9s!) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/005874.html (high availability clustering) http://lists.freeswitch.org/pipermail/freeswitch-users/2008-October/007376.html (outbound call failover) Of these, the two discussing hot failover are closest to what I was referring to. Last night, we upgraded our Acme Packet Session Border Controllers. We upgraded to a new major version, including rebooting each one in turn. No registrations (and presumably no calls) were dropped during the process. They are tied together via ethernet. An SBC's function is admittedly much more narrowly focused. Has there been any work in FS in this area? I presume there is a mechanism for dumping a registration database from a FS box. Is there a way to load that database into another one? Thanks. On 29/09/09 10:34 -0700, Michael Collins wrote: This topic has been beaten to death recently. Search the archives for things like redundancy and failover and you'll see lots of discussions. The bottom line is that your needs will dictate how much time, effort, and money you are willing to sink into this. If you want professional assistance then email consult...@freeswitch.org. If you want to do your own research then I'd say start with the ClueCon videos, specifically Day 3, Presentation #5. Here's the torrent: http://files.freeswitch.org/cluecon_2009/presentations/cluecon_2009.torrent -- Dan White BTC Broadband ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Problem with gateway registration
Hello everyone, I am trying to add a gateway, but after configuring it just like the others gateways I have, it is failing to register with a message like this: 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed with status Operation has no matching challenge [904]. failure #1 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed Registration, setting retry to 10 seconds. I captured the sip traffic and noticed that when trying to register with one gateway (the one that works), I get a Trying reply immediately followed by a 401 Unauthorized which contains a WWW-Authenticate: digest with a qop=auth parameter. Then Freeswitch replies with a second REGISTER including a large Authorization: digest section with cnonce and nc=0001 parameters. The gateway which doesn't register, doesn't send the qop=auth parameter together with the 401 Unauthorized, and then Freeswitch sends a Authorization: digest section on the second REGISTER with no cnonce or nc parameters. I know very little abouth SIP, so I'm wondering what this qop=auth parameter means and how does it affect the registration process. Is there any way to do without the qop=auth parameter? Also, I tried registering with X-Lite directly to the gateway, and it worked, so it appears to be a problem in the Freeswitch/gateway combination. (Note: X-Lite sends an Authorization: digest section on the _first_ REGISTER, apparently this makes a difference) Attached is a sip trace for the registration traffic when doing sofia profile external restart reloadxml on the cli, captured with tshark -i eth0 -o rtp.heuristic_rtp: TRUE -w /tmp/capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or t38' Thanks! Nicolas or_vs_red.pcap Description: Binary data ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 gateway registration
Anthony, thanks. Below are my config files for the two gateways from the sip trace. Both files are located in conf/directory/default. - redvoiss.xml (the one that works) include user id=gateway_redvoiss gateways gateway name=redvoiss-pp param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=pxextmy.redvoiss.net/ param name=realm value=pxextmy.redvoiss.net/ param name=proxy value=pxextmy.redvoiss.net/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2010/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - orange.xml (the one that doesn't work) include user id=gateway_orange gateways gateway name=orange param name=username value=xxx/ param name=password value=xxx/ param name=from-domain value=216.72.10.39/ param name=realm value=216.72.10.39/ param name=proxy value=216.72.10.39/ param name=from-user value=xxx/ param name=caller-id-in-from value=false/ param name=expire-seconds value=600/ param name=register value=true/ param name=retry_seconds value=5/ param name=extension value=2011/ param name=context value=public/ param name=codec-prefs value=G729/ param name=rfc2833-pt value=101/ /gateway /gateways params param name=password value=4321/ /params /user /include - If I remove the register=true param for the non-working gateway, I don't get the registration error on the cli, but then all call attempts get rejected with a 401 Unauthorized, and I get a hangup cause of NORMAL_UNSPECIFIED. Best, Nicolas On Tue, Sep 29, 2009 at 2:22 PM, Anthony Minessale anthony.miness...@gmail.com wrote: 900 level errors are sofia internal errors so probably something is wrong with your gateway config xml. if you want to send it with any critical info replaced with XXX maybe we can see the issue for you. On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner nico...@medularis.comwrote: Hello everyone, I am trying to add a gateway, but after configuring it just like the others gateways I have, it is failing to register with a message like this: 2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed with status Operation has no matching challenge [904]. failure #1 2009-09-29 12:54:40.906798 [WARNING] sofia_reg.c:364 orange Failed Registration, setting retry to 10 seconds. I captured the sip traffic and noticed that when trying to register with one gateway (the one that works), I get a Trying reply immediately followed by a 401 Unauthorized which contains a WWW-Authenticate: digest with a qop=auth parameter. Then Freeswitch replies with a second REGISTER including a large Authorization: digest section with cnonce and nc=0001 parameters. The gateway which doesn't register, doesn't send the qop=auth parameter together with the 401 Unauthorized, and then Freeswitch sends a Authorization: digest section on the second REGISTER with no cnonce or nc parameters. I know very little abouth SIP, so I'm wondering what this qop=auth parameter means and how does it affect the registration process. Is there any way to do without the qop=auth parameter? Also, I tried registering with X-Lite directly to the gateway, and it worked, so it appears to be a problem in the Freeswitch/gateway combination. (Note: X-Lite sends an Authorization: digest section on the _first_ REGISTER, apparently this makes a difference) Attached is a sip trace for the registration traffic when doing sofia profile external restart reloadxml on the cli, captured with tshark -i eth0 -o rtp.heuristic_rtp: TRUE -w /tmp/capture.pcap -b filesize:51200 -b files:100 -R 'sip or rtp or icmp or dns or rtcp or t38' Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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
Re: [Freeswitch-users] ALLOTTED_TIMEOUT hangup cause?
Did a little more digging, ALLOTTED_TIMEOUT has an error code of 602 according to the Wiki (http://wiki.freeswitch.org/wiki/Hangup_causes) nevertheless that code is not covered in RFC 4497 ( http://tools.ietf.org/html/rfc4497) On Mon, Sep 21, 2009 at 8:41 PM, Nicolas Brenner nico...@medularis.comwrote: Hi, Today, while trying to bridge some calls I started to get a ALLOTTED_TIMEOUT hangup cause on the second leg. I looked for info on the Wiki and Google, but I couldn't find a detailed explanation. Does anybody know what does it mean exactly? Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] ALLOTTED_TIMEOUT hangup cause?
Hi, Today, while trying to bridge some calls I started to get a ALLOTTED_TIMEOUT hangup cause on the second leg. I looked for info on the Wiki and Google, but I couldn't find a detailed explanation. Does anybody know what does it mean exactly? Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 compiling esl for use with freepbx v3
I gave up on compiling esl, I got a bunch of errors, there were several people on the list with problems and apparently no straight solution, especially for php-esl. I am now using a ruby library, posted here by Diego Viola I believe. On Tue, Sep 1, 2009 at 2:33 AM, Michael Collins m...@freeswitch.org wrote: Did the simple make in the libs/esl directory run properly? Just curious. I'll have to defer to the Ubuntu gurus out there for thoughts on what else could be wrong. -MC On Mon, Aug 31, 2009 at 10:43 PM, Harondel J. Sibble h...@pdscc.comwrote: Haven't had any responses, anyone have any ideas on the problem with compiling the ESL modules as below? On 23 Aug 2009 at 11:51, Harondel J. Sibble wrote: Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server, then went to install FreePBX v3, I've gotten all the prerequisities in the wizard fixed except for ESL As per http://wiki.freeswitch.org/wiki/Event_Socket_Library http://wiki.freeswitch.org/wiki/Event_Socket I go into my FS source dir /home/sibbleh/freeswitch-1.0.4/libs/esl Run make and then sudo make phpmod-install and I get $ sudo make phpmod-install make MYLIB=../libesl.a SOLINK=-shared -Xlinker -x CFLAGS=- I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused- variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=- I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable CXX_CFLAGS= -C php make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php' g++ -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable - I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM - I/usr/include/php5/Zend -I/usr/include/php5/ext - I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 - Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o cc1plus: warnings being treated as errors esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1047: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1073: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1141: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1172: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1198: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1234: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1269: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1294: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1346: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1403: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1441: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1478: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1508: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1538: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*, zval**, zval*,
[Freeswitch-users] Not receiving DTMF
Hi, I'm trying to get dtmf input, but I'm not getting anything. What I discovered though, is that my provider is at fault, since when I switched to another voip provider, everything started to work beautifully. My question is: since my provider is not doing RC2833 dtmf (even though they say they do), is there another way to get dtmf to work? I'm doing everything in a javascript file, so I tried doing: - session.setVariable(dtmf_type,info); Also: - session.setVariable(dtmf_type,rfc2833); And: - session.execute(start_dtmf); But none worked. The voip provider that works would be ideal, but the calls are twice as expensive, hence besides testing, I wouldn't use them for a real case scenario. BTW, anyone know of a good quality VoIP provider with low rates for termination to Santiago, Chile and Chilean cell phones? Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 trying to use PHP ESL
Hi, I tried the ESL.i modification and when I do make reswig I get:
make -C php reswig
make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php'
rm -f esl_wrap.* ESL.so php_ESL.* ESL.php
swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o esl_wrap.cpp
../ESL.i
swig error : Unrecognized option -php5
Use 'swig -help' for available options.
make[1]: *** [esl_wrap.cpp] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php'
make: *** [reswig] Error 2
On Mon, Aug 10, 2009 at 8:34 AM, Tristan Mahé t.m...@telemaque.fr wrote:
Hello Mike,
No problem to post a patch, but it would break perl/python/etc... as
actually there are some functions defined in ESL.i, wouldn't it ?
I don't know about swig, never used that.
Maybe we could have two swig files instead ? one for generating php and one
for the other languages ?
I'm on IRC if you want to discuss it.
Gled.
Michael Jerris a écrit :
Can you please post a patch to Jira.freswitch.org and assign it to me.
Mike
On Aug 10, 2009, at 5:00 AM, Tristan Mahé t.m...@telemaque.fr wrote:
Morning guys,
Yes the latest make swigall broke php code generation.
Simple workaround:
edit libs/esl/ESL.i to this content:
--Cut--
%{
#include esl.h
#include esl_oop.h
%}
%include esl_oop.h
-Cut---
and make reswig
It should be a swig bug, but I'm not sure.
Regards,
Gled
Andrew Thompson a écrit :
On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
Hi,
I'm trying to get started with the ESL using PHP. I compiled the ESL, then
phpmod according to the wiki instructions, but then when I try the examples
in the libs/esl/php dir, they fail saying:
PHP Fatal error: Cannot redeclare ESLconnection::__construct() in
/usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
Checking ESL.php on line 132, I see there are several different declarations
for the function __construct() with different parameters each, which makes
sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
higher than that or something? What could be the problem?
Someone in the IRC channel mentioned this too. I looked at it briefly
and it looks like the latest 'swigall' screwed it up. The original
reporter said he'd file a jira, but you may want to check yourself and
if not make one yourself. In the meantime, the previous version of the
file was reported to work if you really need it.
Andrew
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Re: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
I changed the script to set hangup_after_bridge to false, but still the same
thing happens, I get this on the console:
2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup
sofia/external/00569xxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/00569xxx [KILL]
2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/00569xxx [BREAK]
2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398
(sofia/external/00569xxx) Running State Change CS_HANGUP
2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL
2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *** Leg2:
NORMAL_CLEARING ***
The second to last line comes from the script, and prints the hangup_cause
of he session, instead of getting ORIGINATOR_CANCEL, I'm getting
NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?
Thanks!
Nicolas
On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner nico...@medularis.comwrote:
Hi Matt,
Actually I'm explicitly setting hangup_after_bridge to true, think setting
it to false would help? I'm going to try that.
Here's the JS code:
(Note: session.getVariable() doesn't work, FS complains saying it is not a
function, also tried self.session.getVariable() - that's what the wiki says
- and FS complains that self does not exist)
var uuid = argv[0]; // Call identifier
var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
var greeting_snd = /var/audio/alert.wav;
console_log(notice, *** STARTING C2C Call ***\n);
timeout = 30;
console_log(notice, *** DIALING +dialstr1+ ***\n);
//var stUsRing = session.getVariable(us-ring); // This doesn't work,
self.session.getVariable doesn't work either
var stUsRing = %(2000,4000,440,480);
// Create new_session
new_session = new Session(originate_str1);
console_log(notice, *** Leg1: + new_session.cause +
***\n);
if (new_session.ready()) {
// log to the console
console_log(notice, *** Leg1 (+dialstr1+) CONNECTED!
***\n);
console_log(notice, *** Playing greeting sound:
+greeting_snd+ ***\n);
new_session.execute(sleep, 100);
new_session.execute(playback, greeting_snd);
// Originate second call and bridge
originate_str2 =
{ignore_early_media=true,originate_timeout=+timeout+,hangup_after_bridge=true,medularis_uuid=+uuid+,c2c_call=true,leg=2}+dialstr2;
// Create new_session
new_session.execute(bridge, originate_str2);
console_log(notice, *** Leg2: + new_session.cause +
***\n);
if (new_session.ready()) {
console_log(notice, *** Leg2 (+dialstr2+)
CONNECTED! ***\n);
}
}
exit();
Thanks!
Nicolas
On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong mattdf...@gmail.com wrote:
Hi Nicolas,
do you have a copy of the .js code you can paste. I would guess tho, that
ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
false. Just a guess tho.
Hangup causes can be found here:
http://wiki.freeswitch.org/wiki/Hangup_causes
http://wiki.freeswitch.org/wiki/Hangup_causes --matt
hello hunter - hosted predictive dialer voice broadcasting
http://www.hellohunter.com
On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner nico...@medularis.comwrote:
I'm bridging 2 calls in a javascript file, I originate the first call and
then execute a bridge with an origination string for the second call. If I
hangup the first call while trying to make the second call, I get this on
the console:
2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL
But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE
Re: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
That variable is not available, it is not included with the
CHANNEL_HANGUP_COMPLETE event info. However I discovered that when the
bridge does not work, there are two CHANNEL_HANGUP_COMPLETE events, one for
each leg, nevertheless for some reason the daemon I have watching the events
misses the second leg event, so I was only seeing the result of the first
leg hangup, which is NORMAL_CLEARING, and the second event's hangup_cause is
ORIGINATOR_CANCEL. I don't know why my daemon is missing the event though.
I'll have to dig into this further.
On Fri, Aug 7, 2009 at 1:28 PM, Phillip Jones pjinthe...@gmail.com wrote:
What does
bridge_hangup_cause
give you?
On Fri, Aug 7, 2009 at 12:43 PM, Nicolas Brenner nico...@medularis.com
wrote:
I changed the script to set hangup_after_bridge to false, but still the
same thing happens, I get this on the console:
2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup
sofia/external/00569xxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/00569xxx [KILL]
2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/00569xxx [BREAK]
2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398
(sofia/external/00569xxx) Running State Change CS_HANGUP
2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL
2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *** Leg2:
NORMAL_CLEARING ***
The second to last line comes from the script, and prints the
hangup_cause of he session, instead of getting ORIGINATOR_CANCEL, I'm
getting NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?
Thanks!
Nicolas
On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner nico...@medularis.com
wrote:
Hi Matt,
Actually I'm explicitly setting hangup_after_bridge to true, think
setting it to false would help? I'm going to try that.
Here's the JS code:
(Note: session.getVariable() doesn't work, FS complains saying it is not
a function, also tried self.session.getVariable() - that's what the wiki
says - and FS complains that self does not exist)
var uuid = argv[0]; // Call identifier
var dialstr1 = argv[1]; // Dial string obtained from previous call to
LCR
var dialstr2 = argv[2]; // Dial string obtained from previous call to
LCR
var greeting_snd = /var/audio/alert.wav;
console_log(notice, *** STARTING C2C Call ***\n);
timeout = 30;
console_log(notice, *** DIALING +dialstr1+ ***\n);
//var stUsRing = session.getVariable(us-ring); // This doesn't work,
self.session.getVariable doesn't work either
var stUsRing = %(2000,4000,440,480);
// Create new_session
new_session = new Session(originate_str1);
console_log(notice, *** Leg1: + new_session.cause +
***\n);
if (new_session.ready()) {
// log to the console
console_log(notice, *** Leg1 (+dialstr1+)
CONNECTED! ***\n);
console_log(notice, *** Playing greeting sound:
+greeting_snd+ ***\n);
new_session.execute(sleep, 100);
new_session.execute(playback, greeting_snd);
// Originate second call and bridge
originate_str2 =
{ignore_early_media=true,originate_timeout=+timeout+,hangup_after_bridge=true,medularis_uuid=+uuid+,c2c_call=true,leg=2}+dialstr2;
// Create new_session
new_session.execute(bridge, originate_str2);
console_log(notice, *** Leg2: + new_session.cause +
***\n);
if (new_session.ready()) {
console_log(notice, *** Leg2 (+dialstr2+)
CONNECTED! ***\n);
}
}
exit();
Thanks!
Nicolas
On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong mattdf...@gmail.com
wrote:
Hi Nicolas,
do you have a copy of the .js code you can paste. I would guess tho,
that ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge
to false. Just a guess tho.
Hangup causes can be found here:
http://wiki.freeswitch.org/wiki/Hangup_causes
--matt
hello hunter - hosted predictive dialer voice broadcasting
http://www.hellohunter.com
On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner nico...@medularis.com
wrote:
I'm bridging 2 calls in a javascript file, I originate the first call
[Freeswitch-users] Error trying to use PHP ESL
Hi, I'm trying to get started with the ESL using PHP. I compiled the ESL, then phpmod according to the wiki instructions, but then when I try the examples in the libs/esl/php dir, they fail saying: PHP Fatal error: Cannot redeclare ESLconnection::__construct() in /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132 Checking ESL.php on line 132, I see there are several different declarations for the function __construct() with different parameters each, which makes sense, but doens't work. I am using PHP 5.1.6, is there a required minimum higher than that or something? What could be the problem? Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Which event contains ORIGINATOR_CANCEL?
I'm bridging 2 calls in a javascript file, I originate the first call and then execute a bridge with an origination string for the second call. If I hangup the first call while trying to make the second call, I get this on the console: 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal sofia/external/005622170039 [KILL] 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal sofia/external/005622170039 [BREAK] 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed. Cause: ORIGINATOR_CANCEL But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see NORMAL_CLEARING. And the variable_originate_disposition has a value of failure. Where can I get the detail of the call/bridge failure due to 'ORIGINATOR_CANCEL' as reported through the console? Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Which event contains ORIGINATOR_CANCEL?
Hi Matt,
Actually I'm explicitly setting hangup_after_bridge to true, think setting
it to false would help? I'm going to try that.
Here's the JS code:
(Note: session.getVariable() doesn't work, FS complains saying it is not a
function, also tried self.session.getVariable() - that's what the wiki says
- and FS complains that self does not exist)
var uuid = argv[0]; // Call identifier
var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
var greeting_snd = /var/audio/alert.wav;
console_log(notice, *** STARTING C2C Call ***\n);
timeout = 30;
console_log(notice, *** DIALING +dialstr1+ ***\n);
//var stUsRing = session.getVariable(us-ring); // This doesn't work,
self.session.getVariable doesn't work either
var stUsRing = %(2000,4000,440,480);
// Create new_session
new_session = new Session(originate_str1);
console_log(notice, *** Leg1: + new_session.cause +
***\n);
if (new_session.ready()) {
// log to the console
console_log(notice, *** Leg1 (+dialstr1+) CONNECTED!
***\n);
console_log(notice, *** Playing greeting sound:
+greeting_snd+ ***\n);
new_session.execute(sleep, 100);
new_session.execute(playback, greeting_snd);
// Originate second call and bridge
originate_str2 =
{ignore_early_media=true,originate_timeout=+timeout+,hangup_after_bridge=true,medularis_uuid=+uuid+,c2c_call=true,leg=2}+dialstr2;
// Create new_session
new_session.execute(bridge, originate_str2);
console_log(notice, *** Leg2: + new_session.cause +
***\n);
if (new_session.ready()) {
console_log(notice, *** Leg2 (+dialstr2+)
CONNECTED! ***\n);
}
}
exit();
Thanks!
Nicolas
On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong mattdf...@gmail.com wrote:
Hi Nicolas,
do you have a copy of the .js code you can paste. I would guess tho, that
ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
false. Just a guess tho.
Hangup causes can be found here:
http://wiki.freeswitch.org/wiki/Hangup_causes
http://wiki.freeswitch.org/wiki/Hangup_causes --matt
hello hunter - hosted predictive dialer voice broadcasting
http://www.hellohunter.com
On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner nico...@medularis.comwrote:
I'm bridging 2 calls in a javascript file, I originate the first call and
then execute a bridge with an origination string for the second call. If I
hangup the first call while trying to make the second call, I get this on
the console:
2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL
But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
NORMAL_CLEARING. And the variable_originate_disposition has a value of
failure. Where can I get the detail of the call/bridge failure due to
'ORIGINATOR_CANCEL' as reported through the console?
Thanks!
Nicolas
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Re: [Freeswitch-users] How to distinguish between the different type of call rejections from Javascript?
I'm bridging 2 calls in a javascript file, I originate the first call and then execute a bridge with an origination string for the second call. If I hangup the first call while trying to make the second call, I get this on the console: 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal sofia/external/005622170039 [KILL] 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal sofia/external/005622170039 [BREAK] 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed. Cause: ORIGINATOR_CANCEL But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see NORMAL_CLEARING. And the variable_originate_disposition has a value of failure. Where can I get the detail of knowing the call/bridge failed because of 'ORIGINATOR_CANCEL' as reported through the console? Thanks! Nicolas the value of variable_originate_disposition at the events level and when I have an origination failure due to 'ORIGINATOR_CANCEL On Wed, Aug 5, 2009 at 6:16 PM, Michael Collins m...@freeswitch.org wrote: On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi raffaele.p.gu...@gmail.com wrote: interesting! what values can contain variable_originate_disposition? And can I set them manually in a script to reject a call simulating user busy or call rejected? A lua example? Thanks, Raffaele Start here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup And note the link to the hangup causes. As far as Lua, I'm not sure there's a good reason to do it there. Could you give us pseudo code example of what you're thinking of doing? -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] Async JS functions?
Thank you Keith, I'll try that as well and let you know. On Fri, Jul 31, 2009 at 9:51 AM, Keith Laaks kei...@voxtelecom.co.zawrote: Hi Nicolas, I wonder if session.execute( sched_broadcast”, “+1 /path/file.wav aleg”) followed immediately by your ‘new Session’ and ‘bridge’ would do the trick ? Not sure if/how “sched_broadcast” functions when the call has not yet been bridged though… Let us know.. Best Regards Keith *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Nicolas Brenner *Sent:* 30 July 2009 20:31 *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Async JS functions? Hi, I have a small JS script that calls a phonenumber, when the call is answered it plays a wave file, then it calls a second phonenumber and bridges the calls. Is it possible to make wave-playing async, so that the second call is generated as soon as the first is picked up? Right now the wave file takes about 2 secs to play, but I need to extend that time, and I don't want to delay the second call. Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Async JS functions?
Hi, I have a small JS script that calls a phonenumber, when the call is answered it plays a wave file, then it calls a second phonenumber and bridges the calls. Is it possible to make wave-playing async, so that the second call is generated as soon as the first is picked up? Right now the wave file takes about 2 secs to play, but I need to extend that time, and I don't want to delay the second call. Thanks! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Async JS functions?
Matthew, Anthony and Michael, thank you very much, seems like you gave me exactly the info I needed! On Thu, Jul 30, 2009 at 4:25 PM, Michael Collins m...@freeswitch.org wrote: On Thu, Jul 30, 2009 at 12:36 PM, Nicolas Brenner nico...@medularis.comwrote: Thanks, I'll try that. How can I play a wav to an active call through the socket? When you are on a socket you can do just about anything you could do at the CLI. Look at all the uuid_XXX commands. Example: uuid_displace uuid -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
[Freeswitch-users] Best way to bridge 2 calls with LCR?
I would like to originate 2 calls from FS and then bridge them. There's no incoming call so I think there's no dialplan involved. What I'd like to do now is apply lcr rules to these calls. I've come up with 2 options so far: 1) call lcr through the socket twice (once for each phonenumber) and then originate the calls through the socket too 2) have a javascript file which runs the actions above, run the script through the socket with 'jsrun' How would you do it? For what I've read on the list, usually the recommended way is to stay away from javascript as much as possible because it is not as efficient as doing everything from the dialplan. Does this mean the first option is the best? or is there a dialplan way of doing it? Thank you very much for your help! Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Best way to bridge 2 calls with LCR?
That looks like a good way to go about it.
How can I access channel variables through the socket using the api? I mean,
how do I recover the value of ${lcr_auto_route}? I would need to add some
other variables, like ignore_early_media=true and a uuid that 'links' the
two calls so I can track it listening for events.
Thanks!
Nicolas
On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker r...@rupa.com wrote:
lcr api command doesn't really return a usable dialstring (it was
originally done for debug purposes). I could add an as xml option if
needed...
Anyway, to do this from the dialplan:
remember that originate's usage is:
-USAGE call url exten|application_name(app_args) [dialplan]
[context] [cid_name] [cid_num] [timeout_sec]
so, the first argument is the call url and the second would be an
extension. so:
1) execute lcr for the first leg of the call
2) execute originate with:
originate ${lcr_auto_route} extension
extension just needs to match something in your dialplan.
In extension, you'd do another lcr lookup and then bridge to that leg's
${lcr_auto_route} value.
On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner
nico...@medularis.comwrote:
I would like to originate 2 calls from FS and then bridge them. There's no
incoming call so I think there's no dialplan involved.
What I'd like to do now is apply lcr rules to these calls. I've come up
with 2 options so far:
1) call lcr through the socket twice (once for each phonenumber) and then
originate the calls through the socket too
2) have a javascript file which runs the actions above, run the script
through the socket with 'jsrun'
How would you do it?
For what I've read on the list, usually the recommended way is to stay
away from javascript as much as possible because it is not as efficient as
doing everything from the dialplan. Does this mean the first option is the
best? or is there a dialplan way of doing it?
Thank you very much for your help!
Nicolas
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Re: [Freeswitch-users] Best way to bridge 2 calls with LCR?
Thank you very much for the offer, but I don't want to bother you with this.
I can just parse the string returned by lcr and get the gateway, that's all
I really need to create my complete originate command.
I am using the socket api because it is easier for me to understand how to
do it, nevertheless I'd really like to know how to do it with the dialplan.
What I don't understand very well about using the dialplan for this, is how
to do the first originate command (which I need to do using the socket api).
What puzzles me is that according to the originate syntax, I need to use an
extension or call an application, yet for the first call I would have to use
a dummy extension as I only need to hit the dialplan section that calls lcr
once to originate the first call with an extension that hits the section of
the dialplan where lcr gets called again and the calls get bridged.
I'm thinking something like this:
1) call originate from socket api to hit dialplan section that does all the
work (this originate command is what I don't understand, is there another
way of hitting the dialplan besides calling originate?)
2) hit dialplan section which calls lcr for first number and bridges to an
extension
3) the extension calls lcr fir the second number and originates the second
call
On steps 2 and 3 I could just use set data to set the additional variables I
need. The first step is what troubles me.
Thank you!
Nicolas
On Tue, Jul 21, 2009 at 12:54 PM, Rupa Schomaker r...@rupa.com wrote:
Ok, if you want to do it from the socket api, then I need to make a 'as
xml' option to mod_lcr and give you lcr_auto_route as one of the nodes in
the returned xml. Then you can do your own substitution in the originate
line... In that case, you'd call lcr twice and do:
originate lcr_auto_route1 bridge(lcr_auto_route2)
How soon do you need this?
On Tue, Jul 21, 2009 at 11:27 AM, Nicolas Brenner
nico...@medularis.comwrote:
That looks like a good way to go about it.
How can I access channel variables through the socket using the api? I
mean, how do I recover the value of ${lcr_auto_route}? I would need to add
some other variables, like ignore_early_media=true and a uuid that 'links'
the two calls so I can track it listening for events.
Thanks!
Nicolas
On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker r...@rupa.com wrote:
lcr api command doesn't really return a usable dialstring (it was
originally done for debug purposes). I could add an as xml option if
needed...
Anyway, to do this from the dialplan:
remember that originate's usage is:
-USAGE call url exten|application_name(app_args) [dialplan]
[context] [cid_name] [cid_num] [timeout_sec]
so, the first argument is the call url and the second would be an
extension. so:
1) execute lcr for the first leg of the call
2) execute originate with:
originate ${lcr_auto_route} extension
extension just needs to match something in your dialplan.
In extension, you'd do another lcr lookup and then bridge to that leg's
${lcr_auto_route} value.
On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner nico...@medularis.com
wrote:
I would like to originate 2 calls from FS and then bridge them. There's
no incoming call so I think there's no dialplan involved.
What I'd like to do now is apply lcr rules to these calls. I've come up
with 2 options so far:
1) call lcr through the socket twice (once for each phonenumber) and
then originate the calls through the socket too
2) have a javascript file which runs the actions above, run the script
through the socket with 'jsrun'
How would you do it?
For what I've read on the list, usually the recommended way is to stay
away from javascript as much as possible because it is not as efficient as
doing everything from the dialplan. Does this mean the first option is the
best? or is there a dialplan way of doing it?
Thank you very much for your help!
Nicolas
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Re: [Freeswitch-users] Best way to bridge 2 calls with LCR?
Now I understand! thank you very much for your explanation, very clear!
On Tue, Jul 21, 2009 at 2:21 PM, Rupa Schomaker r...@rupa.com wrote:
Well, the as xml is something I've been meaning to do, so I'm gonna get
that checked in today sometime anyway. If you want to do any programmatic
processing of the lcr data, the as xml is the way to go rather than parsing
the strings.
As for originate + lcr You can use the loopback endpoint and do it all
in the dialplan:
originate loopback/firstnumber secondnumber
This will hit your dialplan with firstnumber first which you can lcr
route. Then when that call establishes, it'll hit the dialplan with the
second number which will also be routed through lcr.
Is that more what you are looking for?
This way all the 'routing' logic can be done via the dialplan.
On Tue, Jul 21, 2009 at 1:00 PM, Nicolas Brenner nico...@medularis.comwrote:
Thank you very much for the offer, but I don't want to bother you with
this.
I can just parse the string returned by lcr and get the gateway, that's
all I really need to create my complete originate command.
I am using the socket api because it is easier for me to understand how to
do it, nevertheless I'd really like to know how to do it with the dialplan.
What I don't understand very well about using the dialplan for this, is
how to do the first originate command (which I need to do using the socket
api). What puzzles me is that according to the originate syntax, I need to
use an extension or call an application, yet for the first call I would have
to use a dummy extension as I only need to hit the dialplan section that
calls lcr once to originate the first call with an extension that hits the
section of the dialplan where lcr gets called again and the calls get
bridged.
I'm thinking something like this:
1) call originate from socket api to hit dialplan section that does all
the work (this originate command is what I don't understand, is there
another way of hitting the dialplan besides calling originate?)
2) hit dialplan section which calls lcr for first number and bridges to an
extension
3) the extension calls lcr fir the second number and originates the second
call
On steps 2 and 3 I could just use set data to set the additional variables
I need. The first step is what troubles me.
Thank you!
Nicolas
On Tue, Jul 21, 2009 at 12:54 PM, Rupa Schomaker r...@rupa.com wrote:
Ok, if you want to do it from the socket api, then I need to make a 'as
xml' option to mod_lcr and give you lcr_auto_route as one of the nodes in
the returned xml. Then you can do your own substitution in the originate
line... In that case, you'd call lcr twice and do:
originate lcr_auto_route1 bridge(lcr_auto_route2)
How soon do you need this?
On Tue, Jul 21, 2009 at 11:27 AM, Nicolas Brenner nico...@medularis.com
wrote:
That looks like a good way to go about it.
How can I access channel variables through the socket using the api? I
mean, how do I recover the value of ${lcr_auto_route}? I would need to add
some other variables, like ignore_early_media=true and a uuid that 'links'
the two calls so I can track it listening for events.
Thanks!
Nicolas
On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker r...@rupa.com wrote:
lcr api command doesn't really return a usable dialstring (it was
originally done for debug purposes). I could add an as xml option if
needed...
Anyway, to do this from the dialplan:
remember that originate's usage is:
-USAGE call url exten|application_name(app_args) [dialplan]
[context] [cid_name] [cid_num] [timeout_sec]
so, the first argument is the call url and the second would be an
extension. so:
1) execute lcr for the first leg of the call
2) execute originate with:
originate ${lcr_auto_route} extension
extension just needs to match something in your dialplan.
In extension, you'd do another lcr lookup and then bridge to that leg's
${lcr_auto_route} value.
On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner
nico...@medularis.com wrote:
I would like to originate 2 calls from FS and then bridge them.
There's no incoming call so I think there's no dialplan involved.
What I'd like to do now is apply lcr rules to these calls. I've come
up with 2 options so far:
1) call lcr through the socket twice (once for each phonenumber) and
then originate the calls through the socket too
2) have a javascript file which runs the actions above, run the script
through the socket with 'jsrun'
How would you do it?
For what I've read on the list, usually the recommended way is to stay
away from javascript as much as possible because it is not as efficient
as
doing everything from the dialplan. Does this mean the first option is
the
best? or is there a dialplan way of doing it?
Thank you very much for your help!
Nicolas
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Re: [Freeswitch-users] Best way to bridge 2 calls with LCR?
Great! Thanks! On Tue, Jul 21, 2009 at 2:51 PM, Rupa Schomaker r...@rupa.com wrote: Just a note that the as xml syntax has been added to current trunk. On Tue, Jul 21, 2009 at 1:21 PM, Rupa Schomaker r...@rupa.com wrote: Well, the as xml is something I've been meaning to do, so I'm gonna get that checked in today sometime anyway. If you want to do any programmatic processing of the lcr data, the as xml is the way to go rather than parsing the strings. -- -Rupa ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 credit = NETWORK_OUT_OF_ORDER ?
Hi, Today I ran out of credit in one of my voip providers. When this happened, all my outgoing calls started failing with hangup cause NETWORK_OUT_OF_ORDER. Once I got some more credit, the calls kept failing. I restarted freeswitch and then everything worked fine again. Unfortunately this is not something I'd like to reproduce, and the only thing I have is the logs (no SIP trace). But I was wodering if someone here has had a similar experience or could tell if this is plausible or even likely to happen. Another part of the platform I'm running, runs on Asterisk, using the same voip providers, nevertheless the calls originating there only failed during the no credit period, and began working again automatically as soon as credit was added to the account. Thanks, Nicolas ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 credit = NETWORK_OUT_OF_ORDER ?
A little bit more info:
When the calls failed, the following was recorded in the log:
2009-07-17 15:19:07.880175 [ERR] switch_ivr_originate.c:1495 Cannot create
outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER]
2009-07-17 15:19:07.880175 [DEBUG] switch_ivr_originate.c:2123 Originate
Resulted in Error Cause: 38 [NETWORK_OUT_OF_ORDER]
2009-07-17 15:19:07.880175 [WARNING] mod_spidermonkey.c:3013 Cannot Create
Outgoing Channel!
[{ignore_early_media=true,originate_timeout=30,execute_on_answer='sched_hangup
+30 ALLOTED_TIMEOUT'}sofia/gateway/mygateway/005698793046]
2009-07-17 15:19:07.880175 [NOTICE] new_energizer_async.js:15 ***
CAUSE: NETWORK_OUT_OF_ORDER ***
2009-07-17 15:19:34.158980 [NOTICE] sofia_reg.c:319 Registering mygateway
2009-07-17 15:19:34.294518 [ERR] sofia_reg.c:1445 mygateway Registration
Failed with status Operation has no matching challenge [904]. failure #37
2009-07-17 15:19:34.365065 [WARNING] sofia_reg.c:348 mygateway Failed
Registration, setting retry to 190 seconds.
I searched for the Registration Failed with status Operation has no
matching challenge error on the list, and someone else had a similar issue,
but apparently it had something to do with NAT, and in this case there's no
NAT involved. Anyway, I'm running a rev 13973 so I'll update to the latest
svn rev and hope it doesn't happen again.
On Fri, Jul 17, 2009 at 4:35 PM, Nicolas Brenner nico...@medularis.comwrote:
Hi,
Today I ran out of credit in one of my voip providers. When this happened,
all my outgoing calls started failing with hangup cause
NETWORK_OUT_OF_ORDER. Once I got some more credit, the calls kept failing. I
restarted freeswitch and then everything worked fine again.
Unfortunately this is not something I'd like to reproduce, and the only
thing I have is the logs (no SIP trace). But I was wodering if someone here
has had a similar experience or could tell if this is plausible or even
likely to happen.
Another part of the platform I'm running, runs on Asterisk, using the same
voip providers, nevertheless the calls originating there only failed during
the no credit period, and began working again automatically as soon as
credit was added to the account.
Thanks,
Nicolas
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[Freeswitch-users] hangup_cause NONE vs. NORMAL_CLEARING
I have a small JS script that makes a call, plays a sound file and then hangs up. For each call it makes, I log the hangup_cause variable on the CHANNEL_HANGUP_COMPLETE event. Most of the time, when calls are successful, I get a NORMAL_CLEARING cause, but sometimes I'll get a NONE cause. I wanted to know what the difference between these two is, because there is no reference to NONE in the wiki (http://wiki.freeswitch.org/wiki/Hangup_causes ). Thanks, Nicolás ___ Freeswitch-users mailing list Freeswitch-users@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] Buy The FreeSWITCH Developers Dinner!
Thank you for all the patience and effort. You've done a great work! Have a great meal! On Thu, Jun 18, 2009 at 12:48 PM, Michael Collins m...@freeswitch.orgwrote: Thank you so much! The devs are really loving this. -MC On Thu, Jun 18, 2009 at 11:40 AM, Saeed Ahmad saeedahmad1...@gmail.comwrote: Done :) Guten Appetit On Thu, Jun 18, 2009 at 5:19 PM, Michael Collins m...@freeswitch.orgwrote: Hello FreeSWITCHers out there! I have it on good authority that the FreeSWITCH developers have all convened in an undisclosed location. Rumors that they are plotting to take over the world are not yet confirmed but I will keep you updated as information becomes available. :) It would be great for all of us to show our support and appreciation to the guys for all the hard work they've done. How many of us have had a question answered on the IRC channel or here on the list by one of the guys? How many of us use FreeSWITCH every day for work? If you've benefited from their hard work then please give a little. If we can get everyone to hop on the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then we can easily pay for a nice dinner for the guys. Please hit the link and let me know (off list) when you've donated. Let's do this, people! -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 ___ Freeswitch-users mailing list Freeswitch-users@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] Best G729 replacement
I don't think it's a Tier 1. They are middlemen between phone companies and other companies that need voip. In my country (Chile), phone companies don't provide voip services, so you have to buy the service from someone else. On Tue, Apr 21, 2009 at 12:45 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: On Mon, Apr 20, 2009 at 3:10 PM, Nicolas Brenner nico...@medularis.com wrote: Hi, I might be in a position (finally) to ask/suggest one of my voip providers to use an alternative codec to G729. I wanted to know what would be the best replacement for it. Thanks again everybody for your time and info. Regards, Nicolas Nicolas, What do you mean by provider? Is this a Tier 1? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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
[Freeswitch-users] G729 settings
Merely for testing/research purposes I decided to try an open G729 codec posted to this list a couple months ago. I tried it a couple times with FreeSWITCH Version 1.0.trunk (11356M). And everything worked fine, while starting FS I would get: 2009-04-20 13:29:59 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_g729] 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 10ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 20ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 30ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 40ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 50ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 60ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 70ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 80ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 90ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 100ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 110ms 2009-04-20 13:29:59 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 120ms Yesterday I wanted to update to the latest revision on svn, so I did make current, and after a few problems, cleaning everything up and getting the svn source again, I built the latest rev at the time (13083). Then I ran FS, everything seemed to work alright, but when I tried to make a call using G729 I got an error saying it didn't have the right read codec. After investigating a little bit, I discovered my VoIP provider was setting G729 @ 40ms, but FS was only supporting 10ms. This is what shows while starting FS: 2009-04-20 13:33:38 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_g729] 2009-04-20 13:33:38 [NOTICE] switch_loadable_module.c:182 switch_loadable_module_process() Adding Codec 'G729' (G.729) 8000hz 10ms 2009-04-20 13:33:38 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_amr] . There's no support for other configurations other than 10ms. The xml config files are the same, and the G729 module code is the same too, so I guess the only thing that's changed is the code that loads the codec module. Any ideas, help or similar experiences? Thanks! Nicolas ___ Freeswitch-users mailing list Freeswitch-users@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] Best G729 replacement
Hi, I might be in a position (finally) to ask/suggest one of my voip providers to use an alternative codec to G729. I wanted to know what would be the best replacement for it. Thanks again everybody for your time and info. Regards, Nicolas ___ Freeswitch-users mailing list Freeswitch-users@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] Adding Spanish support to say
I'm a native spanish speaker, I can help too! Nicolás Brenner On Tue, Apr 14, 2009 at 2:56 PM, Diego Viola diego.vi...@gmail.com wrote: Hey guys, If you need some Spanish help count with my help also. Diego On Tue, Apr 14, 2009 at 2:12 PM, Michael Collins m...@freeswitch.orgwrote: Cool. We've had several volunteers start translating the phrase files into Spanish and Brazilian Portugese. We'll keep you posted when we have the Spanish one ready. FYI, I committed a stub phrase_es.xml file but it hasn't been translated yet except for the first twenty digits. However, there aren't any audio files associated with it yet... -MC On Tue, Apr 14, 2009 at 11:07 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Brian, For my application I just need to be able to say a string of numbers - Caller ID, etc. Other than the files used there is no syntax or grammar difference (in Spanish) when compared to English. I should just be able to drop the files in. I'll have a problem when I need to handle IVR, voicemail, and other more complex issues but this will solve my immediate needs. For now I'm just trying to figure out how to get language es recognized by say... On Tue, Apr 14, 2009 at 1:57 PM, Brian West br...@freeswitch.org wrote: This also requires you to write all the phrase macros for voicemail, ivr and other things in the demo in lang/en/ /b On Apr 14, 2009, at 12:48 PM, João Mesquita wrote: I know spanish and I would translate it no problem. MC, get in touch with me off-list so we can handle that. I can also translate to portuguese-brazil. jmesquita 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 -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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 ___ Freeswitch-users mailing list Freeswitch-users@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] Replace sqlite with couchDB?
Hi, I am not very familiar with FS internals, but I recently found this new db engine called couchDB. Looks pretty interesting, and its main focus is scalability. Has anybody played with couchDB? does it make sense to replace sqlite with couchDB in FS? Here's a link to the project homepage: - http://couchdb.apache.org/ And here's a video of a presentation given by one of the lead programmers: - http://www.vimeo.com/1992869 ___ Freeswitch-users mailing list Freeswitch-users@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] Replace sqlite with couchDB?
Well, if it's too large compared to sqlite maybe it doesn't make sense. But I was thinking calling data is not always fixed. Depending on what you use FS for, you might want to get a CDR with many different data linked to each call, even different kinds of data linked to different calls, that would make each call very different and variable in its structure, which would fit a document db model. Thinking a bit more now, since couchdb is a document-based DB, it might be good for configuration-generating applications, like the ones consumed by xml_curl. These are external applications, yet they are still very closely related to FS, and might be able to benefit from using something like couchdb. On Mon, Apr 13, 2009 at 1:06 AM, Matthew Fong mattdf...@gmail.com wrote: Hi Nicolas, Just off the top of my head, but I think couchDB is rather large compared to sqlite, and I think it's also geared more towards storing dynamic datasets...rather ones that can be structured...like FS calling data can. But I might be wrong :) your buddy. --matt On Mon, Apr 13, 2009 at 12:00 PM, Nicolas Brenner nico...@medularis.comwrote: Hi, I am not very familiar with FS internals, but I recently found this new db engine called couchDB. Looks pretty interesting, and its main focus is scalability. Has anybody played with couchDB? does it make sense to replace sqlite with couchDB in FS? Here's a link to the project homepage: - http://couchdb.apache.org/ And here's a video of a presentation given by one of the lead programmers: - http://www.vimeo.com/1992869 ___ Freeswitch-users mailing list Freeswitch-users@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] origainate through sofia gateway
Jacek,
I had a similar problem once. It actually depends on your sip gateway,
but I was able to solve the problem by setting the caller id, ie:
session1 = new Session();
session1.setCallerData(caller_id_name, 8280052500);
session1.setCallerData(caller_id_number, 8280052500);
session1.originate(session1,
{ignore_early_media=true}sofia/gateway/sip.ipcorp.cl/0225490317,
60);
In this case, the caller_id was the number assigned to me by the
external gateway.
Hope it helps.
Nicolas
On Tue, Feb 3, 2009 at 10:36 AM, Jacek Sokulski jsokul...@dotsystems.pl wrote:
Hello
I am trying to initiate a call from javascript, it works fine for local
numbers:
session1.originate(session1,
{ignore_early_media=true}user/1...@192.168.1.122);
but when I am trying to connect through sofia gateway, the connection is not
being established:
session2.originate(session2, sofia/gateway/halonet/0225490317);
although I can call to this number from softphone.
I have also tried setting effective_caller_id_number:
session1.originate(session1,
{effective_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317);
with the same result.
A configuration in the dialplan that works is:
extension name=halonet.pl
condition field=destination_number expression=^0095(\d{10})$
action application=set
data=effective_caller_id_number=fixed0248b/
action application=set data=bypass_media=true/
action application=set data=hangup_after_bridge=true/
action application=bridge data=sofia/gateway/halonet/$1/
/condition
/extension
Would appreciate any help.
Jacek
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Re: [Freeswitch-users] origainate through sofia gateway
Oops! Well, fortunately I don't use that voip provider anymore (nor the script). Thanks Brian. Nicolas On Tue, Feb 3, 2009 at 2:25 PM, Brian West br...@freeswitch.org wrote: YOU should NEVER use this method or call setCallerData at all you should use the correct methods to override the callerid. If its a B-Leg born from an A-Leg you use these on the on the A-Leg: http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number If you're originating you use this: http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number /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] Open g729 g723 codec, any expierence
I would love to be a beta tester too! I haven't switched from Asterisk for the same reason. Cheers, Nicolas On Fri, Oct 10, 2008 at 4:38 PM, Michael Collins [EMAIL PROTECTED] wrote: Can I at least be a beta tester or something? Please? I'm desperate!!! Dude, you're hired! :) -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] GUI
On Fri, Aug 1, 2008 at 9:51 AM, Anthony Minessale [EMAIL PROTECTED] wrote: ... A quad woodcrest 2.6ghz can do about 3000 simo media sessions with FS, the same box can just make it to 400 when they are all G729 transcoding calls. If they are bridged calls, that number goes in half, if we take media out of the picture that number quadruples. So I guess I could boast 400 CPS with 3000-6000 simo sessions, but what's the point, I'll let Ken do that.. ;) G729 transcoding? I thought there was no support for that... Anyway I can get it (other than writing it myself)? -- Nicolás Brenner ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH in latin america countries
Brian, Although I agree that it is not a good idea to split the community, this wouldn't much split it as increase it. There's a lot of people who don't understand english, but have the skills to use or learn about FreeSwitch and even help in the development. Creating a latin irc channel, could potentially bring thousands to the FreeSwitch community, people who are now left out because of the language barrier. On Sun, Jun 8, 2008 at 6:24 AM, Brian West [EMAIL PROTECTED] wrote: Arnaldo, I really do not want to split the community up. I highly recommend everyone stay in #freeswitch. /b On Jun 8, 2008, at 1:59 AM, Arnaldo de Moraes Pereira wrote: Hello, For those of you from latin american countries, please join #freeswitch-la. I have plans to knock down the barriers to adopt FreeSWITCH in third-world countries, specially Brazil, so this channel is one of the steps to achieve that. Brazilian portuguese and spanish are also welcomed languages. One of the biggest barriers to use FreeSWITCH as a TDM/SIP gateway, is to have MFC/R2 support, which is being written by Steve Underwood in a generic manner. I'll be focusing on the endpoint for the existing unicall implementation, which we hopefully will merge when Steve has finished his unicall work. So, anyone will be able to use MFC/R2 with a Sangoma and probably other cards. Take care. -- Arnaldo M Pereira [EMAIL PROTECTED] http://www.arnaldopereira.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 Brian West sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Nicolás Brenner ___ Freeswitch-users mailing list Freeswitch-users@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] g729
I guess there's no interest in G729 at all... any good alternatives? What do you guys use to save bandwith and keep a decent audio quality? On Wed, Jun 4, 2008 at 7:34 AM, Nicolas Brenner [EMAIL PROTECTED] wrote: Hi, I'd like to know what's needed to add support for G729, I know there's a bounty, but I couldn't make sense out of what's posted on the wiki. I'm really interested in this, as one of my current VoIP providers restricts me to using only this codec, which limits me to using Asterisk, hence I can't fully move to FS without G729 support. What would it take to make it happen? Thanks, -- Nicolás Brenner Medularis SpA -- Nicolás Brenner Medularis SpA ___ Freeswitch-users mailing list Freeswitch-users@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] g729
How much money is it needed to get something going like the USD 10 license from digium? On Thu, Jun 5, 2008 at 9:05 AM, Ken Rice [EMAIL PROTECTED] wrote: There is a huge interest in G729... The problem is the ammount of money in just getting the priviledge to pay the royalties... Once that has been resolved G729 will show up rather quickly K From: Nicolas Brenner [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Thu, 5 Jun 2008 08:42:50 -0400 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] g729 I guess there's no interest in G729 at all... any good alternatives? What do you guys use to save bandwith and keep a decent audio quality? On Wed, Jun 4, 2008 at 7:34 AM, Nicolas Brenner [EMAIL PROTECTED] wrote: Hi, I'd like to know what's needed to add support for G729, I know there's a bounty, but I couldn't make sense out of what's posted on the wiki. I'm really interested in this, as one of my current VoIP providers restricts me to using only this codec, which limits me to using Asterisk, hence I can't fully move to FS without G729 support. What would it take to make it happen? Thanks, -- Nicolás Brenner Medularis SpA -- Nicolás Brenner Medularis SpA ___ Freeswitch-users mailing list Freeswitch-users@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 -- Nicolás Brenner Medularis SpA website: www.medularis.com cel: +56977584628 skype: nbrenner ___ Freeswitch-users mailing list Freeswitch-users@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] g729
Hi, I'd like to know what's needed to add support for G729, I know there's a bounty, but I couldn't make sense out of what's posted on the wiki. I'm really interested in this, as one of my current VoIP providers restricts me to using only this codec, which limits me to using Asterisk, hence I can't fully move to FS without G729 support. What would it take to make it happen? Thanks, -- Nicolás Brenner Medularis SpA ___ Freeswitch-users mailing list Freeswitch-users@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] Max of 170 channels in the conference room.
Thanks Anthony, this might seem like an innocent question, but I tried running:
originate {ignore_early_media=true,bypass_media=true}sofia/default/[EMAIL
PROTECTED]
sofia/default/[EMAIL PROTECTED] inline
on the console, and FS complained about the syntax, is this equivalent?
originate {ignore_early_media=true,bypass_media=true}sofia/default/[EMAIL
PROTECTED]
bridge({ignore_early_media=true,bypass_media=true}sofia/default/[EMAIL
PROTECTED])
how do I check the media is not actually going through FS?
Thanks!
On Thu, May 29, 2008 at 4:31 PM, Anthony Minessale
[EMAIL PROTECTED] wrote:
you can do originate
{ignore_early_media=true,bypass_media=true}sofia/default/[EMAIL PROTECTED]
sofia/default/[EMAIL PROTECTED] inline
and hairpin 2 calls between the provider
On Thu, May 29, 2008 at 2:55 PM, Nicolas Brenner [EMAIL PROTECTED]
wrote:
Anthony and Ken (specially), thank you very much for your explanations
and figures. About what Ken said, how could I initiate a call in media
mode and then switch it to no_media when the second leg is
bridged/answered? Also, is this something my VoIP provider should be
able to support specially, or is it just standard SIP signaling? Thank
you again very much for your help!
On Thu, May 29, 2008 at 2:48 PM, Ken Rice [EMAIL PROTECTED] wrote:
With FreeSwitch there are a couple of ways to accomplish what you are
doing
with 3 distinct levels of performance
Way 1) Full Media Interaction/Transcoding. This is very similar to the
way
asterisk works and should on modern several give you atleast 2 to 3
times
the performance you see on asterisk (not accounting for any transcoding
load
you may introduce)
Way 2) Media Proxy mode. In this mode you will see a good bit of
performance
gain as FreeSwitch will only proxy the media it will not interact with
the
media stream (ie: no transcoding, no DTMF events etc) but you can still
cut
thru nat, appease providers that don't want to hairpin the media on
their
networks, and still do a full topology hide (not applicable for your
scenario below as you can no jump in and out of proxy only mode)
Way 3) No Media Mode. In this mode FreeSwitch functions more along the
lines
of openser/ser minus the media proxies. Media is passed directly between
the
end points and FreeSwitch is completely out of the media path. This is
the
most efficient mode for routing calls as there is no media load on
freeswitch and the number of concurrent calls is limited by system
memory
resources and speed of the calls coming in (as in how many calls/second
can
freeswitch process) for your particular application originating calls
would
start in media mode and them move to no media mode once the second leg
starts to come online. This would have an impact of performance based on
the
total number of calls doing media with freeswitch at any given time.
Now for some real numbers...
I route calls primarily using the no-media-mode using dell 1950s with
Dual
QuadCore 2Ghz E5335's w/ 4Gs of ram. (admittedly this is a slightly
different method from what you are doing) in our configuration we are
able
route in excess of 200 calls/sec with a concurrent call load in excess
of
3000 calls (6000 legs) per machine. Where we run into problems is not in
the
concurrent call volume, its in the Calls/Sec luckily FreeSwitch has a
Sessions/Second Limiter built in and we can set this and keep the box
from
melting down.
Please Note in the above configuration we are largely routing autodialer
traffic so performace should be much better if you decrease the calls
per
second and increase the average call length. High Call Per Second Rates
are
the bane of any switch
K
From: Nicolas Brenner [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Thu, 29 May 2008 12:54:07 -0400
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Max of 170 channels in the conference
room.
Hi, sorry for my ignorance, but I was wondering if these figures are
in any way comparable to the performance FS would have doing bridged
calls?
I have a web callback app that's currently running on top of Asterisk,
and I'm planning on moving to FS, and use originate/bridge commands
to bridge calls between two actual phones. I'd like to know if (using
the same setup as Johny) I'd be able to hit more than 170 channels?
(or more than 170 calls, I guess that would be 340 channels).
Thanks
On Thu, May 29, 2008 at 12:19 PM, Brian West [EMAIL PROTECTED]
wrote:
These aren't really dual core CPU's they are single core with
hyper-threading. If you disable hyper-threading you'll get more
performance.
You'll never get that great of performance out of these CPU's. The
new
64bit woodcrest/clovertown Xeon's are much better... night and day
difference. (Pinto vs Porsche)
/b
On May 29, 2008, at 11:10 AM
Re: [Freeswitch-users] Max of 170 channels in the conference room.
Anthony and Ken (specially), thank you very much for your explanations and figures. About what Ken said, how could I initiate a call in media mode and then switch it to no_media when the second leg is bridged/answered? Also, is this something my VoIP provider should be able to support specially, or is it just standard SIP signaling? Thank you again very much for your help! On Thu, May 29, 2008 at 2:48 PM, Ken Rice [EMAIL PROTECTED] wrote: With FreeSwitch there are a couple of ways to accomplish what you are doing with 3 distinct levels of performance Way 1) Full Media Interaction/Transcoding. This is very similar to the way asterisk works and should on modern several give you atleast 2 to 3 times the performance you see on asterisk (not accounting for any transcoding load you may introduce) Way 2) Media Proxy mode. In this mode you will see a good bit of performance gain as FreeSwitch will only proxy the media it will not interact with the media stream (ie: no transcoding, no DTMF events etc) but you can still cut thru nat, appease providers that don't want to hairpin the media on their networks, and still do a full topology hide (not applicable for your scenario below as you can no jump in and out of proxy only mode) Way 3) No Media Mode. In this mode FreeSwitch functions more along the lines of openser/ser minus the media proxies. Media is passed directly between the end points and FreeSwitch is completely out of the media path. This is the most efficient mode for routing calls as there is no media load on freeswitch and the number of concurrent calls is limited by system memory resources and speed of the calls coming in (as in how many calls/second can freeswitch process) for your particular application originating calls would start in media mode and them move to no media mode once the second leg starts to come online. This would have an impact of performance based on the total number of calls doing media with freeswitch at any given time. Now for some real numbers... I route calls primarily using the no-media-mode using dell 1950s with Dual QuadCore 2Ghz E5335's w/ 4Gs of ram. (admittedly this is a slightly different method from what you are doing) in our configuration we are able route in excess of 200 calls/sec with a concurrent call load in excess of 3000 calls (6000 legs) per machine. Where we run into problems is not in the concurrent call volume, its in the Calls/Sec luckily FreeSwitch has a Sessions/Second Limiter built in and we can set this and keep the box from melting down. Please Note in the above configuration we are largely routing autodialer traffic so performace should be much better if you decrease the calls per second and increase the average call length. High Call Per Second Rates are the bane of any switch K From: Nicolas Brenner [EMAIL PROTECTED] Reply-To: freeswitch-users@lists.freeswitch.org Date: Thu, 29 May 2008 12:54:07 -0400 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Max of 170 channels in the conference room. Hi, sorry for my ignorance, but I was wondering if these figures are in any way comparable to the performance FS would have doing bridged calls? I have a web callback app that's currently running on top of Asterisk, and I'm planning on moving to FS, and use originate/bridge commands to bridge calls between two actual phones. I'd like to know if (using the same setup as Johny) I'd be able to hit more than 170 channels? (or more than 170 calls, I guess that would be 340 channels). Thanks On Thu, May 29, 2008 at 12:19 PM, Brian West [EMAIL PROTECTED] wrote: These aren't really dual core CPU's they are single core with hyper-threading. If you disable hyper-threading you'll get more performance. You'll never get that great of performance out of these CPU's. The new 64bit woodcrest/clovertown Xeon's are much better... night and day difference. (Pinto vs Porsche) /b On May 29, 2008, at 11:10 AM, Johny Kadarisman wrote: Hi Brian, attached is my cpu info. Rgds, processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Xeon(TM) CPU 2.80GHz stepping: 9 cpu MHz : 2784.780 cache size : 512 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 1 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxs r sse sse2 ss ht tm pbe cid xtpr bogomips: 5573.58 clflush size: 64 processor : 1 vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Xeon(TM) CPU 2.80GHz stepping: 9 cpu MHz : 2784.780 cache size : 512
[Freeswitch-users] Problem playing media
On Wed, Apr 9, 2008 at 7:36 AM, Brian West [EMAIL PROTECTED] wrote: On Apr 9, 2008, at 1:17 AM, Nicolas Brenner wrote: Hello everyone, I'm having some trouble with FS :( apparently with mod_shout. I want to play an mp3 file after answering a call so I compiled mod_shout following the wiki, then configured an extension to answer a call and play an mp3 file I uploaded to the server. The thing is, FS supposedly plays the file, but I can't hear it on the softphone with which I'm calling, and also, after playing the file, FS seems to freeze and then I get a lot of: sofia_event_callback() event [nua_r_bye] status [408][Request Timeout] session: n/a sofia_event_callback() event [nua_i_state] status [408][to BYE] session: n/a lines on the console/log. Well I can't see enough of the log with the two lines you posted to many any kinda of educated guess. Ok, attached is a console log with sip traces as well. What I did was: - start freeswitch - register softphone (xlite) with extension 1000 - call 9998: FS answered the call and played the tetris sound, although I was only able to hear it for 3 secs - hanged up: FS received the hang_up and terminated the call - call 9998 again: FS answered the call and supposedly played the tetris sound, but I didn't hear anything on my side - hanged up: FS did not recieve the hang_up signal, and kept the call 'open' - tried calling 9998 again: FS didn't show any sip packets being received or anything, the call could not be made - shutdown FS Also, after I try to call once, the softphone does not work anymore, and I have to make it register again with FS. After all this, when I shutdown FS, it takes some time, and while it's trying to shutdown it prints a lot of the same lines as above. I can only guess you're using x-lite or eyebeam. Yes, I'm using x-lite I thought it was mod_shout, so I commented it from autoload_configs/modules.conf.xml so it does not load, and replaced the mp3 file with a wav file (FS has permission to read both of them), but I'm getting the same behaviour. Now I'm even getting that behaviour when dialing or 9998 (default dialplan pre-configured extensions). Is this a real server or virtual server? ie vmware or xen? This is a virtual server I think, made with virtuozzo. Is a dedicated virtual server hosted by mediatemple (www.mediatemple.net). Now I can't get FS to work right, everytime I try to make a call, either initiating it from the console, the softphone or js, I get the same weird behaviour. Any clues? I'm sorry if this is trivial or a known issue, but I haven't been able to figure it out, thanks a lot for your time and help. Your best bet at this point is to join the IRC channel and ask for help in realtime... Have you tried make current to ensure you don't have any code skew? I tried make current several times. For the tests above (for which the attachment log file is), I removed /usr/local/freeswwitch, then checked out FS source from svn and compiled with default options (did not add mod_shout), and used default config files with no modifications (also installed audio and music files). Additionally, the server has a public IP address, and I'm connected directly to Internet (not nat involved whatsoever). -- Nicolás Brenner ___ Freeswitch-users mailing list Freeswitch-users@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 sip:[EMAIL PROTECTED] ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org The files attached are: - consolelog.txt: What I saw on the FS console while doing the test - siptrace.log: Output of sip trace to a file using TPORT_DUMP env var - freeswitch.log: log/freeswitch.log on FS folder for this test My email was rejected by the list moderator, the files are here (user/pass: freeswitch/mailing): - http://www.medularis.com/fs/freeswitch.log - http://www.medularis.com/fs/consolelog.txt - http://www.medularis.com/fs/siptrace.log Thanks for your help! -- Nicolás Brenner ___ Freeswitch-users mailing list Freeswitch-users@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 bridge 2 sessions with Javascript?
Thanks! It worked as advertised. The only problem I have now, is my provider (I'm trying gafachi now), I'm getting about one or two seconds delay on the audio, which is pretty bad. One other thing, it takes about 5 or 10 seconds to get a ring tone after answering the first call, is there anyway to fake the tone in the meantine, or just plainly replace the tone until the call is answered? Thanks again! Nicolas ___ Freeswitch-users mailing list Freeswitch-users@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 dialout problems
I made it work! the problem? my sip provider (Gizmo). I changed the configuration to use voipdiscount (no comments), and the problem went away. By the way, I'm looking for good SIP providers. In the coming months I'll need to handle a lot of load, and I'd also like good rates to mobile phones in Chile and Colombia. I'm in the process of starting testing with Net2Phone, any comments or recommendations? Thanks! Brian, thanks a lot for your help on IRC, I really appreciate it! Regards, Nicolas ___ Freeswitch-users mailing list Freeswitch-users@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 trunks with FS
Anyone has experience connecting Freeswitch with Net2Phone? Thanks! On Tue, Mar 25, 2008 at 3:31 PM, Leonardo Alves [EMAIL PROTECTED] wrote: the sip trunk in FS is the gateway. Here is how you dial a gateway: http://wiki.freeswitch.org/wiki/Sofia#Dial_out_of_a_gateway And here is how to configure the gatewaÿ: http://wiki.freeswitch.org/wiki/SIP_Provider_Examples Leonardo From: michael mendel Sent: Tuesday, March 25, 2008 3:12 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunks with FS Hello My Provider isn't in the list. I didn't see in the list an a example for sip trunks. Regards, Michael Mendel On 3/25/08, Michael Collins [EMAIL PROTECTED] wrote: Is your provider on this list? http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing start there… -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of michael mendel Sent: Tuesday, March 25, 2008 6:53 AM To: Freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] SIP trunks with FS Hello I have a question , how I use SIP TRUNKS with FS. Regards, Michael Mendel ___ Freeswitch-users mailing list Freeswitch-users@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 -- Nicolás Brenner Medularis SpA website: www.medularis.com cel: +56977584628 skype: nbrenner ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
