Re: [Freeswitch-users] XML config file parsing
Or, you can use something like Smarty to cache your generated XML on your web server and only invalidate those cached results when you change something that will impact them. On Mon, Nov 23, 2009 at 11:38 AM, Anthony Minessale anthony.miness...@gmail.com wrote: There is a formula to implement caching but it's very complicated and nobody has had time to work on it. You have to take every single input variable into account when caching because who is calling the extension, why they are calling it when they are calling it all make a difference. Web servers are designed to get thousands of hits per second so typically they can handle delivering custom xml instruction quite well. If you do not require such a dynamic setup, you could generate static files instead. On Sun, Nov 22, 2009 at 5:43 PM, Tim Uckun timuc...@gmail.com wrote: On Fri, Nov 20, 2009 at 3:03 AM, Rob Forman rob4manh...@gmail.com wrote: Hi Sam, Take a look at mod_xml_curl. Pretty sure it'll do everything you're looking for. Looking at that diagram it seems like mod_xml_curl makes a call for every SIP connection. That seems like overkill. Is there a way to set it up so that it caches the XML it got for a period of 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Large number of destinations
Performance is not an issue. I clocked 300 calls per second on such a setup using a Dell R710 with two XEON X5570s and 32 GB RAM as the FreeSWITCH server and a Dell 2950 4-core system with 8 GB RAM as the app server. The app server was at 15% - 20% idle at that rate and the Dell R710 was 65% - 70% idle. The main bottleneck I ran into was using the limit application with ODBC. A mutex lock around the ODBC calls meant that I could only pull 160 calls per second, even though the app server was 55% - 60% idle at that rate, because the ODBC call took 1/160th of a second to complete and all the requests were serialized. In theory, you should get better performance using mod_xml_curl because FreeSWITCH will not have to parse a large XML dial plan. One of the drawbacks of the XML dial plan is that any time it tries to locate a route element, it must perform an XML linear search until it finds the correct child (as can be seen in the source code). Thus, searching the XML dialplan is O(n) operation while mod_xml_curl is typically constant time, or at worst, O(log n), depending on how you are storing / querying your data from your database system. Actually, I suppose you could just be a bad programmer and end up making it exponential, but I'm assuming you know how to write code and design your database in a way that avoids that. I have been considering writing a hash cache for the XML dialplan so that lookups can become constant time, but I have no idea when or if I will find the time to do that. :) On Fri, Nov 13, 2009 at 5:23 AM, Robin Vleij vi...@fx-services.com wrote: On 11/13/09 2:49 AM, Eliot Gable wrote: Hi Eliot, Or, of course, there is always mod_xml_curl. Basically, XML dialplan on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a web application server, web application server responds with XML routing response, FreeSWITCH routes the call. Yeah, been looking at that one, really cool idea. Then I could build my routing database in any way I want. I'm just worried about performance and the extra delay it'll introduce. But technically with my complex routing demands this would be the right solution, instead of a mix of modules (which probably brings the same extra load on the machine). I'll fiddle a bit. :) /Robin ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Large number of destinations
Or, of course, there is always mod_xml_curl. Basically, XML dialplan on the fly. Call comes in, FreeSWITCH sends XML request via HTTP to a web application server, web application server responds with XML routing response, FreeSWITCH routes the call. On Thu, Nov 12, 2009 at 5:53 PM, Rupa Schomaker r...@rupa.com wrote: On Thu, Nov 12, 2009 at 4:32 PM, Robin Vleij vi...@fx-services.com wrote: On 11/12/09 9:59 PM, Rupa Schomaker wrote: If I read it right, this is suited for complete nrs. So would I have a system connected with lots of DIDs, I would put them in easyroute. Then for systems with lots of number ranges, I would use mod_lcr. lcr is based on prefix, so the boundaries for which the range is assigned may not match a prefix. You may be better off either: 1) denormalize your ranges and just insert all distinct #s 2) Modify mod_easyroute to support ranges 3) talk to SWK (he is on irc here and there) about his (non free) fancier routing options -- -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 -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP provider with extern rtp server
fsctl loglevel debug console loglevel debug sofia profile internal siptrace on sofia profile external siptrace on sofia loglevel all 9 ^ Then run your call, then do this: sofia loglevel all 0 sofia profile external siptrace off sofia profile internal siptrace off fsctl loglevel warning console loglevel warning On Fri, Oct 30, 2009 at 12:16 PM, Ivan C Myrvold i...@myrvold.org wrote: I have already set debug to 9, on both profiles. Ivan Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable: See that 200 OK that keeps coming in over and over and over and over again? That's because they never received your ACK. If you can turn on sofia loglevel to 9 and then watch where you send the ACK, you will probably have your answer to why the other system did not receive it. If you're still not sure what's going on, post another pastebin with sofia loglevel set to 9. On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold i...@myrvold.org wrote: Oh, what happened to it? Anyway, here is a new pb: http://pastebin.freeswitch.org/10867 Ivan Den 28. okt. 2009 kl. 19:12 skrev Michael Collins: On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org wrote: Here is a debug log from a call from an internal phone out to an external (my iPhone with nbr 91316356): http://pastebin.freeswitch.org/108578 Ivan Uh... you wanna try that PB number again? -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP provider with extern rtp server
See that 200 OK that keeps coming in over and over and over and over again? That's because they never received your ACK. If you can turn on sofia loglevel to 9 and then watch where you send the ACK, you will probably have your answer to why the other system did not receive it. If you're still not sure what's going on, post another pastebin with sofia loglevel set to 9. On Wed, Oct 28, 2009 at 4:51 PM, Ivan C Myrvold i...@myrvold.org wrote: Oh, what happened to it? Anyway, here is a new pb: http://pastebin.freeswitch.org/10867 Ivan Den 28. okt. 2009 kl. 19:12 skrev Michael Collins: On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org wrote: Here is a debug log from a call from an internal phone out to an external (my iPhone with nbr 91316356): http://pastebin.freeswitch.org/108578 Ivan Uh... you wanna try that PB number again? -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] pass arguments into javascript
Should work fine. I use this: var calling_num = argv[0]; var called_num = argv[1]; Are you sure you actually had valid data in $1 and $2? Try to call it from the CLI: jsrun test.js testvar1 testvar2 On Wed, Oct 28, 2009 at 10:22 PM, Erwin Davis davis.er...@gmail.com wrote: Hi, new to javascript. I tried to pass two arguments into javascript, action application=javascript data=test.js $1 $2/ In test.js, I tried to use argv[1] to retrieve $1 and argv[2] to retrieve $2, however, the javascript test.js complained about argv[] as undefined variables. How to retrieve the passing arguments in a javascript same as the case above. Thanks, ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how to config FS with two net interface?
Try setting ext-rtp-ip and ext-sip-ip on both profiles.
On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang lei.tl...@gmail.com wrote:
Hi all, I run FS on a machine with two net interface, each interface has a
ip addr, one of the them connect to public network(has ip addr A), the
other connect to a private network(has ip addr B), FS server as a SIP
server for public through A, all outbound call will bridge to a softswitch
in private network through B. here is my sofia config file and diaplan
config:
sofia internal.xml
param name=rtp-ip value=A/
param name=sip-ip value=A/
sofia external.xml
param name=rtp-ip value=B/
param name=sip-ip value=B/
dialplan
..
extension name=OUTBOUND
condition field=destination_number expression=^(\d+)$
action application=set data=hangup_after_bridge=true/
action application=set
data=continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION/
action application=set
data=effective_caller_id_number=xxx/ !--here change the caller
number --
action application=bridge
data=sofia/external/${destination_numb...@x/
/condition
/extension
.
then call seq is
sipAgent -- [internal --(bridge)--external] --softswith
FREESWITCH
the question is, when sipAgent make a outbound call, FS can't recevie the
caller's up audio stream, I traced the SIP packets, found that FS has return
addr B in SDP when ack the invite request from sipAgent, the ack packet is
===
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
x:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
From: 1000 sip:x...@a;tag=cb4d3c4e
To: 65960581 sip:x...@a;tag=DtvSc0QX01yKN
Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
CSeq: 2 INVITE
Contact: sip:xxx...@b:5060;transport=udp
User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 245
v=0
o=FreeSWITCH 1256598185 1256598186 IN IP4 B ;wrong this is the ip addr
of the adapter connect to the private network
s=FreeSWITCH
c=IN IP4 B ;wrong this is the ip addr of the adapter connect to the
private network
t=0 0
m=audio 31066 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
I think FS should return A in SDP, not the external binding addr (B), does
somebody known how to solve this problem?
--
Lei.Tang
lei.tl...@gmail.com
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Eliot Gable
We do not inherit the Earth from our ancestors: we borrow it from our
children. ~David Brower
I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime. ~David Brower
Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero
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Re: [Freeswitch-users] SIP provider with extern rtp server
Make sure you let their media IPs through your firewall. Also, if you are behind a NAT, check you have things passing to the correct internal address. On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold i...@myrvold.org wrote: I have used a SIP provider for more than a year. A few days ago, he said he was moving to a new server, and asked me to reconfigure. I did, and everything seemed to work fine, until I did an outgoing call to an external telephone. I found out I had no audio, in neither direction. Incoming calls was working fine. My provider said that the rtp is not going through the sip server, as it did earlier, but now through several other IP's. Do I have to do some special configuration to handle that? Ivan ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP provider with extern rtp server
No, the IP address the media originates from does not need to be tied to the SIP IP address. Can you send a Wireshark capture taken on the FreeSWITCH server of both call legs? Or, if you can, pastebin a debug log from FreeSWITCH console with sofia loglevel set to 9 and siptrace on for any Sofia SIP profiles involved. On Tue, Oct 27, 2009 at 11:52 AM, Ivan C Myrvold i...@myrvold.org wrote: The server is on a public IP, so there is no nat issue here. I can also see the rtp messages on wireshark starting just after the 183 Session Progress message on the server, but just in one direction, coming in to the server. So it looks like Freeswitch is stopping the rtp. Is this because the rtp originates from another ip than the sip provider ip? Ivan Den 27. okt. 2009 kl. 14:58 skrev Eliot Gable: Make sure you let their media IPs through your firewall. Also, if you are behind a NAT, check you have things passing to the correct internal address. On Tue, Oct 27, 2009 at 2:46 AM, Ivan C Myrvold i...@myrvold.org wrote: I have used a SIP provider for more than a year. A few days ago, he said he was moving to a new server, and asked me to reconfigure. I did, and everything seemed to work fine, until I did an outgoing call to an external telephone. I found out I had no audio, in neither direction. Incoming calls was working fine. My provider said that the rtp is not going through the sip server, as it did earlier, but now through several other IP's. Do I have to do some special configuration to handle that? Ivan ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Estimating Call Capacity
Although, FYI, I just benchmarked mod_xml_curl on a separate web app server from FS with FS on a Dell R710 with their current best processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32 GB memory. The web app server is less than half the power of the R710. I maxed the web app server at 300 calls per second (both setting up and tearing down) and the R710 running FS was 65% idle. No audio was being proxied through FS, though. If I were running the web app server on an equivalent R710, they probably would have been on-par with each other in performance. Extrapolating, I expect that in such a case I should be able to get at least 650 CPS out of FS, though for production I would probably limit it to 400 CPS or less so I leave room for miscellaneous tasks. I maxed out the R710 at over 16,000 simultaneous calls (again, no audio proxying) but the only reason I couldn't do more was because I hit some sort of thread creation limit in Linux. There was about 17 GB of memory used for this many calls. This should give you some ballpark idea of what you can accomplish with FS. At some point, I will track down and resolve the thread creation issue, at which time I believe call limits will be limited either by a complex combination of available memory, the speed of the processor, the cost of thread context switching, calls per second setup rate, and call duration. -- Eliot Gable -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Monday, October 26, 2009 4:56 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Estimating Call Capacity On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur vinuth.madi...@gmail.com wrote: Here are a few benchmarks that I had stumbled upon. http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x2200+M2 Please remember NO benchmarks are endorsed by the FS community or developers, because there are just too many variables, and a simple figure is just useful for marketing hype, not for real dimensioning. You MUST do your own benchmarking, so you get an idea about how to dimension for your own use case and hardware. Thanks, Vinuth. On Tue, Oct 27, 2009 at 1:43 AM, Brian West br...@freeswitch.org wrote: I highly doubt it... You can wait for someone to post their results but in the end you'll have to do your own load testing because not everyone's numbers will jive with your use case. Which is the reason the project never posts or endorses a set call count. /b On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: Are there any benchmarking test results available publicly? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem with inbound call answered but no sound
FYI, it generally makes debugging easier if you do this: sofia profile external siptrace on sofia profile internal siptrace on That way you can see the actual signaling and it is usually more clear what is going on. In most cases, you will probably be able to figure it out yourself just looking at the signaling. On Mon, Oct 26, 2009 at 10:29 PM, Lars Zeb larc...@yahoo.com wrote: I have tried to update (make current) twice since 15183. All inbound calls are picked up but the caller hears nothing but a couple of clicks. The most recent version I’ve tried is 15241. Any ideas on what may be causing this? http://pastebin.freeswitch.org/10843 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux Thanks Lars ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Eliot Gable We do not inherit the Earth from our ancestors: we borrow it from our children. ~David Brower I decided the words were too conservative for me. We're not borrowing from our children, we're stealing from them--and it's not even considered to be a crime. ~David Brower Esse oportet ut vivas, non vivere ut edas. (Thou shouldst eat to live; not live to eat.) ~Marcus Tullius Cicero ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
