Re: [Freeswitch-users] FreeSWITCH under the Linux 2.6.29 kernel
Just to add data to this: PowerTOP 1.11 (C) 2007, 2008 Intel Corporation Collecting data for 15 seconds Detailed C-state information is not available. P-states (frequencies) 2.34 Ghz 0.0% 2.00 Ghz 100.0% Wakeups-from-idle per second : 405.4interval: 15.0s no ACPI power usage estimate available Top causes for wakeups: 82.2% (1067.3)freeswitch : schedule_hrtimeout_range (hrtimer_wakeup) [snip] ___ Freeswitch-users mailing list Freeswitch-users@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
Hi I can also help with the translation into Spanish. Spanish is my native language. I can be contacted at: rjpereyra (at) gmail (dot) com roberto -- The best dedicated servers - LiquidWeb http://www.liquidweb.com/?RID=contenid ___ Freeswitch-users mailing list Freeswitch-users@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] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?
Hi, I have been testing inbound calls to a Nokia phone with handover to a cellphone number if I get MEDIA_TIMEOUT on the B leg of the call, and had to set rtp-timeout to a very low 6 seconds in order to get fast handover. This introduces an interesting side-effect that hangs up calls even in the ringing state after 6 seconds. Is this the desired behaviour of rtp-timeout-sec? My initial guess was that rtp-timeout-sec should only be valid for established calls where the two endpoints have exchanged rtp at some point but have stopped exchanging media. As far as I know a phone call in ringing state has not shared any RTP with the other endpoint until it gets early media or is answered. Should rtp-timeout-sec really be valid even when ringing? It seems to me that setting rtp-timeout-sec to 60 seconds would add an absolute time limit on ringing phone calls to 60 seconds, which I believe is not the actual purpose of this limit. Could anyone please share their thoughts on this matter? Thanks, Mikael ___ Freeswitch-users mailing list Freeswitch-users@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] uuid_displace FIFO help
Chris Danielson wrote: Excellent thanks, this is what I was looking for. One last question if you don't mind; is there anyway to pull the caller out of a fifo after a certain time either from api or by setting a variable (eg. the destination didn't answer after sometime, so carry on in the dialplan to eg. voicemail)? There is a uuid_transfer that will allow you to route them accordingly. Thanks Chris Adnan ___ Freeswitch-users mailing list Freeswitch-users@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] any way ring fifo members one by one?
learn to think 4th dimensionally =D
Add one member with a | sep list in the dial string.
On Mon, Apr 27, 2009 at 11:35 PM, seven dujinf...@gmail.com wrote:
Hi,
I'm on trunk 13174, and route a call to fifo, but two members ring at
the same time. I want it ring one by one in a round robin manner,
what's wrong with me?
here is fifo.xml
fifo name=sales_f...@$${domain} importance=0
member timeout=60 simo=1
lag=20{fifo_member_wait=wait}user/1...@$${domain}/member
member timeout=60 simo=1
lag=20{fifo_member_wait=wait}user/1...@$${domain}/member
/fifo
We want to implement a call center where agents register to waiting
customers, when a customer calls in, it will drop in a queue and
search one available agent(in round robin manner). Most fifo functions
seems implemented for scenarios where agents dial in and waiting
callers, which is unnecessary on our condition.
Thanks.
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Re: [Freeswitch-users] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?
Are you geting 183+sdp from the nokia? the media timer only operates once media is established and only counts against you if the channel is being read from and that does not happen until you get a 183 or 200 w/sdp try putting a debug line in switch_rtp.c around 1520 printf(MISSED PACKETS %u/%u\n, rtp_session-missed_count, rtp_session-max_missed_packets); but try updating first there was a recent fix that may have prevented a timer surge at the beginning of calls. On Tue, Apr 28, 2009 at 6:20 AM, Mikael Aleksander Bjerkeland mik...@bjerkeland.com wrote: Hi, I have been testing inbound calls to a Nokia phone with handover to a cellphone number if I get MEDIA_TIMEOUT on the B leg of the call, and had to set rtp-timeout to a very low 6 seconds in order to get fast handover. This introduces an interesting side-effect that hangs up calls even in the ringing state after 6 seconds. Is this the desired behaviour of rtp-timeout-sec? My initial guess was that rtp-timeout-sec should only be valid for established calls where the two endpoints have exchanged rtp at some point but have stopped exchanging media. As far as I know a phone call in ringing state has not shared any RTP with the other endpoint until it gets early media or is answered. Should rtp-timeout-sec really be valid even when ringing? It seems to me that setting rtp-timeout-sec to 60 seconds would add an absolute time limit on ringing phone calls to 60 seconds, which I believe is not the actual purpose of this limit. Could anyone please share their thoughts on this matter? Thanks, Mikael ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Internal.xml using Public context/dialplan?
Thanks Brian. I'd like to make sure I finally got how dialplans, contexts, domains, SIP profiles, and extensions in the directory work together: Freeswitch supports different SIP profiles in conf/sip_profiles/ (internal.xml, external.xml, etc.), which are loaded through conf/autoload_configs/sofia.conf.xml. Each profile maps to a domain, which corresponds to the part after @ (eg. 1...@192.168.0.1: The Freeswitch server is listening on 192.168.0.1, and this is the domain to which extension 1000 belongs.) Each profile also maps to a context (dialplan) located under conf/dialplan/. Note that, for security reasons, by default, all SIP profiles are set to use the Public context; Internal extensions must map to the Default (private) context explicitely through the user_context variable. Extensions are located under conf/directory/. Each sub-directory matches the domain to which an extension belongs. As set in conf/vars.xml, the default domain matches the server's IP address which itself maps to the Default domain, so all extensions that belong to this domain are located under conf/directory/default/. Is this correct? -- View this message in context: http://www.nabble.com/Internal.xml-using-Public-context-dialplan--tp23175441p23276533.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [Freeswitch-dev] Compiling the FreeSwitch on Windows XP
Start with installing the windows sdk. error C2143: syntax error : missing ']' before 'constant' C:\Program Files\Microsoft SDKs\Windows\v6.0A\include\ras.h http://msdn.microsoft.com/en-us/windows/bb980924.aspx --dave http://dave.thehorners.com/tech-talk/programming On Tue, Apr 28, 2009 at 4:15 AM, Santosh santosh_tripa...@datamatics.com wrote: Hi, I am compiling the FreeSwitch 1.0.3 on Windows machine and getting hundreds of error.I guess this has something to do with Visual studio Settings.Can you please help me with this.I am attaching the Error file also(in which there are some missing files too).Can you send me a working FreeSwitch code for windows? Regards, Santosh Disclaimer: The information contained in this e-mail and attachments if any are privileged and confidential and are intended for the individual(s) or entity(ies) named in this e-mail. If the reader or recipient is not the intended recipient, or employee or agent responsible for delivering to the intended recipient, you are hereby notified that dissemination, distribution or copying of this communication or attachments thereof is strictly prohibited. IF YOU RECEIVE this communication in error, please immediately notify the sender and return the original message. ___ Freeswitch-dev mailing list freeswitch-...@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev 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] [Freeswitch-dev] Compiling the FreeSwitch on Windows XP
Actually, you might have it it doesn't look like the download scripts are workingcould be file permissions or strange data. I'd delete the directory and do a fresh checkout build too. --dave http://dave.thehorners.com/tech-talk/programming On Tue, Apr 28, 2009 at 8:52 AM, David A. Horner david-mailingli...@tecdev.com wrote: Start with installing the windows sdk. error C2143: syntax error : missing ']' before 'constant' C:\Program Files\Microsoft SDKs\Windows\v6.0A\include\ras.h http://msdn.microsoft.com/en-us/windows/bb980924.aspx --dave http://dave.thehorners.com/tech-talk/programming On Tue, Apr 28, 2009 at 4:15 AM, Santosh santosh_tripa...@datamatics.com wrote: Hi, I am compiling the FreeSwitch 1.0.3 on Windows machine and getting hundreds of error.I guess this has something to do with Visual studio Settings.Can you please help me with this.I am attaching the Error file also(in which there are some missing files too).Can you send me a working FreeSwitch code for windows? Regards, Santosh Disclaimer: The information contained in this e-mail and attachments if any are privileged and confidential and are intended for the individual(s) or entity(ies) named in this e-mail. If the reader or recipient is not the intended recipient, or employee or agent responsible for delivering to the intended recipient, you are hereby notified that dissemination, distribution or copying of this communication or attachments thereof is strictly prohibited. IF YOU RECEIVE this communication in error, please immediately notify the sender and return the original message. ___ Freeswitch-dev mailing list freeswitch-...@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev 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] any way ring fifo members one by one?
Ah, right, that works. I had thought the purpose of members is for
sequential hunting. looks I was wrong.
However, add a | sep'ed dial string is hard to do round robin hunting,
as we don't want the first agent always busy while others have nothing
to do. It is possible to add/delete members using another script to
set different dialstring to emulate a round robin hunt, but why not
implement it in the queue logic?
questions:
1) What's the purpose for members? just for simultaneous ring?
2) What's the best use case of agents dial in a fifo to wait callers?
They just listening to music and waiting if no caller? I guess that
would be for very busy call centers.
3) In my test, other members keep ring after one answered, some times
it even ring a long time after the caller hangup.
I'm currently using trixbox. when a call comes in, it just play a
greeting and ring one free agent and fail over to other agents if no
answer or playing moh if all agents are busy. I just want to implement
the same logic in FS and replace it as * causing a lot of problems.
I know there are already rules of how to pull a call out from a fifo,
and guess it would be possible to add some params to do sequential/
round robin hunting for members, and by using dp tools to dynamically
add/delete members it would me more powerful.
Thanks.
On Apr 28, 2009, at 8:33 PM, Anthony Minessale wrote:
learn to think 4th dimensionally =D
Add one member with a | sep list in the dial string.
On Mon, Apr 27, 2009 at 11:35 PM, seven dujinf...@gmail.com wrote:
Hi,
I'm on trunk 13174, and route a call to fifo, but two members ring at
the same time. I want it ring one by one in a round robin manner,
what's wrong with me?
here is fifo.xml
fifo name=sales_f...@$${domain} importance=0
member timeout=60 simo=1
lag=20{fifo_member_wait=wait}user/1...@$${domain}/member
member timeout=60 simo=1
lag=20{fifo_member_wait=wait}user/1...@$${domain}/member
/fifo
We want to implement a call center where agents register to waiting
customers, when a customer calls in, it will drop in a queue and
search one available agent(in round robin manner). Most fifo functions
seems implemented for scenarios where agents dial in and waiting
callers, which is unnecessary on our condition.
Thanks.
___
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ClueCon http://www.cluecon.com/
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[Freeswitch-users] Sofia doubts
What I want to discuss concerns mostly Sofia-SIP, but not primary FreeSWITCH. Surely it would be better to use Sofia's list instead and I already did it, but have got no response so far. Overall Sofia-SIP is a good library, which works reliable enough on Linux, but Windows port causes me some doubts. Does anybody use FreeSWITCH/Sofia-SIP in production ready environment on Windows? I know it's working, but how far it has been tested. Just to be clear, the issues I've observed concerns UniMRCP/Sofia-SIP 1.12.10 and I just wanted to know how reliable it works on your end. http://sourceforge.net/mailarchive/forum.php?thread_name=708583.92229.qm%40web111308.mail.gq1.yahoo.comforum_name=sofia-sip-devel http://sourceforge.net/mailarchive/forum.php?thread_name=1ae1714d0903290440ree148b5m67fe183da58ee776%40mail.gmail.comforum_name=sofia-sip-devel Thanks, Arsen. www.unimrcp.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] any way ring fifo members one by one?
dujinfang ha scritto: It is possible to add/delete members using another script I'm not such expert but i think you can use variables everywhere in FS. So your member config instead of having a static string us...@domain|us...@domain probably you can use a variables. Then you script just push the user in/out fo that database variable string. Antonio ___ Freeswitch-users mailing list Freeswitch-users@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] any way ring fifo members one by one?
keep in mind mod_fifo is not a call center app. it's a simple *fifo* queue
hence the name.
On Tue, Apr 28, 2009 at 8:30 AM, dujinfang dujinf...@gmail.com wrote:
Ah, right, that works. I had thought the purpose of members is for
sequential hunting. looks I was wrong.
However, add a | sep'ed dial string is hard to do round robin hunting, as
we don't want the first agent always busy while others have nothing to do.
It is possible to add/delete members using another script to set different
dialstring to emulate a round robin hunt, but why not implement it in the
queue logic?
Other strategies to place calls could be added with a patch to the code.
The goal of the module was to be basic and have
most of the control and logic remain outside the module.
questions:
1) What's the purpose for members? just for simultaneous ring?
They are essentially on-hook agents for a setup where you don't need people
waiting on the phone.
2) What's the best use case of agents dial in a fifo to wait callers? They
just listening to music and waiting if no caller? I guess that would be for
very busy call centers.
Yes its so you can call in an pop all the calls off the fifo in the same one
call.
3) In my test, other members keep ring after one answered, some times it
even ring a long time after the caller hangup.
It's not ring-all, there is exactly one outbound call generated for every
one person in the queue who is waiting so sometimes there is collateral
damage.
I'm currently using trixbox. when a call comes in, it just play a greeting
and ring one free agent and fail over to other agents if no answer or
playing moh if all agents are busy. I just want to implement the same logic
in FS and replace it as * causing a lot of problems.
I know there are already rules of how to pull a call out from a fifo, and
guess it would be possible to add some params to do sequential/round robin
hunting for members, and by using dp tools to dynamically add/delete members
it would me more powerful.
You could post it as a bounty, a change like that is a lot to do as a wish
request.
Thanks.
On Apr 28, 2009, at 8:33 PM, Anthony Minessale wrote:
learn to think 4th dimensionally =D
Add one member with a | sep list in the dial string.
On Mon, Apr 27, 2009 at 11:35 PM, seven dujinf...@gmail.com wrote:
Hi,
I'm on trunk 13174, and route a call to fifo, but two members ring at
the same time. I want it ring one by one in a round robin manner,
what's wrong with me?
here is fifo.xml
fifo name=sales_f...@$${domain} importance=0
member timeout=60 simo=1
lag=20{fifo_member_wait=wait}user/1...@$${domain}/member
member timeout=60 simo=1
lag=20{fifo_member_wait=wait}user/1...@$${domain}/member
/fifo
We want to implement a call center where agents register to waiting
customers, when a customer calls in, it will drop in a queue and
search one available agent(in round robin manner). Most fifo functions
seems implemented for scenarios where agents dial in and waiting
callers, which is unnecessary on our condition.
Thanks.
___
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Re: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP?
I also installed the Nokia Configuration Tool (V3). Which settings did you apply for getting the crypto line? So far I only got TLS to work, there is no crypto line so far. Best regards Peter Ognjen Seslija schrieb: Hi, after installing Nokia Configuration Tool I managed to get E61i to offer srtp only it sends crypto line in the RTP/AVP which is not rfc behaviour: 2009-04-27 12:11:12 [ERR] sofia_glue.c:2785 sofia_glue_negotiate_sdp() a=crypto in RTP/AVP, refer to rfc3711 Can anything be done with this? Ognjen On Mon, Apr 27, 2009 at 11:22 AM, Hostinsky Miroslav miroslav.hostin...@sitronicsts.com mailto:miroslav.hostin...@sitronicsts.com wrote: I think SRTP on S60 is not supported in native SIP/IMS client bundled with phone. There is pdf document about RTP/SRTP implementation on the forum-nokia website http://nds2.fds-forum.nokia.com/fdp/interface?fid=A1A1BNIMDIYBst=vmeOzm volzmnda95395bfe31cd07f895a9bd39b45e007033b8900c4f7df925ac9de383011968dc http://nds2.fds-forum.nokia.com/fdp/interface?fid=A1A1BNIMDIYBst=vmeOzm%0Avolzmnda95395bfe31cd07f895a9bd39b45e007033b8900c4f7df925ac9de383011968dc d3b65f0b52fa4cb08fab3705157dc42e3af9022012ab2ac111cbeb00d9079d531933758c b95b6eab7016953987e9489c30075d9e08c51f2857cec41f60409d58ce806fbae1127e2c 87f420efbe962c28eecf16b7c108d6127de0310ea9d5d34c07076ce111ba72f7c91d795e dc2adaa57534d08024eb1061a70aa6ba0c12be59ae7a6c354cd177lid=FN I didn't find any other information about SRTP and S60... -- Miro -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org mailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Saturday, April 25, 2009 7:01 PM To: freeswitch-users@lists.freeswitch.org mailto:freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP? Thanks Jason, I do TLS/SRTP with some other Phones (Snom/X-Lite/Other_Freeswitches). This works nad I have those crypto lines. But I couldn't get it work with my E71 and I am struggling whether I should invest more time into it. I am sure it's on the Symbian side. This was just to ask if anybody manged to get it to work. I you all tell me: Peter I tried, it doesn't work, please do not invest any time, or it works, this would be very helpful information for me. Best regards Peter Jason White schrieb: Peter P GMX prometheus...@gmx.net mailto:prometheus...@gmx.net wrote: has anybody tried successfully to setup a Nokia E71 (or similar symbian S60 3rd phone) with Pjsip and TLS/SRTP? TLS seems to work but what about the SRTP part? Do you have log entries like this? 2009-04-24 11:05:19 [INFO] switch_rtp.c:782 switch_rtp_add_crypto_key() Activati ng Secure RTP SEND 2009-04-24 11:05:19 [INFO] switch_rtp.c:762 switch_rtp_add_crypto_key() Activati ng Secure RTP RECV If so, this establishes that SRTP is being used. Of course, you could also intercept the packets just to be sure, but I think the above log entries should be a reliable indicator. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___
Re: [Freeswitch-users] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?
The scenario I was referring to was actually an outbound call from a locally registered SIP phone to a cellphone. The same thing happens whether I use a SIP or PRI trunk. After 6 s it hangs up. I get SDP on 183 no matter whether I'm calling a cellphone or a fixed line. I also get ringing indication. The 183+sdp is passed on to the Nokia and after 6 s the call is hung up. Both the SIP and PRI trunks claim to send early media but there seems to be no audio/RTP. If I answer the call in 6 s it's not dropped because the media path was established before RTP timeout. The same thing happens on latest trunk. I added the debug line at 1520 and did make /etc/init.d/freeswitch stop make install /etc/init.d/freeswitch start but the debug line didn't show up anywhere in the CLI. Is my upstream provider doing something wrong in sending early media in these cases? Seems pretty odd. It can be avoided by setting a higher rtp-timeout-sec but it will still be an absolute timeout on ringing. A transcript of the log: send 1293 bytes to udp/[1.1.1.1]:5060 at 13:55:56.451865: INVITE sip:21651...@domain.appsvrslip11.prigw.com SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;rport;branch=z9hG4bKm3t6teHv30rBK Route: sip:21651...@1.1.1.1 Max-Forwards: 69 From: someone sip:23695...@2.2.2.2;tag=m2SepeSZ63e3g To: sip:21651...@domain.appsvrslip11.prigw.com Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc CSeq: 114345142 INVITE Contact: sip:mod_so...@2.2.2.2:5060 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13175M 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: 383 P-Asserted-Identity: someone sip:23695...@2.2.2.2 v=0 o=FreeSWITCH 3718974841365302606 4309079514688066219 IN IP4 2.2.2.2 s=FreeSWITCH c=IN IP4 2.2.2.2 t=0 0 m=audio 52706 RTP/AVP 9 8 0 3 101 13 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 52752 RTP/AVP 99 a=rtpmap:99 H264/9 2009-04-28 15:55:56 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel sofia/external-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=sip:21651...@1.1.1.1 entering state [calling][0] recv 305 bytes from udp/[1.1.1.1]:5060 at 13:55:56.482864: SIP/2.0 100 Trying From: someonesip:23695...@2.2.2.2;tag=m2SepeSZ63e3g To: sip:21651...@domain.appsvrslip11.prigw.com Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc CSeq: 114345142 INVITE Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK Content-Length: 0 recv 1035 bytes from udp/[1.1.1.1]:5060 at 13:55:58.296906: SIP/2.0 183 Session Progress From: someonesip:23695...@2.2.2.2;tag=m2SepeSZ63e3g To: sip:21651...@domain.appsvrslip11.prigw.com;tag=20134330840200942815366 Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc CSeq: 114345142 INVITE Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK content-type: application/sdp contact: sip:1.1.1.1:5060;nt_end_pt=YM0 +~K!-.f0vfc830~P68.cio~H9zwgW0VyisWTdcaM26c610Xbo1.nfS.5NQt3mO~~70!-.f0vft815;nt_server_host=1.1.1.1 supported: 100rel x-nt-party-id: -/ allow: ACK allow: BYE allow: CANCEL allow: INVITE allow: OPTIONS allow: INFO allow: SUBSCRIBE allow: REFER allow: NOTIFY allow: PRACK server: CS2000_NGSS/9.0 Content-Length: 300 v=0 o=IWSPM 573585738 573585738 IN IP4 84.20.97.100 s=- e=unkn...@invalid.net t=0 0 m=audio 45954 RTP/AVP 8 0 18 101 c=IN IP4 84.20.97.100 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 0 RTP/AVP 99 c=IN IP4 2.2.2.2 a=rtpmap:99 H264/9 2009-04-28 15:55:58 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel sofia/external-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path=sip:21651...@1.1.1.1 entering state [proceeding][183] 2009-04-28 15:55:58 [DEBUG] sofia.c:2919 sofia_handle_sip_i_state() Remote SDP: v=0 o=IWSPM 573585738 573585738 IN IP4 84.20.97.100 s=- e=unkn...@invalid.net t=0 0 m=audio 45954 RTP/AVP 8 0 18 101 c=IN IP4 84.20.97.100 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=video 0
Re: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP?
Hi, there is a edit online option to use to change the option on the phone. You have to first install SIP Voip Settings Tool from the Nokia forum, and then change the option Secure call to prefered from NCT. Ognjen On Tue, Apr 28, 2009 at 4:32 PM, Peter P GMX prometheus...@gmx.net wrote: I also installed the Nokia Configuration Tool (V3). Which settings did you apply for getting the crypto line? So far I only got TLS to work, there is no crypto line so far. Best regards Peter Ognjen Seslija schrieb: Hi, after installing Nokia Configuration Tool I managed to get E61i to offer srtp only it sends crypto line in the RTP/AVP which is not rfc behaviour: 2009-04-27 12:11:12 [ERR] sofia_glue.c:2785 sofia_glue_negotiate_sdp() a=crypto in RTP/AVP, refer to rfc3711 Can anything be done with this? Ognjen On Mon, Apr 27, 2009 at 11:22 AM, Hostinsky Miroslav miroslav.hostin...@sitronicsts.com mailto:miroslav.hostin...@sitronicsts.com wrote: I think SRTP on S60 is not supported in native SIP/IMS client bundled with phone. There is pdf document about RTP/SRTP implementation on the forum-nokia website http://nds2.fds-forum.nokia.com/fdp/interface?fid=A1A1BNIMDIYBst=vmeOzm volzmnda95395bfe31cd07f895a9bd39b45e007033b8900c4f7df925ac9de383011968dc http://nds2.fds-forum.nokia.com/fdp/interface?fid=A1A1BNIMDIYBst=vmeOzm%0Avolzmnda95395bfe31cd07f895a9bd39b45e007033b8900c4f7df925ac9de383011968dc d3b65f0b52fa4cb08fab3705157dc42e3af9022012ab2ac111cbeb00d9079d531933758c b95b6eab7016953987e9489c30075d9e08c51f2857cec41f60409d58ce806fbae1127e2c 87f420efbe962c28eecf16b7c108d6127de0310ea9d5d34c07076ce111ba72f7c91d795e dc2adaa57534d08024eb1061a70aa6ba0c12be59ae7a6c354cd177lid=FN I didn't find any other information about SRTP and S60... -- Miro -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org mailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: Saturday, April 25, 2009 7:01 PM To: freeswitch-users@lists.freeswitch.org mailto:freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP? Thanks Jason, I do TLS/SRTP with some other Phones (Snom/X-Lite/Other_Freeswitches). This works nad I have those crypto lines. But I couldn't get it work with my E71 and I am struggling whether I should invest more time into it. I am sure it's on the Symbian side. This was just to ask if anybody manged to get it to work. I you all tell me: Peter I tried, it doesn't work, please do not invest any time, or it works, this would be very helpful information for me. Best regards Peter Jason White schrieb: Peter P GMX prometheus...@gmx.net mailto:prometheus...@gmx.net wrote: has anybody tried successfully to setup a Nokia E71 (or similar symbian S60 3rd phone) with Pjsip and TLS/SRTP? TLS seems to work but what about the SRTP part? Do you have log entries like this? 2009-04-24 11:05:19 [INFO] switch_rtp.c:782 switch_rtp_add_crypto_key() Activati ng Secure RTP SEND 2009-04-24 11:05:19 [INFO] switch_rtp.c:762 switch_rtp_add_crypto_key() Activati ng Secure RTP RECV If so, this establishes that SRTP is being used. Of course, you could also intercept the packets just to be sure, but I think the above log entries should be a reliable indicator. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] any way ring fifo members one by one?
Thank you for the detailed implementation, comments follows.
On Apr 28, 2009, at 10:24 PM, Anthony Minessale wrote:
keep in mind mod_fifo is not a call center app. it's a simple *fifo*
queue hence the name.
Yeah but FS is more than a *, it should be able to do a call center
like job.
On Tue, Apr 28, 2009 at 8:30 AM, dujinfang dujinf...@gmail.com
wrote:
Ah, right, that works. I had thought the purpose of members is for
sequential hunting. looks I was wrong.
However, add a | sep'ed dial string is hard to do round robin
hunting, as we don't want the first agent always busy while others
have nothing to do. It is possible to add/delete members using
another script to set different dialstring to emulate a round robin
hunt, but why not implement it in the queue logic?
Other strategies to place calls could be added with a patch to the
code. The goal of the module was to be basic and have
most of the control and logic remain outside the module.
Then maybe better to keep it simple, and make another mod other than
patch the code. What's your suggestion? And, do you think it would be
easy to control my logic outside the module using lua/event_socket ?
questions:
1) What's the purpose for members? just for simultaneous ring?
They are essentially on-hook agents for a setup where you don't need
people waiting on the phone.
2) What's the best use case of agents dial in a fifo to wait
callers? They just listening to music and waiting if no caller? I
guess that would be for very busy call centers.
Yes its so you can call in an pop all the calls off the fifo in the
same one call.
3) In my test, other members keep ring after one answered, some
times it even ring a long time after the caller hangup.
It's not ring-all, there is exactly one outbound call generated for
every one person in the queue who is waiting so sometimes there is
collateral damage.
Will do more test on this.
I'm currently using trixbox. when a call comes in, it just play a
greeting and ring one free agent and fail over to other agents if no
answer or playing moh if all agents are busy. I just want to
implement the same logic in FS and replace it as * causing a lot of
problems.
I know there are already rules of how to pull a call out from a
fifo, and guess it would be possible to add some params to do
sequential/round robin hunting for members, and by using dp tools to
dynamically add/delete members it would me more powerful.
You could post it as a bounty, a change like that is a lot to do as
a wish request.
Maybe I can try to play that, will look the code to see how hard will
be. Thank you again.
Thanks.
On Apr 28, 2009, at 8:33 PM, Anthony Minessale wrote:
learn to think 4th dimensionally =D
Add one member with a | sep list in the dial string.
On Mon, Apr 27, 2009 at 11:35 PM, seven dujinf...@gmail.com wrote:
Hi,
I'm on trunk 13174, and route a call to fifo, but two members ring at
the same time. I want it ring one by one in a round robin manner,
what's wrong with me?
here is fifo.xml
fifo name=sales_f...@$${domain} importance=0
member timeout=60 simo=1
lag=20{fifo_member_wait=wait}user/1...@$${domain}/member
member timeout=60 simo=1
lag=20{fifo_member_wait=wait}user/1...@$${domain}/member
/fifo
We want to implement a call center where agents register to waiting
customers, when a customer calls in, it will drop in a queue and
search one available agent(in round robin manner). Most fifo
functions
seems implemented for scenarios where agents dial in and waiting
callers, which is unnecessary on our condition.
Thanks.
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ClueCon
Re: [Freeswitch-users] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?
as soon as FS sees 183 it expects media. if they send 183 and no media it will most certainly timeout On Tue, Apr 28, 2009 at 9:33 AM, Mikael Aleksander Bjerkeland mik...@bjerkeland.com wrote: The scenario I was referring to was actually an outbound call from a locally registered SIP phone to a cellphone. The same thing happens whether I use a SIP or PRI trunk. After 6 s it hangs up. I get SDP on 183 no matter whether I'm calling a cellphone or a fixed line. I also get ringing indication. The 183+sdp is passed on to the Nokia and after 6 s the call is hung up. Both the SIP and PRI trunks claim to send early media but there seems to be no audio/RTP. If I answer the call in 6 s it's not dropped because the media path was established before RTP timeout. The same thing happens on latest trunk. I added the debug line at 1520 and did make /etc/init.d/freeswitch stop make install /etc/init.d/freeswitch start but the debug line didn't show up anywhere in the CLI. Is my upstream provider doing something wrong in sending early media in these cases? Seems pretty odd. It can be avoided by setting a higher rtp-timeout-sec but it will still be an absolute timeout on ringing. A transcript of the log: send 1293 bytes to udp/[1.1.1.1]:5060 at 13:55:56.451865: INVITE sip:21651...@domain.appsvrslip11.prigw.comsip%3a21651...@domain.appsvrslip11.prigw.comSIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;rport;branch=z9hG4bKm3t6teHv30rBK Route: sip:21651...@1.1.1.1 sip%3a21651...@1.1.1.1 Max-Forwards: 69 From: someone sip:23695...@2.2.2.2 sip%3a23695...@2.2.2.2 ;tag=m2SepeSZ63e3g To: sip:21651...@domain.appsvrslip11.prigw.comsip%3a21651...@domain.appsvrslip11.prigw.com Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc CSeq: 114345142 INVITE Contact: sip:mod_so...@2.2.2.2:5060 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13175M 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: 383 P-Asserted-Identity: someone sip:23695...@2.2.2.2sip%3a23695...@2.2.2.2 v=0 o=FreeSWITCH 3718974841365302606 4309079514688066219 IN IP4 2.2.2.2 s=FreeSWITCH c=IN IP4 2.2.2.2 t=0 0 m=audio 52706 RTP/AVP 9 8 0 3 101 13 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 52752 RTP/AVP 99 a=rtpmap:99 H264/9 2009-04-28 15:55:56 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel sofia/external-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path= sip:21651...@1.1.1.1 sip%3a21651...@1.1.1.1 entering state [calling][0] recv 305 bytes from udp/[1.1.1.1]:5060 at 13:55:56.482864: SIP/2.0 100 Trying From: someonesip:23695...@2.2.2.2 sip%3a23695...@2.2.2.2 ;tag=m2SepeSZ63e3g To: sip:21651...@domain.appsvrslip11.prigw.comsip%3a21651...@domain.appsvrslip11.prigw.com Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc CSeq: 114345142 INVITE Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK Content-Length: 0 recv 1035 bytes from udp/[1.1.1.1]:5060 at 13:55:58.296906: SIP/2.0 183 Session Progress From: someonesip:23695...@2.2.2.2 sip%3a23695...@2.2.2.2 ;tag=m2SepeSZ63e3g To: sip:21651...@domain.appsvrslip11.prigw.comsip%3a21651...@domain.appsvrslip11.prigw.com ;tag=20134330840200942815366 Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc CSeq: 114345142 INVITE Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK content-type: application/sdp contact: sip:1.1.1.1:5060;nt_end_pt=YM0 +~K!-.f0vfc830~P68.cio~H9zwgW0VyisWTdcaM26c610Xbo1.nfS.5NQt3mO~~70!-.f0vft815;nt_server_host=1.1.1.1 supported: 100rel x-nt-party-id: -/ allow: ACK allow: BYE allow: CANCEL allow: INVITE allow: OPTIONS allow: INFO allow: SUBSCRIBE allow: REFER allow: NOTIFY allow: PRACK server: CS2000_NGSS/9.0 Content-Length: 300 v=0 o=IWSPM 573585738 573585738 IN IP4 84.20.97.100 s=- e=unkn...@invalid.net t=0 0 m=audio 45954 RTP/AVP 8 0 18 101 c=IN IP4 84.20.97.100 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 0 RTP/AVP 99 c=IN IP4 2.2.2.2 a=rtpmap:99 H264/9
Re: [Freeswitch-users] any way ring fifo members one by one?
dujinfang ha scritto: Yeah but FS is more than a *, it should be able to do a call center like job. I think if you open a bounty you can get some nice feedback about this. Antonio ___ Freeswitch-users mailing list Freeswitch-users@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] Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?
Thanks! I'll notify them of the problem and see if there's a way around it. 2009/4/28 Anthony Minessale anthony.miness...@gmail.com as soon as FS sees 183 it expects media. if they send 183 and no media it will most certainly timeout On Tue, Apr 28, 2009 at 9:33 AM, Mikael Aleksander Bjerkeland mik...@bjerkeland.com wrote: The scenario I was referring to was actually an outbound call from a locally registered SIP phone to a cellphone. The same thing happens whether I use a SIP or PRI trunk. After 6 s it hangs up. I get SDP on 183 no matter whether I'm calling a cellphone or a fixed line. I also get ringing indication. The 183+sdp is passed on to the Nokia and after 6 s the call is hung up. Both the SIP and PRI trunks claim to send early media but there seems to be no audio/RTP. If I answer the call in 6 s it's not dropped because the media path was established before RTP timeout. The same thing happens on latest trunk. I added the debug line at 1520 and did make /etc/init.d/freeswitch stop make install /etc/init.d/freeswitch start but the debug line didn't show up anywhere in the CLI. Is my upstream provider doing something wrong in sending early media in these cases? Seems pretty odd. It can be avoided by setting a higher rtp-timeout-sec but it will still be an absolute timeout on ringing. A transcript of the log: send 1293 bytes to udp/[1.1.1.1]:5060 at 13:55:56.451865: INVITE sip:21651...@domain.appsvrslip11.prigw.comsip%3a21651...@domain.appsvrslip11.prigw.comSIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;rport;branch=z9hG4bKm3t6teHv30rBK Route: sip:21651...@1.1.1.1 sip%3a21651...@1.1.1.1 Max-Forwards: 69 From: someone sip:23695...@2.2.2.2 sip%3a23695...@2.2.2.2 ;tag=m2SepeSZ63e3g To: sip:21651...@domain.appsvrslip11.prigw.comsip%3a21651...@domain.appsvrslip11.prigw.com Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc CSeq: 114345142 INVITE Contact: sip:mod_so...@2.2.2.2:5060 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13175M 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: 383 P-Asserted-Identity: someone sip:23695...@2.2.2.2sip%3a23695...@2.2.2.2 v=0 o=FreeSWITCH 3718974841365302606 4309079514688066219 IN IP4 2.2.2.2 s=FreeSWITCH c=IN IP4 2.2.2.2 t=0 0 m=audio 52706 RTP/AVP 9 8 0 3 101 13 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 52752 RTP/AVP 99 a=rtpmap:99 H264/9 2009-04-28 15:55:56 [DEBUG] sofia.c:2912 sofia_handle_sip_i_state() Channel sofia/external-eth1/21651...@domain.appsvrslip11.prigw.com;fs_path= sip:21651...@1.1.1.1 sip%3a21651...@1.1.1.1 entering state [calling][0] recv 305 bytes from udp/[1.1.1.1]:5060 at 13:55:56.482864: SIP/2.0 100 Trying From: someonesip:23695...@2.2.2.2 sip%3a23695...@2.2.2.2 ;tag=m2SepeSZ63e3g To: sip:21651...@domain.appsvrslip11.prigw.comsip%3a21651...@domain.appsvrslip11.prigw.com Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc CSeq: 114345142 INVITE Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK Content-Length: 0 recv 1035 bytes from udp/[1.1.1.1]:5060 at 13:55:58.296906: SIP/2.0 183 Session Progress From: someonesip:23695...@2.2.2.2 sip%3a23695...@2.2.2.2 ;tag=m2SepeSZ63e3g To: sip:21651...@domain.appsvrslip11.prigw.comsip%3a21651...@domain.appsvrslip11.prigw.com ;tag=20134330840200942815366 Call-ID: 23302ac7-ae9f-122c-198f-001ec9e8d9bc CSeq: 114345142 INVITE Via: SIP/2.0/UDP 2.2.2.2;rport=5060;branch=z9hG4bKm3t6teHv30rBK content-type: application/sdp contact: sip:1.1.1.1:5060;nt_end_pt=YM0 +~K!-.f0vfc830~P68.cio~H9zwgW0VyisWTdcaM26c610Xbo1.nfS.5NQt3mO~~70!-.f0vft815;nt_server_host=1.1.1.1 supported: 100rel x-nt-party-id: -/ allow: ACK allow: BYE allow: CANCEL allow: INVITE allow: OPTIONS allow: INFO allow: SUBSCRIBE allow: REFER allow: NOTIFY allow: PRACK server: CS2000_NGSS/9.0 Content-Length: 300 v=0 o=IWSPM 573585738 573585738 IN IP4 84.20.97.100 s=- e=unkn...@invalid.net t=0 0 m=audio 45954 RTP/AVP 8 0 18 101 c=IN IP4 84.20.97.100 a=ptime:20 a=fmtp:18 annexb=no
[Freeswitch-users] FreeSWITCH In Blogosphere
I just wanted to let everyone know that there are a few blog posts out there that you FreeSWITCHers might find interesting. Please check out these two stories on the main FreeSWITCH page: Another Interesting Use For FreeSWITCH: OpenSimhttp://www.freeswitch.org/node/175 DIDX Interview With Anthony Minessale and Brian Westhttp://DIDX%20Interview%20With%20Anthony%20Minessale%20and%20Brian%20West Be sure to leave comments where appropriate. I will followup with digg links shortly. -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] Fwd: FreeSWITCH In Blogosphere
FYI, here's the correct link to the DIDX interview story on the main page: http://www.freeswitch.org/node/176 Sorry for the broken link. (Thanks moy!) -MC -- Forwarded message -- From: Michael Collins m...@freeswitch.org Date: Tue, Apr 28, 2009 at 9:27 AM Subject: FreeSWITCH In Blogosphere To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org, freeswitch-...@lists.freeswitch.org I just wanted to let everyone know that there are a few blog posts out there that you FreeSWITCHers might find interesting. Please check out these two stories on the main FreeSWITCH page: Another Interesting Use For FreeSWITCH: OpenSimhttp://www.freeswitch.org/node/175 DIDX Interview With Anthony Minessale and Brian Westhttp://DIDX%20Interview%20With%20Anthony%20Minessale%20and%20Brian%20West Be sure to leave comments where appropriate. I will followup with digg links shortly. -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] DID Provider
Hi All, Could you please refer me to a best DID provider which also works perfectly with FreeSWITCH. I would need 50 channels per DID number. Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DID Provider
FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH works with any major telco. - Original Message From: Saeed Ahmed saeedahmad1...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, April 28, 2009 1:09:42 PM Subject: [Freeswitch-users] DID Provider Hi All, Could you please refer me to a best DID provider which also works perfectly with FreeSWITCH. I would need 50 channels per DID number. Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DID Provider
Please check here for more information. http://www.voip-info.org/ - Original Message From: Saeed Ahmed saeedahmad1...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, April 28, 2009 1:09:42 PM Subject: [Freeswitch-users] DID Provider Hi All, Could you please refer me to a best DID provider which also works perfectly with FreeSWITCH. I would need 50 channels per DID number. Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DID Provider
Hi, But which one would you prefer? In terms of cost and quality. Thanks. Paul wrote: FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH works with any major telco. - Original Message From: Saeed Ahmed saeedahmad1...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, April 28, 2009 1:09:42 PM Subject: [Freeswitch-users] DID Provider Hi All, Could you please refer me to a best DID provider which also works perfectly with FreeSWITCH. I would need 50 channels per DID number. Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@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] DID Provider
https://teliax.com/?referral_code=11 Teliax can provide 50 Channels per did. Contact me for more information. Thanks, Geoff Love 303-629-8304 On Tue, Apr 28, 2009 at 11:09 AM, Saeed Ahmed saeedahmad1...@gmail.comwrote: Hi All, Could you please refer me to a best DID provider which also works perfectly with FreeSWITCH. I would need 50 channels per DID number. 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 -- Geoff Love Sales Engineer gl...@teliax.com 303-629-8304 Referral Code 11 ___ Freeswitch-users mailing list Freeswitch-users@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] DID Provider
Go with Verizon. They're reliable and available anywhere in the US. - Original Message From: Saeed Ahmed saeedahmad1...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, April 28, 2009 1:21:46 PM Subject: Re: [Freeswitch-users] DID Provider Hi, But which one would you prefer? In terms of cost and quality. Thanks. Paul wrote: FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH works with any major telco. - Original Message From: Saeed Ahmed saeedahmad1...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, April 28, 2009 1:09:42 PM Subject: [Freeswitch-users] DID Provider Hi All, Could you please refer me to a best DID provider which also works perfectly with FreeSWITCH. I would need 50 channels per DID number. Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@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] DID Provider
The best quality and cheapest for all applications serving all markets around the world, all at the same time? Tall order. It's easier for people to give meaningful answers if you give some hints as to little details such as what country you're talking about (though the working assumption of the U.S. is probably relatively safe...). However, since you're talking about 50 channels per DID, there's a chance that you're talking about an application of the type that is starting to drive telcos a bit nuts. For example, Verizon, a perfectly fine provider of SIP trunking that I happen to use, just announced on the 22nd that high volume predictive auto-dialers and mass calling applications have resulted in such an increase in traffic so far in 2009 that they've put their wholesale customers on notice that certain calling patterns are liable to get your contract canceled. For example, an average call length per account over a given month of less than 78 seconds is no longer allowed. The ratio of concurrent channels used to call attempts per second also has to meet their new standards. A long way of saying that if you're talking about any type of call center or mass calling you're much better off doing your own, detailed research, especially on cost. -- --Jon Radel j...@radel.com Paul wrote: Go with Verizon. They're reliable and available anywhere in the US. - Original Message From: Saeed Ahmed saeedahmad1...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, April 28, 2009 1:21:46 PM Subject: Re: [Freeswitch-users] DID Provider Hi, But which one would you prefer? In terms of cost and quality. Thanks. Paul wrote: FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH works with any major telco. - Original Message From: Saeed Ahmed saeedahmad1...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, April 28, 2009 1:09:42 PM Subject: [Freeswitch-users] DID Provider Hi All, Could you please refer me to a best DID provider which also works perfectly with FreeSWITCH. I would need 50 channels per DID number. Thanks. smime.p7s Description: S/MIME Cryptographic Signature ___ Freeswitch-users mailing list Freeswitch-users@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] DID Provider
Yup, I would need it for inbound call center services. Thanks for your detailed reply. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jon Radel Sent: Tuesday, April 28, 2009 8:07 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] DID Provider The best quality and cheapest for all applications serving all markets around the world, all at the same time? Tall order. It's easier for people to give meaningful answers if you give some hints as to little details such as what country you're talking about (though the working assumption of the U.S. is probably relatively safe...). However, since you're talking about 50 channels per DID, there's a chance that you're talking about an application of the type that is starting to drive telcos a bit nuts. For example, Verizon, a perfectly fine provider of SIP trunking that I happen to use, just announced on the 22nd that high volume predictive auto-dialers and mass calling applications have resulted in such an increase in traffic so far in 2009 that they've put their wholesale customers on notice that certain calling patterns are liable to get your contract canceled. For example, an average call length per account over a given month of less than 78 seconds is no longer allowed. The ratio of concurrent channels used to call attempts per second also has to meet their new standards. A long way of saying that if you're talking about any type of call center or mass calling you're much better off doing your own, detailed research, especially on cost. -- --Jon Radel j...@radel.com Paul wrote: Go with Verizon. They're reliable and available anywhere in the US. - Original Message From: Saeed Ahmed saeedahmad1...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, April 28, 2009 1:21:46 PM Subject: Re: [Freeswitch-users] DID Provider Hi, But which one would you prefer? In terms of cost and quality. Thanks. Paul wrote: FreeSWITCH works with any SIP trunk provider. In other words, FreeSWITCH works with any major telco. - Original Message From: Saeed Ahmed saeedahmad1...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, April 28, 2009 1:09:42 PM Subject: [Freeswitch-users] DID Provider Hi All, Could you please refer me to a best DID provider which also works perfectly with FreeSWITCH. I would need 50 channels per DID number. Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Voice mail: skip announcement
Is there a way to skip the person at extension... announcement when forwarding a call to voice mail, so that it starts recording immediately, or just says something like start recording...? Also, is it possible to turn off voice mail play back feature, meaning when somebody tries to leave a voice mail one needs to press a button to end message and then VM suddenly plays it back? I see there's sounds.xml but not sure it help me to do what I need , is there any write up on how to customize it? Would appreciate some directions. ___ Freeswitch-users mailing list Freeswitch-users@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] Voice mail: skip announcement
This macro: macro name=voicemail_play_greeting input pattern=^(.*)$ match action function=play-file data=voicemail/vm-person.wav/ action function=say data=$1 method=pronounced type=name_spelled/ action function=play-file data=voicemail/vm- not_available.wav/ /match /input /macro in conf/lang/en/vm/sounds.xml This is what you would modify to do that. On Apr 28, 2009, at 3:00 PM, paul.degt wrote: Is there a way to skip the person at extension... announcement when forwarding a call to voice mail, so that it starts recording immediately, or just says something like start recording...? Also, is it possible to turn off voice mail play back feature, meaning when somebody tries to leave a voice mail one needs to press a button to end message and then VM suddenly plays it back? I see there's sounds.xml but not sure it help me to do what I need , is there any write up on how to customize it? Would appreciate some directions. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available
The link in the post refers to pre5, not pre6. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, April 28, 2009 12:44 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available The latest pre-release of 1.0.4 is available. More information here: http://www.freeswitch.org/node/177 The FreeSWITCH development team would like to thank everyone for their help in reporting bugs and testing features. Please keep the feedback coming and spread the word about what a cool project FreeSWITCH really is! -Michael See you at ClueCon - http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Voice mail: skip announcement
But is there a way to keep original macro as well and trigger them on demand from a dial plan? Brian West wrote: This macro: macro name=voicemail_play_greeting input pattern=^(.*)$ match action function=play-file data=voicemail/vm-person.wav/ action function=say data=$1 method=pronounced type=name_spelled/ action function=play-file data=voicemail/vm-not_available.wav/ /match /input /macro in conf/lang/en/vm/sounds.xml This is what you would modify to do that. On Apr 28, 2009, at 3:00 PM, paul.degt wrote: Is there a way to skip the person at extension... announcement when forwarding a call to voice mail, so that it starts recording immediately, or just says something like start recording...? Also, is it possible to turn off voice mail play back feature, meaning when somebody tries to leave a voice mail one needs to press a button to end message and then VM suddenly plays it back? I see there's sounds.xml but not sure it help me to do what I need , is there any write up on how to customize it? Would appreciate some directions. Brian West br...@freeswitch.org mailto:br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com http://www.cluecon.com/ ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available
Fixed. /b On Apr 28, 2009, at 3:38 PM, Alex Rambau wrote: The link in the post refers to pre5, not pre6. Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Voice mail: skip announcement
You could just record a greeting and copy a file thats 1ms long over the top of it. /b On Apr 28, 2009, at 3:46 PM, paul.degt wrote: But is there a way to keep original macro as well and trigger them on demand from a dial plan? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] any way ring fifo members one by one?
oh, thank you Antonio. I think it would be better to collect more ideas before open a bounty. And I more interested in playing(including patching the code) with that than use the function. On Apr 28, 2009, at 11:39 PM, Antonio Gallo wrote: dujinfang ha scritto: Yeah but FS is more than a *, it should be able to do a call center like job. I think if you open a bounty you can get some nice feedback about this. Antonio ___ Freeswitch-users mailing list Freeswitch-users@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] SNMP or Cacti to graph perrformance?
Hi List, I have searched and searched and have found nothing. Id like to be able to keep track of current calls and the CPS and then chart these via Cacti. Is their some a SNMP plugin that I have not seen for FS that can do some of these things? Also, if I have to make some custom plugin that montiors output the of show channels /show calls via one of the API's, anyway we can get a realtime stream so I don't have to query it every second to get correct data? Thanks! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SNMP or Cacti to graph perrformance?
every second is way too high of a resolution. 10 seconds 20 seconds are more sane! /b On Apr 28, 2009, at 9:11 PM, Ron McCarthy wrote: lso, if I have to make some custom plugin that montiors output the of show channels /show calls via one of the API's, anyway we can get a realtime stream so I don't have to query it every second to get correct data? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SNMP or Cacti to graph perrformance?
will hitting it every 10 or 20 seconds hurt performance / eat a lot of CPU? Only reason id like to hit every second so I could get some sort of attempt on the call setups and call transactions. Has anyone even done anything like this before? Thanks On Tue, Apr 28, 2009 at 7:16 PM, Brian West br...@freeswitch.org wrote: every second is way too high of a resolution. 10 seconds 20 seconds are more sane! /b On Apr 28, 2009, at 9:11 PM, Ron McCarthy wrote: lso, if I have to make some custom plugin that montiors output the of show channels /show calls via one of the API's, anyway we can get a realtime stream so I don't have to query it every second to get correct data? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SNMP or Cacti to graph perrformance?
Connect to the socket and subscribe to the heartbeat. The heartbeat event fires every 20 seconds and contains the session count. On Tue, Apr 28, 2009 at 21:19, Ron McCarthy ronmc...@gmail.com wrote: will hitting it every 10 or 20 seconds hurt performance / eat a lot of CPU? Only reason id like to hit every second so I could get some sort of attempt on the call setups and call transactions. Has anyone even done anything like this before? Thanks On Tue, Apr 28, 2009 at 7:16 PM, Brian West br...@freeswitch.org wrote: every second is way too high of a resolution. 10 seconds 20 seconds are more sane! /b On Apr 28, 2009, at 9:11 PM, Ron McCarthy wrote: lso, if I have to make some custom plugin that montiors output the of show channels /show calls via one of the API's, anyway we can get a realtime stream so I don't have to query it every second to get correct data? Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Shannon ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SNMP or Cacti to graph perrformance?
On Apr 28, 2009, at 9:19 PM, Ron McCarthy wrote: will hitting it every 10 or 20 seconds hurt performance / eat a lot of CPU? No its just silly in my opinion if you want that then hook on event socket using ESL and write something to register callbacks for each event and collect stats. Then you have realtime stats collection you can query or dump out any way you see fit. Only reason id like to hit every second so I could get some sort of attempt on the call setups and call transactions. we already collect those stats on the sofia profiles. calls-in14/calls-in calls-out11/calls-out failed-calls-in2/failed-calls-in failed-calls-out3/failed-calls-out sofia xmlstatus profile ;) Has anyone even done anything like this before? Thanks Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SNMP or Cacti to graph perrformance?
That's really I want that looks perfect, anyways I can get stats for a gateway? Far as the calls-in and calls out? If not the heartbeat looks like I could writ esomething up and use that to. On Tue, Apr 28, 2009 at 7:33 PM, Brian West br...@freeswitch.org wrote: On Apr 28, 2009, at 9:19 PM, Ron McCarthy wrote: will hitting it every 10 or 20 seconds hurt performance / eat a lot of CPU? No its just silly in my opinion if you want that then hook on event socket using ESL and write something to register callbacks for each event and collect stats. Then you have realtime stats collection you can query or dump out any way you see fit. Only reason id like to hit every second so I could get some sort of attempt on the call setups and call transactions. we already collect those stats on the sofia profiles. calls-in14/calls-in calls-out11/calls-out failed-calls-in2/failed-calls-in failed-calls-out3/failed-calls-out sofia xmlstatus profile ;) Has anyone even done anything like this before? Thanks Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@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] TLS initialization failures
I know this isn't the place to report bugs; unfortunately, the Jira Web interface isn't working for me due to accessibility issues. (If there is an alternative way to submit reports that could be efficiently handled by the developers, let me know). A few weeks ago I reported problems with the initialization of multiple SIP profiles for which TLS was enabled. At the time, I suspected the underlying cause was an attempt to bind to port 5061 under IPv4 and IPv6 separately. It turns out, however, that even if the TLS ports in all of the profiles are distinct, one or more of the profiles often fails to start when FreeSWITCH is first loaded. Running sofia profile start profile-name thereafter always succeeds; thus I suspect a subtle timing problem. I am having trouble reproducing this with the Sofia debugging variables enabled. I also have a core file that was generated when FreeSWITCH crashed while attempting to initialize TLS during startup. I can supply a backtrace if this would help. The segfault is not reliably reproducable, but it has happened more than once, most recently under rev. 13081, for which I kept the core file. I am running Debian Sid. A colleague with Fedora 10 has also been able to reproduce the TLS initialization bug. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
