Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
you are right for the regex. This is part of an old setup, correct with:
extension name=PEER_01
condition field=${ROUTE_GW} expression=PEER_01
action application=set data=hangup_after_bridge=true/
Hristo Benev a écrit :
Hello Rod,
I did the change.
Here is extract of console:
-
variable_continue_on_fail: [true]
variable_sip_h_X-ROUTE: [LOOKUP]
variable_export_vars: [sip_h_X-ROUTE]
variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20]
variable_sip_redirect_contact_0: [sip:fra...@peer_01]
variable_sip_redirected_to: [sip:fra...@peer_01]
variable_sip_redirect_contact_user_0: [France]
---
variable_sip_redirect_contact_host_0: [PEER_01]
variable_sip_redirect_dialstring_0: [sofia/internal/sip:fra...@peer_01]
variable_sip_redirect_dialstring: [sofia/internal/sip:fra...@peer_01]
variable_proto_specific_hangup_cause: [sip:503]
variable_sip_hangup_phrase: [DNS Error]
variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE]
variable_ROUTE_GW: [France]
variable_AREA: [France]
variable_current_application: [info]
--
I have different value it is actually the description field as shown here:
/opt/kamailio/sbin/kamctl cr show
cr carrier names
++-+
| id | carrier |
++-+
| 1 | default |
++-+
cr domain names
++-+
| id | domain |
++-+
| 1 | default |
++-+
cr routes
++-++-+---+--+--+---+--+++-+
| id | carrier | domain | scan_prefix | flags | mask | prob | strip |
rewrite_host | rewrite_prefix | rewrite_suffix | description |
++-++-+---+--+--+---+--+++-+
| 1 | 1 | 1 | 1000| 0 |0 |1 | 0 | PEER_01
||| France |
++-++-+---+--+--+---+--+++-+
-
And here is what I have in kamailio:
-
### Routing Logic
# main request routing logic
route{
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
}
t_check_trans();
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply(484,Address Incomplete);
exit;
}
# LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP
if (is_method(INVITE) $hdr(X-ROUTE)==LOOKUP){
if(!cr_route(default, default, $rU, $rU,
call_id,$avp(s:route_desc))){
#xlog(ROUTING FAILED: no route found for $rU);
sl_send_reply(604, Unable to route this call);
exit;
} else {
xlog(LOOKUP FOUND: $rd $avp(s:route_desc));
avp_pushto($ru/username, $avp(s:route_desc));
sl_send_reply(302, $rd);
exit;
}
}
}
--
Another question...
In that part of FreeSwitch dialplan.xml
---
context name=ROUTING
extension name=PEER_01
condition field=${sip_h_X-ROUTE} expression=PEER_01
action application=set data=hangup_after_bridge=true/
---
X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the
regex is true.
As for thanks - for sure by default they are also for the developers of both
apps.
I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning
so it will be nice to have something working to start with.
Оригинално писмо
От: rod
Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Петък, 2009, Август 28 09:54:00 EEST
Hello,
the trace seems good.
If you check the answer from Kamailio, you'll see that Kamailio answers
with 302 PEER_01.
As Michael Collins stated before, you can get the variable containing
PEER_01, then this variable is stored in a custom variable.
In your dialplan, may you please add:
, just before the transfer line, eg:
Using application Info,
[Freeswitch-users] FreeSWITCH environmental variables
Dear all, In the case of asterisk PBX. I can get all the information about the call from the environmental variable itself like Caller id ,called id, channel no. But in freeswitch where all the environmental variables are resides? How can I access all the variables? Please Help me? -- Regards, Thangappan.M ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 environmental variables
Check out INFO http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info Throw that in your dialplan the look at your logs... You should find what your after.. Jay On 29/08/2009, at 16:45, Thangappan.M thangappan...@gmail.com wrote: Dear all, In the case of asterisk PBX. I can get all the information about the call from the environmental variable itself like Caller id ,called id, channel no. But in freeswitch where all the environmental variables are resides? How can I access all the variables? Please Help me? -- Regards, Thangappan.M ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FreeSWITCH environmental variables
Try to read docuemntation first: http://wiki.freeswitch.org/wiki/Channel_Variables 2009/8/29 Thangappan.M thangappan...@gmail.com Dear all, In the case of asterisk PBX. I can get all the information about the call from the environmental variable itself like Caller id ,called id, channel no. But in freeswitch where all the environmental variables are resides? How can I access all the variables? Please Help me? -- Regards, Thangappan.M ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FreeSWITCH HA + Loadbalancing
On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola diego.vi...@gmail.com wrote: Yes, FreeSWITCH is a system that you can trust 100%. I have switched my Asterisk servers to FreeSWITCH and have peace now. If I were you I would get rid of Asterisk and use FreeSWITCH, FS will handle all what you want very well. And I agree with David, fail-over is kinda irrelevant since the FS doesn't crash like Asterisk does. You still have hardware failures and fail-over is also useful for hit-less maintenance on boxes. I'd be interested to know how Brian West was approaching his live migration work. Steve ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Success! and an issue.
well, just made my first extension to extension call using freeswitch, so that was pretty painless. Huge fan of the config layout at least (having the extensions defined each in their own file is going to be so much nicer than asterisks huge jumble). However, I did run into one issue, not sure if its somehow related to the phones I'm using (softphone of unknown quality (weephone) on iphones)... I'm using the default config, when I dial 4000 for voicemail, it works, however at apparently random times during the menu navigation I get call waiting beeps and I have an incoming call from 4000. If I decline or leave this call unanswered, my call into voicemail gets disconnected. If I accept this call, it apparently disconnects immediately (IE I only have 1 active call), however I can then continue navigating the menu, etc... Very strange to me, I will be setting up a couple of polycom 501s on this system tomorrow and will report if I see similar issues. Anyway, so far I'm pleased with freeswitch, I've got a list of features I need to support before I can deploy it, so I'll be bugging people (sorry), but so far thanks! -Tom ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] newbie questions
hi, totally new to freeswitch, but not to asterisk. just installed it, made samples but do not have 1000-1019 as extensions. so i thought i create: include user id=1000 mailbox=1000 params param name=password value=1234/ param name=vm-password value=1000/ /params variables variable name=accountcode value=1000/ variable name=user_context value=default/ variable name=effective_caller_id_name value=Extension 1000/ variable name=effective_caller_id_number value=1000/ /variables /user /include and save it as 1000.xml in the internal-sip folder. cli tells me now: 2009-08-29 09:43:55.858101 [WARNING] sofia_reg.c:1771 Can't find user [ 1...@192.168.1.4] You must define a domain called '192.168.1.4' in your directory and add a user with the id=1000 attribute and you must configure your device to use the proper domain in it's authentication credentials. what do i have to do now to get it a) internally running b) talk to an external cllient? thx ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Disabling Core Dumping
Hi, Whenever i stop freeswich, it creates a core dump. How can i disable that? thanks, Max. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_limit and memcache
Ok, it was a bit more work than expected, but I have a first cut at mod_limit using memcache checked in. Test if you can. I've not put docs on the wiki 'cause some of the implementation / setup details are going to change with regards to mod_memcache. Oh, and mod_limit's api may be changing to make it more pluggable. Basically, Setup mod_memcache (on wiki) It uses the same user facing api as limit_hash and friends except it is limit_memcache and limit_memcache_usage. so follow mod_limit's docs with suitable substitutions. On Thu, Aug 27, 2009 at 9:42 AM, Rupa Schomakerr...@rupa.com wrote: limit and memcache haven't been introduced to each other yet -- it is on my (semi-long) list of things to do. If you want it you can: 1) do it yourself and submit the patches 2) open a jira and hope someone does it 3) open a jira + bounty and someone will probably do it It will get done eventually, just hasn't been a itch for ME to scratch yet. To do it: 1) I need to make it possible to call inc/dec methods of mod_memcache to support an expiration time. 2) mod_limit.c - use the hash limit as a guide Initial pitfalls: hash limits concurrent access/modification of the hash and by implication limit_hash_item_t (hash data) by using a mutex. We can't mutex across FS instances. So perhaps split up limit_has_item_t and spread it across multiple keys. So instead of one key marked as realm_id, we could have realm_id_total_usage realm_id_rate_usage and realm_id_last_check. This does mean that rate_usage and total_usage can inc/dec independent of each other, but I think the logic will still be ok *IF* we remember to decrement earlier incremented items in the event a later item is failed. (so, say we increment rate but fail on total we need to remember to decrement rate so that we have no net effect on the counters) Alternatively, we could use CAS support and pull the limit_hash_item_t item from memcache, twiddle it and then try to put it back only if the check info is the same (no one else has changed the entry). If the entry has changed, pull the new version, do the limit logic, and try again. Loop that a few times until you succeed or give up. Problem is that CAS needs to be explicitly turned on in memcache (some distros compile with it off), is relatively new in memcache (hint: may have issues) and has some performance/memory downsides though by how much I'm not sure. Thoughts? On Thu, Aug 27, 2009 at 8:49 AM, Woody Dicksonwoodydick...@gmail.com wrote: Hello, I read something that talks about using memcache for mod_limit before. Is it something that is available now? If I have multiple instances of freeswitch that need to share the same limit status, it there any existing solution? If no existing solution is available, what is the best way to go about modifying mod_limit to accomplish limiting for multiple freeswitch servers together? Thanks, Woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -Rupa -- -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
Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Thinking about it, maybe we can create a solution, if some of us work together: My strength are in virtualization, linux, development, databases, integration, etc. What I do not now much about is how SIP (and everything else for that matter in the Voice world) works under the hood, and how it's implemented in FS. I know that the state information for a call has to be stored and retrieved somewhere and somehow, only I do not know that part. What I know is that it hast to be do-able to store all the stream information (ip's, port's, current state's, etc.) in a very fast database (e.g. my idea would be memcached) so another FS could just take this information and take over the call, maybe you loose a second of voice, maybe you loose the recorded call file or a part of it, but that should be it. (SipFoundry has a boxed opensource PBX, which, of course is not flexible like FreeSWITCH or Asterisk, but has Call Live Migration and Call Live Failover integrated!). What I want is for my company to be able to sell a 99.99 uptime PBX (we do mostly call-center related stuff), which can scale well, and can grow with the company without lot's of hassles, my Dream would be: To begin with: One Hardware Node with the essential hardware (digium cards for example). On this node are OpenVZ virtualized containers: [VirtCnt1: FS which only talks to the Hardware and forwards everything] = Could be replaced with hardware media gateway, etc. [VirtCnt2: FS which handles the PBX] \___ Loadbalanced, with odbc or xml, Failover, Livetakeover [VirtCnt3: FS which handles the PBX] / [VirtCnt4: Database for state information] (maybe something as resource-friendly as memcached? ressource heavvy database?) With this we can achieve all this: Problem with VirtCnt2 (e.g. crash, lock, ...) * VirtCnt3 can take over. - You are free without stress to investigate the problem, you can debug and analyze whyle the machine is still running - you can also create a machine-state-dump of the virtual container, dump the container as well, copy the data to your lab and restore the machine up the state which it was running with the problem, so you can liveinvestigate it in the lab (some prerequirements given, but easy doable) - just think about the possibility of better bugreports because someone can take the time to read out all the data with GDB to investigate the proper cause of a machine Lock! You want to upgrade to a new FreeSWITCH version? * Take VirtCnt2 out of the LoadBalancing Scheme, * Stop it, Clone it, * Upgrade FreeSWITCH in the cloned Container * Start the cloned container * if there's something wrong, stop it and restart the original VirtCnt2 - No problem at all, you can Test on the Live Hardware, with part of the Live users (maybe a low-volume queue) to be sure everything works out fine before you activate the full loadbalance Server on it's own can't handle the load * Buy new machine * Setup Hardware Node * Livemigrate VirtCnt3 (no downtime) Now the first Server with the VrtCnt1 and VirtCnt2 as well has to much load * Buy new machine * Setup Hardware Node * Livemigrate VirtCnt2 (no downtime) - Now you have a 3 server solution (1 mediaprox, 2 loadbalanced / failover PBXes) out of the first box you bought, without headaches, because the system was built for it from the beginning! The Database drains to much? * Buy new machine * Setup Hardware Node * Livemigrate database VirtCnt4 (no downtime) You want to upgrade Hardware/Kernel in Hardware node 1? * Livemigrate VirtCnt2 to a hotstandby machine, or to the other PBX machine, upgrade the hardware, Re-Livemigrate the containers. (no downtime) * OR just break the loadbalancing, wait until all current calls are teared down correctly, upgrade machine, reenable the loadbalancer You want an exact copy of the first server for Hardware HA? * Buy new machine * Setup Hardware node * Buy hardware PRI switchover box * Clone VirtCnt1 - VirtCnt4 to the new machine * Make basic failover configuration - the sky's the limit, as the saying goes ... So, I can do all the openvz stuff and the integration with database / memcached / heartbeat / whatever is needed here, someone there to be willing to work with me on this on the FreeSWITCH side? or at least provide me with the necessary information about what's needed / how to talk / what states from FreeSWITCH? I know this seems very ambitious but if this could be made in a rather relativly easy to setup package, with good documentation, it would be a boost for FreeSWITCH, i am sure, because after all this is what everyone is grown accustomed to from good old phone companys and the good old pbx's: carrier grade uptimes ... Thanks for everyone reading up until here, all the best, Ray -- Raimund Sacherer - RunSolutions Open Source It Consulting - Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca
Re: [Freeswitch-users] Group Call
Don't confuse this with the actual group/ endpoint. /b On Aug 25, 2009, at 11:41 AM, João Mesquita wrote: According to wiki: group,[insert|delete|call]:group name:url,group [insert|delete| call ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Disabling Core Dumping
report the issue on jira If its dumping a core something is wrong and we need to fix it. /b On Aug 29, 2009, at 9:07 AM, Max Bridgewater wrote: Hi, Whenever i stop freeswich, it creates a core dump. How can i disable that? thanks, Max. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Success! and an issue.
Can you get a sip trace and console trace because this is highly unlikely. /b On Aug 29, 2009, at 4:28 AM, Christensen Tom wrote: However, I did run into one issue, not sure if its somehow related to the phones I'm using (softphone of unknown quality (weephone) on iphones)... I'm using the default config, when I dial 4000 for voicemail, it works, however at apparently random times during the menu navigation I get call waiting beeps and I have an incoming call from 4000. If I decline or leave this call unanswered, my call into voicemail gets disconnected. If I accept this call, it apparently disconnects immediately (IE I only have 1 active call), however I can then continue navigating the menu, etc... Very strange to me, I will be setting up a couple of polycom 501s on this system tomorrow and will report if I see similar issues. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] newbie questions
yes u are right, i was mistaken. everything in there by default...still the same domain error msg. what am i supposed to do? tjx ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 HA + Loadbalancing
I was able to do this using OpenVZ, You can get away with it on smaller instances... like if you're doing one instance per company but don't expect live migration to work as well on large instances with thousands of calls up at once. You need a fast network, fast disks and to follow the howto on the wiki. /b On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: You still have hardware failures and fail-over is also useful for hit-less maintenance on boxes. I'd be interested to know how Brian West was approaching his live migration work. Steve ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] newbie questions
Are you on the latest configs? This message is unlikely in the default configs. Please double check that you have the internal.xml from the default install it forces the register domain to match the IP so you won't get this error. /b On Aug 29, 2009, at 10:19 AM, tom wrote: yes u are right, i was mistaken. everything in there by default...still the same domain error msg. what am i supposed to do? tjx ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] newbie questions
mh, a search for domain= doesnt brin anything up ininternal.xml... im total new to this...sorry do i ahev to add this? //yes im using the 1.0.4 tar On Sat, Aug 29, 2009 at 11:22 AM, Brian West br...@freeswitch.org wrote: Are you on the latest configs? This message is unlikely in the default configs. Please double check that you have the internal.xml from the default install it forces the register domain to match the IP so you won't get this error. /b On Aug 29, 2009, at 10:19 AM, tom wrote: yes u are right, i was mistaken. everything in there by default...still the same domain error msg. what am i supposed to do? tjx ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] newbie questions
Tom,
You should not have to do anything other than make samples. If you are
having these kinds of problems then something went very wrong during
the install. Try starting from scratch just to be sure. Also, what
distro are you running?
-MC
Sent from my iPhone
On Aug 29, 2009, at 8:40 AM, tom tomabr...@gmail.com wrote:
ok, what u sent is fine, i have that. do i have to do somethign in
that section:?
domains
!-- indicator to parse the directory for domains with
parse=true to get gateways--
domain name=$${domain} parse=true/
!-- indicator to parse the directory for domains with
parse=true to get gateways and alias every domain to this profile
--
!--domain name=all alias=true parse=true/--
domain name=all alias=true parse=false/
/domains
?thx
On Sat, Aug 29, 2009 at 11:33 AM, Brian West br...@freeswitch.org
wrote:
If you don't have these
!--all inbound reg will look in this domain for the users --
param name=force-register-domain value=$${domain}/
!--all inbound reg will stored in the db using this domain --
param name=force-register-db-domain value=$${domain}/
Then you need to wipe your conf dir and make samples again please.
/b
On Aug 29, 2009, at 10:28 AM, tom wrote:
mh, a search for domain= doesnt brin anything up
ininternal.xml... im total new to this...sorry
do i ahev to add this?
//yes im using the 1.0.4 tar
On Sat, Aug 29, 2009 at 11:22 AM, Brian West br...@freeswitch.org
wrote:
Are you on the latest configs? This message is unlikely in the
default configs. Please double check that you have the internal.xml
from the default install it forces the register domain to match the
IP
so you won't get this error.
/b
On Aug 29, 2009, at 10:19 AM, tom wrote:
yes u are right, i was mistaken. everything in there by
default...still the same domain error msg. what am i supposed to
do?
tjx
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Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
We have previously estimated the development of live fail over (after a box dies where live migration is no longer possible) to exceed 100k in development costs. It requires several additions to the sofia sip library, freeswitch and a dependancy on some other code we would have to implement to manage it. It may or may not be worth it to raise that kind of funding just to avoid an occasional disaster. Then there is a matter of securing the time of the developers necessary to carry out the implementation. On Aug 29, 2009 10:19 AM, Brian West br...@freeswitch.org wrote: I was able to do this using OpenVZ, You can get away with it on smaller instances... like if you're doing one instance per company but don't expect live migration to work as well on large instances with thousands of calls up at once. You need a fast network, fast disks and to follow the howto on the wiki. /b On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: You still have hardware failures and fail-over... ___ FreeSWITCH-users mailing list freeswitch-us...@lists... ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] external sip profile
No, enable sip trace I suspect your config is wrong in the end point. /b On Aug 29, 2009, at 11:05 AM, e schmidbauer wrote: I am using a fresh install of freeswitch trunk and I am unable to register a phone on port 5080. I set the the registrar of the phone to server.host.com:5080 but when i do a sofia status profile external, there are no registrations. when i do a sofia status profile internal, it shows the phone registered to port 5060. Is this a bug in freeswitch or is there some configuration I need to change? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] newbie questions
No you shouldn't have to touch ANYTHING in the default configs at all
to get 1000 with password 1234 to register... I recommend you:
cd /usr/local/freeswitch
mv conf conf.old
then do make samples again.
/b
On Aug 29, 2009, at 10:40 AM, tom wrote:
ok, what u sent is fine, i have that. do i have to do somethign in
that section:?
domains
!-- indicator to parse the directory for domains with
parse=true to get gateways--
domain name=$${domain} parse=true/
!-- indicator to parse the directory for domains with
parse=true to get gateways and alias every domain to this profile
--
!--domain name=all alias=true parse=true/--
domain name=all alias=true parse=false/
/domains
?thx
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Re: [Freeswitch-users] Success! and an issue.
This sounds to me like a phone that doesn't know that a re-invite is not a new call and session timers are turned on. A sip trace will show you if this is the case, and if it is, that phone is indeed badly broken. Mike On Aug 29, 2009, at 11:15 AM, Brian West wrote: Can you get a sip trace and console trace because this is highly unlikely. /b On Aug 29, 2009, at 4:28 AM, Christensen Tom wrote: However, I did run into one issue, not sure if its somehow related to the phones I'm using (softphone of unknown quality (weephone) on iphones)... I'm using the default config, when I dial 4000 for voicemail, it works, however at apparently random times during the menu navigation I get call waiting beeps and I have an incoming call from 4000. If I decline or leave this call unanswered, my call into voicemail gets disconnected. If I accept this call, it apparently disconnects immediately (IE I only have 1 active call), however I can then continue navigating the menu, etc... Very strange to me, I will be setting up a couple of polycom 501s on this system tomorrow and will report if I see similar issues. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FreeSWITCH HA + Loadbalancing
On PRI circuits failover with ha is possible using a red tone appliance, but I m sure it's going to be almost impossible to transfer the running calls during the event of fs box going down. Thanks Regards, Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 29-Aug-2009, at 11:35 PM, David Knell d...@3c.co.uk wrote: Hi Raimund, The difficult bit in all of this is having calls seamlessly transferred from one box to another when the first box dies. There's a lot of state associated with a call, not all of which is easy to replicate across. 99.99% uptime implies an average of no more than 15 seconds downtime during a 40 hour business week (or about one unscheduled reboot every couple of months), which is easily achievable using FreeSWITCH (and, in my experience, Asterisk) on standard hardware. People agonize about their four or five nines too much, in my opinion. Folk are used to their phones crashing, needing rebooting and dropping calls - we've cellphones to thank for that - and a half-decent VoIP solution will knock the spots off your favourite mobile carrier for reliability. Plus there's all the external factors - 99.999% uptime on your PRI means that someone can only drive a digger through the cable once every 273 years, assuming that it takes a day to fix, and no telco's going to give you that in an SLA. And your power can't go off. And so on. Lastly, I'm afraid that virtualization and VoIP don't play well together, at least not if you want to achieve a sensible density. The large number of small packets being moved around the network interfaces - both physical and virtual - will quickly chew up your CPU. Cheers -- Dave Thinking about it, maybe we can create a solution, if some of us work together: My strength are in virtualization, linux, development, databases, integration, etc. What I do not now much about is how SIP (and everything else for that matter in the Voice world) works under the hood, and how it's implemented in FS. I know that the state information for a call has to be stored and retrieved somewhere and somehow, only I do not know that part. What I know is that it hast to be do-able to store all the stream information (ip's, port's, current state's, etc.) in a very fast database (e.g. my idea would be memcached) so another FS could just take this information and take over the call, maybe you loose a second of voice, maybe you loose the recorded call file or a part of it, but that should be it. (SipFoundry has a boxed opensource PBX, which, of course is not flexible like FreeSWITCH or Asterisk, but has Call Live Migration and Call Live Failover integrated!). What I want is for my company to be able to sell a 99.99 uptime PBX (we do mostly call-center related stuff), which can scale well, and can grow with the company without lot's of hassles, my Dream would be: To begin with: One Hardware Node with the essential hardware (digium cards for example). On this node are OpenVZ virtualized containers: [VirtCnt1: FS which only talks to the Hardware and forwards everything] = Could be replaced with hardware media gateway, etc. [VirtCnt2: FS which handles the PBX] \___ Loadbalanced, with odbc or xml, Failover, Livetakeover [VirtCnt3: FS which handles the PBX] / [VirtCnt4: Database for state information] (maybe something as resource-friendly as memcached? ressource heavvy database?) With this we can achieve all this: Problem with VirtCnt2 (e.g. crash, lock, ...) * VirtCnt3 can take over. - You are free without stress to investigate the problem, you can debug and analyze whyle the machine is still running - you can also create a machine-state-dump of the virtual container, dump the container as well, copy the data to your lab and restore the machine up the state which it was running with the problem, so you can liveinvestigate it in the lab (some prerequirements given, but easy doable) - just think about the possibility of better bugreports because someone can take the time to read out all the data with GDB to investigate the proper cause of a machine Lock! You want to upgrade to a new FreeSWITCH version? * Take VirtCnt2 out of the LoadBalancing Scheme, * Stop it, Clone it, * Upgrade FreeSWITCH in the cloned Container * Start the cloned container * if there's something wrong, stop it and restart the original VirtCnt2 - No problem at all, you can Test on the Live Hardware, with part of the Live users (maybe a low-volume queue) to be sure everything works out fine before you activate the full loadbalance Server on it's own can't handle the load * Buy new machine * Setup Hardware Node * Livemigrate VirtCnt3 (no downtime) Now the first Server with the VrtCnt1 and VirtCnt2 as well has to much load * Buy new machine * Setup Hardware Node * Livemigrate
Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Hmm, so basically 100 interested companys which each chip in 1000 bucks :-) sounds like lots of manpower, but on the other hand, I do not know the issues regarding the SIP protocoll, but, basically, isn't it *just* to tell another FS box to listen on port x for voicetraffic, forward it to ip on port y? ok, i understand there's a lot going on under the hood, i guess it would mean to setup a call, but take care to not really set up the call, just the internal state ... hmm, could it theoretically be done with the event system? ok, i guess I have to dive further into the internals to fully understand the scope. But a live migration, where the box is available, be possible right now? Would be a step i would like to implement just to be able to do work on a hardware node if necesary without interrupting the service ... -- Raimund Sacherer - RunSolutions Open Source It Consulting - Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares On Aug 29, 2009, at 6:01 PM, Anthony Minessale wrote: We have previously estimated the development of live fail over (after a box dies where live migration is no longer possible) to exceed 100k in development costs. It requires several additions to the sofia sip library, freeswitch and a dependancy on some other code we would have to implement to manage it. It may or may not be worth it to raise that kind of funding just to avoid an occasional disaster. Then there is a matter of securing the time of the developers necessary to carry out the implementation. On Aug 29, 2009 10:19 AM, Brian West br...@freeswitch.org wrote: I was able to do this using OpenVZ, You can get away with it on smaller instances... like if you're doing one instance per company but don't expect live migration to work as well on large instances with thousands of calls up at once. You need a fast network, fast disks and to follow the howto on the wiki. /b On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: You still have hardware failures and fail-over... ___ FreeSWITCH-users mailing list freeswitch-us...@lists... ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FreeSWITCH HA + Loadbalancing
It is not possible to do a live migration at all right now... There is no way to move a call and have all the states required for that call to magically re-appear on a different instance. This will require a fair bit of work to get there. It is possible to configure fs via some back end DB magic to share configurations ie: n+1 or ³warm standby² style fault tolerance but you are going to loose the calls that are up when the failure occurs. From: Raimund Sacherer r...@runsolutions.com Reply-To: freeswitch-users@lists.freeswitch.org Date: Sat, 29 Aug 2009 20:33:21 +0200 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing Hmm, so basically 100 interested companys which each chip in 1000 bucks :-) sounds like lots of manpower, but on the other hand, I do not know the issues regarding the SIP protocoll, but, basically, isn't it *just* to tell another FS box to listen on port x for voicetraffic, forward it to ip on port y? ok, i understand there's a lot going on under the hood, i guess it would mean to setup a call, but take care to not really set up the call, just the internal state ... hmm, could it theoretically be done with the event system? ok, i guess I have to dive further into the internals to fully understand the scope. But a live migration, where the box is available, be possible right now? Would be a step i would like to implement just to be able to do work on a hardware node if necesary without interrupting the service ... -- Raimund Sacherer - RunSolutions Open Source It Consulting - Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares On Aug 29, 2009, at 6:01 PM, Anthony Minessale wrote: We have previously estimated the development of live fail over (after a box dies where live migration is no longer possible) to exceed 100k in development costs. It requires several additions to the sofia sip library, freeswitch and a dependancy on some other code we would have to implement to manage it. It may or may not be worth it to raise that kind of funding just to avoid an occasional disaster. Then there is a matter of securing the time of the developers necessary to carry out the implementation. On Aug 29, 2009 10:19 AM, Brian West br...@freeswitch.org wrote: I was able to do this using OpenVZ, You can get away with it on smaller instances... like if you're doing one instance per company but don't expect live migration to work as well on large instances with thousands of calls up at once. You need a fast network, fast disks and to follow the howto on the wiki. /b On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: You still have hardware failures and fail-over... ___ FreeSWITCH-users mailing list freeswitch-us...@lists... ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FreeSWITCH HA + Loadbalancing
Is itnpossible to have a db cluster know the state of each and every call and then use Heartbeat on this db + fs cluster so that clients see only one ip where as internally all fs boxes refer db for call states, db again is under replication. This in the thioery can be written, but I am sure if we think bit more on this direction the problem seem to be getting addressed. Other guys also chip in their 2 cents, we just need 50 of em to make a full dollar. Thanks Regards, Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 30-Aug-2009, at 12:14 AM, Ken Rice kr...@freeswitch.org wrote: It is not possible to do a live migration at all right now... There is no way to move a call and have all the states required for that call to magically re-appear on a different instance. This will require a fair bit of work to get there. It is possible to configure fs via some back end DB magic to share configurations ie: n+1 or “warm standby” style fault tolerance but you are going to loose the calls that are up when the failure oc curs. From: Raimund Sacherer r...@runsolutions.com Reply-To: freeswitch-users@lists.freeswitch.org Date: Sat, 29 Aug 2009 20:33:21 +0200 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing Hmm, so basically 100 interested companys which each chip in 1000 bucks :-) sounds like lots of manpower, but on the other hand, I do not know the issues regarding the SIP protocoll, but, basically, isn't it *just* to tell another FS box to listen on port x for voicetraffic, forward it to ip on port y? ok, i understand there's a lot going on under the hood, i guess it would mean to setup a call, but take care to not really set up the call, just the internal state ... hmm, could it theoretically be done with the event system? ok, i guess I have to dive further into the internals to fully understand the scope. But a live migration, where the box is available, be possible right now? Would be a step i would like to implement just to be able to do work on a hardware node if necesary without interrupting the service ... -- Raimund Sacherer - RunSolutions Open Source It Consulting - Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares On Aug 29, 2009, at 6:01 PM, Anthony Minessale wrote: We have previously estimated the development of live fail over (after a box dies where live migration is no longer possible) to exceed 100k in development costs. It requires several additions to the sofia sip library, freeswitch and a dependancy on some other code we would have to implement to manage it. It may or may not be worth it to raise that kind of funding just to avoid an occasional disaster. Then there is a matter of securing the time of the developers necessary to carry out the implementation. On Aug 29, 2009 10:19 AM, Brian West br...@freeswitch.org wrote: I was able to do this using OpenVZ, You can get away with it on smaller instances... like if you're doing one instance per company but don't expect live migration to work as well on large instances with thousands of calls up at once. You need a fast network, fast disks and to follow the howto on the wiki. /b On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: You still have hardware failures and fail-over... ___ FreeSWITCH-users mailing list freeswitch-us...@lists... ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FreeSWITCH HA + Loadbalancing
The sip stack needs to be modified to spin that data up into the state machine so that it can take over calls once the fail over takes place... its not an easy task. /b On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote: Is itnpossible to have a db cluster know the state of each and every call and then use Heartbeat on this db + fs cluster so that clients see only one ip where as internally all fs boxes refer db for call states, db again is under replication. This in the thioery can be written, but I am sure if we think bit more on this direction the problem seem to be getting addressed. Other guys also chip in their 2 cents, we just need 50 of em to make a full dollar. Thanks Regards, Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.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 HA + Loadbalancing
I am also in the process of playing around with FS running inside Xen Virtual Machines with "mirrored" VMs on a second failover system. So far, the initial tests are promising. I can see a 2-3 sec "hiccup" in the net traffic during the live migration of the Xen VM. My calls do not drop, but I will stress that I'm only running test cases at this time, I am not using real world traffic.Once I figure it all out, I'll report it here.-pete Original Message Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing From: Michael Collins m...@freeswitch.org Date: Fri, August 28, 2009 12:37 pm To: freeswitch-users@lists.freeswitch.org On Fri, Aug 28, 2009 at 12:11 PM, Giovanni Maruzzelli gmar...@celliax.org wrote: Usually you don't need to worry about stability issues with FS. For scalability, peoples tend to use openser or some other sip loadbalancer in fron of fs, but you probably would not need that. Live migration of calls is not yet possible, tough. Brian West has done some testing with live migrations but I don't know where he left off. Brian, were you using OpenVZ? I forget... In any case, FS allows you to try to do this with the hope that it will actually work in a production environment. As for the other things - yes, FS can work with the TDM card and the queues, etc. If you are in a position to install FS on a sandbox machine for testing then that would be your best bet. I recommend diving in, which is probably what you did when you first started learning Asterisk... Have fun!-MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
I guess I should also mention that Xen is a side-project. When considering this issue for an existing production systems, we chose to put as much HA into hardware as we can. We are not concerned with FS crashing, as so far we've never seen that happen (except when our module caused it :) So for each of our systems:- We have dual NIC cards (onboad NIC + PCI card) both bridged together in case one fails- We have redundant power supplies.- We use Mirrored Solid State Disks for local storage (far better MTBF than HDD, a lot faster too)- All but OS and speed-critical data is stored on a NAS device - We have redundant DBs with Memcache infront for speedAt the same time we chose to use COTS hardware (SuperMicro chassis/MoBo) rather than the big-boys like IBM or Dell. This kept the overall cost per machine low. Initially some were concerned that not having a name like IBM on our servers would be concerning to some potential clients. The solution was to pay a company to deisgn and build a custom face plate for the SuperMicro boxes. Which oddly looks more impressive to clients that a rack full of IBM faceplates. It was suprisingly low cost for the faceplates too.For scalability, OpenSIPS was our choice. There's a very nice tutorial on their website on how to configure Load Balancing.-pete Original Message Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing From: "Pete Mueller" p...@privateconnect.com Date: Sat, August 29, 2009 2:25 pm To: freeswitch-users@lists.freeswitch.org I am also in the process of playing around with FS running inside Xen Virtual Machines with "mirrored" VMs on a second failover system. So far, the initial tests are promising. I can see a 2-3 sec "hiccup" in the net traffic during the live migration of the Xen VM. My calls do not drop, but I will stress that I'm only running test cases at this time, I am not using real world traffic.Once I figure it all out, I'll report it here.-pete Original Message Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing From: Michael Collins m...@freeswitch.org Date: Fri, August 28, 2009 12:37 pm To: freeswitch-users@lists.freeswitch.org On Fri, Aug 28, 2009 at 12:11 PM, Giovanni Maruzzelli gmar...@celliax.org wrote: Usually you don't need to worry about stability issues with FS. For scalability, peoples tend to use openser or some other sip loadbalancer in fron of fs, but you probably would not need that. Live migration of calls is not yet possible, tough. Brian West has done some testing with live migrations but I don't know where he left off. Brian, were you using OpenVZ? I forget... In any case, FS allows you to try to do this with the hope that it will actually work in a production environment. As for the other things - yes, FS can work with the TDM card and the queues, etc. If you are in a position to install FS on a sandbox machine for testing then that would be your best bet. I recommend diving in, which is probably what you did when you first started learning Asterisk... Have fun!-MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Group Call
This seems to be a problem in the 1.0.4 tarball release. A friend with SVN 14305 the group_call from fs_cli works. Shawn On Sat, Aug 29, 2009 at 10:12 AM, Brian West br...@freeswitch.org wrote: Don't confuse this with the actual group/ endpoint. /b On Aug 25, 2009, at 11:41 AM, João Mesquita wrote: According to wiki: group,[insert|delete|call]:group name:url,group [insert|delete| call ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Shawn L. Djernes SD Consulting sh...@djernes.org | sdjer...@gmail.com MSN: wizard...@hotmail.com 402.345.7734 | 402.350.6973 Cell ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Group Call
Group call works for me on mine... I suspect your config is not correct... tar up your directory and mail it to me off list if you can please. freeswi...@default version FreeSWITCH Version 1.0.trunk (14665M) freeswi...@default group_call default [presence_id=1...@example.com]sofia/internal/sip:1...@192.168.1.31,[presence_id=1...@example.com ]sofia/internal/sip:gw...@192.168.1.112:6060;transport=udp,[presence_id=1...@example.com ]sofia/internal/sip:1...@192.168.1.22, [presence_id=1...@example.com]sofia/internal/sip:1...@192.168.1.101:5061;fs_nat=yes;fs_path=sip%3A1015%4080.93.247.26%3A1100 freeswi...@default group_call sales [presence_id=1...@example.com]sofia/internal/sip:1...@192.168.1.31,[presence_id=1...@example.com ]sofia/internal/sip:gw...@12.237.205.134:6060;transport=udp freeswi...@default group_call billing [presence_id=1...@example.com]sofia/internal/sip:gw...@192.168.1.112:6060;transport=udp ,[presence_id=1...@example.com]sofia/internal/sip:1...@192.168.1.22 freeswi...@default Thanks, /b On Aug 29, 2009, at 4:55 PM, Shawn L. Djernes wrote: This seems to be a problem in the 1.0.4 tarball release. A friend with SVN 14305 the group_call from fs_cli works. Shawn ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] newbie questions
hi and thx for all ur help! i basically re-svned installed it internally is fine now. playtime ;-) ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] sipdroid
hi, sipdroid was working with asterisk, anyone running it with FS? thx ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FS - external users
hi, as newbie here a simple one: based on the std-installation, what do i need to do to have a user connected from the outside. lets assume i want to leave 1000-1020 as they are, i make a copy of 1020, call it 1021 and then? should i leave it where it is? do i have to move the file? and what about he conetxt? thx ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] What is the difference between preAnswer and ring_ready?
Hi, Assuming an inbound call, I have trouble understanding what the supposed difference between the following two set of instructions is: session.execute(ring_ready); session.set(set, ringback=/home/ring.wav); and session.preAnswer(); session.streamFile(/home/ring.wav) In practice, however, the first scenario doesn't work for me although the example here suggests otherwise: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready Any idea? Thanks, Max. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] What is the difference between preAnswer and ring_ready?
Max Bridgewater max.bridgewa...@gmail.com wrote: Hi, Assuming an inbound call, I have trouble understanding what the supposed difference between the following two set of instructions is: session.execute(ring_ready); session.set(set, ringback=/home/ring.wav); Ring_ready sends a SIP ringing message and will use your device's default ringback sound - at least, that has been my experience. I should remember the message number in the SIP protocol, but I don't (the SIP experts here will undoubtedly have the details at the ready). and session.preAnswer(); session.streamFile(/home/ring.wav) I think that will play the file as early media. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] liverpie as gui-proxy?
hi, i was actually looking for something completely else, but reading wiki led me to the tought: if mod_event can basically do all kinds of FS-things, why not having a standalone ruby-on-rails as as gui which interacts with the evnet-socket via liverpie? im a total newbie to FS, just writing down my thoughts... thx for ur replies ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FS: possible to create conference + password on the fly?
at best, from an external application, language is not important. as far i have seen, could liverpie (aka the mod_event) do that? thx ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Group Call
The two systems in question are Debian with Freeswitch build using the dpkg-buildpackage. One of the systems has been customized but the other has a plain build and install on it, so it is not the customization. Is there a debian / ubuntu user on the list that can test and tell me if this works for you. The 14305 system is Fedora build with RPM so something may be broken in the build system. Thanks, Shawn On Sat, Aug 29, 2009 at 7:14 PM, Brian West br...@freeswitch.org wrote: Group call works for me on mine... I suspect your config is not correct... tar up your directory and mail it to me off list if you can please. freeswi...@default version FreeSWITCH Version 1.0.trunk (14665M) freeswi...@default group_call default [presence_id=1...@example.com]sofia/internal/sip:1...@192.168.1.31sip%3a1...@192.168.1.31 ,[presence_id=1...@example.com ]sofia/internal/sip:gw...@192.168.1.112:6060;transport=udp,[presence_id= 1...@example.com ]sofia/internal/sip:1...@192.168.1.22 sip%3a1...@192.168.1.22, [presence_id=1...@example.com]sofia/internal/sip:1...@192.168.1.101:5061 ;fs_nat=yes;fs_path=sip%3A1015%4080.93.247.26%3A1100 freeswi...@default group_call sales [presence_id=1...@example.com]sofia/internal/sip:1...@192.168.1.31sip%3a1...@192.168.1.31 ,[presence_id=1...@example.com ]sofia/internal/sip:gw...@12.237.205.134:6060;transport=udp freeswi...@default group_call billing [presence_id=1...@example.com]sofia/internal/sip:gw...@192.168.1.112:6060 ;transport=udp ,[presence_id=1...@example.com]sofia/internal/sip:1...@192.168.1.22sip%3a1...@192.168.1.22 freeswi...@default Thanks, /b On Aug 29, 2009, at 4:55 PM, Shawn L. Djernes wrote: This seems to be a problem in the 1.0.4 tarball release. A friend with SVN 14305 the group_call from fs_cli works. Shawn ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Shawn L. Djernes SD Consulting sh...@djernes.org | sdjer...@gmail.com MSN: wizard...@hotmail.com 402.345.7734 | 402.350.6973 Cell ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 HA + Loadbalancing
This sounds like so many redundancy projects that will probably offer nothing in the real world. On 08/30/2009 05:52 AM, Pete Mueller wrote: I guess I should also mention that Xen is a side-project. When considering this issue for an existing production systems, we chose to put as much HA into hardware as we can. We are not concerned with FS crashing, as so far we've never seen that happen (except when our module caused it :) So for each of our systems: - We have dual NIC cards (onboad NIC + PCI card) both bridged together in case one fails NICs hardly ever fail. Its the wiring which is the vulnerable area. How independent can you make the two wiring paths, when they come from the same box? - We have redundant power supplies. Even with a good UPS, power fails more often than a high quality power supply. Just how independent are the two power sources feeding your two power supplies? Do you have two completely independent UPS sets? Do you have spacially diverse wiring from them? - We use Mirrored Solid State Disks for local storage (far better MTBF than HDD, a lot faster too) My experience so far is that SSD reliability is very poor, with entire drives disappearing, rather than just getting the odd bad sector. I guess to balance this, hard disk drive reliability seems to have plummeted in the last year or so, after several good years. - All but OS and speed-critical data is stored on a NAS device NAS == more wiring. More wiring == more vulnerabilities. Are you sure your setup is a win? NAS tends to help keep the data secure, but it isn't good for reliable access to that data. - We have redundant DBs with Memcache infront for speed At the same time we chose to use COTS hardware (SuperMicro chassis/MoBo) rather than the big-boys like IBM or Dell. This kept the overall cost per machine low. Initially some were concerned that not having a name like IBM on our servers would be concerning to some potential clients. The solution was to pay a company to deisgn and build a custom face plate for the SuperMicro boxes. Which oddly looks more impressive to clients that a rack full of IBM faceplates. It was suprisingly low cost for the faceplates too. Some years ago we made an entire custom chassis for off the shelf boards. The quotes for fabricating that in small numbers were all over the place, but we ended with a good quality chassis at low cost. Most off the shelf rack mount enclosures are really pricy, so it isn't that hard to match their price with a custom build. We ended up with a better design (at least for our purposes) that cost us no more. It can really make your stuff stand out. A simple respray of the front panel can achieve a distinctive look at low cost too. :-) For scalability, OpenSIPS was our choice. There's a very nice tutorial on their website on how to configure Load Balancing. Regards, Steve ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] prompt playback volume using mod_shout
Hi, I'm finding that when I record a caller's voice using mod_shout in MP3 format, the recording is too soft when I play it back. I see there's a volume setting in shout.conf.xml, but it says, Dont change these unless you are insane. When I insanely try to increase this value, I don't here any difference in volume. Is this the right knob, or is there some other way to do this? Thanks, Marc ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] What is the difference between preAnswer and ring_ready?
Hi, In SIP, ring_ready will send a 180 ringing, pre answer will send a 183 with a session description, so that some audio can be exchanged while the call isnt answered yet (used a lot for inband error messages etc). Regards, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 29-Aug-09, at 9:50 PM, Jason White wrote: Max Bridgewater max.bridgewa...@gmail.com wrote: Hi, Assuming an inbound call, I have trouble understanding what the supposed difference between the following two set of instructions is: session.execute(ring_ready); session.set(set, ringback=/home/ring.wav); Ring_ready sends a SIP ringing message and will use your device's default ringback sound - at least, that has been my experience. I should remember the message number in the SIP protocol, but I don't (the SIP experts here will undoubtedly have the details at the ready). and session.preAnswer(); session.streamFile(/home/ring.wav) I think that will play the file as early media. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FreeSWITCH HA + Loadbalancing
On Sun, 2009-08-30 at 11:17 +0800, Steve Underwood wrote: This sounds like so many redundancy projects that will probably offer nothing in the real world. On 08/30/2009 05:52 AM, Pete Mueller wrote: I guess I should also mention that Xen is a side-project. When considering this issue for an existing production systems, we chose to put as much HA into hardware as we can. We are not concerned with FS crashing, as so far we've never seen that happen (except when our module caused it :) So for each of our systems: - We have dual NIC cards (onboad NIC + PCI card) both bridged together in case one fails NICs hardly ever fail. Its the wiring which is the vulnerable area. How independent can you make the two wiring paths, when they come from the same box? This is one area where you can do quite well. A simple setup: two machines (1, 2), two NICs (A, B) in each, two switches (S1, S2) - wire up 1A - S1 - 2A, 1B - S2 - 2B - run OSPF across the links allows you to unplug any cable or any switch without interrupting communications for more than a second or two if the OSPF timers are suitably set. This generalises nicely - we used to run two machines as web servers, each advertising the same IP address via OSPF to the routers via a setup like the one above. Unplug any one thing, and the whole still worked. Three complete power outages in the data center we were in in 18 months, one of which took out a number of power supplies, neatly illustrated Steve's point: our real-world reliability was determined elsewhere. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: d...@3c.co.uk W: http://www.3c.co.uk ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FS: possible to create conference + password on the fly?
I think you need to read documentation around a week, try to do something with FS. And then most of our questions go out. Yes, you can do that - dynamic conference with pass. 2009/8/30 tom tomabr...@gmail.com at best, from an external application, language is not important. as far i have seen, could liverpie (aka the mod_event) do that? thx ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] FreeSWITCH HA + Loadbalancing
There is a need for ensuring that calls do not drop, but we must balance that with the cost of making the system redundant. We took some small, inexpensive measures, to improve our odds, but we could spend a lot more, for basically nothing more than giving some client a warm fuzzy. To expand on what's mentioned below, The biggest cause for downtime that we've experienced is human accidents. We only have our solution in Tier 1 co-location facilities, so power/net dying isn't really an issue. (If the power does go out, 1000s of systems are down, and everyone notices). What we do end up with is IT admins tripping over power cords, pulling the wrong Ethernet cable, blowing a fuse on one side of the rack, etc. So we've doubled-up on all our cables. After that, the next biggest cause has been MoBo/CPU failure due to fan failure. This issue doesn't really have a good solution, and is why we began looking at Xen. This is where SUN systems look attractive, as systems like the E1 can shut down one CPU or board and keep the rest of the system running. But the cost for that solution is high, and I think that's SPARC-only. I'd love to head others take on a solution for this, as Xen is really a lot of overhead for a rare problem. Though it is at least technically interesting for me :) As for storage, this was completely personal experience. Our SSD have had no issues, while SATA drives seem to fail about 1 every month. The NAS storage is connected via dual NICs as well (again for the cabling) and is completely separated from the network, very close to DAS, just using GigE as the "cable". We are always looking for ways to improve, but the newest and greatest from EMC and others just doesn't seem to offer anything significant and cost a LOT more. I like the idea about the complete custom chassis. I hadn't considered that due to my thinking it would be expensive. Sounds like it's worth a look. As we consider creating an appliance offering, this may become more important.-pete Original Message Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing From: Steve Underwood ste...@coppice.org Date: Sat, August 29, 2009 8:17 pm To: freeswitch-users@lists.freeswitch.org This sounds like so many "redundancy" projects that will probably offer nothing in the real world. On 08/30/2009 05:52 AM, Pete Mueller wrote: I guess I should also mention that Xen is a side-project. When considering this issue for an existing production systems, we chose to put as much HA into hardware as we can. We are not concerned with FS crashing, as so far we've never seen that happen (except when our module caused it :) So for each of our systems: - We have dual NIC cards (onboad NIC + PCI card) both bridged together in case one fails NICs hardly ever fail. Its the wiring which is the vulnerable area. How independent can you make the two wiring paths, when they come from the same box? - We have redundant power supplies. Even with a good UPS, power fails more often than a high quality power supply. Just how independent are the two power sources feeding your two power supplies? Do you have two completely independent UPS sets? Do you have spacially diverse wiring from them? - We use Mirrored Solid State Disks for local storage (far better MTBF than HDD, a lot faster too) My experience so far is that SSD reliability is very poor, with entire drives disappearing, rather than just getting the odd bad sector. I guess to balance this, hard disk drive reliability seems to have plummeted in the last year or so, after several good years. - All but OS and speed-critical data is stored on a NAS device NAS == more wiring. More wiring == more vulnerabilities. Are you sure your setup is a win? NAS tends to help keep the data secure, but it isn't good for reliable access to that data. - We have redundant DBs with Memcache infront for speed At the same time we chose to use COTS hardware (SuperMicro chassis/MoBo) rather than the big-boys like IBM or Dell. This kept the overall cost per machine low. Initially some were concerned that not having a name like IBM on our servers would be concerning to some potential clients. The solution was to pay a company to deisgn and build a custom face plate for the SuperMicro boxes. Which oddly looks more impressive to clients that a rack full of IBM faceplates. It was suprisingly low cost for the faceplates too. Some years ago we made an entire custom chassis for off the shelf boards. The quotes for fabricating that in small numbers were all over the place, but we ended with a good quality chassis at low cost. Most off the shelf rack mount enclosures are really pricy, so it isn't that hard to match their price with a custom build. We ended up with a better design (at least for our purposes) that cost us no more. It can really make your stuff stand out. A simple respray of the front panel can achieve a distinctive look at low cost too. :-) For
Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Pete Mueller p...@privateconnect.com wrote: There is a need for ensuring that calls do not drop, but we must balance that with the cost of making the system redundant. We took some small, inexpensive measures, to improve our odds, but we could spend a lot more, for basically nothing more than giving some client a warm fuzzy. I think this is one area where, as indicated earlier in the thread, a lot of development effort would be needed to obtain that extra degree of reliability. From a broader perspective, the question is whether, over the next decade or two, VoIP can compete with the PSTN in reliability. My (limited) understanding is that PSTN equipment typically achieves 99.9% uptime, and if VoIP systems are going to play in that arena, it would be desirable for free/open-source software to do so. If FreeSWITCH itself is working correctly, all you need is a hardware failure or a kernel panic or a network outage to drop that up-time substantially, not to mention dropping the calls as well, which I've never experienced as a user of the PSTN due to equipment at the telephone exchange. I have, however, experienced some rather low-quality PSTN calls over international lines, which have the added disadvantage of being expensive to use. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
