Re: [Freeswitch-users] forcing ptime settings

They don't operate their own voip gateways, just run an SBC in front of a bunch of other providers. So someone they are reselling is using Sonus gear. I use them to originate to some destinations but in the US I avoid them due to the sonus stuff that pops up on certain routes. On Wed, Dec 23, 200

Re: [Freeswitch-users] RTP/RTCP media whilst recording

http://wiki.freeswitch.org/wiki/Variable_record_waste_resources On Wed, Dec 23, 2009 at 8:26 AM, TTNC - Technical wrote: > Hi There > > Our Freeswitch cluster receives inbound calls via a SIP trunk from our > supplier. I currently have an issue where when a call is sent to voicemail > using ses

Re: [Freeswitch-users] tigerjet 560C USB-to-rj11: incorperate usbhid/usbsnd device into freeswitch

Interesting. It would have to do more than just dialtone/dtmf though. Need call control, caller id, etc. What do they ship with it as far as drivers go? On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne wrote: > Hi all, > > > This weekend, I got the chance to buy a "profoon IP-150 RJ11-to-USB" >

Re: [Freeswitch-users] Codecs and things

On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji wrote: > Hello people, > > Can someone please clear the following ambiguities with codecs: > > Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy > Media mode, or does FS need to be running in bypass-media ? the Wiki is not > clear

Re: [Freeswitch-users] build errors :(

I think you are thinking of SWK. He has checked in mod_easyroute for easy routing of DIDs. [intra]lanman and I have mod_lcr for (umm) lcr routing. SWK still has a (probably) more capable lcr - though not open source. On Wed, Dec 16, 2009 at 1:30 PM, RR wrote: > BTW, at that time there was no

Re: [Freeswitch-users] Does FS support STUN by default?

STUN is not a way to open ports in a manner in which sheilds up would detect. http://en.wikipedia.org/wiki/STUN UPNP is what you want if you want to open ports. STUN is just a method for figuring out how to do nat traversal. STUN "method" is initiated by the p

Re: [Freeswitch-users] How to run a JS script periodically

If doing this, I'd suggest checking for a global var to see if the script should terminate itself. Otherwise, you'll have to bring down the whole freeswitch to stop this script. On Thu, Dec 3, 2009 at 7:28 AM, Seven Du wrote: > Not sure about js, but in lua, you can use luarun to run a > long-r

Re: [Freeswitch-users] B Leg Account Code on Fail Over dialing

You need to use brackets [] not braces {} for per-leg variables. On Thu, Nov 26, 2009 at 6:08 AM, Michael Toop < micha...@voxcore.voxtelecom.co.za> wrote: > Hi Everyone, > > How do I get the outbound sofia SIP route that the call took into a CDR > when using fail over dialing with the 'bridge' a

Re: [Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript

Stephen, I think you've jumped into the middle of a thread about sip 302, not about http. Anyway, you might want to look at using mod_curl instead of mod_spidermonkey_curl. mod_curl can give you a json response which you can then parse easily in javascript or any other language. The json response

Re: [Freeswitch-users] Recording with Native File PCMU

Yes On Wed, Nov 25, 2009 at 9:44 AM, Matthew Fong wrote: > so is using session_record with .wav my best option for recording bridged > calls? > > --matt > > > On Wed, Nov 25, 2009 at 7:18 AM, Brian West wrote: > >> These two options attach media bugs on to the session. Which doesn't >> work w

Re: [Freeswitch-users] Simplest of Conference Setup questions

The behavior of not being able to change the default caller controls are documented on the wiki: http://wiki.freeswitch.org/wiki/Mod_conference#.3Ccaller-controls.3E *Reserved Group Names* - none - Use this name to prevent installing caller-controls for callers of a conference. - defaul

Re: [Freeswitch-users] How to implement mod_lcr + mod_limit

Using lcr_auto_route + limit isn't really possible at this point. It is on the list of things to do but is more complex than it seems on it's surface. mod_lcr just constructs dial strings, it doesn't do any call control. It does provide enough information to do what you want via a scripting langu

Re: [Freeswitch-users] store registration info in memcache

I'd have to double check all the sql used for registration, but I doubt memcache is expressive enough to act as the registration store. For instance, you can't get a list of registrations from it (sofia status profile internal). memcache is a keystore only. That being said, one could use memcache

Re: [Freeswitch-users] Problem with Siemens A580 IP Phones

Just use the siemens to do the transfer. Hit the menu key, select internal (if you want to do an siemens-siemens transfer), select the extension, hit talk, then menu and choose conference or transfer. If youw ant another extension, then choose external, dial the extension, and the rest is the sam

Re: [Freeswitch-users] Large number of destinations

On Thu, Nov 12, 2009 at 4:32 PM, Robin Vleij wrote: > On 11/12/09 9:59 PM, Rupa Schomaker wrote: > If I read it right, this is suited for "complete" nrs. So would I have a > system connected with lots of DIDs, I would put them in easyroute. Then > for systems with lots of

Re: [Freeswitch-users] Large number of destinations

Take a look at mod_easyroute. On Thu, Nov 12, 2009 at 1:14 PM, Robin Vleij wrote: > Hi all, > > I'm currently building a proof-of-concept box using Freeswitch. Coming > from Asterisk/Kamalio/OpenSER it looks very cool so far, very complete. > > The plan is to make some sort of SIP router, some wo

Re: [Freeswitch-users] Cordless VOIP Phones

On Tue, Nov 10, 2009 at 6:56 AM, Rupa Schomaker wrote: > Oh, and it looks like a new firmware came out for the Siemens today. > Wonder what it fixes (and breaks).  Hmm.. wonder where I can find a > list of whats new. Well, it seems to totally break g722. I haven't had a chance to

Re: [Freeswitch-users] Cordless VOIP Phones

2009/11/9 João Mesquita : > Beat me with a dead cat all you want but I rather the snom m3 than the > Siemens A580IP Siemens has very low volume which makes its call quality > suck despite of being ergonomic and all... Did you flip hte option in the base station that tells it to make the audio

Re: [Freeswitch-users] Cordless VOIP Phones

I agree about the M3. I have the handset and it is not ergonomic at all. I also have a Siemens A580-IP. It does do G722 but has a few bugs related to G722 that I normally run it with G711 only. There is a quality difference between G722 and G711 when talking among the A580 handsets or the hands

Re: [Freeswitch-users] Another Concurrent Calls Monitor Question

Use mod limit to do this. You can choose to use it in count only mode if you want (no limit). On Mon, Nov 9, 2009 at 11:05 AM, DJB wrote: > I am also curious whether you can recommend how I can get the info if I want > to see concurrent calls by account code.  Let's say if I am running FS as SB

Re: [Freeswitch-users] Setting up Conference with Moderator

ublic.xml if you want to do it all in dialplan. > Sorry: I am still learning the Freeswitch world. > > > > -Original Message- > From: freeswitch-users-boun...@lists.freeswitch.org > [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa > Schomaker

Re: [Freeswitch-users] Fwd: Microsoft Exchange 2007 - Moved Temporarily

I don't believe this is a supported configuration. If you really must run both on the same server, put FS and Exchange on different IP addresses and see if that works. SIP client/server on the same IP has issues. On Sat, Nov 7, 2009 at 5:24 AM, Rudá Cunha wrote: > I'm having a problem when usin

Re: [Freeswitch-users] IP Phones with FreeSwitch

These phones work with FS, come by irc and you can talk to sekil about his use of them. In general, if you haven't invested in a bunch of phones, I'd recommend: Polycom 330,450,550 -- pick your price point snom - again, pick your price point These are generally well supported over the rest. On

Re: [Freeswitch-users] Error checking for PMP [general error]

If you don't have a router with NAT-PMP enabled then it is expected. Same if you don't have upnp. If you are behind a NAT, it is in your best interest to enable one or the other in your router. It will save you a bunch of headaches... On Tue, Nov 3, 2009 at 3:25 PM, Jerry Richards wrote: > > Whe

Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

Nothing jumps out at me on this one. mod_cdr_csv and/or mod_xml_cdr should be fine. On Sat, Oct 31, 2009 at 1:24 PM, Seven Du wrote: > I also got some zombie channels, if someone can help me take a look that's > really nice. > > http://pastebin.freeswitch.org/10912 > > I only loaded mod_cdr_csv

Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

fscore_pb has been updated. Next time put in the pastebin url. This one was 10911. Ok, a suspicious thread looks below. If I read it right, you are doing an originate from within the hangup handler. That doesn't look normal or sane to me. Thread 41 (process 30732): #0 0xb7a5e007 in select ()

Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

I believe this means you are hung up in CDR reporting. You can get a core of the running system and look at the stack traces to see where exactly it is hung up. Try using: support-d/fscore_pb gcore yourname This will put a copy of the stacktrace on the pastebin for poeple (or you) to look at.

Re: [Freeswitch-users] reload from managed

Is it necessary for you to call the reload function directly? Why not make an api call using switch_api_execute and the command "reloadxml". On Tue, Oct 27, 2009 at 4:32 AM, srinivasula reddy wrote: > Hi, > > how can i call reload from managed code? and i called > Csharp_switch_xml_open_root, t

Re: [Freeswitch-users] Address Rupa: Database for Audio Data

I use the dialplan app session_record: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session I call that in the appropriate parts before bridging to call. For incoming it is just before the bridge to user/username. For outgoing it is before I bridge to the outgoing provider. On T

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179

cond would be helpful here? I updated the wiki on this one just now with a bit more detail. It is a api call. so, you'd use it like: ${cond(eval ? trueval : falseval)} so to get a value of ERR if the var my myvar is > 15 you could: ${cond(${myvar} > 15 ? ERR : OK)} If both sides of the compar

Re: [Freeswitch-users] check sip client availability

This applies only to clients that are detected as nat, but maybe: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#nat-options-ping This is applied to the profile, not an app or a var setting. On Wed, Oct 21, 2009 at 8:26 AM, Woody Dickson wrote: > Hi, > > Is there any API to tell freeswitch to s

Re: [Freeswitch-users] NOT in dialplan expression

You can also look at using anti-action rather than action after the condition. condition == if action == then anti-action == else On Wed, Oct 21, 2009 at 6:34 AM, Mark Campbell-Smith wrote: > Hi! > > How do I do a NOT equal to in a dialplan expression > > Normaly in regex I would use the ! chara

Re: [Freeswitch-users] Status of ubuntu/debian packages.

I was under the impression that if you mark the files as conffiles you get the default behavior which prompts for diff/overwrite/edit options. If you want to get real fancy, look into using ucf to do three-way diffs between installed default, user modifications and new default. Finally, I have ch

Re: [Freeswitch-users] macros for performing repeated dialplan actions?

On Sat, Oct 17, 2009 at 1:44 AM, Simon J Mudd wrote: > I've other plans like: > > 1. choose the cheapest gateway based on some scripts which download > rates from the various gateways web pages and let me compare rates. > Hard partly as the "charges pages" aren't designed to allow me to > match a

Re: [Freeswitch-users] conference call

The wiki has a pretty good explanation of how eavesdrop works. Enabling a talk path to A or B or both A and B requires dtmf. So, if C hits the 1 button on the phone they can talk to the UUID you bound the eavesdrop to. 2 talks to the other party. 3 talks to both parties. http://wiki.freeswitch

Re: [Freeswitch-users] Some help with my post-paid billing project

I would still suggest using mod_lcr for this... If you have any real volume, use postgresql with the prefix module. It also supports IN lists, OR lists, optional quoting (since mysql is retarded), and custom sql so you can interface with whatever stored proc or deal with whatever database table y

Re: [Freeswitch-users] Some help with my post-paid billing project

Why not use mod_lcr to determine your rates? It already does all this work for you. I even added a feature recently that allows one to calculate/lookup end user rates (in addition to the carrier rate you are paying for a given route). On Tue, Oct 13, 2009 at 4:53 PM, Diego Viola wrote: > Hello,

Re: [Freeswitch-users] Database for Audio Data

What I do is record all calls and store the call with the UUID as the filename. Then when the call is hung up a CDR entry is sent to my web server. This CDR contains callerid and other info I might want to query by. The service on the web server inserts appropriate record(s) into the database.

Re: [Freeswitch-users] FreeSWITCH Update: Valet Parking

Oh, definitely. I think both use cases (existing, and park in next avail slot and read back slot #) are useful. On Fri, Oct 9, 2009 at 8:25 AM, Anthony Minessale wrote: > or twist that into a feature... =D > > alice is bob's secretary so she calls him on his cell when he has an appt > and parks

Re: [Freeswitch-users] FreeSWITCH Update: Valet Parking

Perhaps a "put in next empty slot" mode and then a say of the slot so Alice and make a note of the lot #? On Fri, Oct 9, 2009 at 6:43 AM, Seven Du wrote: > It's very cool. But think about the following scenario: > > Alice answers a call from Bob. > Alice transfers Bob to 6001 and hangs up. > Bob

Re: [Freeswitch-users] mod_opal - call charged before H.225 connect

That code is GPL, not license compatible. ;( 2009/10/8 Tihomir Culjaga : > Hi Yuriy, did you manage to do something with H323plus and FS ? > > btw: have you checked Objective OpenH323 > http://www.obj-sys.com/telephony-objective.shtml ? > This looks better to me as it is lighter and can be easily

Re: [Freeswitch-users] NAT problems - sorry

Double check your firewall and: 1) ensure you've actually enabled UPNP and 2) Ensure that any mention of a SIP ALG (application level gateway) is turned off. SIP ALGs tend to really screw things up. On Wed, Oct 7, 2009 at 12:48 PM, Andy wrote: > I've now set the external ips manually to be my e

Re: [Freeswitch-users] stun not working in fs 1.0.4?

Yes, the stun thing was fixed after 1.4 I believe. On Mon, Oct 5, 2009 at 6:43 AM, Vladimir Elizarov wrote: > > How do I fix this? Upgrade fs to trunk? -- -Rupa ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.fr

Re: [Freeswitch-users] FS Does Not Relay PresencePUBLISHToSubscribing Phones

You are missing the trailing ; On Fri, Oct 2, 2009 at 1:28 PM, Jerry Richards wrote: > I put the sqlite3 select query in the paste bin, and prior to that, I > entered the .dump command.  The select command came back with a "...>" > prompt which I don't understand.  I don't know enough about sqlit

Re: [Freeswitch-users] Need to know

and (if using custom_sql + the right query) you can now have the channelvar lcr_user_rate populated with your end-user-rate tables which can then be used by mod_nibblebill or other billing engine. Just committed and wiki updated. On Wed, Sep 30, 2009 at 11:30 AM, William Suffill wrote: > You cou

Re: [Freeswitch-users] Automagic Phone Provisioning

On Wed, Sep 23, 2009 at 10:19 AM, digilord wrote: > Hello all, >        I know this is done and I think I figured out how to do it but I don't > want to reinvent the wheel so here goes.  I am looking for a program > that will sit on the PBX.  This program will intercept DHCP reply > packets destin

Re: [Freeswitch-users] Multitenancy

Use something like mod_easyroute to consult a database of DIDs. If you host the DID it'll give you a route to dial out. If not, route out your gateway. Or load your lcr tables with your own DIDs. Consult mod_lcr and use it's dialstring. It'll prefer longer prefix matches, so you will always wi

Re: [Freeswitch-users] mod_lcr and indexes

On Fri, Sep 18, 2009 at 11:31 AM, Marcelo Sosa - LST wrote: > Hello, > >>> I know that mysql is dumb, but i´m postgresql-fobic :) (i was happy using >>> pgsql, until i found a bug in the restore of backups, that makes the >>> backup >>> unable to be restored, very bad day) :-) >> >> huh, really? >

Re: [Freeswitch-users] mod_lcr and indexes

rying both ways, what is the measured difference in performance? Does mysql have a way to analyze the table to ensure it's statistics are up to date? > > Regards, > Marcelo Sosa > > - Original Message - > From: "Rupa Schomaker" > To: > Sent: Friday

Re: [Freeswitch-users] mod_lcr and indexes

Hmm This is because mysql is "dumb" :( Anyway, if you wanted quoted digits, there is an option to enable that in the mod_lcr config file. http://wiki.freeswitch.org/wiki/Mod_lcr#Advanced_Usage Specifically, look at the parameter: quote_in_list The most efficient way (that I know of) to use

Re: [Freeswitch-users] Build problems with Shoutcast module under Debian

did you try to rerun configure? I think that'll fix it for you. On Thu, Sep 17, 2009 at 9:55 PM, Jason White wrote: > It turns out that Debian recently removed the libogg.la file, deliberately, > from the package. > > > ___ > FreeSWITCH-users mailing l

Re: [Freeswitch-users] memory leak - outbound socket

Either: 1) Provide a simple self-contained example that demonstrates the leak or 2) Run your application with FreeSWITCH under valgrind and provide the final output. To run freeswitch under valgrind: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_With_Valgrind_.28Linux.2

Re: [Freeswitch-users] Limit_Hash

On Wed, Sep 16, 2009 at 5:40 AM, Matt Riddell wrote: > My question then becomes, how do I set a hash for an originated call? > > It seems that limit_hash is an application rather than a channel > variable, and so far I've been doing most things without touching the > dialplan. Yes, it is a dialpl

Re: [Freeswitch-users] memory leak

On Wed, Sep 9, 2009 at 4:58 AM, Benedikt Fraunhofer wrote: > Hello *, > > the latest bugfixes for luarun (pool allocation) fixed it. > It's working now with luarun (friday-monday) and bgapi (monday-today). > The last test with the new "&" operator for sched_api is currently running. > > Thx! > >  B

Re: [Freeswitch-users] memory leak

There are other smaller leakers. xmlrpc is leaking, but the leaks are very small compared to the lua leak. Same with spidermonkey_curl - it is leaking but not too terribly much. I'll hop on #freeswitch in a bit and see if anyone has an idea. On Fri, Sep 4, 2009 at 9:35 AM, Benedikt Fraunhofer w

Re: [Freeswitch-users] memory leak

Doesn't that look like a pool that isn't being destroyed? On Fri, Sep 4, 2009 at 9:10 AM, Rupa Schomaker wrote: > Worst offenders (leakers over 100K).  The last one is the worst (672M) > -- looks like a lua script.  What are you doing in lua again? > ==28624== 672,268,288 byt

Re: [Freeswitch-users] memory leak

Worst offenders (leakers over 100K). The last one is the worst (672M) -- looks like a lua script. What are you doing in lua again? ==28624== 105,725 bytes in 1,804 blocks are still reachable in loss record 497 of 529 ==28624==at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624==by 0x4

Re: [Freeswitch-users] stability problems

25, Brian West wrote: >> Can you try NOT using a package?  I have a theory that the package has >> a few optimization flags in it that breaks things. >> >> /b >> >> On Sep 3, 2009, at 10:56 AM, Rupa Schomaker wrote: >> >>> Since y

Re: [Freeswitch-users] stability problems

On Thu, Sep 3, 2009 at 10:02 AM, Anthony Minessale wrote: > 1) Remove any binary files which may get mixed in from an older build >    rm /usr/local/freeswitch/bin/* >    rm /usr/local/freeswitch/lib/* >    rm /usr/local/freeswitch/mod Since you are using the debian package, the files will be in /

Re: [Freeswitch-users] memory leak

Isn't there a known issue with lua+sql leaking memory on some platforms? On Mon, Aug 31, 2009 at 8:32 AM, Brian West wrote: > Use valgrind. > > /b > > On Aug 31, 2009, at 7:22 AM, Benedikt Fraunhofer wrote: > >> >> The other two freeswitch-machines in this scenario play well, they're >> dumb load-

Re: [Freeswitch-users] mod_limit and memcache

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,

Re: [Freeswitch-users] mod_limit and memcache

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 eventual

Re: [Freeswitch-users] hanging problem with switch_scheduler_add_task

Use gcore to get a core dump. It will pause the process for the duration of the dump, but will not kill the process. On Fri, Aug 21, 2009 at 11:24 AM, mark morreny wrote: > Hi Mathieu, > > Thanks for your help. > > How can I intentionally crash a hanging freeswitch and obtain the core file > to r

Re: [Freeswitch-users] Loopback and bypass_media

On Thu, Aug 13, 2009 at 6:54 PM, Mathieu Rene wrote: > Hi All, > > The reason it works when you wait 3 seconds is that mod_loopback bails [snip] Thanks for that explanation. It umm.. explains a lot. :) > On another note, mod_sofia will behave differently when it detects its > being bridge with

Re: [Freeswitch-users] random route selection

mod_lcr will do random route selection if the rates are the same. But that gives an equal distribution. There is no weighting/percentage supported. On Wed, Aug 12, 2009 at 12:21 PM, Juan Backson wrote: > Hi, > > I would like to implement a random route selection based on some arbitrary > percent

Re: [Freeswitch-users] Loopback and bypass_media

On Wed, Aug 12, 2009 at 10:22 AM, Phillip Jones wrote: > Hi there, > >>> application="originate" >>> data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)" > > I agree. However, perhaps the ideal is not to specify the carriers at > this level, as carriers are added and removed fairly o

Re: [Freeswitch-users] Loopback and bypass_media

perhaps we need to add some syntax + logic to originate: application="originate" data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)" This would acomplish the equiv of loopback/bar,loopback/yum where bar and yum are then further expanded in the dialplan as sofia/foo/bar|sofia/baz

Re: [Freeswitch-users] Calling Multiple Destinations with Failover

On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker wrote: > > That of course, should be: -- -Rupa ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIB

Re: [Freeswitch-users] Calling Multiple Destinations with Failover

While any use of loopback can be considered abuse, that is how I solve this issue. I use loopback for each dialed # and then in the dialplan use mod_lcr to build a dialstring that does failover in order of cost. So: becomes In the dialplan you then match on those (just regular 10digit dial

Re: [Freeswitch-users] core dump

That backtrace is useless since you are not giving the right path to freeswitch. Look at line 9 of the pastebin. Until you fix that we can't look at the backtrace. On Mon, Jul 27, 2009 at 8:00 AM, Baskar wrote: > *Hi Rupa, > > I get core dump segmentation fault in freeswitch machine frequentl

Re: [Freeswitch-users] core dump

That backtrace is not useful because gdb was unable to locate the freeswitch binary. Since you are running it from /opt/freeswitch/bin directory, don't use 'bin/freeswitch' to point gdb to the binary: gdb freeswitch corefile On Mon, Jul 27, 2009 at 1:40 AM, Baskar wrote: > Hi, > > I get core d

Re: [Freeswitch-users] Confusing handling of incoming calls

-pete > > Original Message > Subject: Re: [Freeswitch-users] Confusing handling of incoming calls > From: Rupa Schomaker > Date: Wed, July 22, 2009 2:12 am > To: freeswitch-users@lists.freeswitch.org > > > > On Tue, Jul 21, 2009 at 11:35 PM, Pet

Re: [Freeswitch-users] Confusing handling of incoming calls

On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller wrote: > My goal is: > 0) figure out why the bandwidth gateway is being processed as "internal" > (this is more of a security thing) > they are probably terminating traffic on port 5060 rather than 5080. 5060 is internal, 5080 is external. > > 1)

Re: [Freeswitch-users] Call confirm ivr

Maybe look at the group_confirm_* stuff. http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation There is a way to get it to execute a script as well which is probably waht you want. This would be simpler than doing the work you are saying below. On Tue, Jul 21, 2009 at 2:0

Re: [Freeswitch-users] Best way to bridge 2 calls with LCR?

Just a note that the "as xml" syntax has been added to current trunk. On Tue, Jul 21, 2009 at 1:21 PM, Rupa Schomaker wrote: > Well, the "as xml" is something I've been meaning to do, so I'm gonna get > that checked in today sometime anyway. If you want to

Re: [Freeswitch-users] Best way to bridge 2 calls with LCR?

for first number and bridges to an > extension > > 3) the extension calls lcr fir the second number and originates the second > call > > On steps 2 and 3 I could just use set data to set the additional variables > I need. The first step is what troubles me. > > > Thank

Re: [Freeswitch-users] Best way to bridge 2 calls with LCR?

e of ${lcr_auto_route}? I would need to add > some other variables, like ignore_early_media=true and a uuid that 'links' > the two calls so I can track it listening for events. > > Thanks! > > Nicolas > > > On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker wrote: &g

Re: [Freeswitch-users] Best way to bridge 2 calls with LCR?

lcr api command doesn't really return a usable dialstring (it was originally done for debug purposes). I could add an "as xml" option if needed... Anyway, to do this from the dialplan: remember that originate's usage is: -USAGE |&() [] [] [] [] [] so, the first argument is the call url and th

Re: [Freeswitch-users] Getting error without continuation

Didn't you just set max_calls=1? Maybe takes that out so you can handle more than 1 call On Tue, Jul 21, 2009 at 3:12 AM, Eli Hayun wrote: > Hi > I set continue_on_fail=true but I keep getting error : > 2009-07-21 11:00:53.284148 [INFO] mod_dialplan_xml.c:252 Processing > phone-1->limit_exc

Re: [Freeswitch-users] Freeswitch Windows Issues

mod_spidermonkey_odbc is a mod for spidermonkey, not freeswitch. So, you need to enable this mod in the spidermonkey config file, not freeswitch's. This is in conf/spidermonkey.conf.xml. The default has mod_spidermonkey_odbc commented out. On Mon, Jul 20, 2009 at 11:25 PM, Baskar wrote: > *Hi

Re: [Freeswitch-users] SIP TLS (and SRTP)

hmm... When I put that section in, I put the wrong filename. It should be directory/default.xml. I'll update the wiki. Also, it is a valid configuration to support tls but not srtp. I'll put a bit of a discussion in there talking about that. Setting sip_secure_media to true requires the endpo

Re: [Freeswitch-users] Documentation error?

Thanks. At least a few of us monitor every change to the wiki. I especially monitor stuff related to my modules. So... feel free to make changes, we can always roll back to an earlier version of the page or clarify further. The wiki is a community resource, everyone should feel like they can m

Re: [Freeswitch-users] Error in my dialplan

moved that ability to login without password from the > > default configs. > > > > /b > > > > On Jul 8, 2009, at 4:48 PM, Rupa Schomaker wrote: > > > > > the problem is the + is coming from the network... > > > > > > On Wed, Jul 8, 20

Re: [Freeswitch-users] Error in my dialplan

the problem is the + is coming from the network... On Wed, Jul 8, 2009 at 4:46 PM, Brian West wrote: > You have to escape the + with \+ > /b > > On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote: > > 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: 1 > [nothing to repeat][^+1310555121

Re: [Freeswitch-users] How to get the hook state?

I think you can (mostly) solved this using mod_limit and some dialplan work. Something like: if you want the second call to ring, limit should be set to 2 or higher. for either VM or busy, limit should be set to 1 if limit was reached and the user wants busy, send back busy if limit was reached

Re: [Freeswitch-users] Is there any license G729?

And unless you are directly connecting to the cell phone provider you are going to be converted to ULAW/ALAW to traverse the PSTN. So there is no advantage going to a native GSM codec only to have it expanded out to G711 and then back to GSM codec. On Tue, Jun 30, 2009 at 10:05 PM, Steve Underwoo

Re: [Freeswitch-users] Polycom configuration problems?

hone rather than FS. I > would never have figured this out. > > > > Are Polycoms the only SIP phones which have this feature? > > > > Lars > > > > *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: > freeswitch-users-boun...@lists.freeswitch.org] *O

Re: [Freeswitch-users] Polycom configuration problems?

is in the polycom manuals which you can donwload from polycom. On Tue, Jun 23, 2009 at 12:25 PM, Lars Zeb wrote: > Via a web browser. > > > > *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: > freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Rupa &

Re: [Freeswitch-users] Polycom configuration problems?

How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb wrote: > I’m sorry Chris, but I don’t know where the look for the “global sip.cfg > and mac/phone specific cfg” settings. I also looked for digitmap but could > find nothing. > > > > Can you be more specific? > > >

Re: [Freeswitch-users] Controlling Conference Controls

I added the ability to call into your dialplan from the caller controls in conferences a while back. Depending on your goal, that might be an easy way to get your problem resolved. You can keep state using the hash api and hash on the conference name or some other useful thingie. On Wed, Jun 17,

Re: [Freeswitch-users] Can't hear outbound calls

see any 0.0.0.0 if it were > present. > > > > The [ERR] was due to me removing example.xml from sip_profiles/internal. I > put it back after this. I don’t understand the following command in > conf/sofia.conf.xml. > > > > > > > > Lars > > > >

Re: [Freeswitch-users] Can't hear outbound calls

gt; What options do I have for setting up logging? I’m sorry, but I don’t know > anything about this. > > > > Thanks, Lars > > > > *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: > freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Rupa > Schoma

Re: [Freeswitch-users] mod_lcr for a-z rates

Diego, Here is how I'd go about doing what I think you want. As payment, add a section to the wiki when you have it working. Create two profiles in lcr.conf. the first profile is your callingcard rate deck. Give it a profile id of 1. Then load your data into the lcr tables. carriers = define

Re: [Freeswitch-users] Can't hear outbound calls

On Tue, Jun 9, 2009 at 6:43 PM, Lars Zeb wrote: > > It looks like the error message only appears on the console when started > without the nc option; and it does not appear in log/freeswitch.log in any > case. > You might want to review how you have your logging setup then. The example I gave yo

Re: [Freeswitch-users] Can't hear outbound calls

If we get 0.0.0.0 as a public address, upnp support is disabled since we're getting the info from a gateway that can't route us to the internet. It is broken, so we don't trust it. The message you should see in the logs is something like: 2009-06-08 13:51:44.587812 [ERR] switch_nat.c:126 uPNP De

Re: [Freeswitch-users] Few questions

2009/6/9 freeswitch list > 1) How do you setup a gateway where the provider doesn't require a user > name and password? For freeswitch gateways username and password are > mandatory. > Use the param: > > 2) Is there anyway to have xml_curl send the password for directory entry > requests. > d

Re: [Freeswitch-users] Freeswitch creating more then two sessions for one call

Some thoughts: 1) look at mod_lcr 2) comments on script below... On Sat, Jun 6, 2009 at 2:53 AM, Shoaib Khanzada wrote: > Following is the js i am using to select route and bridge the call > > > use("ODBC"); > > var DSN = "myodbc"; > var DB_USER = "neo"; > var DB_PASS = "."; > >

Re: [Freeswitch-users] mod_lcr caller id

Update to at least rev 13611 and mod_lcr will check "effective_caller_id_number" prior to the real callerid on the channel. This works in the same way that bridge does so no surprises. On Thu, Jun 4, 2009 at 8:15 AM, Yuriy Ivzhenko wrote: > On Thursday 04 June 2009 15:54:24 R

Re: [Freeswitch-users] mod_lcr caller id

The API mthod of setting the callerid is really for testing purposes. Since you have no callerid from the FS command line interface I added a parameter you can pass to specify the CID. When in the dialplan you actually have the CID, so none is used. mod_lcr is just grabbing the callerid on the c

Re: [Freeswitch-users] effective_caller_id_number on bridge dialstring

I've fixed mod_lcr now. It should have been setting origination_caller_id_number not effective_caller_id_number. On Tue, Jun 2, 2009 at 6:04 AM, Yuriy Ivzhenko wrote: > Hello all. > > I have test the lcr overriding the Caller ID functionality. > > It return dialstring, that contains 'effective

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