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

this will be perfect ... but it is up to Yuriy if he is willing to donate his work... T. On Tue, Oct 13, 2009 at 8:08 AM, Brian West wrote: > Does anyone see a problem with hosting mod_h323 in our SVN? I would > like to centralize everything we can to reuse our issue tracking > resources and

Re: [Freeswitch-users] Groups information in sqllite

Its based on the directory and who is in the group... check out the defaults it does exactly this on the 2000 range if I recall correctly. /b On Oct 12, 2009, at 11:32 PM, srinivasula reddy wrote: > OK, then when i call to group number(911) how it will call to all > the registered members in

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

i come across the valet_park application when i just finished an improved-version of the call parking (using mod_fifo) such that it is parked to different ext'n numbers when the caller is att_xfer'd to ext 777. i used strftime(%s) to generate 700~759 parking numbers. i also added feature that if

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

that will make life easier. 2009/10/13 Brian West > Does anyone see a problem with hosting mod_h323 in our SVN? I would > like to centralize everything we can to reuse our issue tracking > resources and not fragment the community if possible. > > /b > > On Oct 12, 2009, at 2:43 PM, Tihomir Culj

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

On Tue, Oct 13, 2009 at 8:31 AM, Brian West wrote: > I wouldn't call it donating per se... Its just giving it a place to > live with easy access for end users without having to do anything > extra go get it! ;) > > /b > > I agree with you Brian. > > > __

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

hello, yes, host please to let users test it and report bug YATE/Asterisk fully support but freeswitch no fully support it Brian West a écrit : Does anyone see a problem with hosting mod_h323 in our SVN? I would like to centralize everything we can to reuse our issue tracking resources and no

Re: [Freeswitch-users] Groups information in sqllite

hi brain, thank u very much for your valuable time, can u please tell me where the groups data will maintain thought the session, Thanks Srinivas On Tue, Oct 13, 2009 at 12:18 PM, Brian West wrote: > Its based on the directory and who is in the group... check out the > defaults it does exactly

[Freeswitch-users] 606 error

Hi, two users are registered in freeswitch, when i making call to another user i am getting 606 error, any help -- Srinivasula Reddy K ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/

Re: [Freeswitch-users] 606 error

Hi, Console user1181 attempted to call console user1171 resulted in failure. Sip server returned "Temporarily unavailable" with reason header cause=606; text="user-not-registered". This also happened with other consoles. Thanks SRINIVAS On Tue, Oct 13, 2009 at 2:35 PM, Tihomir Culjaga wrot

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

Meftah Tayeb wrote: > hello, > yes, host please to let users test it and report bug > YATE/Asterisk fully support but freeswitch no fully support it I would disagree about YATE & Asterisk fully supporting H323. :) They both have some support for H323 for years now, but only for voice calls. Non

[Freeswitch-users] dingaling: Destination out of order

Hi! I am trying to call from FS to gtalk. This used to work, so not sure if there is a problem with my build (FreeSWITCH Version 1.0.trunk (15126)) freeswi...@internal> dingaling status --DingaLing status-- login | connected mygmai...@gmail.com/gtalk| AUTHORIZED It looks okay

[Freeswitch-users] sofia gateways and linux multipath routing

Hello all, I'm interested in using mod_sofia with multiple Internet connections (configured as a unique load-balancing route using multipath). One solution would be to define a different profile for each connection, but it would be more practical having a unique external profile that would automa

Re: [Freeswitch-users] dingaling: Destination out of order

Mark Campbell-Smith wrote: > When I dial (which is to call my gtalk user), I get the following > in the console: [snip] Could you turn on debug logging in the console and post the output? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.

[Freeswitch-users] Registering a large number of SIP users

Hi, I am creating a load test setup for FreeSWITCH using Sofia SIP. I have two machines both configured with mod_xml_curl, one machine (lets call it SIP Server) has 100 SIP accounts. Now i want to register second machine (lets call it SIP Client) to all these 100 SIP accounts on first machine. How

Re: [Freeswitch-users] 606 error

what about some console logs & sip traces ? T. On Tue, Oct 13, 2009 at 10:56 AM, srinivasula reddy < srinivas.ksvre...@gmail.com> wrote: > Hi, > > two users are registered in freeswitch, when i making call to another user > i am getting 606 error, > any help > > -- > Srinivasula Reddy K > >

Re: [Freeswitch-users] Mod_fifo posision in queue

Has anybody managed to get this to work already? How do you play the announcements dependent on the variable in the dialplan? Best regards Peter Michael Collins schrieb: > > > On Thu, Sep 10, 2009 at 12:32 PM, Diego Viola > wrote: > > Lets make sure we add it o

[Freeswitch-users] sched_api doesn't get launched

Hi: I am using mod_java. And in my script I was able to achieve using: execute("sched_hangup", "+300 alloted_timeout"); However, when I try to run sched_api in the same way, system log returns that it's an invalid application. I have also tried to trigger it with many conditional channel variable

Re: [Freeswitch-users] 606 error

and you are sure both users are registered to the same context and your dialplan is correct ? T. On Tue, Oct 13, 2009 at 11:13 AM, srinivasula reddy < srinivas.ksvre...@gmail.com> wrote: > > Hi, > > > > Console user1181 attempted to call console user1171 resulted in failure. > Sip server returne

Re: [Freeswitch-users] dingaling: Destination out of order

This is all I see: console loglevel 9 +OK console log level set to DEBUG freeswi...@internal> 2009-10-13 21:33:05.578863 [NOTICE] switch_channel.c:613 New Channel sofia/internal_nat/1...@192.168.1.120 [cad049fe-b7e3-11de-94a7-1dd4d003eac8] 2009-10-13 21:33:05.634924 [INFO] mod_dialplan_xml.c:391

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

On 2009-10-13 15:05 +0800, Seven Du wrote freeswitch-users@lists.freeswitch.org: hm, host it if you wont, i has nothing against it. SD>that will make life easier. SD> SD>2009/10/13 Brian West SD> SD>> Does anyone see a problem with hosting mod_h323 in our SVN? I would SD>> like to centralize ev

Re: [Freeswitch-users] dingaling: Destination out of order

I've fixed the problem. My dialplan for outbound calling had a typo: The gtalk was gtallk somehow . On Tue, Oct 13, 2009 at 8:56 PM, Mark Campbell-Smith wrote: > Hi! > > I am trying to call from FS to gtalk.  This used to work, so not sure > if there is a problem with my build (FreeSW

Re: [Freeswitch-users] 606 error

Hi, thank u very much for your valuable time, s am sure they are both in same it is not occur continuously, i dont know the reason, i am having the wireshark file, any help? thanks srinivas On Tue, Oct 13, 2009 at 4:02 PM, Tihomir Culjaga wrote: > and you are sure both users are registered to

Re: [Freeswitch-users] 606 error

wireshark image thanks srinivas On Tue, Oct 13, 2009 at 4:33 PM, srinivasula reddy < srinivas.ksvre...@gmail.com> wrote: > Hi, > > thank u very much for your valuable time, > s am sure they are both in same it is not occur continuously, i dont know > the reason, > i am having the wireshark file,

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

static const char* h323_formats[] = { "G.711-*A*Law-64k", "PCM*U*", "G.711-*u*Law-64k", "PCM*A*", "GSM-06.10", "gsm", "MS-GSM", "msgsm", I've changed this to meed desired caps ... need more tests ... 2009/10/13 Georgiewskiy Yuriy > On 2009-10-13 15:05 +0800, Seven Du wrote >

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

On 2009-10-13 13:35 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: this morning me bring in hospital, and now i cannot make much work, i think return to the ranks in 1-2 week. TC>static const char* h323_formats[] = { TC>"G.711-*A*Law-64k", "PCM*U*", TC>"G.711-*u*Law-64k", "P

Re: [Freeswitch-users] 606 error

of course, if you can send it thi will be great... T. On Tue, Oct 13, 2009 at 1:03 PM, srinivasula reddy < srinivas.ksvre...@gmail.com> wrote: > Hi, > > thank u very much for your valuable time, > s am sure they are both in same it is not occur continuously, i dont know > the reason, > i am havi

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

2009/10/13 Georgiewskiy Yuriy > On 2009-10-13 13:35 +0200, Tihomir Culjaga wrote > freeswitch-us...@lists.fre...: > > this morning me bring in hospital, and now i cannot make much work, > i think return to the ranks in 1-2 week. > > damn, hope you will recover soon... take it easy. T. __

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

Georgiewskiy Yuriy wrote: > On 2009-10-13 13:35 +0200, Tihomir Culjaga wrote > freeswitch-us...@lists.fre...: > > this morning me bring in hospital, and now i cannot make much work, > i think return to the ranks in 1-2 week. I wish you a speedy recovery, Yuriy. Regards, Vlasis. _

Re: [Freeswitch-users] Registering a large number of SIP users

You may try this tool, "sipp", to execute a load test. http://sipp.sourceforge.net/ 2009/10/13 Muhammad Shahzad > Hi, > > I am creating a load test setup for FreeSWITCH using Sofia SIP. I have two > machines both configured with mod_xml_curl, one machine (lets call it SIP > Server) has 100 SIP a

Re: [Freeswitch-users] Registering a large number of SIP users

Yes i have used this tool before, but sip registration is just first part of my load test, i will be make SIP call load testing too. Also, i want to use FreeSWITCH against FreeSWITCH to test its capability both as SIP Server and SIP Client. Thank you. On Tue, Oct 13, 2009 at 6:21 PM, Ryanny Lin

Re: [Freeswitch-users] openzap Failure opening channel error

We are using Reliance as the Carrier. I think, with this same Reliance carrier, in my office, they are able to make outgoing calls through asterisk+libpri. On Tue, Oct 13, 2009 at 12:19 AM, Michael Collins wrote: > Lak, > > Okay I will need a little bit of time to dig into the IE's and what they

[Freeswitch-users] SIP Overlap support?

hi there, i would like to ask, if fs has support for something like "SIP Overlap"? instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digi

[Freeswitch-users] FS Extension Groups Documentation

Is there a link to the documentation of FS groups and what can be done with a group (i.e. capabilities)? Thanks And Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinf

Re: [Freeswitch-users] SIP Overlap support?

have you tried it? I *think* either we did support it or we would with a small patch to sofia lib that I cannot recall if we ever got committed. On Tue, Oct 13, 2009 at 8:51 AM, Dennis wrote: > hi there, > > i would like to ask, if fs has support for something like "SIP Overlap"? > > instead of

Re: [Freeswitch-users] SIP Overlap support?

how could we try? we played arround with a snom phone (snom seems to support something in this direction, but are not shure, how we can test it and how we can see if it is supported or not. any hint? 2009/10/13 Anthony Minessale : > have you tried it? > I *think* either we did support it or we w

Re: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT event

Sorry for my foolishness but I stil can't grasp it. I'm developing app based on inbound mod_event_socket and I don't know how to run "enable_heartbeat" or " sched_heartbeat" without specifying any alive session uuid. I tried to use "create_uuid" and send mentioned commands to the created uuid but

Re: [Freeswitch-users] Fwd: Groups information in sqllite

Group information is not stored in sqlite, it is pulled from the xml registry (switch_xml_locate_group function can find them) . Also, please do not cross post between lists. http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups http://wiki.freeswitch.org/wiki/Mod_commands#in_group

Re: [Freeswitch-users] openzap Failure opening channel error

On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy wrote: We are using Reliance as the Carrier. I think, with this same Reliance carrier, in my office, they are able to make outgoing calls through asterisk+libpri. If that's the case I would be very interested in seeing a pri debug from

[Freeswitch-users] Database for Audio Data

Hi, Does anyone know whether FreeSWITCH has a function to automatically record every call as an audio file in a server or forward them to be stored in a database with additional parameters such as caller ID, date, starting time and ending time? So that these recorded audio data can be q

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] Some documentation thoughts

I will add this thought to the weekly discussion. Perhaps we can crowdsource this one. -MC On Mon, Oct 12, 2009 at 10:52 PM, Mark Crane wrote: > "In many cases the issue with the docs isn't that they aren't complete but > rather that they are hard to find." > > Agreed! The biggest problem with t

Re: [Freeswitch-users] Some documentation thoughts

Under special pages there is ways to get a list of all the wiki pages. ( http://wiki.freeswitch.org/wiki/Special:SpecialPages) http://wiki.freeswitch.org/wiki/Special:AllPages Due to the number of pages it's broken into sub pages based in alphabetical order of the page names. -- W __

Re: [Freeswitch-users] SIP Overlap support?

you need a softswitch i'm afraid a SIP phone is not designed for overlap... T. On Tue, Oct 13, 2009 at 5:26 PM, Dennis wrote: > how could we try? we played arround with a snom phone (snom seems to > support something in this direction, but are not shure, how we can > test it and how we can

Re: [Freeswitch-users] SIP Overlap support?

i do think some softphone can do it but i forgot which one it was either snom or grandstream On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga wrote: > you need a softswitch i'm afraid a SIP phone is not designed for > overlap... > > T. > > > On Tue, Oct 13, 2009 at 5:26 PM, Dennis wrote: >

Re: [Freeswitch-users] sched_api doesn't get launched

You need to pass the UUID to sched_hangup. Usage: sched_hangup [+] [] http://wiki.freeswitch.org/wiki/Mod_commands#sched_hangup On Tue, Oct 13, 2009 at 10:14 AM, Henry Huang wrote: > Hi: > I am using mod_java. And in my script I was able to achieve using: > > execute("sched_hangup", "+300 all

Re: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT event

On Tue, Oct 13, 2009 at 8:32 AM, Artem Shiyanov wrote: > Sorry for my foolishness but I stil can't grasp it. > I'm developing app based on inbound mod_event_socket and I don't know how > to run "enable_heartbeat" or " sched_heartbeat" without specifying any alive > session uuid. I tried to use "c

Re: [Freeswitch-users] SIP Overlap support?

Both support it. In the Grandstream, I believe it is called Early Dial (vs. SNOM's Overlap Dialing). It can be problematic if you have a device somewhere in the middle that doesn't support 484s. -metik - Original Message - From: Anthony Minessale To: freeswitch-users@lists.free

Re: [Freeswitch-users] SIP Overlap support?

i never found it working properly... i always had some interoperability issues and i finished having a "dialplan" on my phones being delivered through a config file via tftp or http .. depending of the phone capability. BTW: using overlap can lead to a greater system load... be careful when settin

[Freeswitch-users] lua script causing FreeSwitch to crash?

http://pastebin.freeswitch.org/10686 I am trying to use a lua script for inbound calls. The caller hears a busy signal. After this call fails to go through, FreeSwitch is no longer in memory - it does not appear in 'ps -ef' output. I don't have a clue what I might be doing incorrectly in t

Re: [Freeswitch-users] lua script causing FreeSwitch to crash?

On Tue, Oct 13, 2009 at 12:38 PM, Lars Zeb wrote: > http://pastebin.freeswitch.org/10686 > > > > I am trying to use a lua script for inbound calls. The caller hears a busy > signal. After this call fails to go through, FreeSwitch is no longer in > memory – it does not appear in ‘ps –ef’ output.

Re: [Freeswitch-users] SIP Overlap support?

As evidenced by various DTMF interop issues (with RFC2833, inband, etc) over the years, I would avoid it if at all possible. What does it particularly do that can not accomplished by using RFC 2833 or (less ideal) inband DTMF? Or are you attempting to use it as a band-aid to address some sor

Re: [Freeswitch-users] FS Extension Groups Documentation

On Tue, Oct 13, 2009 at 7:54 AM, Jerry Richards wrote: > > Is there a link to the documentation of FS groups and what can be done with > a group (i.e. capabilities)? > > Thanks And Best Regards, > Jerry > http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups http://wiki.freeswitch.org/w

[Freeswitch-users] FreeSWITCH Conf Call Agenda For October 16th

FYI, The agenda for this week's call is here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_16 Please feel free to add agenda items and updates as you see fit. -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http:/

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

Hello, I'm trying to write a post-paid billing script, I have the CDR on my database and also a "rates" table, the CDR contains fields like caller_destination_number, variable_duration, etc. and the rates table contains: destination, prefix, rate (cost). The problem is that I can't just strip the

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

Should I try to pass the whole DNIS and compare with teh rates list/table and when a prefix from the DNIS matches with the rates list I get hte rate that way? On Tue, Oct 13, 2009 at 9:53 PM, Diego Viola wrote: > Hello, > > I'm trying to write a post-paid billing script, I have the CDR on my >

[Freeswitch-users] Dingaling: using a hostname instead of stun for rtp

Hi! I have a hostname set in vars.conf.xml for the parameters external_rtp_ip and the external_sip_ip instead of the usual stun. I found that stun was timing out and was causing some problems. And as I have a hostname, it makes sense to use that instead of relying on stun. However, when I use D

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

What database are you using? You could do this with regular SQL, but it would by a costly operation, for PostgreSQL we're using the prefix module: http://pgfoundry.org/projects/prefix/ You can then match the closest prefix by using something like "WHERE myprefix_col @> caller_destination_number O

Re: [Freeswitch-users] FS Slide deck?

If that's the case, does this also apply to VoIP gateway services? If so, I'll consider phrasing this differently: http://wiki.freeswitch.org/wiki/Voicepulse.xml --begin-- Voicepulse Warning WARNING!!! Recent changes (Sep 2008) in VoicePulse have limited connect03 to only IAX termination. Pl

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,

[Freeswitch-users] Preannouncing a message before making a call

I'm looking to migrate to Freeswitch from Asterisk (SOHO usage). One thing I have configured at the moment which I'm unsure how to do in FreeSwitch is the following. When the call's dial pattern is recognised I've been getting Asterisk to say: "Dialing via " where might be a country, or area and

[Freeswitch-users] Building Freeswitch with VPATH (src and obj directories are different)?

I was hoping to be able to build Freeswitch using VPATH, that is have the src and object directories in different locations. This ensures that the src tree is unspoilt and I don't have to reclean it if I do updates. However this does not seem to work: [sjm...@mad07 src]$ pwd /Users/sjmudd/src [

Re: [Freeswitch-users] Building Freeswitch with VPATH (src and obj directories are different)?

On Tue, Oct 13, Simon J Mudd wrote: > I was hoping to be able to build Freeswitch using VPATH, that is have the src > and object directories in different locations. > This ensures that the src tree is unspoilt and I don't have to reclean it if > I do updates. You can use svn export. Not sure if

Re: [Freeswitch-users] FS Slide deck?

Karl, Yeah, this post is wrong. You've got "acronym" misspelled. Once you fix that then I think it's all good! ;) Seriously, it's okay to put critical information into the wiki, but perhaps you could remove the tongue-in-cheek comments. -MC On Tue, Oct 13, 2009 at 3:59 PM, Karl Vesterling wrot

Re: [Freeswitch-users] Dingaling: using a hostname instead of stun for rtp

I don't think mod_dingaling will do a lookup for host: like sofia will as it doesn't have the code for that last I checked... I could be wrong but I don't recall it doing that. /b On Oct 13, 2009, at 3:02 PM, Mark Campbell-Smith wrote: > I have a hostname set in vars.conf.xml for the paramet

Re: [Freeswitch-users] Building Freeswitch with VPATH (src and obj directories are different)?

You shouldn't have to make clean usually ... doing so might break your tree... You can usually get by with "make current" that will ensure the critical things are cleaned and built correctly... every now and they you'll hit a snafu but we'll usually tell you about it. /b On Oct 13, 2009, at

Re: [Freeswitch-users] FS Slide deck?

Michael; Roger that. I'll check my schedule to see when I might have some time available to make the change. It seems as if I have some time available the tuesday after (bleep) freezes over. Acronym corrected. ;-) Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Oct 1

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

I'm using MySQL now but I will try PostgreSQL with the prefix module, is there a way to do that without the prefix module and with regular SQL? Any examples? Diego On Tue, Oct 13, 2009 at 10:45 PM, Even André Fiskvik wrote: > What database are you using? > You could do this with regular SQL, b

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

Wrong question. Is there a way to compare numbers with prefixes without using the prefix module? Diego On Wed, Oct 14, 2009 at 1:36 AM, Diego Viola wrote: > I'm using MySQL now but I will try PostgreSQL with the prefix module, is > there a way to do that without the prefix module and with regu

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

Diego Viola wrote: > I'm using MySQL now but I will try PostgreSQL with the prefix module, is > there a way to do that without the prefix module and with regular SQL? > > Any examples? SELECT * FROM rates WHERE prefix = SUBSTRING('$NUMBER$', 1, LENGTH(prefix)) LIMIT 1 Adnan > Diego > > On Tue,

[Freeswitch-users] T.38 in sip header

Dear all, Is posible to check T.38 in sip header? i want to disconnect call when T.38 not support in gateway and try next gateway BG Dome C. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.o

Re: [Freeswitch-users] sched_api doesn't get launched

Diego: You probably miss understood me. I said I was able to make "sched_hangup" work, but not the "sched_api" in the same way I script for "sched_hangup" The problem was on the second paragraph. thanks, On Wed, Oct 14, 2009 at 2:45 AM, Diego Viola wrote: > You need to pass the UUID to sched_h

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

Diego: Didn't wrote a prepaid system already? how did you compare the prefix in that system then? On Wed, Oct 14, 2009 at 10:07 AM, TTNC - Adnan Barakat wrote: > Diego Viola wrote: > > I'm using MySQL now but I will try PostgreSQL with the prefix module, is > > there a way to do that without the

Re: [Freeswitch-users] sched_api doesn't get launched

On Oct 13, 2009, at 8:30 PM, Henry Huang wrote: > Diego: > > You probably miss understood me. I said I was able to make > "sched_hangup" work, but not the "sched_api" in the same way I > script for "sched_hangup" > > The problem was on the second paragraph. Henry, can you capture the debu

Re: [Freeswitch-users] T.38 in sip header

Well if you look at the SDP you can look for "image" usually but thats not something I think you can see PRE call setup. /b On Oct 13, 2009, at 7:39 PM, Dome Charoenyost wrote: > Dear all, > > Is posible to check T.38 in sip header? i want to > disconnect call when T.38 not support

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

In our testing with SQL Server, we found that executing several queries for direct matches yielded far better performance than one query trying to check prefixes. (The column was also part of the clustered index, but AFAIK MySQL doesn't support defining your own clustered indexes; you get the PK

Re: [Freeswitch-users] openzap Failure opening channel error

But that setup is not there right now. It was an year back they used that. I'll try to make those setup and send you the log. I think it may take some 2 days time, and I'm not very familiar with asterisk. In the mean time, I request you to kindly go-through freeswitch debug log. You may still get

Re: [Freeswitch-users] openzap Failure opening channel error

Hi. I made the setup quickly with one of my colleague, and here is the debug log given by asterisk Steps done by me : + Call Made from 04443902743 to 04439114600 + When call lands to 04439114600, I just route it to 09176454982. *CLI> *CLI> *CLI> pri debug span 1 Enabled debugging

[Freeswitch-users] Perl event socket library problem

Dear All, I am implementing an IVR framework using Perl event socket libraries. At first I set socket mode async full. I have faced problem that the Perl statements executing before dailplan execution. So, I couldn't control my process. Next I have tried without async mode, it solved

Re: [Freeswitch-users] openzap Failure opening channel error

Lak, Okay, it stood out right away: FS is trying u-law but Asterisk is trying A-law. I'm not sure where the codec for openzap gets selected but you can try modifying this line in vars.xml: Try putting PCMA first. Also, I've never tried this but perhaps in your dialplan you can set absolute_code

Re: [Freeswitch-users] mod_socket: custom timeout for HEARTBEAT event

Finally!! Thank you Michael, I didn't know about "status" app. It satisfies all my desires. And again, thanks for all the community for the strong support! Artem On Tue, Oct 13, 2009 at 10:48 PM, Michael Collins wrote: > > > On Tue, Oct 13, 2009 at 8:32 AM, Artem Shiyanov wrote: > >> Sorry f

Re: [Freeswitch-users] Building Freeswitch with VPATH (src and obj directories are different)?

br...@freeswitch.org (Brian West) writes: > You shouldn't have to make clean usually ... doing so might break your > tree... Why? In any case on my mac (Leopard) neither make clean or distclean fully cleans up afterwards. I would prefer to get a completely "untouched" source tree after doing

Re: [Freeswitch-users] SIP Overlap support?

I suppose he want to have a central dialplan and a dummy phone instead... something as a MGCP phone behavior. T. On Tue, Oct 13, 2009 at 10:22 PM, Metik wrote: > As evidenced by various DTMF interop issues (with RFC2833, inband, > etc) over the years, I would avoid it if at all possible. > > Wh