Re: [Freeswitch-users] embedded freeswitch compatable hardware
voyage linux is a stripped debian and i was using it on an alix board some time ago... Asterisk was compiling on that without any issue. I beleive FS will do the same. T. On Fri, Dec 11, 2009 at 2:57 AM, Brian May br...@microcomaustralia.com.auwrote: On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote: Lack of OpenZAP support might be an issue, I assume that would be required to connect to an onboard analogue port... I assume I could just install Debian or another distribution instead though. This is another distribution I found: http://linux.voyage.hk/ It comes with Asterisk out of the box, although I suspect it wouldn't be too hard to get Freeswitch working instead. -- Brian May br...@microcomaustralia.com.au ___ 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] embedded freeswitch compatable hardware
Kristian, from your experience, supposed we go for net5501 + a 4 - 8 FXS card, what is the maximum simultaneous calls that this box can handle of course using g729 codec? I used blackgin (IP08), alix2d3... and all of them were giving up on 6-7 simultaneous calls. To be honest, i didnt run AstLinux on alix i used voyage instead but anyhow... this seems to be the limit. what i'm looking for it an appliance to run 2-16 FXS on it any suggestion? T. On Thu, Dec 10, 2009 at 4:47 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Brian, I have been making efforts to fully support FreeSWITCH in AstLinux. Our primary targets are low powered x86 boards like the Soekris and Alix. x86, powerful enough, cheap enough (as low as $100), and about 12 watts. Not bad. The Soekris net5501 and standard case will (I believe) take a full height card. Then again you could use any board and get an external SIP gateway (ATA). We don't currently support OpenZAP with FS in AstLinux but I'd love to add support for it eventually. I'm currently working with the FS devs on getting some issues in trunk resolved to get cross compiling working again. Until then you can find ISOs with FreeSWITCH and AstLInux here if you'd like to check it out: http://mirror.astlinux.org/freeswitch/daily/ Let me know what you think. On Wed, Dec 9, 2009 at 7:55 PM, Brian May br...@microcomaustralia.com.au wrote: Hello, I asked this question on my local linux user group mailing list, and got the recommendation to ask here. Anyway, at the moment I am running Asterisk on an IP04 embedded system. http://www.rowetel.com/ucasterisk/ip04.html It works well most of the time, however there are some bugs that do, under circumstances lead to less then desirable behaviour (such as on some occasions which I don't fully understand sometimes the remote system fails to generate any audio packets when there is no audio - almost like silence suppression was supported by the remote system - and asterisk fails to generate any audio packets in return; on another slower computer running the same SIP software and on the same network everything works fine; as far as I can tell the software - twinkle - doesn't even support silence suppression). I suspect at least some - if not all - of the issues I have encountered may be resolved with Freeswitch, however I don't really want to replace my small, energy efficient, embedded system, with a large, power hungry computer system. Overkill. An added complication is I need at least 1 analogue port to connect to the Australian based telephone line (2 ports exchange ports and 1 extension port would be ideal but not essiential). Unfortunately, I have been told that the IP04 hardware isn't compatable with the requirements of Freeswitch. Such as not having a MMU. So there doesn't appear to be much effort porting Freeswitch to IP04 as a result. I do have a spare TDM400p card, although as it is full height, suspect this isn't going to help. Are there any other good alternatives? Thanks. -- Brian May br...@microcomaustralia.com.au ___ 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 -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] embedded freeswitch compatable hardware
ok, but how much smultaneous calls did you get on an alix board using astlinux... for istnace, this is the question? T. On Thu, Dec 10, 2009 at 5:12 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: On Thu, Dec 10, 2009 at 9:26 AM, Frank Carmickle fr...@carmickle.com wrote: The 330 boards are a little more power hungry but you get a dual core 64 bit processor. As far as I'm concerned the performance increase is well worth the extra money. You still well below the power consumption of any other 64 bit dual core machines. http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383 --FC While these are low power when compared to traditional desktop/server systems, they're not what I would consider to be embedded. The CPU requires a fan (embedded no-no) and between the chipset and CPU they draw several times more power than a traditional embedded system. The ALIX and Soekris boards run with 12 watt power supplies (12v, 1 amp). The Atom 330 alone can draw 8 watts. This is still impressive for a processor of this class but it's not what I would consider to be embedded, yet... I think of embedded systems like this: Blackfin - Very low power, good performance (especially for DSP), very difficult porting (usually) ARM/MIPS - Very low power, decent performance depending on application, mild difficulty in porting X86 (Geode, etc) - Pretty low power, decent performance, relative ease in porting (often none) Everything else - You should probably call it an appliance, not an embedded system With the correct target application and design ARM and Geode systems can provide more than enough CPU power for many, many practical applications. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] HA questions.
Hi Mike, Lets suppose we have: - 2 machines configured for high availability (LAN HA) in a master/slave configuration with a floating public address on the master. ( http://www.ultramonkey.org/3/topologies/ha-overview.html) - freeswitch installed on every machine configured to use mysql in the core via odbc - both freeswitch have identical dialplan and directory configuration - mysql installed on every machine (with replication between the DBs) - SIP Trunks towards the upper provider (without registration but i should work with registration) - SIP Phones/Terminals registering to the active freeswitch When a terminal registers to the active freeswitch, the registration is propagated to the inactive one via DB replication. Now, lets suppose we have a switchover ... of course we will lose the ongoing calls but new calls (from SIP Phones) should be able to establish. The same applies to incoming calls from the upper provider. Im just talking about HA here not loadbalancing and performance scaling... what do you think about that? On Fri, Dec 4, 2009 at 1:56 AM, Michael Jerris m...@jerris.com wrote: so your registering to the provider to get the calls? If so, this gets tricky, the provider likely does not support multiple registrations, even if they did they probably send the call to both registered endpoints. With this big unknown its not very easy to suggest a good solution. If I were looking to set this up without needing proxies I would want to use srv records and naptr records and a provider that would balance using these including failiover. Mike On Dec 3, 2009, at 3:40 PM, Tim Uckun wrote: On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris m...@jerris.com wrote: The easiest place to do this is at the point you send the calls to FreeSWITCH. How are the calls coming in? From an as of now unkown SIP trunk provider (we are still in negotiations with a couple of companies). ___ 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] Bypass_media and re_invite
On Fri, Nov 27, 2009 at 11:00 AM, Steve Kurzeja steve.kurz...@gmail.comwrote: Is that USD ? :) i believe these are not Turkish liras :P ___ 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] odbc FLAG_MULTI_STATMENTS
On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com wrote: http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445 http://bugs.mysql.com/7445) so, is this the right patch ? http://bugs.mysql.com/file.php?id=6994 T. ___ 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] Handling the 302 Moved Temporarily response from JavaScript
this is how i do it from the dialplan:
extension name=ServiceLookup
condition field=destination_number
expression=^(300030)(.*)|^\+(300030)(.*)
action application=set data=bPfx=$1$3/
action application=set data=bNum=$2$4/
action inline=true application=set
data=intf=${regex(${caller_id_number}|^i\+(..)(.*) |%1)}/
action application=set
data=caller_id_number=${cond(${intf}==true ? ${caller_id_number:1:32} :
${caller_id_number})}/
action inline=true application=set
data=aPfx=${caller_id_number:0:6}/
action inline=true application=set
data=aNum=${caller_id_number:6:16}/
action inline=true application=set
data=IP_ADDR=${network_addr}:5060/
action application=lookup_service_destination data=in ${aNum},
in ${aPfx},
in ${bNum},
in ${bPfx},
in
${IP_ADDR},
out
redContact,
out
authResult/
action application=log data=INFO
ServiceLookup \n/
action application=log data=INFO
contact = '${redContact}' ##\n/
action application=log data=INFO
CallerNum = '${caller_id_number:6:16}' ##\n/
action application=log data=INFO
RADIUS auth = '${authResult}' ##\n/
action application=execute_extension data=doRedirect XML
public/
/condition
/extension
extension name=doRedirect
condition field=destination_number expression=^doRedirect$/
condition field=${authResult} expression=^0$|
action application=log data=INFO
RADIUS auth OK!!!' ##\n/
action application=redirect data=${red_contact}/
anti-action application=log data=INFO
RADIUS auth NOK!! ##\n/
anti-action application=respond data=403 Forbidden/
/condition
/extension
On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris m...@jerris.com wrote:
In trunk there is a sofia profile setting to allow dialplan processing of
302 responses. This won't get you back into your same javascript, but you
can probably do something clever from there.
Mike
On Nov 24, 2009, at 5:04 PM, John Platts wrote:
I have considered writing JavaScript code to bridge two calls together.
However, I would like to perform custom handling of the 302 Moved
Temporarily response. How do I handle the 302 Moved Temporarily response if
I use JavaScript?
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Re: [Freeswitch-users] Media got stuck after attended transfer...
it is better to enhance mod_fax with t.38 support... we have done sometihng and it is close to be work... T. On Sat, Nov 21, 2009 at 2:17 AM, Michael Jerris m...@jerris.com wrote: I think a better approach here is to use spandsp. We already have some groundwork done for this. If you are interested in contributing, please email consult...@freeswitch.org and we can discuss further. Mike On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote: Hi, one of my customers is willing to contribute for t38 integration. The basic idea is to connect HylaFAX to FS: t38modem - FreeSWITCH - Media Gateway with t38 support All this without media proxy. Another idea might be to implement t38 origination/termination with a class 1 modem input/output for use with HylaFAX. Do you know how much money we need to collect for t38 support? How much time is needed for implementing this? Thanks, Klaus *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Friday, October 16, 2009 2:10 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Media got stuck after attended transfer... On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga tculj...@gmail.com wrote: hi, any clue when can t38 be added? Eventually. :) Of course, if we could get more to add to the bounty it might grease the wheels of innovation. http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch Of course, I was listening to my A.M radio the other day and they said that there was this new invention called the Internet that would let people send documents to each other electronically. Maybe you should look into that. Next thing you know they'll come up with telephones that people don't have to plug into the wall and can take with them in the car. ;) -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
[Freeswitch-users] siptrace/debug log timestamp difference
Hi, just a thing i noticed... the debug log and sip trace have different time ... one hour difference ... looks like UTC/GMT issue. where do i set the time for siptrace correctly ? 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:411 (sofia/external/00010038516659...@10.4.5.107:5060) State Change CS_REPORTING - CS_DESTROY 2009-11-16 09:47:13.779070 [DEBUG] switch_core_session.c:1068 Session 31 (sofia/external/00010038516659...@10.4.5.107:5060) Locked, Waiting on external entities 2009-11-16 09:47:13.779070 [NOTICE] switch_core_session.c:1086 Session 31 (sofia/external/00010038516659...@10.4.5.107:5060) Ended 2009-11-16 09:47:13.779070 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/00010038516659...@10.4.5.107:5060 [CS_DESTROY] 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:564 (sofia/external/00010038516659...@10.4.5.107:5060) State DESTROY 2009-11-16 09:47:13.779070 [DEBUG] mod_sofia.c:255 sofia/external/ 00010038516659...@10.4.5.107:5060 SOFIA DESTROY 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:60 sofia/external/00010038516659...@10.4.5.107:5060 Standard DESTROY 2009-11-16 09:47:13.779070 [DEBUG] switch_core_state_machine.c:564 (sofia/external/00010038516659...@10.4.5.107:5060) State DESTROY going to sleep recv 462 bytes from udp/[10.4.5.107]:5060 at 08:47:13.799578: ACK sip:30003038515000...@l01sipindir2.ot.hr:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.4.5.107:5060 ;branch=z9hG4bKterm-13e-30003038515000403-00010038516659280-59521 From: 00010038516659280 sip:00010038516659...@10.4.5.107:5060 ;user=phone;tag=261638185 To: 30003038515000403 sip:30003038515000...@l01sipindir2.ot.hr:5060 ;user=phone;tag=9v58macH5mNNH Call-ID: 3189ce3b-3da37db2-3ac943f-...@10.4.5.107 CSeq: 1 ACK Max-Forwards: 10 Content-Length: 0 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP Overlap support?
Brian is right, pls, lets stop with exceptions and get stick to RFCs... otherwise it will be a big mess ... T. On Wed, Nov 4, 2009 at 3:03 PM, Brian West br...@freeswitch.org wrote: I'm going to say No! /b On Nov 4, 2009, at 2:23 AM, Dennis wrote: is there a way to send something like 484 (or something else), which does not make it a final answer and keep the call/socket alive? so we can ask the cirpack for further digits and decide what to do, if the cirpack does not send any digits. ___ 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] 3Com 3102 (3C10402B) Phone with FreeSwitch
well, if it is a sip phone than you should be able to input your usernamepassword somewhere. Usually, SIP phones downloads their configuration using dhcp/tftp|http method... the FW is downloaded just once if you need to upgrade the phone... I don't have any of these phones on my desk, just found the manual on the web. anyhow, freeswitch is expecting a SIP phone to register and thats it :P ... there is no specific phone provisioning from FS side. T. On Tue, Nov 3, 2009 at 9:03 PM, Dave Stevenson steve...@primrosebank.netwrote: Tihomir, thanks for the link, but actually, I had already found/downloaded/read and almost understood that document ! However, the options to log into the phone and configure the extension number etc. do not appear on my phone. From reading another post on the web, I don't think that the phone has the SIP software loaded until it is downloaded from the Server - I think that there is a special version of Asterix for 3Com that does this, maybe the same functionality does not exist in FreeSwitch ? Maybe I should have been clearer in the post below, but I think that this is the root of the problem. I think that the 3Com phone is looking for the Switch to download the SIP firmware to it and FreeSwitch does not seem to do that. Given that you have pointed me in the direction of that document, are you using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but please let me know how you've made it work regards Dave - Original Message - *From:* Tihomir Culjaga tculj...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tuesday, November 03, 2009 7:53 PM *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson steve...@primrosebank.netwrote: Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave ___ 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] 3Com 3102 (3C10402B) Phone with FreeSwitch
pity,the phone looks quite nice... On Tue, Nov 3, 2009 at 9:18 PM, Chris Chen chris.chen2...@gmail.com wrote: I think you are most likely on the wrong track, 3COM phones are locked to either 3COM PBX or the special Asterisk edition locked-down by 3COM. You cannot make them work with either FreeSWITCH or any other open SIP server other than 3COM IP PBX systems. I learned this over one year ago by playing with 3COm 3102 phones myself. Chris On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson steve...@primrosebank.netwrote: Tihomir, thanks for the link, but actually, I had already found/downloaded/read and almost understood that document ! However, the options to log into the phone and configure the extension number etc. do not appear on my phone. From reading another post on the web, I don't think that the phone has the SIP software loaded until it is downloaded from the Server - I think that there is a special version of Asterix for 3Com that does this, maybe the same functionality does not exist in FreeSwitch ? Maybe I should have been clearer in the post below, but I think that this is the root of the problem. I think that the 3Com phone is looking for the Switch to download the SIP firmware to it and FreeSwitch does not seem to do that. Given that you have pointed me in the direction of that document, are you using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but please let me know how you've made it work regards Dave - Original Message - *From:* Tihomir Culjaga tculj...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tuesday, November 03, 2009 7:53 PM *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson steve...@primrosebank.net wrote: Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave ___ 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 ___ 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] Sipura Codec Problem
just an off-topic question but it concenns mass provissioning ... does anyone know if there is an open TR069 platform we can work on? T. On Wed, Nov 4, 2009 at 1:16 AM, Michael Collins m...@freeswitch.org wrote: On Tue, Nov 3, 2009 at 11:11 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: It appears that Tony has already added an option (amazing) BUT you should really be setup for central provisioning with an installed base that large... You'll eventually have issues that *NO* amount of Tony/FreeSWITCH magic can fix. Kristian is correct. Listen to him because he's familiar with having lots and lots of units out in the field. The bandage Tony applied will eventually wear off. The long-term solution is to treat the malady and not the symptom. I'm certain that members of the FS community could point you toward some resources to assist with central provisioning. -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] mod_t38gateway
i tought so :PP T. On Sun, Nov 1, 2009 at 6:34 AM, Michael Jerris m...@jerris.com wrote: This is a non working module, just a shell for development. Mike On Oct 30, 2009, at 5:52 PM, Tihomir Culjaga wrote: does anybody know how does it work and how to use it in a dialplan? ___ 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] mod_t38gateway
does anybody know how does it work and how to use it in a dialplan? freeswi...@nemesis freeswi...@nemesis freeswi...@nemesis load mod_t38gateway API CALL [load(mod_t38gateway)] output: +OK 2009-10-30 22:44:38.204268 [NOTICE] mod_t38gateway.c:147 T.38 gateway enabled 2009-10-30 22:44:38.204268 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_t38gateway] 2009-10-30 22:44:38.204268 [NOTICE] switch_loadable_module.c:248 Adding Application 't38gateway' 2009-10-30 22:44:38.205374 [NOTICE] switch_loadable_module.c:270 Adding API Function 't38gateway' freeswi...@nemesis freeswi...@nemesis freeswi...@nemesis freeswi...@nemesis t38gateway API CALL [t38gateway()] output: -USAGE: uuid command freeswi...@nemesis T. ___ 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] Fwd: mod_opal - call charged before H.225 connect
Handling of fastStart in CallProceeding is commented out in h323plus library, this is exploration from h323plus developers about this: Yes that should be mera. The problem is that Callproceeding does not always come from the remote it may be generated by the gatekeeper. this is a feature .. called force_callproceeding. It means MERA will send a provisional CallProceeding in order not to timeout on calls that don't respond with that message on time. If this message contains a faststart element it is certanly a bug and it has to be reported to them. MERA where sending fast start elements in the Call proceeding and connect. The call proceeding where not valid and causing the media to fail. well if there is a correct faststart element within a connect message (or alerting or facility or progress), the originator should adjust the media resources accordingly. Here what could went wrong is just the media before the next faststart element in the row. Normally (although valid) EP's do not set Fast Start in Call proceeding so the code was disabled to resolve the MERA issue. well, this is unlikely as fast start element can be included in call proceeding message. The developer's task is to determine whether a call proceeding message is to be trusted or not. Also, provisional call proceeding messages don't have faststart element included! There are equipment (Cisco PGW / HSI) that are sending call proceeding with faststart element and h245Controll (OLC + TCS/MSD) that has to be treated correctly. Unfortunately, just disabling handling of callproceeding faststart element is not a real option... if you wont read bugs file in mod_h323, there is explaned how to enable it. of course i can enable it during build time but this will not solve interop issues later we can encounter... Do you maybe have some sniffs/traces of the wrong call proceeding message ? ...anyhow this is the expected behaviour when a GK/Proxy sends a provisional Call Proceding to the terminator and later it receives the real Call Proceeding carring faststart and h245Controll element within. Entities in the signalling path shall also use the Facility message or the Progress message to convey any new information (such as Q.931 information elements, CallProceeding-UUIE fields, tunnelled non-H.323 protocols, and encapsulated H.245 messages) received in a Call Proceeding message to the other endpoint if the entity has already sent a Call Proceeding message. This will allow the entity, for example, to transmit the fastStart element to facilitate proper establishment of a Fast Connect call and/or a Progress Indicator to indicate the presence of in-band tones and announcements. When using the Facility message to carrying such information extracted from the Call Proceeding message, the reason in the Facility should be set to forwardedElements. in other words: ORIG GKTERM - Setup OLC = Call proceeding (prov)= Setup OLC = Call proceeding (OLC+TCS/MSD) = Facility (OLC+TCS/MSD)= --- normal call establishment scenario follows --- ___ 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] Fwd: mod_opal - call charged before H.225 connect
P.S. people from russian community report what current version of module work fine on fs trunk version. that's strange that they report it working as m_txAudioOpened is never gonna be ready in pre_answer :P... i had to comment it to make it working. anyhow, i moved everything to trunk and will do some tests on Monday. T Hello Yuriy, I tried the trunk (FreeSWITCH Version 1.0.trunk (15216M)) and i'm getting some nice coredumps... FS crashes when placing outgoing calls. coredump on outbound call: FS log and backtrace http://pastebin.freeswitch.org/10834 FS crashes on incoming calls. coredump on inbound call: FS log and backtrace http://pastebin.freeswitch.org/10835 FS crashes when i try to load mod_h323. I need 3-4 attempts to load it without a crash. coredump on mod_h323 load: FS log and backtrace: http://pastebin.freeswitch.org/10833 FS crashes on shutdown procedure if mod_h323 was loaded previously. coredump on mod_h323 load: FS log and backtrace: http://pastebin.freeswitch.org/10836 It is quite bad :) ___ 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] Downloaded tar vs latest SVN - 200 OK has more headers
On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale anthony.miness...@gmail.com wrote: The headers are used to pass the callee-id info back to the other side so you have the id of who you called. The standards have failed us in this case as everything does it differently to the point that there is no standard thus we have invented our own way to carry this across from one FreeSWITCH box to another, but of course we can never make anybody happy. =/ I agree with you, X headers should be ignored by the equipment normally. Anyhow Kristian has a point here; there will be a lot of complains because of broken SIP stack on many vendor equipments So, can you consider some customizable a config option for such headers? T. ___ 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] Downloaded tar vs latest SVN - 200 OK has more headers
On Mon, Oct 26, 2009 at 7:26 PM, Anthony Minessale anthony.miness...@gmail.com wrote: This is ridiculous but here it is try r15230 add the profile param param name=pass-callee-id value=false/ sorry for that but, this will save you a lot of e-mail explaining why calls are not going through... thanks man! T. ___ 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] Setup advice on small LAN
If one of the computers does a big download, it messes with FS in two ways. If a connection is made, the voices are broken up, intermittent and difficult to understand. If the download is long enough, the connection to Flowroute is no longer usable due to registration failure. In any case, regardless if you are using a dedicated or mixed dsl line you should flag your voice traffic properly. signaling AF41, RTP EF... your voice traffic must never be flagged as pure date when sending it through open internet! T. ___ 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] Fwd: mod_opal - call charged before H.225 connect
TCBTW: it really doesn't have sense to develop on 1.0.4 ... the proof of TCconcept was done. I'm able to place calls in both directions so, lets move TCto trunk now. i have my own voip infrastructure on my work, and it's better for me to use it for tests, and use my personal on work too, i have problem with various hardware and call making, there is i have dialup connection and all what i can - make one call whit only g729 codec, or receive one call, no more, it's is a bandwich limitation. from hardware i have only ip phone artdio ipf2000 and one addpac gateway. also ssh sesssion is very slow. I see... it is not the ideal development environment :)... Indeed, a dialup is not a good thing... i guess your g729 is compressed to the maximum :) P.S. people from russian community report what current version of module work fine on fs trunk version. that's strange that they report it working as m_txAudioOpened is never gonna be ready in pre_answer :P... i had to comment it to make it working. anyhow, i moved everything to trunk and will do some tests on Monday. T ___ 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] Fwd: mod_opal - call charged before H.225 connect
TC TCit is gonna be easier to track. TC TCTomorrow i will test on 1.0.4 but please lets move to trunk. i make it a bit later, to move tickets to jira and source to svn i need some time to undertand how this system is works, especially jira. audio issue is better now :) however i have a few questions: 1. can we control codec framing size via config file setting (e.g. PCMA:20, PCMU:20)? 2. can we control tunneling via config file setting? 3. can we control mediaWaitForConnect flag within setup message via config file setting? Now, when i can test more and place outgoing calls to different equipment, i found that there is an issue when we get h.225 progress without a fastStart element. Here is a tshark: 5.24252610.4.62.7 - 10.4.4.254 H.225.0 CS: setup OpenLogicalChannel 5.243982 10.4.4.254 - 10.4.62.7H.225.0 CS: callProceeding 10.512617 10.4.4.254 - 10.4.62.7H.225.0 CS: progress 10.983697 10.4.4.254 - 10.4.62.7H.225.0 CS: alerting 20.002796 10.4.4.254 - 10.4.62.7H.225.0 CS: connect 20.002981 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility terminalCapabilitySet 20.003210 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility masterSlaveDetermination 31.472362 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: releaseComplete endSessionCommand the terminating GW didn't include a faststart element within a returning h.225 message we didn't match the capabilities (framing of them) in our setup (and you are waiting an open LC to start pre_answer) so now, the terminator is waiting for the originator to start exchanging TCS/MSD. As tunneling is true, this should be done using h.225 Facility messages. your behavior should be like this: 5.24252610.4.62.7 - 10.4.4.254 H.225.0 CS: setup OpenLogicalChannel g711A with 30 ms 5.243982 10.4.4.254 - 10.4.62.7H.225.0 CS: callProceeding 10.512617 10.4.4.254 - 10.4.62.7H.225.0 CS: progress 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility terminalCapabilitySet 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility masterSlaveDetermination 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility terminalCapabilitySet 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility masterSlaveDetermination 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility terminalCapabilitySetAck 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility masterSlaveDeterminationAck 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility terminalCapabilitySetAck 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility masterSlaveDeterminationAck 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility openlogicalchannel (g711A) 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility openlogicalchannel (g711A) 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility openlogicalchannelAck 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility openlogicalchannelAck now you can start pre_answer! 10.983697 10.4.4.254 - 10.4.62.7H.225.0 CS: alerting ... ... ... ___ 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] Fwd: mod_opal - call charged before H.225 connect
btw you are back with an old issue:
static const char modulename[] = h323;
static const char* h323_formats[] = {
G.711-ALaw-64k, PCMU,
G.711-uLaw-64k, PCMA,
GSM-06.10, GSM,
MS-GSM, msgsm,
SpeexNarrow, speex,
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Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
i meant you switched PCMA and PCMU... T. 2009/10/23 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-23 10:16 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TC TC TC TCit is gonna be easier to track. TC TC TC TCTomorrow i will test on 1.0.4 but please lets move to trunk. TC TC i make it a bit later, to move tickets to jira and source to svn i TC need some time to undertand how this system is works, especially jira. TC TC TCaudio issue is better now :) TC TChowever i have a few questions: TC TC1. can we control codec framing size via config file setting (e.g. PCMA:20, TCPCMU:20)? at this time i think no, there is a number issues in codec part now. TC2. can we control tunneling via config file setting? at this time no, i implement it later. TC3. can we control mediaWaitForConnect flag within setup message via config TCfile setting? what is mediaWaitForConnect flag, may be another trmin in h323? TCNow, when i can test more and place outgoing calls to different equipment, i TCfound that there is an issue when we get h.225 progress without a fastStart TCelement. TC TCHere is a tshark: TC TC 5.24252610.4.62.7 - 10.4.4.254 H.225.0 CS: setup OpenLogicalChannel TC TC 5.243982 10.4.4.254 - 10.4.62.7H.225.0 CS: callProceeding TC 10.512617 10.4.4.254 - 10.4.62.7H.225.0 CS: progress TC 10.983697 10.4.4.254 - 10.4.62.7H.225.0 CS: alerting TC 20.002796 10.4.4.254 - 10.4.62.7H.225.0 CS: connect TC 20.002981 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility TCterminalCapabilitySet TC 20.003210 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility TCmasterSlaveDetermination TC 31.472362 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: releaseComplete TCendSessionCommand TC TC TCthe terminating GW didn't include a faststart element within a returning TCh.225 message we didn't match the capabilities (framing of them) in our TCsetup (and you are waiting an open LC to start pre_answer) so now, the TCterminator is waiting for the originator to start exchanging TCS/MSD. As TCtunneling is true, this should be done using h.225 Facility messages. TC TC TCyour behavior should be like this: TC TC 5.24252610.4.62.7 - 10.4.4.254 H.225.0 CS: setup TCOpenLogicalChannel g711A with 30 ms TC 5.243982 10.4.4.254 - 10.4.62.7H.225.0 CS: callProceeding TC 10.512617 10.4.4.254 - 10.4.62.7H.225.0 CS: progress TC TC 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility TCterminalCapabilitySet TC 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility TCmasterSlaveDetermination TC TC 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility TCterminalCapabilitySet TC 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility TCmasterSlaveDetermination TC TC 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility TCterminalCapabilitySetAck TC 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility TCmasterSlaveDeterminationAck TC TC 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility TCterminalCapabilitySetAck TC 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility TCmasterSlaveDeterminationAck TC TC 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility TCopenlogicalchannel (g711A) TC 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility TCopenlogicalchannel (g711A) TC TC 10.4.62.7 - 10.4.4.254H.225.0/H.245 CS: facility TCopenlogicalchannelAck TC 10.4.4.254 - 10.4.62.7H.225.0/H.245 CS: facility TCopenlogicalchannelAck TC TC now you can start pre_answer! TC TC 10.983697 10.4.4.254 - 10.4.62.7H.225.0 CS: alerting TC... may bee, while i in hospital i have a very limited ways for testing, especially for inbound calls throuce h323. i find one issues in signaling part in h323plus, src/h323.cxx grep Very Frustrating - S.H. try uncomment fast start handling there, may be it help. C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE ___ 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
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
TC3. can we control mediaWaitForConnect flag within setup message via
config
TCfile setting?
what is mediaWaitForConnect flag, may be another trmin in h323?
If the calling endpoint sets the mediaWaitForConnect element to TRUE in the
Setup message, then
the called endpoint shall not send any media until after the Connect message
is sent.
The calling endpoint may begin transmitting media (according to the channels
opened) immediately
upon receiving a Q.931 message containing fastStart. Thus, the called
endpoint must be prepared to
immediately receive media on the channels it accepted in the Q.931 message
containing fastStart.
Note that national requirements may prohibit calling endpoints from
transmitting media prior to
receipt of a Connect message; it is the responsibility of the endpoint to
comply with applicable
requirements.
check H225_Setup_UUIE H323SignalPDU::BuildSetup within src/h323pdu.cxx
(H323plus)
TC...
may bee, while i in hospital i have a very limited ways for testing,
especially for inbound calls throuce h323. i find one issues in signaling
part in h323plus, src/h323.cxx grep Very Frustrating - S.H. try uncomment
fast start handling there, may be it help.
I'm not sure it is gonna help. This is only for CallProceeding having a
faststart element... What i have is a progress message without a faststart
element but with h245 address... it should go to StartControlChannel but i
think it is stuck since you call pre_answer before it actually opens a LC.
PBoolean H323Connection::OnReceivedProgress(const H323SignalPDU pdu)
{
if (pdu.m_h323_uu_pdu.m_h323_message_body.GetTag() !=
H225_H323_UU_PDU_h323_message_body::e_progress)
return FALSE;
const H225_Progress_UUIE progress =
pdu.m_h323_uu_pdu.m_h323_message_body;
SetRemoteVersions(progress.m_protocolIdentifier);
SetRemotePartyInfo(pdu);
SetRemoteApplication(progress.m_destinationInfo);
// Check for fastStart data and start fast
if (progress.HasOptionalField(H225_Progress_UUIE::e_fastStart))
HandleFastStartAcknowledge(progress.m_fastStart);
// Check that it has the H.245 channel connection info
if (progress.HasOptionalField(H225_Progress_UUIE::e_h245Address))
return StartControlChannel(progress.m_h245Address);
return TRUE;
}
you should handle this and postpone pre_answer until you get an open LC.
T.
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Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
TCcheck H225_Setup_UUIE H323SignalPDU::BuildSetup within
src/h323pdu.cxx
TC(H323plus)
i think it can be implemented later, but, why it may be needed? can you
explain some
situation where it need?
TC
TCyou should handle this and postpone pre_answer until you get an open LC.
pre_answer is not complete at this time, i say it a some kinde of hack,
there is
another issues with it ans sofia in case proxy-media true.
bool FSH323Connection::OnReceivedProgress(const H323SignalPDU pdu)
{
PTRACE(4, mod_h323\t==FSH323Connection::OnReceivedProgress);
m_txAudioOpened.Wait();
switch_channel_mark_pre_answered(m_fsChannel);
return true;
}
so for me the workaround for this was:
bool FSH323Connection::OnReceivedProgress(const H323SignalPDU pdu)
{
PTRACE(4, mod_h323\t==FSH323Connection::OnReceivedProgress);
PTRACE(4, mod_h323\t==FSH323Connection::OnReceivedProgress -
disabled pre_answer);
//m_txAudioOpened.Wait();
//switch_channel_mark_pre_answered(m_fsChannel);
return true;
}
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Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
TC
TCbool FSH323Connection::OnReceivedProgress(const H323SignalPDU pdu)
TC{
TCPTRACE(4,
mod_h323\t==FSH323Connection::OnReceivedProgress);
TC
TCPTRACE(4, mod_h323\t==FSH323Connection::OnReceivedProgress
-
TCdisabled pre_answer);
TC
TC//m_txAudioOpened.Wait();
TC//switch_channel_mark_pre_answered(m_fsChannel);
TCreturn true;
TC}
TC
in that chase wee are not hear anything going inband if receive progress
ind from called h323 endpoint,
there will bee ringback, for exmaple mobule fone then it out of network. if
you dont need
this make this changes until i fix it.
not true, because you have mediaWaitForConnect = false... the terminating
endpoint can send media before H.225 connect message and this actually
works well :P
7.317880 10.4.62.89 - 10.4.62.7SIP/SDP Request: INVITE
sip:00914392...@singtel, with session description
7.31831910.4.62.7 - 10.4.62.89 SIP Status: 100 Trying
7.33143010.4.62.7 - 10.4.62.89 SIP Status: 407 Proxy Authentication
Required
7.339420 10.4.62.89 - 10.4.62.7SIP Request: ACK
sip:00914392...@singtel
7.345078 10.4.62.89 - 10.4.62.7SIP/SDP Request: INVITE
sip:00914392...@singtel, with session description
7.34537810.4.62.7 - 10.4.62.89 SIP Status: 100 Trying
7.38716610.4.62.7 - 10.4.4.254 H.225.0 CS: setup OpenLogicalChannel
7.388636 10.4.4.254 - 10.4.62.7H.225.0 CS: callProceeding
9.389852 10.4.4.254 - 10.4.62.7H.225.0 CS: progress
10.639897 10.4.4.254 - 10.4.62.7H.225.0 CS: alerting
10.65132210.4.62.7 - 10.4.62.89 SIP Status: 180 Ringing
10.65393210.4.62.7 - 10.4.198.113 H.245 terminalCapabilitySet
10.65456510.4.62.7 - 10.4.198.113 H.245 masterSlaveDetermination
10.659757 10.4.198.113 - 10.4.62.7H.245 terminalCapabilitySet
10.659814 10.4.198.113 - 10.4.62.7H.245 masterSlaveDetermination
10.660161 10.4.198.113 - 10.4.62.7H.245 terminalCapabilitySetAck
10.660238 10.4.198.113 - 10.4.62.7H.245 masterSlaveDeterminationAck
10.66602810.4.62.7 - 10.4.198.113 H.245 terminalCapabilitySetAck
10.67038810.4.62.7 - 10.4.198.113 H.245 masterSlaveDeterminationAck
10.674693 10.4.198.113 - 10.4.62.7H.245 openLogicalChannel (g711A)
10.68241010.4.62.7 - 10.4.62.7RTP Unknown RTP version 1
#678: OLC found 10.4.62.7/10.4.198.113/129
10.68390210.4.62.7 - 10.4.198.113 H.245 openLogicalChannelAck
10.68737810.4.62.7 - 10.4.198.113 H.245 openLogicalChannel (g711A)
#723: OLC found 10.4.198.113/10.4.62.7/108
10.691579 10.4.198.113 - 10.4.62.7H.245 openLogicalChannelAck
10.778413 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=0, Time=24640
10.798476 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=1, Time=24800
10.818432 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=2, Time=24960
-- snip -
13.298358 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=126, Time=44800
13.318460 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=127, Time=44960
13.338405 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=128, Time=45120
13.358353 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=129, Time=45280
13.369984 10.4.4.254 - 10.4.62.7H.225.0 CS: connect
13.378381 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=130, Time=45440
13.38233010.4.62.7 - 10.4.62.89 SIP/SDP Status: 200 OK, with session
description
13.38883310.4.62.7 - 10.4.62.7RTP Unknown RTP version 3
13.38912310.4.62.7 - 10.4.62.7RTP Unknown RTP version 3
13.396419 10.4.62.89 - 10.4.62.7SIP Request: ACK
sip:00914392...@10.4.62.7:5060;transport=udp
13.398457 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=131, Time=45600
13.405954 10.4.62.89 - 10.4.62.7RTP PT=ITU-T G.711 PCMU,
SSRC=0xDEF4B36, Seq=27943, Time=991142687
13.418401 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=132, Time=45760
13.425864 10.4.62.89 - 10.4.62.7RTP PT=ITU-T G.711 PCMU,
SSRC=0xDEF4B36, Seq=27944, Time=991142847
13.438360 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=133, Time=45920
13.43857010.4.62.7 - 10.4.62.89 RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46377, Time=640
13.446202 10.4.62.89 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=27945, Time=991143007
13.458320 10.4.142.38 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0x1EC68E26, Seq=134, Time=46080
13.45846710.4.62.7 - 10.4.62.89 RTP PT=ITU-T G.711 PCMA,
SSRC=0x172DD4B, Seq=46378, Time=800
13.45900810.4.62.7 - 10.4.142.38 RTP PT=ITU-T G.711 PCMA,
SSRC=0xB9D8D8, Seq=1379, Time=991143007
13.466010 10.4.62.89 - 10.4.62.7RTP PT=ITU-T G.711 PCMA,
SSRC=0xDEF4B36, Seq=27946, Time=991143167
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
a solution to H323 endpoint = FS = SIP user no audio issue
is to disable a wait for tx Audio ... for case
SWITCH_MESSAGE_INDICATE_ANSWER:{
//m_txAudioOpened.Wait();
case SWITCH_MESSAGE_INDICATE_ANSWER:{
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, ANSWER: we got Answer event\n);
if (switch_channel_test_flag(channel, CF_OUTBOUND))
{
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, ANSWER: we got Answer event - CF_OUTBOUND
\n);
return SWITCH_STATUS_FALSE;
}
AnsweringCall(H323Connection::AnswerCallNow);
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, ANSWER: suppose the call is Answered Now\n);
PTRACE(4, mod_h323\tMedia started on connection
*this);
// test
//switch_channel_mark_answered(m_fsChannel);
m_rxAudioOpened.Wait();
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, ANSWER: wait for m_rxAudioOpened\n);
//m_txAudioOpened.Wait();
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, ANSWER: we disable wait for m_txAudioOpened\n);
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, ANSWER: were waiting for rx/tx AudioOpen\n);
if (!switch_channel_test_flag(m_fsChannel,
CF_EARLY_MEDIA)) {
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, ANSWER: we have early media\n);
PTRACE(4,
mod_h323\tswitch_channel_mark_answered(m_fsChannel)
*this);
switch_channel_mark_answered(m_fsChannel);
switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_CONSOLE, ANSWER: answered in early Media\n);
}
break;
}
Now, I'm able to both originate and terminate cals with 2-way audio...
the signaling looks correct...
outgoing:
1369.42504610.4.62.7 - 10.4.62.89 SIP/SDP Request: INVITE
sip:1...@10.4.62.89 sip%3a1...@10.4.62.89;transport=udp, with session
description
1369.42625510.4.62.7 - 10.4.62.31 H.225.0 CS: alerting
1369.435950 10.4.62.89 - 10.4.62.7SIP Status: 100 Trying
1369.449065 10.4.62.89 - 10.4.62.7SIP Status: 180 Ringing
1369.60510910.4.62.7 - 10.4.62.31 H.225.0 CS: progress
OpenLogicalChannel
1369.609788 10.4.62.31 - 10.4.62.7H.225.0/H.245 CS: facility
terminalCapabilitySet
1369.610489 10.4.62.31 - 10.4.62.7H.225.0/H.245 CS: facility
masterSlaveDetermination
1369.61907110.4.62.7 - 10.4.62.31 H.225.0/H.245 CS: empty
terminalCapabilitySet
1369.62034910.4.62.7 - 10.4.62.31 H.225.0/H.245 CS: empty
terminalCapabilitySetAck
1369.623215 10.4.62.31 - 10.4.62.7H.225.0/H.245 CS: facility
terminalCapabilitySetAck
1369.62559110.4.62.7 - 10.4.62.31 H.225.0/H.245 CS: empty
masterSlaveDeterminationAck
1369.628174 10.4.62.31 - 10.4.62.7H.225.0/H.245 CS: facility
masterSlaveDeterminationAck
1370.966958 10.4.62.89 - 10.4.62.7SIP/SDP Status: 200 OK, with
session description
1370.96743110.4.62.7 - 10.4.62.89 SIP Request: ACK
sip:1...@10.4.62.89 sip%3a1...@10.4.62.89;transport=udp
1370.97517210.4.62.7 - 10.4.62.31 H.225.0 CS: connect
1372.354383 10.4.62.89 - 10.4.62.7SIP Request: BYE
sip:mod_so...@10.4.62.7:5060
1372.35514710.4.62.7 - 10.4.62.89 SIP Status: 200 OK
1372.39290410.4.62.7 - 10.4.62.31 H.225.0/H.245 CS: releaseComplete
endSessionCommand
1372.397302 10.4.62.31 - 10.4.62.7H.225.0 CS: releaseComplete
incoming:
1502.817154 10.4.62.31 - 10.4.62.7H.225.0 CS: setup
OpenLogicalChannel
1502.83373210.4.62.7 - 10.4.62.31 H.225.0 CS: callProceeding
1502.85090910.4.62.7 - 10.4.62.89 SIP/SDP Request: INVITE
sip:1...@10.4.62.89 sip%3a1...@10.4.62.89;transport=udp, with session
description
1502.85175810.4.62.7 - 10.4.62.31 H.225.0 CS: alerting
1502.861828 10.4.62.89 - 10.4.62.7SIP Status: 100 Trying
1502.875127 10.4.62.89 - 10.4.62.7SIP Status: 180 Ringing
1503.03325810.4.62.7 - 10.4.62.31 H.225.0 CS: progress
OpenLogicalChannel
1503.037908 10.4.62.31 - 10.4.62.7H.225.0/H.245 CS: facility
terminalCapabilitySet
1503.038608 10.4.62.31 - 10.4.62.7H.225.0/H.245 CS: facility
masterSlaveDetermination
1503.05015410.4.62.7 - 10.4.62.31 H.225.0/H.245 CS: empty
terminalCapabilitySet
1503.05138110.4.62.7 - 10.4.62.31 H.225.0/H.245 CS: empty
terminalCapabilitySetAck
1503.054297 10.4.62.31 - 10.4.62.7H.225.0/H.245 CS: facility
terminalCapabilitySetAck
1503.05491710.4.62.7 - 10.4.62.31 H.225.0/H.245 CS: empty
masterSlaveDeterminationAck
1503.057933 10.4.62.31 - 10.4.62.7
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
2009/10/23 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-23 10:37 -0500, Anthony Minessale wrote freeswitch-us...@lists.f...: i have no way to install trunk at this time, i will go out of hospital about one week later, after this i will can try it on trunk. AMif you were on trunk that line of code would be gone. AMyou really can't do development on 1.0.4 its 6 months old and it will cause AMyou more trouble than you think when you eventually upgrade if you do not do uh... you are still in there ... damn sorry to hear that. I know what the feeling is... i was in the same position a couple of months ago... anyhow at least i had plenty of time to do my private research.. hope it is the case with you as well. so, if you have that time i can offer you my development server with trunk installed and all the hands on. let me know... BTW: it really doesn't have sense to develop on 1.0.4 ... the proof of concept was done. I'm able to place calls in both directions so, lets move to trunk now. T. ___ 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] Fwd: mod_opal - call charged before H.225 connect
TC
TCI have enabled crash-protection and when i do SIP = H323 call it
doesn't
TCgenerate coredumps... it is just this thread that is crashing ... pls
check
TCthe log downbelow:
core dump in case enabled crash-protection may be unusable, i have a case
then
my module crash silently, after this crash-protection is killing sip leg
and after
this i get core dump where backtrace show me segfault in libc6, i spent one
day to
understand this situation, and then i disable crash-protection i see there
is actualy
it crashes. disable crash-protection and show backtrace of crash, i think
result will
be different.
TC2009-10-21 17:35:28.691688 [DEBUG] mod_h323.cpp:600
TC==FSH323Connection::decodeCapability
TC
TC
TC
TCWell, I'm not sure if the backtrace is from 1.0.4 ... i will disable
TCcrass-protection and will send new logs to you.
TC
TC
TCAlso, if you like i can give you access to the machine itself...
TC
TCT.
TC
Hi, here is the FS log without crash-protection:
http://pastebin.freeswitch.org/10796 and here is the backtrace:
http://pastebin.freeswitch.org/10797
my dialplan looks ok, so i guess it is up to the module.
extension name=ENYTHING_ELSE
condition field=destination_number
expression=^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
!--action application=set data=bypass_media=false/--
action application=set data=proxy_media=true/
!--action application=bridge data=opal/h323:0...@${ncx_ip}/--
action application=bridge data=h323/0...@${ncx_ip}/
/condition
/extension
please advice,
T.
C уважением With Best Regards
Георгиевский Юрий.Georgiewskiy Yuriy
+7 4872 711666+7 4872 711666
факс +7 4872 711143 fax +7 4872 711143
Компания ООО Ай Ти Сервис IT Service Ltd
http://nkoort.ru http://nkoort.ru
JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru
YG129-RIPEYG129-RIPE
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Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
TCHi, here is the FS log without crash-protection: TChttp://pastebin.freeswitch.org/10796 and here is the backtrace: TChttp://pastebin.freeswitch.org/10797 i fix this crash already, please download latest version from same link as previous, recompile and try again. outgoing works, I can place an end-to-end call ... except the RTP is realy delayed ... after approx 30 sec of conversation the audio is delayed more than 10 seconds but i have 2 way audio for outgoing calls:) Do you need some logs ? Inbound cals still the same... i suppose you didn't have a chance working on that as well ... T. ___ 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] Fwd: mod_opal - call charged before H.225 connect
TC
TCDo you need some logs ?
try disable medai-proxy, there is issue with rtp now then medai-proxy or
transcoding enabled.
Outbound calls:
disabled rtp proxy and it is still the same issue ... audio delay H323 =
SIP endpoint.
Inbound calls:
This is the extension i use to register my Avaya SIP phone to FS.
include
user id=1001
params
param name=password value=$${default_password}/
param name=vm-password value=1001/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=1001/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension 1001/
variable name=effective_caller_id_number value=1001/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
/variables
/user
/include
This is my h323.conf.xml
configuration name=h323.conf description=H323 Endpoints
settings
param name=trace-level value=4/
param name=context value=default/
param name=dialplan value=XML/
param name=codec-prefs value=PCMU,PCMA,GSM,G729,G726/
param name=gk-address value=/!-- empty to disable, * to
search LAN --
param name=gk-identifer value=/ !-- optional name of gk --
param name=gk-interface value=/ !-- optional listener interface
name --
/settings
listeners
listener name=default
param name=h323-ip value=10.4.62.7/
param name=h323-port value=1720/
/listener
/listeners
/configuration
I'm using default context and an inbound call looks for a registered user in
default context where 1001 user is registered to.
here is the log for an outgoing call:
http://pastebin.freeswitch.org/10799and here is a tshark output:
http://pastebin.freeswitch.org/10800
there are 2 thing that are not working here:
1. no audio at all!
2. hangup from SIP User side doesn't release the H323 leg
two points for your reference in the logs:
1. Here, SIP User disconnected the SIP leg, but nothing was triggered in
mod_h323 ... as the callback function on_hangup (in mod_h323.cpp) was
never triggered!
freeswi...@subzero
freeswi...@subzero
freeswi...@subzero recv 371 bytes from udp/[10.4.62.89]:5060 at
14:39:36.714521:
BYE sip:mod_so...@10.4.62.7:5060 SIP/2.0
From: sip:1...@10.4.62.89 sip%3a1...@10.4.62.89
;transport=udp;tag=-22166b474ae08abf-7_T10.4.62.89
To: sip:1001282...@10.4.62.7 sip%3a1001282...@10.4.62.7
;tag=Qpc53NZ2cZc1N
Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
CSeq: 127 BYE
Via: SIP/2.0/UDP
10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
Content-Length: 0
Max-Forwards: 70
Supported: replaces
2009-10-22 16:39:36.714604 [NOTICE] sofia.c:322 Hangup sofia/internal/
sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [CS_CONSUME_MEDIA]
[NORMAL_CLEARING]
2009-10-22 16:39:36.714604 [DEBUG] switch_channel.c:1683 Send signal
sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [KILL]
2009-10-22 16:39:36.714604 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [BREAK]
send 520 bytes to udp/[10.4.62.89]:5060 at 14:39:36.715258:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.4.62.89;branch=z9hG4bK-7e5dc720_442d0f8-2d8bf1174f235bec_B
From: sip:1...@10.4.62.89 sip%3a1...@10.4.62.89
;transport=udp;tag=-22166b474ae08abf-7_T10.4.62.89
To: sip:1001282...@10.4.62.7 sip%3a1001282...@10.4.62.7
;tag=Qpc53NZ2cZc1N
Call-ID: 8aa825c6-39bb-122d-bb89-00110a5be1f0
CSeq: 127 BYE
User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:503
(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) State
CONSUME_MEDIA going to sleep
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) Running State
Change CS_HANGUP
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:434
(sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) State HANGUP
2009-10-22 16:39:36.721097 [DEBUG] mod_sofia.c:338 Channel sofia/internal/
sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 hanging up, cause:
NORMAL_CLEARING
2009-10-22 16:39:36.721097 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 Standard HANGUP,
cause: NORMAL_CLEARING
2009-10-22 16:39:36.721097
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-22 09:27 -0500, Anthony Minessale wrote freeswitch-us...@lists.f...: AMcrash protection has been completely removed from FreeSWITCH, I certianly AMhope you are working on this against SVN trunk? i don't have trunk at this time, my current work is based on 1.0.4 version. Yuriy, it is better if we move this through a jira ticket, this way it is a mess. So if you agree, we can open a ticket where we can follow up all issues with mod_h323. Also, the same applies to FS trunk... first i wanted to see if i was doing something wrong when i tried your module. Now, when you fixed outgoing calls it is time to go on trunk as when we finish this 1.0.4 will be outdated and obsolete. so, to continue on this topic i suggest: 1. open a jira ticket 2. move to fs-trunk 3. upload the current src of mod_h323 to the FSSVN do you agree ? Tihomir. ___ 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] Proxy media mode with T.38 re-invite
On Thu, Oct 22, 2009 at 5:44 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: An update for Tony, Brian, Mike, and everyone on the list... I was able to get some phone time with the team yesterday. Tony worked on my machine, found the issue, and had it committed within 30 minutes. I've been testing T.38 all morning between the fax machines in the office with few issues. and what these few issues are? :P THANKS AGAIN GUYS! ___ 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] Proxy media mode with T.38 re-invite
indeed, this looks like a dialect problem between your fax machine and your T.38 device. Anyhow, T.38 doesn't work well with SG3... I Always have to disable v.34 in order to have a reliable fax service. Also, cisco uses to suppress CM so that SG3 timeouts on ANSam the communication fallbacks to ordinary G3. Kristian, just for fun, what are you using to send the fax ? T. On Thu, Oct 22, 2009 at 8:16 PM, Gabriel Kuri gk...@ieee.org wrote: Out of curiosity, is it a Super G3 (ie v.34) capable FAX? We've had nothing but intermittent problems with Super G3 FAXes over T.38, unless v.34 is strictly turned off on the machine. Gabe Kristian Kielhofner wrote: On Thu, Oct 22, 2009 at 11:58 AM, Tihomir Culjaga tculj...@gmail.com wrote: One fax machine here in the office (still testing others) correctly sends all fax pages. A minute or so after the fax is marked successful on both sides it hangs up, redials, and resends the last page... It never did it while connected to the PSTN but then again my other fax machine isn't doing it either. I'm going to test with more fax machines to see if it's an issue with that specific machine. ___ 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] Fwd: mod_opal - call charged before H.225 connect
2009/10/22 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-22 16:53 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: finally i fix this rtp bug, check new wersion please. if course i can do that, but tomorrow morning ... i'm not in the office anymore. BTW: can we please move the tickets to jira? it is gonna be easier to track. Tomorrow i will test on 1.0.4 but please lets move to trunk. T. ___ 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] Fwd: mod_opal - call charged before H.225 connect
simple:
action application=bridge data=h323/${number}/
if fs not registered on gk then data=h323/${numb...@xxx.xxx.xxx.xxx.
TC
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability:
TCUserInput/PointDevice 14
TC2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external
thread
TC0xb6eb60a0 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external
thread
TC0xb6ebafa8 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread
TC0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90)
TC2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability:
TCUserInput/Modal 15
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TCexpiries=0
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0,
TCexpiries=0
TC2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external
thread
TC0xb6eba910 for id 3048876944
TC2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached
TCSegmentation fault (core dumped)
TCtculj...@subzero:~/freeswitch-trunk$
TC
TCpls check: http://pastebin.freeswitch.org/10769
look strange, what version of libpt/h323plus you use and freeswitch itself
?
TC
TC
I was using latest libpt.so.2.7-beta1.
Now I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you mentioned...)
and FS is crashing on every call :P .. regardless if it is inbound or
outbound...
FreeSWITCH Version 1.0.trunk (15079M)
H323Plus is from cvs
so, what i did is:
create a directory e.g. h323
mkdir -p ~/h323
cd ~/h323
svn co
http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/branches/v2_6ptlib-2.6
export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig
export LD_LIBRARY_PATH=/usr/local/lib
cd ptlib-2.6
./configure
make
sudo make install
cd ~/h323
cvs -d:pserver:anonym...@h323plus.cvs.sourceforge.net:/cvsroot/h323plus
checkout h323plus
export PTLIBDIR=~/h323plus/ptlib
cd h323plus
./configure
make
sudo make install
assuming you have FS src in your home
cd ~/freeswitch-trunk
make mod_h323-clean
make mod_h323
sudo make mod_h323-install
cd /usr/local/freeswitch/lib/
sudo ln -sf /usr/local/lib/libpt.so.2.6-beta6 libpt.so.2.6-beta6
start FS and load mod_h323
Please, can you advice what exact revisions of ptlib you are using so i can
do svn so -r xxx, also what exact revision of freeswitch and H323Plus you
are using ?
Now with ptlib-2.6-beta6 can't even.
T.
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Re: [Freeswitch-users] Call custom variable in condition
consider this:
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS} XML
Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=${AUTHENTICATION_STATUS} expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
here is one of my dialplan. I'm using execute_extension but it is quite the
same...
extension name=ServiceLookup
condition field=destination_number expression=(^300030)(.*)
action application=lookup_service_destination data=in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult/
action application=log data=INFO
ServiceLookup \n/
action application=log data=INFO
contact = '${red_contact}' ##\n/
action application=log data=INFO
CallerNum = '${caller_id_number:6:16}' ##\n/
action application=log data=INFO
RADIUS auth = '${authResult}' ##\n/
action application=execute_extension data=doRedirect XML
public/
/condition
/extension
extension name=doRedirect
condition field=destination_number expression=^doRedirect$/
condition field=${authResult} expression=^0$|^60$
action application=log data=INFO
RADIUS auth OK!!!' ##\n/
action application=redirect data=${red_contact}/
anti-action application=log data=INFO
RADIUS auth NOK!! ##\n/
anti-action application=respond data=403 Forbidden/
/condition
/extension
On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
Hi,
I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I
want to use it in condition, as I'm listing down the configuration below;
context name=SIP_incoming
extension name=call-sip-extensions
condition field=destination_number expression=^(\d+)$
action application=set data=AUTHENTICATION_STATUS=0/
action application=transfer data=${AUTHENTICATION_STATUS}
XML Authen_Status/
/condition
/extension
/context
context name=Authen_Status
extension name=exten-auth-status
condition field=AUTHENTICATION_STATUS expression=^0$
action application=answer/
action application=playback data=play.wav/
/condition
/extension
/context
But unfortunately it is not working. Kindly advise me how to do implement
it(Note: I don't want to call script). And one more thing to ask how can I
transfer the values within the same context?
--
Regards,
Ahmed Munir
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Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
2009/10/21 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-21 09:34 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TC TC TC TCI was using latest libpt.so.2.7-beta1. TC TCNow I went back to libpt.so.2.6-beta6 (can't find 2.6.5 you mentioned...) TCand FS is crashing on every call :P .. regardless if it is inbound or TCoutbound... http://www.opalvoip.org/ first link into Lalande Stable 5 Released. On some version of cvs ptlib i get crash on module loading:) TC TCFreeSWITCH Version 1.0.trunk (15079M) hm, i don't test it on trunk, may be there some isues, try get stack backtrace from core file to see where it crash. I use 1.0.4 module load crash: http://pastebin.freeswitch.org/10783 FreeSWITCH backtrace: http://pastebin.freeswitch.org/10784 now, the only different thing is FS trunk ... :P ___ 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] NOT in dialplan expression
it depends of what you are trying to acheave one approach is with regex
check this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_regex
you can set a different variable and have it true or false ... than you can
compare for false state...
well .. it is up to you ...
T.
On Wed, Oct 21, 2009 at 1:34 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How do I do a NOT equal to in a dialplan expression
Normaly in regex I would use the ! character. This doesn't seem to work in
FS..
ie
condition field=${variable} expression=!^1
Shouldn't that match when the variable is not starting with one?
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Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
FS = 10.4.62.7
SIP phone = 10.4.62.89
H323 endpoint = 10.1.14.153
TC2. hangup from sip side doesn't release the h323 leg (now the difference
is
TCthat FS is not complaining about thread mismatch ant it looks clean but
FS
TCdoesn't send any releasecomplete message... strange)
TC3. coredumps when i place outgoing calls
btw,
TC 70.552157 10.1.14.153 - 10.4.62.7H.245 endSessionCommand
TC 70.552401 10.1.14.153 - 10.4.62.7H.225.0 CS: releaseComplete
TC 70.55398110.4.62.7 - 10.4.62.7RTP Unknown RTP version 3
TC 70.55448810.4.62.7 - 10.1.14.153 H.245 endSessionCommand
TC 70.55505610.4.62.7 - 10.1.14.153 H.225.0 CS: releaseComplete
it send, now i have no way to test h323-sip transit, i will have it
tomorow.
sip-h323 for me work fine now, give backtrace from code dump of 1.0.4
where it die?
this endSession is when i hangup from H232 side as well :P ... if i don't
hangup on H323 side the H323 leg is not released. Pls chec the time the
packets were sent ...
Here i hangup on the SIP Phone:
68.374916 10.4.62.89 - 10.4.62.7SIP Request: BYE
sip:mod_so...@10.4.62.7:5060
68.37534210.4.62.7 - 10.4.62.7RTP Unknown RTP version 3
68.37562010.4.62.7 - 10.4.62.89 SIP Status: 200 OK
2 sec delay
Here i hangup on the H323 endpoint (releaseComplete comes from H323 endpoint
first here )
70.552157 10.1.14.153 - 10.4.62.7H.245 endSessionCommand
70.552401 10.1.14.153 - 10.4.62.7H.225.0 CS: releaseComplete
FS just acknowlages it here:
70.55448810.4.62.7 - 10.1.14.153 H.245 endSessionCommand
70.55505610.4.62.7 - 10.1.14.153 H.225.0 CS: releaseComplete
I have enabled crash-protection and when i do SIP = H323 call it doesn't
generate coredumps... it is just this thread that is crashing ... pls check
the log downbelow:
Dialplan: sofia/internal/1...@singtel Regex (FAIL) [EMERGENCY]
destination_number(05492122) =~ /^0(112|9[23456])$/ break=on-false
Dialplan: sofia/internal/1...@singtel parsing [default-SPECIAL_SERVICES]
continue=false
Dialplan: sofia/internal/1...@singtel Regex (FAIL) [SPECIAL_SERVICES]
destination_number(05492122) =~ /^0(9[01789]\d{3,4})$/ break=on-false
Dialplan: sofia/internal/1...@singtel parsing [default-ENYTHING_ELSE]
continue=false
Dialplan: sofia/internal/1...@singtel Regex (PASS) [ENYTHING_ELSE]
destination_number(05492122) =~
/^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$/ break=on-false
Dialplan: sofia/internal/1...@singtel Action
set(effective_caller_id_number=1001282122)
Dialplan: sofia/internal/1...@singtel Action set(NCX_IP=10.4.4.254)
Dialplan: sofia/internal/1...@singtel Action set(call_timeout=30)
Dialplan: sofia/internal/1...@singtel Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1...@singtel Action set(bypass_media=false)
Dialplan: sofia/internal/1...@singtel Action set(proxy_media=true)
Dialplan: sofia/internal/1...@singtel Action bridge(h323/05492...@${ncx_ip})
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:114
(sofia/internal/1...@singtel) State Change CS_ROUTING - CS_EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_session.c:932 Send signal
sofia/internal/1...@singtel [BREAK]
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:484
(sofia/internal/1...@singtel) State ROUTING going to sleep
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:398
(sofia/internal/1...@singtel) Running State Change CS_EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/1...@singtel) State EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] mod_sofia.c:173
sofia/internal/1...@singtel SOFIA EXECUTE
2009-10-21 17:35:28.682475 [DEBUG] switch_core_state_machine.c:151
sofia/internal/1...@singtel Standard EXECUTE
EXECUTE sofia/internal/1...@singtel set(open=true)
2009-10-21 17:35:28.682475 [DEBUG] mod_dptools.c:748
sofia/internal/1...@singtel SET [open]=[true]
EXECUTE
sofia/internal/1...@singtelhash(insert/10.4.62.7-spymap/1001/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
EXECUTE
sofia/internal/1...@singtelhash(insert/10.4.62.7-last_dial/1001/05492122)
EXECUTE
sofia/internal/1...@singtelhash(insert/10.4.62.7-last_dial/global/5c3ebda2-be57-11de-a6dd-e7de0b74bdc9)
EXECUTE sofia/internal/1...@singtelset(effective_caller_id_number=1001282122)
2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
sofia/internal/1...@singtel SET [effective_caller_id_number]=[1001282122]
EXECUTE sofia/internal/1...@singtel set(NCX_IP=10.4.4.254)
2009-10-21 17:35:28.685219 [DEBUG] mod_dptools.c:748
sofia/internal/1...@singtel SET [NCX_IP]=[10.4.4.254]
EXECUTE sofia/internal/1...@singtel set(call_timeout=30)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1...@singtel SET [call_timeout]=[30]
EXECUTE sofia/internal/1...@singtel set(hangup_after_bridge=true)
2009-10-21 17:35:28.686292 [DEBUG] mod_dptools.c:748
sofia/internal/1...@singtel SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1...@singtel set(bypass_media=false)
2009-10-21
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
TC TCcall flow is SIP_user = FS = H323_endpoint is failing .. coredumped TChttp://pastebin.freeswitch.org/10703 i fix some bugs now, ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this is updated version, try it, if you experience no audio try enable rtp proxy in you sip profile. Hi, there are several issues... lets start with top 4 :) 1. I'm still stuck with no audio: I have this parameter in the sip profile set: param name=inbound-proxy-media value=true/ ...tried with both with slow start and fast start... any idea ? pls check: http://pastebin.freeswitch.org/10771 2. outgoing calls still failing in coredumps: what is your dialplan ? ... how do you call bridge application? 2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3252 Found capability: UserInput/PointDevice 14 2009-10-20 10:08:18.424083 [DEBUG] h323caps.cxx:3248 FindCapability: 15 2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread 0xb6eb60a0 for id 3048876944 2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:65 Destroyed external thread 0xb6ebafa8 for id 3048876944 2009-10-20 10:08:18.426608 [DEBUG] tlibthrd.cxx:406 Destroyed thread 0xb6ebafa8 PExternalThread:0xb5ba2b90(id = b5ba2b90) 2009-10-20 10:08:18.426608 [DEBUG] h323caps.cxx:3252 Found capability: UserInput/Modal 15 2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0, expiries=0 2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:880 MONITOR: timers=0, expiries=0 2009-10-20 10:08:18.426608 [DEBUG] osutils.cxx:60 Created external thread 0xb6eba910 for id 3048876944 2009-10-20 10:08:18.426608 [DEBUG] h4601.cxx:1725 Endpoint Attached Segmentation fault (core dumped) tculj...@subzero:~/freeswitch-trunk$ pls check: http://pastebin.freeswitch.org/10769 3. when you hangup from SIP side, the call is not released end-to-end (the H323 endpoint doesn't get any releaseComplete message) 2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2, expiries=3 2009-10-20 10:10:51.264527 [DEBUG] h323neg.cxx:432 Received MasterSlaveDeterminationAck: state=Incoming 2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=1, expiries=3 2009-10-20 10:10:51.264527 [DEBUG] osutils.cxx:880 MONITOR: timers=2, expiries=4 2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138 InternalEstablishedConnectionCheck: connectionState=EstablishedConnection fastStartState=FastStartAcknowledged 2009-10-20 10:10:51.264527 [DEBUG] h323caps.cxx:3264 FindCapability: T.120 2009-10-20 10:10:51.264527 [DEBUG] h323.cxx:4138 InternalEstablishedConnectionCheck: connectionState=EstablishedConnection fastStartState=FastStartAcknowledged 2009-10-20 10:10:51.264527 [DEBUG] tlibthrd.cxx:1023 PThread::PXBlockOnIO(45,0) 2009-10-20 10:10:54.405479 [NOTICE] sofia.c:328 Hangup sofia/internal/ sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [CS_CONSUME_MEDIA] [NORMAL_CLEARING] 2009-10-20 10:10:54.405479 [DEBUG] switch_channel.c:1726 Send signal sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [KILL] 2009-10-20 10:10:54.405479 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 [BREAK] 2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:437 thread mismatch skipping state handler. 2009-10-20 10:10:54.405479 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) Running State Change CS_HANGUP 2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89) State HANGUP 2009-10-20 10:10:54.406530 [DEBUG] mod_sofia.c:338 Channel sofia/internal/ sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 hanging up, cause: NORMAL_CLEARING 2009-10-20 10:10:54.406530 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1...@10.4.62.89 sip%3a1...@10.4.62.89 Standard HANGUP, cause: NORMAL_CLEARING pls check: http://pastebin.freeswitch.org/10771 4. sometimes when i shutdown FS i get coredimps - from my experience it looks like you don't wait for a FS thread to finish when you exit... 2009-10-20 10:05:59.493306 [CONSOLE] switch_event.c:508 Stopping queue thread 2 2009-10-20 10:05:59.493339 [CONSOLE] switch_core.c:1693 Finalizing Shutdown. 2009-10-20 10:05:59.493379 [CONSOLE] switch_log.c:310 Logger Ended. 2009-10-20 10:05:59.494472 [CONSOLE] switch_core_memory.c:567 Stopping memory pool queue. Segmentation fault (core dumped) tculj...@subzero:~/freeswitch-trunk$ tculj...@subzero:~/freeswitch-trunk$ Please advice your FS/mod_h323.conf.xml settings... Tihomir. C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE
Re: [Freeswitch-users] Troubles with proxy media mode
you are making FS to play wav file when sending a call in G711 or GSM or some other codec. you might use mod_native_filehttp://wiki.freeswitch.org/wiki/Mod_native_fileto avoid transcoding. T. On Tue, Oct 20, 2009 at 9:56 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Hello everyone, I'm trying to use proxy media across two profiles. The codec settings are identical, they both have late negotiation enabled, and they both have inbound-proxy-media set to true (I also tried setting proxy_media from the dialplan). FreeSWITCH ends up clearing the call with TRANSCODING_NECESSARY but I can't figure out why it thinks it needs to transcode for this call. I've attached a level 7 debug and an ngrep siptrace showing traffic from both profiles. FreeSWITCH trunk rev. 15180 running on Debian 5.0.2. See anything interesting? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Media got stuck after attended transfer...
Of course, I was listening to my A.M radio the other day and they said that there was this new invention called the Internet that would let people send documents to each other electronically. Maybe you should look into that. Next thing you know they'll come up with telephones that people don't have to plug into the wall and can take with them in the car. ;) yes, in that galaxy far far away :P -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] Media got stuck after attended transfer...
hi, any clue when can t38 be added? T. On Thu, Oct 15, 2009 at 3:57 PM, Anthony Minessale anthony.miness...@gmail.com wrote: This is a known limitation until we add actual t38 support to the project. On Wed, Oct 14, 2009 at 6:56 PM, Klaus Hochlehnert maili...@kh-dev.dewrote: Hi, sometimes I have the problem that after doing an attended transfer the media got stuck in FS. Meaning the call goes through, but I don’t hear anything and the caller still hears music. Now I found out that setting the sofia parameter media-option=resume-media-on-hold helps here. But after setting this parameter I always get the error “Codec PROXY PASS-THROUGH encoder error!” when using t38modem with proxy_media=true. So I stuck here. I can either use fax or do attended transfers. Does anyone have a solution for this? Thanks, Klaus ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP 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 freeswitch-users-l...@metik.comwrote: 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 sort of interop issue with the carrier involved that is wrecking havoc with your particular application? -metik - Original Message - *From:* Tihomir Culjaga tculj...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tuesday, October 13, 2009 3:24 PM *Subject:* 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 setting the minimum number of digits you will send in 1st message. I wish you luck... T. On Tue, Oct 13, 2009 at 9:03 PM, Metik freeswitch-users-l...@metik.comwrote: 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 anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tuesday, October 13, 2009 2:01 PM *Subject:* 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 tculj...@gmail.comwrote: 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 oderm...@googlemail.com 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 see if it is supported or not. any hint? 2009/10/13 Anthony Minessale anthony.miness...@gmail.com: 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 oderm...@googlemail.com wrote: 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 digit-by-digit, but what about the fs-side? thanks and kind regards dennis ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.commsn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.orgsip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK
Re: [Freeswitch-users] Fwd: mod_opal - call charged before H.225 connect
2009/10/14 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-14 08:59 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: try sow start on h323 channel, there is a bug in faststart, i will fix it later. there are few things, 1. capability PCMU/PCMA needs to be inverted 2. when you place outgoing calls SIP_user = FS = H323_endpoint FS corediumps: what is the data format for bridge? Is that correct = data=h323/1...@10.1.1.1 ? 3. when you place incoming calls H323_endpoint = FS = SIPUser, the call goes through but there is no audio. After H.225 connect and TCS/MSD, FS stops sending RTP back to te originator. 4. when you place incoming calls H323_endpoint = FS = SIPUser and you hangup from SIPUser side, the call is not released on H323 side... switch_core_state_machine complains about a wrong thread. i will send you logs for: 1. slow start 2. slow start early h245 3. fast start tunneling true 4. faststart tunneling false 5. faststart with early h245 tunneling true 6. faststart with early h245 tunneling false do you need a tcpdump for every scenario as well ? T. ___ 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] Fwd: mod_opal - call charged before H.225 connect
2009/10/14 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: i need trace level 4 from mod_h323 and debug log of entire call, tcpdump may be needed later, i have no way to test it on this time, i do it later. Ok, will generate this logs... hope you will recover soon. T. ___ 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] Fwd: mod_opal - call charged before H.225 connect
On Wed, Oct 14, 2009 at 10:16 AM, Tihomir Culjaga tculj...@gmail.comwrote: 2009/10/14 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: i need trace level 4 from mod_h323 and debug log of entire call, tcpdump may be needed later, i have no way to test it on this time, i do it later. Here are the logs: call flow is H323_endpoint = FS = SIP_user slow start= http://pastebin.freeswitch.org/10693 slow start w eH245= http://pastebin.freeswitch.org/10694 fastStart w tunneling true = http://pastebin.freeswitch.org/10701 fastStart w tunneling false= http://pastebin.freeswitch.org/10702 fastStart w tunneling true eH245= http://pastebin.freeswitch.org/10699 fastStart w tunneling false eH245= http://pastebin.freeswitch.org/10700 In all cases, there is no audio on SIP User side... i see the SIP Phone (terminator) sending RTP to FS but FS is not forwarding it back to the originator. Also, i see originator (H323) sending RTP to FS but FS doesn't forward it to the terminator (SIP). call flow is SIP_user = FS = H323_endpoint is failing .. coredumped http://pastebin.freeswitch.org/10703 T. hope you will recover soon. T. ___ 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_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 br...@freeswitch.org 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 not fragment the community if possible. /b On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote: hi, finally i compiled it right ... had a stupid issue with ekiga and wrong ptlib in place... anyhow, i loaded the module and will continue the tests tomorrow ...first thing i arrive in my office :P ___ 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] mod_opal - call charged before H.225 connect
On Tue, Oct 13, 2009 at 8:31 AM, Brian West br...@freeswitch.org 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. ___ 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] 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 ___ 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] 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 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 tculj...@gmail.comwrote: 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 ___ 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 -- Srinivasula Reddy K ___ 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] 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 bottle...@icf.org.ru
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.
SDthat will make life easier.
SD
SD2009/10/13 Brian West br...@freeswitch.org
SD
SD Does anyone see a problem with hosting mod_h323 in our SVN? I would
SD like to centralize everything we can to reuse our issue tracking
SD resources and not fragment the community if possible.
SD
SD /b
SD
SD On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
SD
SD hi,
SD
SD finally i compiled it right ... had a stupid issue with ekiga and
SD wrong ptlib in place...
SD
SD anyhow, i loaded the module and will continue the tests
SD tomorrow ...first thing i arrive in my office :P
SD
SD
SD
SD
SD ___
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http://lists.freeswitch.org/mailman/options/freeswitch-users
SD http://www.freeswitch.org
SD
SD
C уважением With Best Regards
Георгиевский Юрий.Georgiewskiy Yuriy
+7 4872 711666+7 4872 711666
факс +7 4872 711143 fax +7 4872 711143
Компания ООО Ай Ти Сервис IT Service Ltd
http://nkoort.ru http://nkoort.ru
JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru
YG129-RIPEYG129-RIPE
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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 having the wireshark file, any help? thanks srinivas On Tue, Oct 13, 2009 at 4:02 PM, Tihomir Culjaga tculj...@gmail.comwrote: 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 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 tculj...@gmail.comwrote: 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 ___ 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 -- Srinivasula Reddy K ___ 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 -- Srinivasula Reddy K ___ 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] mod_opal - call charged before H.225 connect
2009/10/13 Georgiewskiy Yuriy bottle...@icf.org.ru 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. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP 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 oderm...@googlemail.com 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 see if it is supported or not. any hint? 2009/10/13 Anthony Minessale anthony.miness...@gmail.com: 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 oderm...@googlemail.com wrote: 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 digit-by-digit, but what about the fs-side? thanks and kind regards dennis ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] 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 setting the minimum number of digits you will send in 1st message. I wish you luck... T. On Tue, Oct 13, 2009 at 9:03 PM, Metik freeswitch-users-l...@metik.comwrote: 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 anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tuesday, October 13, 2009 2:01 PM *Subject:* 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 tculj...@gmail.comwrote: 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 oderm...@googlemail.com 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 see if it is supported or not. any hint? 2009/10/13 Anthony Minessale anthony.miness...@gmail.com: 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 oderm...@googlemail.com wrote: 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 digit-by-digit, but what about the fs-side? thanks and kind regards dennis ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.commsn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.orgsip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 -- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org
Re: [Freeswitch-users] mod_opal - call charged before H.225 connect
2009/10/12 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TCHi Yuriy, TC TCcan you share what you have so far, I'm sure we can help with RTP part... ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but seems it work, but should be buggy, to build need libpt 2.6.5 and h323plus cvs version, i test it now on fs 1.0.4. TC TCT. TC TC2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru got it and building it right now... T. ___ 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_opal - call charged before H.225 connect
hi, can't make it... subZero:~/freeswitch-trunk$ make mod_h323 making all mod_h323 Compiling mod_h323.cpp... quiet_libtool: compile: g++ -g -ggdb -I/usr/local/include/ptlib -I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/home/tculjaga/freeswitch-trunk/src/include -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_h323.cpp -fPIC -DPIC -o .libs/mod_h323.o In file included from /usr/local/include/openh323/h323.h:493, from mod_h323.h:8, from mod_h323.cpp:3: /usr/local/include/openh323/h323ep.h: In member function ‘virtual void NATFactoryStartup::OnShutdown()’: /usr/local/include/openh323/h323ep.h:2731: error: ‘NatFactory’ has not been declared make[4]: *** [mod_h323.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_h323-all] Error 1 make[1]: *** [mod_h323] Error 2 make: *** [mod_h323] Error 2 what exact ptlib and h323plus versions did you use? .. can you send us a link so we can use the exact ones. T. 2009/10/12 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-12 09:43 -0500, Brian West wrote freeswitch-us...@lists.freeswit...: BWWe can host this in our SVN if you wish? If in fs svn i think yes. But i think may be little time later? i don't known is it builds on trunk because i develop it on 1.0.4. BW/b BW BWOn Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote: BW BW ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but seems BW it work, but should be buggy, BW to build need libpt 2.6.5 and h323plus cvs version, i test it now on fs BW 1.0.4. BW C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE ___ 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] how to match '#' in XML dialplan ?
this is up to your phone # means address complete and you phone sends the number you dialed into an INVITE message. if you want to support FAC with # you should modify the phone's dialplan and make it expect more digits... for certain prefixes. T. On Sun, Oct 11, 2009 at 12:10 PM, Henry Huang red.rain.se...@gmail.comwrote: Daqiang: How do you make your IP phone not dial right after you press #? Usually the IP phone will dial the number already once you pushed # On Sun, Oct 11, 2009 at 10:45 AM, daqiang wang wangdq@gmail.comwrote: it's work . Thank you very much . 2009/10/11 Michael Collins m...@freeswitch.org Some characters need a backslash to match in a regular expression. However, # is not one of them. I think your regex is wrong: condition field=destination_number expression=^1#(d+)#(d+)$/ It should probably be: condition field=destination_number expression=^1#(\d+)#(\d+)$/ Note the backslashes in front of the d+ entries. \d means match a digit whereas a bare d means make a lowercase d character. Hope that helps. -MC P.S. - The * character does need to be escaped in regexes. See the default.xml dialplan file for some obvious examples. On Sat, Oct 10, 2009 at 6:24 AM, Milena testeado...@gmail.com wrote: escape character is '\'try condition field=destination_number expression=^1\#(d+)\#(d+)$/ 2009/10/10 daqiang wang wangdq@gmail.com hello every one : I want to match the # in XML dialplan , how to do ? example : 1## . how to do ? I do this : condition field=destination_number expression=^1#(d+)#(d+)$/ but it's not work ___ 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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] mod_opal - call charged before H.225 connect
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 customized. T. 2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-07 15:09 -0500, Brian West wrote freeswitch-us...@lists.freeswit...: opal have addition abstraction layer called opalmgr, and it implementation is not so good in this case, for example to implemet pre_answer in mod_opal i need patch libopal, because there is no way to send progress inicator throuch opalmgr. and there is many another issues like this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is my work on mod_opal before i start moving to h323plus, may be this help somebody there. BW From what I have been told h323plus is a based/fork of OpenH323 which BWOPAL is just a continuation of OpenH323. So why not support the BWdevelopers of OPAL/OpenH323 ? BW BW/b BW BWOn Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote: BW BW We are developing module to handle h323 proto now, we try to use BW mod_opal and try improve it, but no luck, BW there is many issues in libopal, and finaly we now move to h323plus BW library. BW BW BW___ BWFreeSWITCH-users mailing list BWFreeSWITCH-users@lists.freeswitch.org BWhttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users BWUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users BWhttp://www.freeswitch.org BW C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE ___ 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] mod_opal - call charged before H.225 connect
yep, you made the point :P T. 2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-08 16:20 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TCHi Yuriy, did you manage to do something with H323plus and FS ? i already doing it, but now it not in usable state. TCbtw: have you checked Objective OpenH323 TChttp://www.obj-sys.com/telephony-objective.shtml ? TCThis looks better to me as it is lighter and can be easily customized. i see this library later in asterisk module, h323plus is a successor of opanh323, i use it many yars and i think it more complete mature and stable than objective systems stack, and finally h323plus not depend in its development from some kinde of Objective System Inc/any other xxx Inc. TC TC2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru TC TC On 2009-10-07 15:09 -0500, Brian West wrote TC freeswitch-us...@lists.freeswit...: TC TC opal have addition abstraction layer called opalmgr, and it implementation TC is not so good in TC this case, for example to implemet pre_answer in mod_opal i need patch TC libopal, because TC there is no way to send progress inicator throuch opalmgr. and there is TC many another issues like TC this in that layer. ftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is TC my work on mod_opal before TC i start moving to h323plus, may be this help somebody there. TC TC BW From what I have been told h323plus is a based/fork of OpenH323 which TC BWOPAL is just a continuation of OpenH323. So why not support the TC BWdevelopers of OPAL/OpenH323 ? TC BW TC BW/b TC BW TC BWOn Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote: TC BW TC BW We are developing module to handle h323 proto now, we try to use TC BW mod_opal and try improve it, but no luck, TC BW there is many issues in libopal, and finaly we now move to h323plus TC BW library. TC BW TC BW TC BW___ TC BWFreeSWITCH-users mailing list TC BWFreeSWITCH-users@lists.freeswitch.org TC BWhttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users TC BWUNSUBSCRIBE: TC http://lists.freeswitch.org/mailman/options/freeswitch-users TC BWhttp://www.freeswitch.org TC BW TC TC C уважением With Best Regards TC Георгиевский Юрий.Georgiewskiy Yuriy TC +7 4872 711666+7 4872 711666 TC факс +7 4872 711143 fax +7 4872 711143 TC Компания ООО Ай Ти Сервис IT Service Ltd TC http://nkoort.ru http://nkoort.ru TC JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru TC YG129-RIPEYG129-RIPE TC TC ___ TC FreeSWITCH-users mailing list TC FreeSWITCH-users@lists.freeswitch.org TC http://lists.freeswitch.org/mailman/listinfo/freeswitch-users TC UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users TC http://www.freeswitch.org TC TC TC C уважением With Best Regards Георгиевский Юрий.Georgiewskiy Yuriy +7 4872 711666+7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО Ай Ти Сервис IT Service Ltd http://nkoort.ru http://nkoort.ru JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru YG129-RIPEYG129-RIPE ___ 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] mod_opal - call charged before H.225 connect
Hi Yuriy, can you share what you have so far, I'm sure we can help with RTP part... T. 2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote freeswitch-us...@lists.freesw...: TzHi, Tz Tz2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru: Tz On 2009-10-08 10:43 -0500, Anthony Minessale wrote freeswitch-us...@lists.f...: Tz Tz AMIf you are going to make that alternate module are you going to host it in Tz AMthe FS tree along side mod_opal? Tz Tz Yes, but then it be useful, now i have working only signaling part and some Tz kinde of not working rtp part :) Tz TzIf you dont use fs rtp stack, its unlikely that it will be accepted Tzinto the tree. Tz Tz Tz AMalso if were working on mod_opal why did you not try to involve us and the Tz AMopal team? Tz Tz Because i made patches for libopal, one is a bugfix in rtp part, there is a race condition Tz in inicialisation in jitter buffer, another patch implements method to send progress indicator, Tz and i don't wont spent my time to incorporate this changes into libopal. Tz TzThats bad. TzAny bugfixes from fs, goes to upstream on any of the used libraries. TzYou should be doing the same. TzAnd Opal developers, will either include or refuse your patches. If Tzthey refuse it, they will give you the reason. i make this fix only to freeze my current mod_opal work on working state, while it now work for me i work on my new implementation of h323 proto for fs, i think opal developers will fix this rtp bug himself becouse it crashes and make library unuseful. Tz Tz without this changes Tz my work on mod_opal in freeswitch don't useful at all, i provide link to my work with all Tz patches, if somebody wont incorporate it in libopal tree and fs - go on, but i think Tz better and more elegant make new module based on h323plus. Tz TzIf you dont publish your changes, all those you are trying to achieve, Tzwont happen. Tz Tz Tz AMHow far away from what is in tree are these patches you have? Tz AM Tz AM2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru Tz AM Tz AM On 2009-10-07 15:09 -0500, Brian West wrote Tz AM freeswitch-us...@lists.freeswit...: Tz AM Tz AM opal have addition abstraction layer called opalmgr, and it implementation Tz AM is not so good in Tz AM this case, for example to implemet pre_answer in mod_opal i need patch Tz AM libopal, because Tz TzThe patch you have in there, adds a method to the OpalCall, it does Tznot touch any parts of OpalManager Tzso, i dont understand why opalmanager would be the cause of your pain? Tz Tz AM there is no way to send progress inicator throuch opalmgr. and there is Tz AM many another issues like Tz AM this in that layer. Tz TzPlease point me to the issues you have in opal, their bug reports , traces etc. TzI dont think any of the opal people has psychic abilities to detect Tz-your- problems Tzand solve them. Tz Tzftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is Tz AM my work on mod_opal before Tz AM i start moving to h323plus, may be this help somebody there. Tz AM Tz AM BW From what I have been told h323plus is a based/fork of OpenH323 which Tz AM BWOPAL is just a continuation of OpenH323. So why not support the Tz AM BWdevelopers of OPAL/OpenH323 ? Tz AM BW Tz AM BW/b Tz AM BW Tz AM BWOn Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote: Tz AM BW Tz AM BW We are developing module to handle h323 proto now, we try to use Tz AM BW mod_opal and try improve it, but no luck, Tz AM BW there is many issues in libopal, and finaly we now move to h323plus Tz AM BW library. Tz TzDid any of you try to report those issues? Tz TzRegards Tz Tz/tyn Tz Tz AM BW Tz AM BW Tz AM BW___ Tz AM BWFreeSWITCH-users mailing list Tz AM BWFreeSWITCH-users@lists.freeswitch.org Tz AM BWhttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users Tz AM BWUNSUBSCRIBE: Tz AM http://lists.freeswitch.org/mailman/options/freeswitch-users Tz AM BWhttp://www.freeswitch.org Tz AM BW Tz AM Tz AM C уважением With Best Regards Tz AM Георгиевский Юрий.Georgiewskiy Yuriy Tz AM +7 4872 711666+7 4872 711666 Tz AM факс +7 4872 711143 fax +7 4872 711143 Tz AM Компания ООО Ай Ти Сервис IT Service Ltd Tz AM http://nkoort.ru http://nkoort.ru Tz AM JID: ghh...@jabber.tula-ix.net.ru JID: ghh...@jabber.tula-ix.net.ru Tz AM YG129-RIPEYG129-RIPE Tz AM Tz AM ___ Tz AM FreeSWITCH-users mailing list Tz AM FreeSWITCH-users@lists.freeswitch.org Tz AM http://lists.freeswitch.org/mailman/listinfo/freeswitch-users Tz AM UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users Tz AM http://www.freeswitch.org Tz AM Tz AM Tz AM Tz AM Tz AM Tz Tz C уважением
Re: [Freeswitch-users] mod_opal - call charged before H.225 connect
k 2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...: TCHi Yuriy, TC TCcan you share what you have so far, I'm sure we can help with RTP part... I think there is a few days and i make it work, after this i start to test and share it. TC TCT. TC TC2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru TC TC On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote TC freeswitch-us...@lists.freesw...: TC TC TzHi, TC Tz TC Tz2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru: TC Tz On 2009-10-08 10:43 -0500, Anthony Minessale wrote TC freeswitch-us...@lists.f...: TC Tz TC Tz AMIf you are going to make that alternate module are you going to TC host it in TC Tz AMthe FS tree along side mod_opal? TC Tz TC Tz Yes, but then it be useful, now i have working only signaling part and TC some TC Tz kinde of not working rtp part :) TC Tz TC TzIf you dont use fs rtp stack, its unlikely that it will be accepted TC Tzinto the tree. TC Tz TC Tz TC Tz AMalso if were working on mod_opal why did you not try to involve us TC and the TC Tz AMopal team? TC Tz TC Tz Because i made patches for libopal, one is a bugfix in rtp part, there TC is a race condition TC Tz in inicialisation in jitter buffer, another patch implements method to TC send progress indicator, TC Tz and i don't wont spent my time to incorporate this changes into TC libopal. TC Tz TC TzThats bad. TC TzAny bugfixes from fs, goes to upstream on any of the used libraries. TC TzYou should be doing the same. TC TzAnd Opal developers, will either include or refuse your patches. If TC Tzthey refuse it, they will give you the reason. TC TC i make this fix only to freeze my current mod_opal work on working state, TC while it now work for me i work on TC my new implementation of h323 proto for fs, i think opal developers will TC fix this rtp bug himself becouse TC it crashes and make library unuseful. TC TC Tz TC Tz without this changes TC Tz my work on mod_opal in freeswitch don't useful at all, i provide link TC to my work with all TC Tz patches, if somebody wont incorporate it in libopal tree and fs - go TC on, but i think TC Tz better and more elegant make new module based on h323plus. TC Tz TC TzIf you dont publish your changes, all those you are trying to achieve, TC Tzwont happen. TC Tz TC Tz TC Tz AMHow far away from what is in tree are these patches you have? TC Tz AM TC Tz AM2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru TC Tz AM TC Tz AM On 2009-10-07 15:09 -0500, Brian West wrote TC Tz AM freeswitch-us...@lists.freeswit...: TC Tz AM TC Tz AM opal have addition abstraction layer called opalmgr, and it TC implementation TC Tz AM is not so good in TC Tz AM this case, for example to implemet pre_answer in mod_opal i need TC patch TC Tz AM libopal, because TC Tz TC TzThe patch you have in there, adds a method to the OpalCall, it does TC Tznot touch any parts of OpalManager TC Tzso, i dont understand why opalmanager would be the cause of your pain? TC Tz TC Tz AM there is no way to send progress inicator throuch opalmgr. and TC there is TC Tz AM many another issues like TC Tz AM this in that layer. TC Tz TC TzPlease point me to the issues you have in opal, their bug reports , TC traces etc. TC TzI dont think any of the opal people has psychic abilities to detect TC Tz-your- problems TC Tzand solve them. TC Tz TC Tzftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is TC Tz AM my work on mod_opal before TC Tz AM i start moving to h323plus, may be this help somebody there. TC Tz AM TC Tz AM BW From what I have been told h323plus is a based/fork of TC OpenH323 which TC Tz AM BWOPAL is just a continuation of OpenH323. So why not support TC the TC Tz AM BWdevelopers of OPAL/OpenH323 ? TC Tz AM BW TC Tz AM BW/b TC Tz AM BW TC Tz AM BWOn Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote: TC Tz AM BW TC Tz AM BW We are developing module to handle h323 proto now, we try to TC use TC Tz AM BW mod_opal and try improve it, but no luck, TC Tz AM BW there is many issues in libopal, and finaly we now move to TC h323plus TC Tz AM BW library. TC Tz TC TzDid any of you try to report those issues? TC Tz TC TzRegards TC Tz TC Tz/tyn TC Tz TC Tz AM BW TC Tz AM BW TC Tz AM BW___ TC Tz AM BWFreeSWITCH-users mailing list TC Tz AM BWFreeSWITCH-users@lists.freeswitch.org TC Tz AM BW http://lists.freeswitch.org/mailman/listinfo/freeswitch-users TC Tz AM BWUNSUBSCRIBE: TC Tz AM http://lists.freeswitch.org/mailman/options/freeswitch-users TC Tz AM BWhttp://www.freeswitch.org TC Tz AM BW TC Tz AM TC Tz AM C уважением With Best Regards TC Tz AM Георгиевский Юрий.Georgiewskiy Yuriy TC Tz AM +7 4872 711666+7 4872 711666 TC Tz AM факс +7 4872 711143 fax +7
Re: [Freeswitch-users] mod_opal - call charged before H.225 connect
Anthony,
of course, nobody wants to start anything... we are all here to help making
FS a better product.
so, regarding the founding for mod_opal ... what is the amount you need?
Tihomir.
On Wed, Oct 7, 2009 at 2:58 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
I didn't mean to start anything. I'm just saying we work very long hours
and barely get anybody asking about h.323.
I wanted to support it and that's why we took up a collection to get
funding for mod_opal but when only 1 donor showed any interest we were
forced to proceed in our spare time which is very limited.
The developers of opal are not part of our project and they need financial
compensation to be motivated to work on it. Its not even related to me its
only fair that an outside developer who makes his living as a consultant
would want money to integrate his work into our project.
Like I said, I will do my best to point your issue to the opal devs but I
cannot force them to work on it.
On Tue, Oct 6, 2009 at 7:22 PM, Diego Viola diego.vi...@gmail.com wrote:
Yeah I understand your point of view, but saying I want a H.323 module
or I want a Ferrari wont magically make it happen.
We need to work on it ourselves or pay to the people that knows how to do
it to do it for us.
There is no other way I think.
Diego
On Tue, Oct 6, 2009 at 11:41 PM, Tihomir Culjaga tculj...@gmail.comwrote:
Diego,
what i'm pointing here is the situation where you have a great product
that lacks in one of most common protocol. It is true H323 is going to
disappear (eventually), it is true that the community prefers SIP/IAX
instead ... but the reality still remains. H323 is going to be used for
quite a long time to exchange a lot of traffic while FS will be left aside.
Today, when you setup an IP peering interconnection 80% of carriers will
prefer H323.
Of course, developing something costs time (and we all know what time
stands for...) and as i said, i understand the financial point of view and i
really understand if nobody is going to work on that, but let's face it FS
doesn't have any usable module to reliably handle H323 protocol.
said that, i don't intend to offend anyone... just facing the reality.
regarding the h323 module, we don't have any issue fixing the existing or
developing a new one... but before we go developing something it is always
better check if the thing you want already exists in an usable state or
not... that's what i did today.
So, I'm interested in a reliable module handling H323v4... anyone else?
T.
On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola diego.vi...@gmail.comwrote:
Instead of complaining and demanding things for free, people should
start to put their money where their mouth is.
Diego
On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga tculj...@gmail.comwrote:
hi Anthony,
it is somewhere here:
switch_status_t
FSConnection::receive_message(switch_core_session_message_t *msg)
anyhow, i will open an issue jira of course.
I understand your financial point of view, but anyhow while the entire
world is wants sip and trying to move to sip, the reality is quite
different. The majority of voice traffic exchanged via IP is still H323.
This means a working SIP - H323 interworking is really needed... pity
nobody
wants/has time to work in this direction to produce a decent mod_h323.
T.
On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
pcap is not as useful as FS console log on debug with:
sofia profile internal siptrace on
you should be reporting issues to jira under mod_opal not to the
mailing list.
http://jira.freeswitch.org
FYI
There is little financial support from the community for h323 which
prevents the mod_opal from getting much attention.
We actually have to contract the author of opal to help with these
issues including the original writing of the module that he did with very
little funding and nobody ever wants to pay him to improve it.
That does not mean your issue will not be addressed but there is no
promise how fast it will be.
On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga
tculj...@gmail.comwrote:
hello guys,
i was playing with mod_opal to see if i can make it working ... well
it seems SIP-H323 interworking is not tuned at all.
I have a call from a registered sip user (1001) to PSTN via mod_opal
include
extension name=EMERGENCY
condition field=destination_number
expression=^0(112|9[23456])$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=bridge data=opal/h323:0...@${ncx_ip}/
/condition
/extension
extension name=SPECIAL_SERVICES
condition field=destination_number
expression=^0(9[01789]\d{3,4
Re: [Freeswitch-users] mod_opal - call charged before H.225 connect
On Wed, Oct 7, 2009 at 2:40 PM, Claudiu Filip clau...@globtel.ro wrote:
Hi Tihomir,
I've done some tests to see how suitable is freeswitch as a
SIP/H323 translator and you are right about the fact that H323
'alert+open logical channel' will generate a SIP '200 OK'. I was
able to fix that with a couple of changes in mod_opal.cpp, however
some things were changed on mod_sofia in the latest svn. (on this
particular issue, open_logical_channel is processed BEFORE the
alerting, so the call is in SetupPhase when the proc
OnOpenMediaStream is triggered)
yep, thats correct ... i was just wondering why it hangs in SetUpPhase
2009-10-07 16:50:11.690451 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[n03f409711]-EPh323[localhost/3263],OpalRTPMediaStream-Source-G.711-ALaw-64k
2009-10-07 16:50:11.690451 [INFO] mod_opal.cpp:1283 opal/
h323:05492...@10.4.4.254 h323%3a05492...@10.4.4.254 initialise
opal/h323:05492...@10.4.4.254read audio codec G.711-ALaw-64k for connection
FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 16:50:11.690451 [DEBUG] mod_opal.cpp:1313 Set read audio codec to
G.711-ALaw-64k for connection FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 16:50:11.691525 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[n03f409711]-EPlocal[1],FSMediaStream-Sink-G.711-ALaw-64k
*2009-10-07 16:50:11.691525 [CONSOLE] mod_opal.cpp:852 SetUpPhase =
GetPhase() = '1'*
2009-10-07 16:50:11.691525 [DEBUG] connection.cxx:561 Opened sink stream
n03f409711_1 with format G.711-ALaw-64k
2009-10-07 16:50:11.691525 [DEBUG] patch.cxx:341 Created Sink:
format=G.711-ALaw-64k
2009-10-07 16:50:11.691525 [DEBUG] mediastrm.cxx:666 RTP data size cannot be
changed to 160, fixed at 2048
2009-10-07 16:50:11.691525 [DEBUG] patch.cxx:179 Added direct media stream
sink FSMediaStream-Sink-G.711-ALaw-64k
this is the original code, and it never triggers eraly_media as never
reaches AlertingPhase.
if (GetMediaStream(stream.GetSessionID(), stream.IsSink()) != NULL) {
// Have open media in both directions.
if (GetPhase() == AlertingPhase) {
switch_channel_mark_pre_answered(m_fsChannel);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, LOG
==\t Alerting = GetPhase() = '%d'\n,GetPhase());
} else if (GetPhase() ReleasingPhase) {
switch_channel_mark_answered(m_fsChannel);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, LOG
==\t GetPhase() = '%d'\n,GetPhase());
}
}
I tried this, it works for early media but i still need to open a full media
path and say the call actually connected
if (GetMediaStream(stream.GetSessionID(), stream.IsSink()) != NULL) {
// Have open media in both directions.
if (GetPhase() ConnectedPhase) {
switch_channel_mark_pre_answered(m_fsChannel);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, EARLY
MEDIA = GetPhase() = '%d'\n,GetPhase());
} else if (GetPhase() ReleasingPhase) {
switch_channel_mark_answered(m_fsChannel);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CONSOLE, FULL
MEDIA = GetPhase() = '%d'\n,GetPhase());
}
}
this is when i'm dong early_media:
2009-10-07 17:45:26.788082 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[c8dce50981]-EPh323[localhost/26906],OpalRTPMediaStream-Source-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [INFO] mod_opal.cpp:1279 opal/
h323:05492...@10.4.4.254 h323%3a05492...@10.4.4.254 initialise
opal/h323:05492...@10.4.4.254read audio codec G.711-ALaw-64k for connection
FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [DEBUG] mod_opal.cpp:1309 Set read audio codec to
G.711-ALaw-64k for connection FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [DEBUG] manager.cxx:718 OnOpenMediaStream
Call[c8dce50981]-EPlocal[1],FSMediaStream-Sink-G.711-ALaw-64k
2009-10-07 17:45:26.789158 [NOTICE] mod_opal.cpp:887 Pre-Answer opal/
h323:05492...@10.4.4.254 h323%3a05492...@10.4.4.254!
2009-10-07 17:45:26.789158 [DEBUG] switch_channel.c:1822 Send signal
sofia/internal/1...@10.4.62.7 [BREAK]
*2009-10-07 17:45:26.789158 [CONSOLE] mod_opal.cpp:888 EARLY MEDIA =
GetPhase() = '1'*
2009-10-07 17:45:26.789158 [DEBUG] connection.cxx:561 Opened sink stream
c8dce50981_1 with format G.711-ALaw-64k
2009-10-07 17:45:26.789158 [DEBUG] patch.cxx:341 Created Sink:
format=G.711-ALaw-64k
2009-10-07 17:45:26.790236 [DEBUG] switch_ivr_originate.c:2154
sofia/internal/1...@10.4.62.7 receive message [PROGRESS]
2009-10-07 17:45:26.790236 [INFO] switch_ivr_originate.c:2154 Sending early
media
2009-10-07 17:45:26.790236 [DEBUG] sofia_glue.c:2329 AUDIO RTP
[sofia/internal/1...@10.4.62.7] 10.4.62.7 port 19594 - 10.4.62.89 port 5004
codec: 8 ms: 20
2009-10-07 17:45:26.790236 [DEBUG] switch_rtp.c:1155 Starting timer [soft]
160 bytes per 20ms
2009-10-07 17:45:26.790236 [DEBUG] mediastrm.cxx:666 RTP data size cannot be
changed to 160,
[Freeswitch-users] mod_opal - call charged before H.225 connect
hello guys,
i was playing with mod_opal to see if i can make it working ... well it
seems SIP-H323 interworking is not tuned at all.
I have a call from a registered sip user (1001) to PSTN via mod_opal
include
extension name=EMERGENCY
condition field=destination_number expression=^0(112|9[23456])$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=bridge data=opal/h323:0...@${ncx_ip}/
/condition
/extension
extension name=SPECIAL_SERVICES
condition field=destination_number expression=^0(9[01789]\d{3,4})$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=bridge data=opal/h323:0...@${ncx_ip}/
/condition
/extension
extension name=ENYTHING_ELSE
condition field=destination_number
expression=^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=bridge data=opal/h323:0...@${ncx_ip}/
/condition
/extension
/include
One of the many issues i sow is that FS connects the call on SIP leg before
it actually receives H.225 connect from H323 leg... as it is configured to
send 200 OK on the 1st H.225 message containing a FastStart element/OLC.
Attached is the tcpdump i took on FS machine... just use this filter: h225
or h245 or q931 or sip
Also, you can check the attac CDR this is an unanswered call i placed to
PSTN and FS billed it 23 seconds.
Can anyone tell where i can do correct SIP - H323 message mappings to avoid
this?
T.
r...@subzero:/usr/local/freeswitch/log/xml_cdr# cat a_9db67edc-b29a-11de-bcf9-1fb6bf4c98f1.cdr.xml
?xml version=1.0?
cdr
variables
sip_received_ip10.4.62.89/sip_received_ip
sip_received_port5060/sip_received_port
sip_via_protocoludp/sip_via_protocol
sip_authorizedtrue/sip_authorized
sip_number_alias1001/sip_number_alias
sip_auth_username1001/sip_auth_username
sip_auth_realm10.4.62.7/sip_auth_realm
number_alias1001/number_alias
user_name1001/user_name
domain_name10.4.62.7/domain_name
toll_allowdomestic,international,local/toll_allow
accountcode1001/accountcode
user_contextdefault/user_context
effective_caller_id_nameExtension%201001/effective_caller_id_name
outbound_caller_id_nameFreeSWITCH/outbound_caller_id_name
outbound_caller_id_number00/outbound_caller_id_number
callgrouptechsupport/callgroup
record_stereotrue/record_stereo
default_gatewayexample.com/default_gateway
default_areacode918/default_areacode
transfer_fallback_extensionoperator/transfer_fallback_extension
sip_from_user1001/sip_from_user
sip_from_uri1001%4010.4.62.7/sip_from_uri
sip_from_host10.4.62.7/sip_from_host
sip_from_user_stripped1001/sip_from_user_stripped
sip_from_tag-1058464acb9540-4_F10.4.62.89/sip_from_tag
sofia_profile_nameinternal/sofia_profile_name
sip_req_user05492122/sip_req_user
sip_req_uri05492122%4010.4.62.7/sip_req_uri
sip_req_host10.4.62.7/sip_req_host
sip_to_user05492122/sip_to_user
sip_to_uri05492122%4010.4.62.7/sip_to_uri
sip_to_host10.4.62.7/sip_to_host
sip_contact_paramstransport%3Dudp/sip_contact_params
sip_contact_user051494197/sip_contact_user
sip_contact_uri051494197%4010.4.62.89/sip_contact_uri
sip_contact_host10.4.62.89/sip_contact_host
channel_namesofia/internal/1001%4010.4.62.7/channel_name
sip_call_id15_344db6d7ed3814aceda20_I%4010.4.62.89/sip_call_id
sip_via_host10.4.62.89/sip_via_host
max_forwards70/max_forwards
presence_id1001%4010.4.62.7/presence_id
switch_r_sdpv%3D0%0D%0Ao%3Dsip%3A051494197%4010.4.62.89%201%2022%20IN%20IP4%2010.4.62.89%0D%0As%3Dsip%3A051494197%4010.4.62.89%0D%0Ac%3DIN%20IP4%2010.4.62.89%0D%0At%3D0%200%0D%0Am%3Daudio%205004%20RTP/AVP%20101%208%2018%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000/1%0D%0Aa%3Drtpmap%3A8%20PCMA/8000/1%0D%0Aa%3Drtpmap%3A18%20G729/8000/1%0D%0Aa%3Dfmtp%3A18%20annexb%3Dno%0D%0A/switch_r_sdp
remote_media_ip10.4.62.89/remote_media_ip
remote_media_port5004/remote_media_port
read_codecPCMA/read_codec
read_rate8000/read_rate
write_codecPCMA/write_codec
write_rate8000/write_rate
effective_caller_id_number1001282122/effective_caller_id_number
NCX_IP10.4.4.254/NCX_IP
call_timeout30/call_timeout
hangup_after_bridgetrue/hangup_after_bridge
current_application_dataopal/h323%3A05492122%4010.4.4.254/current_application_data
Re: [Freeswitch-users] mod_opal - call charged before H.225 connect
hi Anthony,
it is somewhere here:
switch_status_t
FSConnection::receive_message(switch_core_session_message_t *msg)
anyhow, i will open an issue jira of course.
I understand your financial point of view, but anyhow while the entire world
is wants sip and trying to move to sip, the reality is quite different. The
majority of voice traffic exchanged via IP is still H323. This means a
working SIP - H323 interworking is really needed... pity nobody wants/has
time to work in this direction to produce a decent mod_h323.
T.
On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
pcap is not as useful as FS console log on debug with:
sofia profile internal siptrace on
you should be reporting issues to jira under mod_opal not to the mailing
list.
http://jira.freeswitch.org
FYI
There is little financial support from the community for h323 which
prevents the mod_opal from getting much attention.
We actually have to contract the author of opal to help with these issues
including the original writing of the module that he did with very little
funding and nobody ever wants to pay him to improve it.
That does not mean your issue will not be addressed but there is no promise
how fast it will be.
On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga tculj...@gmail.comwrote:
hello guys,
i was playing with mod_opal to see if i can make it working ... well it
seems SIP-H323 interworking is not tuned at all.
I have a call from a registered sip user (1001) to PSTN via mod_opal
include
extension name=EMERGENCY
condition field=destination_number expression=^0(112|9[23456])$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=bridge data=opal/h323:0...@${ncx_ip}/
/condition
/extension
extension name=SPECIAL_SERVICES
condition field=destination_number
expression=^0(9[01789]\d{3,4})$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=bridge data=opal/h323:0...@${ncx_ip}/
/condition
/extension
extension name=ENYTHING_ELSE
condition field=destination_number
expression=^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=bridge data=opal/h323:0...@${ncx_ip}/
/condition
/extension
/include
One of the many issues i sow is that FS connects the call on SIP leg
before it actually receives H.225 connect from H323 leg... as it is
configured to send 200 OK on the 1st H.225 message containing a FastStart
element/OLC.
Attached is the tcpdump i took on FS machine... just use this filter:
h225 or h245 or q931 or sip
Also, you can check the attac CDR this is an unanswered call i placed
to PSTN and FS billed it 23 seconds.
Can anyone tell where i can do correct SIP - H323 message mappings to
avoid this?
T.
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] mod_opal - call charged before H.225 connect
Diego,
what i'm pointing here is the situation where you have a great product that
lacks in one of most common protocol. It is true H323 is going to disappear
(eventually), it is true that the community prefers SIP/IAX instead ... but
the reality still remains. H323 is going to be used for quite a long time to
exchange a lot of traffic while FS will be left aside. Today, when you setup
an IP peering interconnection 80% of carriers will prefer H323.
Of course, developing something costs time (and we all know what time
stands for...) and as i said, i understand the financial point of view and i
really understand if nobody is going to work on that, but let's face it FS
doesn't have any usable module to reliably handle H323 protocol.
said that, i don't intend to offend anyone... just facing the reality.
regarding the h323 module, we don't have any issue fixing the existing or
developing a new one... but before we go developing something it is always
better check if the thing you want already exists in an usable state or
not... that's what i did today.
So, I'm interested in a reliable module handling H323v4... anyone else?
T.
On Tue, Oct 6, 2009 at 11:45 PM, Diego Viola diego.vi...@gmail.com wrote:
Instead of complaining and demanding things for free, people should start
to put their money where their mouth is.
Diego
On Tue, Oct 6, 2009 at 8:47 PM, Tihomir Culjaga tculj...@gmail.comwrote:
hi Anthony,
it is somewhere here:
switch_status_t
FSConnection::receive_message(switch_core_session_message_t *msg)
anyhow, i will open an issue jira of course.
I understand your financial point of view, but anyhow while the entire
world is wants sip and trying to move to sip, the reality is quite
different. The majority of voice traffic exchanged via IP is still H323.
This means a working SIP - H323 interworking is really needed... pity nobody
wants/has time to work in this direction to produce a decent mod_h323.
T.
On Tue, Oct 6, 2009 at 9:25 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
pcap is not as useful as FS console log on debug with:
sofia profile internal siptrace on
you should be reporting issues to jira under mod_opal not to the mailing
list.
http://jira.freeswitch.org
FYI
There is little financial support from the community for h323 which
prevents the mod_opal from getting much attention.
We actually have to contract the author of opal to help with these issues
including the original writing of the module that he did with very little
funding and nobody ever wants to pay him to improve it.
That does not mean your issue will not be addressed but there is no
promise how fast it will be.
On Tue, Oct 6, 2009 at 12:37 PM, Tihomir Culjaga tculj...@gmail.comwrote:
hello guys,
i was playing with mod_opal to see if i can make it working ... well it
seems SIP-H323 interworking is not tuned at all.
I have a call from a registered sip user (1001) to PSTN via mod_opal
include
extension name=EMERGENCY
condition field=destination_number
expression=^0(112|9[23456])$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=bridge data=opal/h323:0...@${ncx_ip}/
/condition
/extension
extension name=SPECIAL_SERVICES
condition field=destination_number
expression=^0(9[01789]\d{3,4})$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=bridge data=opal/h323:0...@${ncx_ip}/
/condition
/extension
extension name=ENYTHING_ELSE
condition field=destination_number
expression=^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$
action application=set
data=effective_caller_id_number=1001282122/
action application=set data=NCX_IP=10.4.4.254/
action application=set data=call_timeout=30/
action application=set data=hangup_after_bridge=true/
action application=bridge data=opal/h323:0...@${ncx_ip}/
/condition
/extension
/include
One of the many issues i sow is that FS connects the call on SIP leg
before it actually receives H.225 connect from H323 leg... as it is
configured to send 200 OK on the 1st H.225 message containing a FastStart
element/OLC.
Attached is the tcpdump i took on FS machine... just use this filter:
h225 or h245 or q931 or sip
Also, you can check the attac CDR this is an unanswered call i
placed to PSTN and FS billed it 23 seconds.
Can anyone tell where i can do correct SIP - H323 message mappings to
avoid this?
T.
___
FreeSWITCH-users mailing list
Re: [Freeswitch-users] mod_opal - call charged before H.225 connect
thanks for your e-mail, H323 is mainly used for trunking purpose, inter-carrier traffic exchange... it is not used to control IP phones :P well, believe me, I've heard enough of H323 that i'm sick of it :P What i can tell you comes from my own experience on daily activities i'm doing for living... Of course, there might be part of the world where H323 dispersed completely but over here in Europe things tend to stick on tradition :P Yep, you are right... the forum wants SIP and that's understandable... anyhow you might check this: http://www.dailypayload.com/content/3111 T. On Wed, Oct 7, 2009 at 12:58 AM, Jason White ja...@jasonjgw.net wrote: Tihomir Culjaga tculj...@gmail.com wrote: I understand your financial point of view, but anyhow while the entire world is wants sip and trying to move to sip, the reality is quite different. The majority of voice traffic exchanged via IP is still H323. Is there any evidence in support of the above assertion (e.g., survey results of VoIP traffic)? I've heard of H323 but I don't know anyone who uses it, or any phones that implement it. The lack of interest in this forum and the absence of financial support to improve the H323 support in FreeSWITCH suggest that the level of demand for this is quite low, relative to SIP. Of course, improvements are always welcome, so if you're interested in funding better H323 support, or helping with the module I'm sure the FreeSWITCH community would welcome your efforts. ___ 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] configure FS: config.status: error: cannot find input file: Makefile.in
it works, thx! T. On Mon, Oct 5, 2009 at 12:31 AM, Michael Jerris m...@jerris.com wrote: I updated the tiff lib to build better inline, try make tiff-reconf Mike On Oct 2, 2009, at 8:05 AM, Tihomir Culjaga wrote: hello, i just got the last trunk and tried to compile it on one of my development machines... Well configure fails on tiff-3.8.2 where it is unable to find Makefile.in ... Can someone advice? checking if g++ static flag -static works... yes checking if g++ supports -c -o file.o... yes checking if g++ supports -c -o file.o... (cached) yes checking whether the g++ linker (/usr/bin/ld) supports shared libraries... yes checking dynamic linker characteristics... GNU/Linux ld.so checking how to hardcode library paths into programs... immediate checking for OpenGL Utility library... no checking for GLUT library... no configure: creating ./config.status config.status: error: cannot find input file: Makefile.in tculj...@subzero:~/freeswitch-trunk/libs/tiff-3.8.2$ ls -l total 2224 -rw-r--r-- 1 tculjaga tculjaga 23741 2009-10-02 13:19 acinclude.m4 -rw-r--r-- 1 tculjaga tculjaga 316978 2009-10-02 13:28 aclocal.m4 -rwxr-xr-x 1 tculjaga tculjaga121 2009-10-02 13:19 autogen.sh -rw-r--r-- 1 tculjaga tculjaga 124047 2009-10-02 13:19 ChangeLog drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:28 config -rw-r--r-- 1 tculjaga tculjaga 103926 2009-10-02 14:00 config.log -rwxr-xr-x 1 tculjaga tculjaga 73065 2009-10-02 14:00 config.status -rwxr-xr-x 1 tculjaga tculjaga 740145 2009-10-02 13:28 configure -rw-r--r-- 1 tculjaga tculjaga 20492 2009-10-02 13:19 configure.ac -rwxr-xr-x 1 tculjaga tculjaga 56 2009-10-02 13:19 configure.gnu -rwxr-xr-x 1 tculjaga tculjaga 737794 2009-10-02 13:57 configure.lineno drwxr-xr-x 16 tculjaga tculjaga 4096 2009-10-02 13:19 contrib -rw-r--r-- 1 tculjaga tculjaga 1146 2009-10-02 13:19 COPYRIGHT -rw-r--r-- 1 tculjaga tculjaga 1570 2009-10-02 13:19 HOWTO-RELEASE drwxr-xr-x 5 tculjaga tculjaga 4096 2009-10-02 13:19 html drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:28 libtiff drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 m4 -rw-r--r-- 1 tculjaga tculjaga 1908 2009-10-02 13:19 Makefile.am -rw-r--r-- 1 tculjaga tculjaga 1724 2009-10-02 13:19 Makefile.vc drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 man -rw-r--r-- 1 tculjaga tculjaga 6270 2009-10-02 13:19 nmake.opt drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 port -rw-r--r-- 1 tculjaga tculjaga 2363 2009-10-02 13:19 README -rw-r--r-- 1 tculjaga tculjaga 9 2009-10-02 13:19 RELEASE-DATE -rw-r--r-- 1 tculjaga tculjaga 5893 2009-10-02 13:19 SConstruct drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 test -rw-r--r-- 1 tculjaga tculjaga433 2009-10-02 13:19 TODO drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 tools -rw-r--r-- 1 tculjaga tculjaga 6 2009-10-02 13:19 VERSION tculj...@subzero:~/freeswitch-trunk/libs/tiff-3.8.2$ tculj...@subzero:~/freeswitch-trunk/libs/tiff-3.8.2$ ___ 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] Detecting a fax
hi Mark,
This is an inbound call leg and media channel (so far) is open in reverse
direction only (application ringback). I'm afraid you have to answer the
call to be able to hear the fax tone.
T.
On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris m...@jerris.com wrote:
Fax tones are not played by the remote machine until after answer, the
tone_detect application starts a media bug that listens for the tone,
can you confirm the tone is happening at all. Maybe the issue here is
the timeout, try making that longer, or doing the tone_detect in
execute_on_answer
Mike
On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:
Thanks for the response Mike,
I read that page and this one (among others)
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
I'm still lost. This is an extract of my dialplan
extension name=Local
condition field=destination_number expression=^(10[01][0-9])
$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=set data=ringback=${au-ring}/
action application=fax_detect/
action application=tone_detect data=fax 1100 r +5000
transfer fax XML features /
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge data=user/${dialed_extensi...@$
{domain}/
I would assume that on detecting a fax, the dialplan 'fax' is called
in context features. This never happens.
When is the fax tone detected? Is it while the call is ringing or
can it be detected after the call is answered? My goal is to be able
to have the same extension for a voice and fax call. i assume that
the fax 'tones' are standardised and the ones on the wiki are correct?
Also, I guess this doesn't work with media bypass (which I don't
use).
Thanks!
On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris m...@jerris.com
wrote:
check out
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
Note, you can't just have tone_detect as your last iten in the
dialplan as the call will just get hung up.
Mike
On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
Hi
I was hoping someone could help me to setup the fax detection / tone
detection application.
I want to be able to transfer an incoming fax to a specific
extension.
In my default.xml file, I have the following (extracted):
extension name=1000
condition field=destination_number expression=^(10[01]
[0-9])
$
action application=fax_detect/
action application=tone_detect data=fax 1100 r +5000
transfer fax XML features /
I can't get the fax to be detected and transferred. Is there any
way
this can be done?
Thanks!
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Re: [Freeswitch-users] wav files compression
also, you can store files in PCMA/PCMU format and avoid transcoding at all... and as said disk space is cheap.. go get some... On Sat, Oct 3, 2009 at 7:07 PM, Diego Viola diego.vi...@gmail.com wrote: Why is not recommended? On Sat, Oct 3, 2009 at 2:52 PM, Brian West br...@freeswitch.org wrote: MP3 is NOT recommend and if WAV files are too large you can mosey on down to the local Best Buy and snag 1.5TB of disk for like $119 dollars. Disk is cheap. /b On Oct 3, 2009, at 1:44 AM, Keith Wood wrote: I am working on an implementation for managing thousands of IVR within an organization. Right now, I am storing all audio files in wav format, but it quickly become unmanagable because the size of these wav files ( 8 bits mono ) quickly consuming a lot of the disk space. Is there anyway I can store those audio files and still have high quality audio for IVR? I know mp3 is smaller but freeswitch does not support it. any ideas? keith ___ 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
[Freeswitch-users] configure FS: config.status: error: cannot find input file: Makefile.in
hello, i just got the last trunk and tried to compile it on one of my development machines... Well configure fails on tiff-3.8.2 where it is unable to find Makefile.in ... Can someone advice? checking if g++ static flag -static works... yes checking if g++ supports -c -o file.o... yes checking if g++ supports -c -o file.o... (cached) yes checking whether the g++ linker (/usr/bin/ld) supports shared libraries... yes checking dynamic linker characteristics... GNU/Linux ld.so checking how to hardcode library paths into programs... immediate checking for OpenGL Utility library... no checking for GLUT library... no configure: creating ./config.status config.status: error: cannot find input file: Makefile.in tculj...@subzero:~/freeswitch-trunk/libs/tiff-3.8.2$ ls -l total 2224 -rw-r--r-- 1 tculjaga tculjaga 23741 2009-10-02 13:19 acinclude.m4 -rw-r--r-- 1 tculjaga tculjaga 316978 2009-10-02 13:28 aclocal.m4 -rwxr-xr-x 1 tculjaga tculjaga121 2009-10-02 13:19 autogen.sh -rw-r--r-- 1 tculjaga tculjaga 124047 2009-10-02 13:19 ChangeLog drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:28 config -rw-r--r-- 1 tculjaga tculjaga 103926 2009-10-02 14:00 config.log -rwxr-xr-x 1 tculjaga tculjaga 73065 2009-10-02 14:00 config.status -rwxr-xr-x 1 tculjaga tculjaga 740145 2009-10-02 13:28 configure -rw-r--r-- 1 tculjaga tculjaga 20492 2009-10-02 13:19 configure.ac -rwxr-xr-x 1 tculjaga tculjaga 56 2009-10-02 13:19 configure.gnu -rwxr-xr-x 1 tculjaga tculjaga 737794 2009-10-02 13:57 configure.lineno drwxr-xr-x 16 tculjaga tculjaga 4096 2009-10-02 13:19 contrib -rw-r--r-- 1 tculjaga tculjaga 1146 2009-10-02 13:19 COPYRIGHT -rw-r--r-- 1 tculjaga tculjaga 1570 2009-10-02 13:19 HOWTO-RELEASE drwxr-xr-x 5 tculjaga tculjaga 4096 2009-10-02 13:19 html drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:28 libtiff drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 m4 -rw-r--r-- 1 tculjaga tculjaga 1908 2009-10-02 13:19 Makefile.am -rw-r--r-- 1 tculjaga tculjaga 1724 2009-10-02 13:19 Makefile.vc drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 man -rw-r--r-- 1 tculjaga tculjaga 6270 2009-10-02 13:19 nmake.opt drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 port -rw-r--r-- 1 tculjaga tculjaga 2363 2009-10-02 13:19 README -rw-r--r-- 1 tculjaga tculjaga 9 2009-10-02 13:19 RELEASE-DATE -rw-r--r-- 1 tculjaga tculjaga 5893 2009-10-02 13:19 SConstruct drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 test -rw-r--r-- 1 tculjaga tculjaga433 2009-10-02 13:19 TODO drwxr-xr-x 3 tculjaga tculjaga 4096 2009-10-02 13:19 tools -rw-r--r-- 1 tculjaga tculjaga 6 2009-10-02 13:19 VERSION tculj...@subzero:~/freeswitch-trunk/libs/tiff-3.8.2$ tculj...@subzero:~/freeswitch-trunk/libs/tiff-3.8.2$ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?
what if you are running some huge traffic e.g. 2000 calls with media?
a typical application for that is an IVR system handling several different
services. I'd like to dedicate some capacity for inbound on per service
basis.
e.g.
DID 10001 limit to 500 calls
DID 10002 limit to 400 calls
DID 10003 limit to 100 calls
DID 10005 limit to 1000 calls
This will be a total of 2000 calls.
don't you think js is simply too weak for that? It should cont
calls/channels, brake counts per service/DID and update the counters on
every call hit.
in the DP you would have something like this for every DID:
include
extension name=MY_DID_NUM
condition field=destination_number expression=^MY_DID_NUMBER$
action application=set data=SERVICE_LIMIT=500/
!--
count number of active channels going towards MY_DID_NUMBER and
store it into COUNT_MY_DID_NUMBER
--
action application=transfer data=do_MY_SERVICE XML public/
/condition
/extension
/include
include
extension name=SERVICE1
condition field=destination_number expression=^do_MY_SERVICE$/
condition field=${COUNT_MY_DID_NUMBER} expression=^SERVICE_LIMIT$
!-- do your service here --
action application=playback data=I_Accept_Your_Call.wav/
action application=hangup data=NORMAL_CLEARING/
!-- do your limitation here --
anti-action application=respond data=403 Forbidden/ = put your
response here!
/condition
/extension
/include
but the question is ... how powerful a JavaScript can be? Will it be enough
to handle that load?
Tihomir.
On Fri, Oct 2, 2009 at 1:11 PM, Alberto Escudero aep.li...@it46.se wrote:
You can use the api and check that the channel is occupied with show
channels?
You can write a small javascript that checks if the channel is occupied by
means of session.execute api.
/aep
--
Stopping junk mailers is good for the environment
My SIP provider allows only one call (incoming or outgoing) via one
SIP account. For FreeSWITCH I have configured it as public DID
extension and outgoing gateway. Now I would like to transfer to
another gw (or generate limit exceded) when one tries to place an
outgoing call while incoming call is in progress. How tho do that?
Limiting the number of outgoing calls is easy (mod_limit), but how to
take into account incoming one?
- Dmitry Bely
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Re: [Freeswitch-users] Dialplan Issue
anyhow, this is how it works for me!
include
context name=public
extension name=LNP
condition field=destination_number
expression=(^30)(.*)
action application=lnp_getprefix data=in $2, out
reroutingalias/
action application=redirect data=sip:${
reroutingali...@10.4.13.11:5060/
/condition
/extension
extension name=LBS
condition field=destination_number
expression=(^300010)(.*)
action application=lbs_getpublicphone data=in
${caller_id_number}, in $2, out reroutingalias/
action application=redirect data=sip:${
reroutingali...@10.4.13.11:5060/
/condition
/extension
extension name=CPS
condition field=destination_number
expression=(^300020)(.*)
action application=cps_verifyphone data=in
${caller_id_number}, in $2, out radiusacc/
/condition
condition field=radiusacc expression=1
action application=redirect data=sip:${
caller_id_numb...@10.4.13.11:5060/
anti-action application=respond data=403
Forbidden/
/condition
/extension
extension name=ServiceLookup
condition field=destination_number expression=(^300030)(.*)
action application=lookup_service_destination data=in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $
1, in ${network_addr}:5060, out red_contact, out authResult/
action application=log data=INFO
ServiceLookup \n/
action application=log data=INFO
contact = '${red_contact}' ##\n/
action application=log data=INFO
CallerNum = '${caller_id_number:6:16}' ##\n/
action application=log data=INFO
RADIUS auth = '${authResult}' ##\n/
action application=execute_extension data=doRedirect XML
public/
/condition
/extension
extension name=doRedirect
condition field=destination_number expression=^doRedirect$/
condition field=${authResult} expression=^0$|^60$
action application=log data=INFO
RADIUS auth OK!!!' ##\n/
action application=redirect data=${red_contact}/
anti-action application=log data=INFO
RADIUS auth NOK!! ##\n/
anti-action application=respond data=403 Forbidden/
/condition
/extension
/context
/include
On Thu, Oct 1, 2009 at 6:18 PM, Shelby Ramsey sicfsl...@gmail.com wrote:
Just to confirm ... works like a champ.
Thanks again!!!
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Re: [Freeswitch-users] mod_cdr_csv missing timestamps in A-LEG
Hi Michael, thanks for your feedback but it's late now :(
I had to moved back to 1.0.3 because it is in production. On that version it
works as a charm.
for some reason i cannot get it right in 1.0.4 and trunk.
Actually, what i'm doing is to subscribe to events (within a custom module)
and try to get timestamps... I started having issues when i moved to trunk.
To be sure that i'm not doing something wrong, i configured mod_cdr_csv to
dump CDRs. Well it turned out this module doesn't work as well in the trunk.
Can it be because of AMD opteron + Debian 5.0 enviorment?
There is something in the 1.0.4/trunk version that is wrong for that kind of
event/CDR.
T.
On Fri, Sep 25, 2009 at 6:44 AM, Michael Jerris m...@jerris.com wrote:
Can you get these same values in xml-cdr? I don't think csv was ever
intended to work with different cdrs for a and b leg, it was more intended
as a more familiar interface for those coming over from asterisk.
Mike
On Sep 24, 2009, at 6:10 AM, Tihomir Culjaga wrote:
hello,
i'm on latest trunk and for some reason i cannot get timestamps dumped in
my cdrs. I use mod_cdr_csv with default settings plus i enabled to get both
a and b legs dumped.
cdr_csv.conf.xml:
configuration name=cdr_csv.conf description=CDR CSV Format
settings
!-- 'cdr-csv' will always be appended to log-base --
!--param name=log-base value=/var/log/--
param name=default-template value=example/
!-- This is like the info app but after the call is hung up --
!--param name=debug value=true/--
param name=rotate-on-hup value=true/
!-- may be a b or ab --
param name=legs value=ab/
/settings
templates
template name=sqlINSERT INTO cdr VALUES
(${caller_id_name},${caller_id_number},${destination_number},${context},${s
tart_stamp},${answer_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},
${accountcode}
);/template
template
name=example${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${
write_codec}/template
template
name=snom${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answer_
stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},
${accountcode},${read_codec},${wr
ite_codec},${sip_user_agent},${call_clientcode},${sip_rtp_rxstat},${sip_rtp_txstat},${sofia_record_file}/template
template
name=linksys${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${
write_codec},${sip_user_agent},${sip_p_rtp_stat}/template
template
name=asterisk${accountcode},${caller_id_number},${destination_number},${context},${caller_id},${channel_
name},${bridge_channel},${last_app},${last_arg},${start_stamp},${answer_stamp},${end_stamp},${duration},${billsec},
${hangup_cause},${amaflags},${uuid},${userfield}/template
/templates
/configuration
call flow is the following:
CALLER = FS = CALLED
FS answers the call from CALLER, plays an announcement and bridges towards
CALLED.
I get different behavior when the call is released by Caller and by Called.
Released by Caller: the CDR is ok having all timestamps
OK CDR:
Outbound LEG = 016659280,016659280,0914392122,public,2009-09-24
12:02:48,2009-09-24 12:02:54,2009-09-24
12:03:01,13,7,NORMAL_CLEARING,699cc2d0-a8f1-11de-962a-e328afdb9d8d,,,PCMA,PCMA
Inbound LEG = 016659280,016659280,05000403,public,2009-09-24
12:02:27,2009-09-24 12:02:41,2009-09-24
12:03:01,34,20,NORMAL_CLEARING,5d530192-a8f1-11de-962a-e328afdb9d8d,699cc2d0-a8f1-11de-962a-e328afdb9d8d,,PCMA,PCMA
Released by Called: the CDR is NOT OK as timestamps are missing
NOT OK CDR:
Inbound LEG = 016659280,016659280,0914392122,public,2009-09-24
12:05:20,2009-09-24 12:05:30,2009-09-24
12:05:39,19,9,NORMAL_CLEARING,c479411a-a8f1-11de-962a-e328afdb9d8d,,,PCMA,PCMA
Outbound LEG =016659280,016659280,015000403,public,*,,,*
0,0,NORMAL_CLEARING,b82f2046-a8f1-11de-962a-e328afdb9d8d,c479411a-a8f1-11de-962a-e328afdb9d8d,,PCMA,PCMA
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Re: [Freeswitch-users] mod_cdr_csv missing timestamps in A-LEG
should i move this to the DEV mailing list ?
T.
On Fri, Sep 25, 2009 at 4:12 PM, Michael Jerris m...@jerris.com wrote:
nothing I can think of, set up a test box that is not in production and
lets figure out what is wrong.
Mike
On Sep 25, 2009, at 7:22 AM, Tihomir Culjaga wrote:
Hi Michael, thanks for your feedback but it's late now :(
I had to moved back to 1.0.3 because it is in production. On that version
it works as a charm.
for some reason i cannot get it right in 1.0.4 and trunk.
Actually, what i'm doing is to subscribe to events (within a custom module)
and try to get timestamps... I started having issues when i moved to trunk.
To be sure that i'm not doing something wrong, i configured mod_cdr_csv to
dump CDRs. Well it turned out this module doesn't work as well in the trunk.
Can it be because of AMD opteron + Debian 5.0 enviorment?
There is something in the 1.0.4/trunk version that is wrong for that kind
of event/CDR.
T.
On Fri, Sep 25, 2009 at 6:44 AM, Michael Jerris m...@jerris.com wrote:
Can you get these same values in xml-cdr? I don't think csv was ever
intended to work with different cdrs for a and b leg, it was more intended
as a more familiar interface for those coming over from asterisk.
Mike
On Sep 24, 2009, at 6:10 AM, Tihomir Culjaga wrote:
hello,
i'm on latest trunk and for some reason i cannot get timestamps dumped in
my cdrs. I use mod_cdr_csv with default settings plus i enabled to get both
a and b legs dumped.
cdr_csv.conf.xml:
configuration name=cdr_csv.conf description=CDR CSV Format
settings
!-- 'cdr-csv' will always be appended to log-base --
!--param name=log-base value=/var/log/--
param name=default-template value=example/
!-- This is like the info app but after the call is hung up --
!--param name=debug value=true/--
param name=rotate-on-hup value=true/
!-- may be a b or ab --
param name=legs value=ab/
/settings
templates
template name=sqlINSERT INTO cdr VALUES
(${caller_id_name},${caller_id_number},${destination_number},${context},${s
tart_stamp},${answer_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},
${accountcode}
);/template
template
name=example${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${
write_codec}/template
template
name=snom${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answer_
stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},
${accountcode},${read_codec},${wr
ite_codec},${sip_user_agent},${call_clientcode},${sip_rtp_rxstat},${sip_rtp_txstat},${sofia_record_file}/template
template
name=linksys${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${
write_codec},${sip_user_agent},${sip_p_rtp_stat}/template
template
name=asterisk${accountcode},${caller_id_number},${destination_number},${context},${caller_id},${channel_
name},${bridge_channel},${last_app},${last_arg},${start_stamp},${answer_stamp},${end_stamp},${duration},${billsec},
${hangup_cause},${amaflags},${uuid},${userfield}/template
/templates
/configuration
call flow is the following:
CALLER = FS = CALLED
FS answers the call from CALLER, plays an announcement and bridges towards
CALLED.
I get different behavior when the call is released by Caller and by
Called.
Released by Caller: the CDR is ok having all timestamps
OK CDR:
Outbound LEG = 016659280,016659280,0914392122,public,2009-09-24
12:02:48,2009-09-24 12:02:54,2009-09-24
12:03:01,13,7,NORMAL_CLEARING,699cc2d0-a8f1-11de-962a-e328afdb9d8d,,,PCMA,PCMA
Inbound LEG = 016659280,016659280,05000403,public,2009-09-24
12:02:27,2009-09-24 12:02:41,2009-09-24
12:03:01,34,20,NORMAL_CLEARING,5d530192-a8f1-11de-962a-e328afdb9d8d,699cc2d0-a8f1-11de-962a-e328afdb9d8d,,PCMA,PCMA
Released by Called: the CDR is NOT OK as timestamps are missing
NOT OK CDR:
Inbound LEG = 016659280,016659280,0914392122,public,2009-09-24
12:05:20,2009-09-24 12:05:30,2009-09-24
12:05:39,19,9,NORMAL_CLEARING,c479411a-a8f1-11de-962a-e328afdb9d8d,,,PCMA,PCMA
Outbound LEG =016659280,016659280,015000403,public,*,,,*
0,0,NORMAL_CLEARING,b82f2046-a8f1-11de-962a-e328afdb9d8d,c479411a-a8f1-11de-962a-e328afdb9d8d,,PCMA,PCMA
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FreeSWITCH-users
Re: [Freeswitch-users] Ringback when running G729 codec
does it mean, if i encode my voice files in g729 i can use mod_nativefile to playback to a call using 729 codec? T. On Fri, Sep 25, 2009 at 8:30 PM, Anthony Minessale anthony.miness...@gmail.com wrote: fixed in latest trunk, please test thank you On Fri, Sep 25, 2009 at 6:17 AM, Hound Dog d_ho...@ymail.com wrote: Hi, very happy with freeswitch as a PBX/softswitch/SBC system its working solidly for a few weeks now - just great I have a question regarding ringback tones - custom or regular - I cant get freeswitch to send ringback using G729 I used the following settings ( it will just play one of the IVR prompts as ringback (filename ivr-to_repeat_these_options) - I took it from the G729 encoded files package , it has PCMA , G729 G723 extensions ) extension name=inbound_routing condition field=destination_number expression=^4420885767(0\d)$ action application=set data=ringback=/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options/ action application=set data=instant_ringback=true/ action application=bridge data=user/10$1/ /condition /extension when I call with G711 enabled , it plays the file no problems - see log 2009-09-25 11:29:58.641361 [DEBUG] sofia.c:3289 Channel sofia/external/ 442078562...@80.80.80.80 entering state [early][183] 2009-09-25 11:29:58.641361 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562...@80.80.80.80 [BREAK] 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success l...@8000hz 1 channel 20ms 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1745 Play Ringback File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 2009-09-25 11:29:58.641361 [INFO] mod_native_file.c:82 Opening File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 8000hz 2009-09-25 11:29:58.889369 [DEBUG] sofia.c:3289 Channel sofia/internal/ sip:1...@82.80.131.233:40505 entering state [proceeding][180] when I call with G729 only - I get silence , and freeswitch only send the comfort noise packet and no RTP , see log 2009-09-25 11:28:57.437537 [DEBUG] sofia.c:3289 Channel sofia/external/ 442078562...@80.80.80.80 entering state [early][183] 2009-09-25 11:28:57.437537 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562...@80.80.80.80 [BREAK] 2009-09-25 11:28:57.685536 [DEBUG] sofia.c:3289 Channel sofia/internal/ sip:1...@82.80.131.233:40505 entering state [proceeding][180] 2009-09-25 11:28:57.685536 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:1...@82.80.131.233:40505! mod_native_file works well for me when used in applications and plays G729 files no problem any ideas why is that happening , any suggestions on how to resolve ? thanks Ori ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] mod_cdr_csv missing timestamps in A-LEG
hello,
i'm on latest trunk and for some reason i cannot get timestamps dumped in my
cdrs. I use mod_cdr_csv with default settings plus i enabled to get both a
and b legs dumped.
cdr_csv.conf.xml:
configuration name=cdr_csv.conf description=CDR CSV Format
settings
!-- 'cdr-csv' will always be appended to log-base --
!--param name=log-base value=/var/log/--
param name=default-template value=example/
!-- This is like the info app but after the call is hung up --
!--param name=debug value=true/--
param name=rotate-on-hup value=true/
!-- may be a b or ab --
param name=legs value=ab/
/settings
templates
template name=sqlINSERT INTO cdr VALUES
(${caller_id_name},${caller_id_number},${destination_number},${context},${s
tart_stamp},${answer_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},
${accountcode}
);/template
template
name=example${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${
write_codec}/template
template
name=snom${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answer_
stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},
${accountcode},${read_codec},${wr
ite_codec},${sip_user_agent},${call_clientcode},${sip_rtp_rxstat},${sip_rtp_txstat},${sofia_record_file}/template
template
name=linksys${caller_id_name},${caller_id_number},${destination_number},${context},${start_stamp},${answ
er_stamp},${end_stamp},${duration},${billsec},${hangup_cause},${uuid},${bleg_uuid},${accountcode},${read_codec},${
write_codec},${sip_user_agent},${sip_p_rtp_stat}/template
template
name=asterisk${accountcode},${caller_id_number},${destination_number},${context},${caller_id},${channel_
name},${bridge_channel},${last_app},${last_arg},${start_stamp},${answer_stamp},${end_stamp},${duration},${billsec},
${hangup_cause},${amaflags},${uuid},${userfield}/template
/templates
/configuration
call flow is the following:
CALLER = FS = CALLED
FS answers the call from CALLER, plays an announcement and bridges towards
CALLED.
I get different behavior when the call is released by Caller and by Called.
Released by Caller: the CDR is ok having all timestamps
OK CDR:
Outbound LEG = 016659280,016659280,0914392122,public,2009-09-24
12:02:48,2009-09-24 12:02:54,2009-09-24
12:03:01,13,7,NORMAL_CLEARING,699cc2d0-a8f1-11de-962a-e328afdb9d8d,,,PCMA,PCMA
Inbound LEG = 016659280,016659280,05000403,public,2009-09-24
12:02:27,2009-09-24 12:02:41,2009-09-24
12:03:01,34,20,NORMAL_CLEARING,5d530192-a8f1-11de-962a-e328afdb9d8d,699cc2d0-a8f1-11de-962a-e328afdb9d8d,,PCMA,PCMA
Released by Called: the CDR is NOT OK as timestamps are missing
NOT OK CDR:
Inbound LEG = 016659280,016659280,0914392122,public,2009-09-24
12:05:20,2009-09-24 12:05:30,2009-09-24
12:05:39,19,9,NORMAL_CLEARING,c479411a-a8f1-11de-962a-e328afdb9d8d,,,PCMA,PCMA
Outbound LEG =016659280,016659280,015000403,public,*,,,*
0,0,NORMAL_CLEARING,b82f2046-a8f1-11de-962a-e328afdb9d8d,c479411a-a8f1-11de-962a-e328afdb9d8d,,PCMA,PCMA
What can be wrong?
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Re: [Freeswitch-users] Can this be done in FreeSWITCH?
when i said inline ... i just meant to define some variables in your DP ... this is not a solution for you ... it is rather a proof of concept instead. you need to do a DB lookup (sqlite or mysql). T. On Wed, Sep 23, 2009 at 1:32 AM, Francis Vidal francisv.l...@gmail.comwrote: Yes, this is the desired outcome. I was planning of using FreeSWITCH + MySQL to do this. How do I do this inline? On Wed, Sep 23, 2009 at 12:49 AM, Tihomir Culjaga tculj...@gmail.comwrote: so, you say ... CallingParty = AS5300 A: aNum B: didNum AS5300 = PSTN A: 1 + didNum B: prefix (actually the PSTN subscriber's number) well, without a doubt... you can manipulate whatever number you want ... you just need to find the best way to do it. This depends of the number of DIDs you would like to host. You can do a DB lookup to retrieve the prefix / Subscriber Number... or you can do it inline in your dialplan. It really depends of how much you need to scale. ___ 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] Gateways in Freeswitch
endpoints that you are sending/receiving calls to/from It is useful to have a separate configuration (other than dialplan) when you need to specify credentials for GW to register somewhere, to specify domain, etc, etc ... T. On Wed, Sep 23, 2009 at 9:30 AM, Anil Kumar S. R. sra...@gmail.com wrote: Hi All, Can anybody please tell me what are the gateways in Freeswitch ? Thanks, -- Anil Kumar S. R. http://sranil.googlepages.com/ The best way to succeed in this world is to act on the advice you give to others. ___ 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] Unable to set internal call to registered sip user
and this is not enough for you?
!--- The *%* behind the username tells FS to lookup the user in it's
local sip_registration database --
action application=bridge data=user/${dialed_extension}@
${domain_name}/
!--- x.x.x.x in the line above is the IP address to the FreeSWITCH
server/device --
!--- If you don't want to bridge a call to a local registered user,
but to a SIP URI, use the @ instead of %:
action application=bridge data=sofia/profilename/5...@x.x.x.x/
--
T.
On Tue, Sep 22, 2009 at 1:52 PM, Filip Lyncker lync...@lyth.de wrote:
Dear List,
I read the documentation, but Im still confused about how to dial a
internal registered sip user.
I configured the both sip phones in the directory in my local.xml file :
include
domain name=$${domain}
user id=22 mailbox=22
params
param name=password value=Xk21%/param
param name=vm-password value=22/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=22/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
22/variable
variable name=effective_caller_id_number value=22/variable
/variables
/user
user id=24 mailbox=24
params
param name=password value=dudeldum/param
param name=vm-password value=24/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=24/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
24/variable
variable name=effective_caller_id_number value=24/variable
/variables
/user
/domain
/include
It seems, that they can connect to the freeswitch.
I configured the dialplan like following :
include
context name=default
extension name=diallocal
condition field=destination_number expression=^(2[0-9])$
!--- The % behind the username tells FS to lookup the user in
it's local sip_registration database --
action application=bridge
data=user/${dialed_extensi...@${domain_name}/action
!--- x.x.x.x in the line above is the IP address to the
FreeSWITCH server/device --
!--- If you don't want to bridge a call to a local registered
user, but to a SIP URI, use the @ instead of %:
action application=bridge
data=sofia/profilename/5...@x.x.x.x/ --
/condition
/extension
...
If I call from the sip user 24 to 22 , freeswitch logs the following and
gives an busy tone immediately:
freeswi...@bigfish 2009-09-22 13:50:29.367114 [NOTICE]
switch_channel.c:602 New Channel sofia/internal/2...@192.168.1.34
[decc119c-a973-6b4c-bf11-ec251c653cda]
2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing
24-22 in context default
2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user
[...@192.168.1.34]
2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot
create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed.
Cause: SUBSCRIBER_ABSENT
2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup
sofia/internal/2...@192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT]
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session
13 (sofia/internal/2...@192.168.1.34) Ended
2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/internal/2...@192.168.1.34 [CS_DESTROY]
thanks again for your help ...
regards,
Filip
--
_
Filip Lyncker, Dipl.-Inform. (FH)
Lyncker Theis GmbH
Wilhelmstr. 16
65185 Wiesbaden
Germany
Fon +49 611/9006951
Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden
Steuernummer: 4023897051
USt-IdNr.: DE255806399
Geschäftsführer:
Filip Lyncker,
Armin Theis
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Re: [Freeswitch-users] Some Newbie questions about dialplan and local Sip registration
hmmm .. can you register using x-lite or some other softphone with the same
credentials?
can you paste a siptrace of the failed registration?
BTW: Make sure nothing is already registered with this credentials when you
try with FS
T.
On Tue, Sep 22, 2009 at 12:56 PM, Filip Lyncker lync...@lyth.de wrote:
Hi Tihomir,
Thanks for your help , I added the Asteriskparameters as you described
below, but I still get the same timeout error:
2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed
Registration, setting retry to 270 seconds.
2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk Registration
Failed with status Request Timeout [408]. failure #9
Now, my gateway entry looks like the following :
include
gateway name=asterisk
param name=username value=28/
param name=realm value=192.168.1.119/
param name=proxy value=192.168.1.119/
param name=password value=test/
param name=register value=true/
param name=caller-id-in-from value=true/
param name=sip-port value=5060/param
/gateway
/include
What can be still wrong here?
Regards,
Filip
Tihomir Culjaga schrieb:
hi Filip,
for calling a user... please read this first:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User
for making a GW register into e.g. asterisk please use this:
include
gateway name=gw01
param name=username value=USERNAME_ON_ASTERISK/
param name=realm value=ASTERISK_IP_ADDRESS/
param name=password value=PASSWORD_ON_ASTERISK/
param name=register value=true/
param name=caller-id-in-from value=true/
/gateway
/include
this should be enough to register the GW... after that please read
this:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways
in your case it will be something like this:
extension name=dialGW
condition field=destination_number
expression=^(NUMBER_TO_SEND_TO_ASTERISK)$
action application=bridge data=sofia/gateway/gw01/$1/
/condition
/extension
On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker lync...@lyth.de
mailto:lync...@lyth.de wrote:
Hi List,
for the first experiments with freeswitch I downloaded the Windows
installation.
Now Im trying to get my 2 Sipphones get connected to. Later I want
connect the freeswitch to my asterisk gateway.
I find the examples pretty complex therfore Im trying to build up a
simple solution to understand the functions from the scratch ..
my current problem is , that I cant route my local sips to each
other (
registration seems to work now).
the next is , that freeshwitch is not able to connect to asterisk.
but I
will describe this later.
I installed in the Directory a xml file ( called 22.xml) with the
following content :
include
domain name=$${domain}
user id=22 mailbox=22
params
param name=password value=Xk21%/param
param name=vm-password value=22/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=22/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
22/variable
variable name=effective_caller_id_number
value=22/variable
/variables
/user
user id=24 mailbox=24
params
param name=password value=dudeldum/param
param name=vm-password value=24/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=24/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
24/variable
variable name=effective_caller_id_number
value=24/variable
/variables
/user
/domain
/include
This seems to be ok now. Now I want to dial from 22 to 24 ,
wherefore I
configured this dialplan :
include
context name=any
condition field=destination_number expression=^(2[0-9])$
action application=bridge data=user/${dialed_extension}/
/condition
/include
wich doesnt work , mybe b/c the user/${dialed_extension} I dont
know...
Freeswitch says:
[INFO] switch_core_state_machine.c:136 No Route, Aborting
[NOTICE] switch_core_state_machine.c:137 Hangup
sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34
[CS_ROUTING] [NO_ROUTE_DESTINATION]
[NOTICE] switch_core_session.c:1086 Session 17
(sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34) Ended
[NOTICE] switch_core_session.c:1088 Close Channel
sofia/internal/2...@192.168.1.34 mailto:2...@192.168.1.34 [CS_DESTROY]
Im sure , for you
Re: [Freeswitch-users] Can this be done in FreeSWITCH?
well .. it is AS .. it can be SIP or H323 ... well if it is hooked to a PGW it is MGCP but i doubt... so it is either SIP or H323. i will put a nickel for H323 :P T. On Tue, Sep 22, 2009 at 6:49 PM, Tihomir Culjaga tculj...@gmail.com wrote: so, you say ... CallingParty = AS5300 A: aNum B: didNum AS5300 = PSTN A: 1 + didNum B: prefix (actually the PSTN subscriber's number) well, without a doubt... you can manipulate whatever number you want ... you just need to find the best way to do it. This depends of the number of DIDs you would like to host. You can do a DB lookup to retrieve the prefix / Subscriber Number... or you can do it inline in your dialplan. It really depends of how much you need to scale. T. On Tue, Sep 22, 2009 at 6:17 AM, Francis Vidal francisv.l...@gmail.comwrote: Hi all, Consider the following scenario: Calling party -- DID provider -- Cisco AS5300 -- POTS provider -- Called party The Calling party calls a number provided by the DID provider. This is then processed by the AS5300 facing the POTS provider to do the following number translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS prefixed with 1). The Cisco AS5300 then sends a prefix which is actually the number of the Called party in their system (of the POTS provider). However, the Cisco AS5300 has a finite limit on the number of translations (approx. 128-300 translations). Can the number translation be done on FreeSWITCH instead? Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300 -- POTS provider -- Called party This can also evolve into: Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300[1] -- POTS provider -- Called party \ / +- Cisco AS5300[2] ---+ If we wanted to increase the number of ports the POTS provider. ___ 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] Can this be done in FreeSWITCH?
so, you say ... CallingParty = AS5300 A: aNum B: didNum AS5300 = PSTN A: 1 + didNum B: prefix (actually the PSTN subscriber's number) well, without a doubt... you can manipulate whatever number you want ... you just need to find the best way to do it. This depends of the number of DIDs you would like to host. You can do a DB lookup to retrieve the prefix / Subscriber Number... or you can do it inline in your dialplan. It really depends of how much you need to scale. T. On Tue, Sep 22, 2009 at 6:17 AM, Francis Vidal francisv.l...@gmail.comwrote: Hi all, Consider the following scenario: Calling party -- DID provider -- Cisco AS5300 -- POTS provider -- Called party The Calling party calls a number provided by the DID provider. This is then processed by the AS5300 facing the POTS provider to do the following number translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS prefixed with 1). The Cisco AS5300 then sends a prefix which is actually the number of the Called party in their system (of the POTS provider). However, the Cisco AS5300 has a finite limit on the number of translations (approx. 128-300 translations). Can the number translation be done on FreeSWITCH instead? Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300 -- POTS provider -- Called party This can also evolve into: Calling party -- DID provider -- FreeSWITCH -- Cisco AS5300[1] -- POTS provider -- Called party \ / +- Cisco AS5300[2] ---+ If we wanted to increase the number of ports the POTS provider. ___ 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] recompile with gdb
Hi, Nope, I'm still on Debian 5.0... in transit to CentOS 5.3 but it needs to wait a bit. i was talking about gdb, not gcc and was trying to recompile FS with debug symbols on: CFLAGS=-g -ggdb MOD_CFLAGS=-g -ggdb. yes, I understand that gcc segfault most probably means only one thing... HW isues. This is sometihng that I'm going to check tomorrow running memtest to see what i get. Also, I will repeat the same test with a new block of RAM. Maybe i didn't explain myself well... apologize. T. On Tue, Sep 22, 2009 at 8:42 PM, Anthony Minessale anthony.miness...@gmail.com wrote: One of the things that -hp does is call memlockall which disables swapping which uses more memory which makes hitting a land mine in your ram chip much more likely. On the other hand: Since you are talking about with and without gcc support I am going to guess you are on Solaris which you probably should have mentioned before. it's possible that some of the more aggressive things activated by -hp is not possible on that platform. If so we either have to identify that and disable it or disable hp completely for Solaris. Either way, gcc randomly crashing is never ok and is a symptom of a pretty serious issue. Are you using 2 separate fresh checkouts for both suncc and gcc builds because it's not possible to switch the same source tree once it's already configured for one of them. On Tue, Sep 22, 2009 at 11:29 AM, Tihomir Culjaga tculj...@gmail.comwrote: Hi Anthony, it is not the machine ... and yep there was some memory related issue ... but this was caused by my module So, to summarize.. i had two issues: 1. FS crashing without any notice (at 5 CPS) 2. Unable to recompile FS with gdb support The first issue was actually related to -hp switch i was using in my startup script. With it, FS was crashing without any notice (even on low traffic) and regardless if i load my custom modules or not. The second issue was related to many FS crashes having my module loaded... I found it later and fixed that. So, after the machine cleanup I rebuild FS with gdb support without any issues. Of course i sow this log .. but i didn't realize for a while... and after that i was fighting with crashes caused by -hp ... also, it was quite late as well ended up at 3 AM :P Anyhow, the poit is; FS works well with my custom module. It just finished 2 mil. calls (with media) at 100 CPS having ~1600 simultaneous calls... well, thats something :P. T. On Tue, Sep 22, 2009 at 4:35 PM, Anthony Minessale anthony.miness...@gmail.com wrote: see this from your own log? make[2]: Entering directory `/opt/freeswitch-trunk/libs/ pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 This is a FATAL error to have on your machine. It's failing during the build. This is your compiler crashing while trying to build the software. This is very bad. You most likely have a hardware failure and need to replace the machine or at the very least all of the memory chips. On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga tculj...@gmail.comwrote: hi Brian, well, there is no coredump at all... and when i start FS with gdb it doesn't crash :P I need to do some more testing and will come back to you. T. On Tue, Sep 22, 2009 at 1:22 AM, Brian West br...@freeswitch.orgwrote: This looks like gcc is segfaulting can you provide me a complete backtrace of the core file that dumps from FreeSWITCH? http://wiki.freeswitch.org/wiki/Reporting_Bugs It sounds like you might have bad ram or bad hardware... gcc crashing is usually a sign something is really wrong with your machine. /b On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: but without luck... ode1:/opt/freeswitch-trunk# node1:/opt/freeswitch-trunk# sudo make make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' make all-am make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 node1:/opt
Re: [Freeswitch-users] recompile with gdb
well ... shame on me :P thx anyway... T. On Tue, Sep 22, 2009 at 10:12 PM, Diego Viola diego.vi...@gmail.com wrote: He's doing an extra effort... just compile it as you would normally and you will have the debug symbols. On Tue, Sep 22, 2009 at 8:11 PM, Diego Viola diego.vi...@gmail.comwrote: Then why is Tihomir trying to compile with debug symbols? On Tue, Sep 22, 2009 at 8:00 PM, Brian West br...@freeswitch.org wrote: yes On Sep 22, 2009, at 2:32 PM, Diego Viola wrote: Doesn't FS already compiles with debug symbols by default? ___ 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] Affordable GSM gateway for one cellphone?
I didn't say i have a working FS on blackfin... i just said i've ported a lot of software to blackfin and it was always floating point, fork vs vfork ... main issues... but why do you think it cannot be done? T. On Mon, Sep 21, 2009 at 6:08 AM, Hadley Rich h...@nice.net.nz wrote: On Mon, 21 Sep 2009 15:58:33 Juan Backson wrote: Are you able to have freeswitch working on blackfin platform? This has been covered many times on the list now, currently the answer is no. hads -- https://nicegear.co.nz VoIP and Open Source Hardware ___ 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] Affordable GSM gateway for one cellphone?
its a waste of time ... i doubt it can be done. T. On Mon, Sep 21, 2009 at 10:56 AM, Fred-145 codecompl...@free.fr wrote: Or as a more affordable solution... is it possible to connect an entry-level GSM phone to a PC running Freeswitch and use this as a poor man's gateway? -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530241.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] recompile with gdb
Hi Guys,
I have an issue running FS... it crashes apparently without leaving any log
... not even a core dump is left.
The machine is dual AMD opteron quad core with 8 GB RAM and i'm running 75
simultaneous calls (with media) with a rate of 5 calls per second.
As i was not able to reproduce the issue on a real traffic so i went back to
sipp and started generating some... sipp scenario files are ok.
after a while (few minutes)... on sipp i start getting retransmissions and
when i check FS i see two situations:
1. freeswitch has died
2. freeswitch process is running but it doesn't respond to any call... as
nothing has been sent ... and after a while it dies too.
I'm using sip profile external (moved to port 5060) with some semi-complex
dialplan... attached.
well .. the point is that i cannot even tell where it crashes as there is no
log.
I have:
param name=loglevel value=debug/
X-PRE-PROCESS cmd=set data=call_debug=true/
X-PRE-PROCESS cmd=set data=console_loglevel=debug/
fs is dumping the log to the log directory ... but nothing special can't bee
seen there...
I tried to recompile with gdb
export CFLAGS=-g -ggdb
export MOD_CFLAGS=-g -ggdb
./configure
but without luck...
ode1:/opt/freeswitch-trunk#
node1:/opt/freeswitch-trunk# sudo make
make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre'
make all-am
make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre'
g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF
.deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc
g++: Internal error: Segmentation fault (program cc1plus)
Please submit a full bug report.
See file:///usr/share/doc/gcc-4.3/README.Bugs for instructions.
make[2]: *** [pcrecpp_unittest.o] Error 1
make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre'
make: *** [libs/pcre/libpcre.la] Error 2
node1:/opt/freeswitch-trunk#
node1:/opt/freeswitch-trunk#
Of course I'm using the latest trunk...
Can anyone help?
include
extension name=VAS
condition field=destination_number expression=^0(\d+)$
action application=log data=INFO Entering VAS \n/
action application=execute_extension data=0$1_priceAdvice XML public/
action application=execute_extension data=0$1_serviceDiscriminator XML public/
action application=hangup data=NORMAL_CLEARING/
/condition
/extension
extension name=priceAdvice
condition field=destination_number expression=(\d+)_priceAdvice$
action application=log data=INFO Price Adviced \n/
!--action application=getServiceTypeID_db data=in $1, out service_type_id/--
action application=set data=service_type_id=1/
action application=pre_answer/
!--action application=getPricePrompt_db data=in $1, in ${caller_id_number} , out price_prompt/--
action application=set data=price_prompt=4.93kn_novo_upozorenje.wav/
action application=playback data=vas/${price_prompt}/
!--action application=sched_hangup data=+${cond(${regex($1|3856(\d)\d+|%1)} == 8 ? 120 : 3600)}/--
action application=sleep data=2000/
/condition
/extension
extension name=ServiceDiscriminator
condition field=destination_number expression=(\d+)_serviceDiscriminator$
action application=log data=INFO Service Discriminator \n/
!--action application=getServiceTypeID_db data=in $1, out service_type_id/--
action application=set data=dialed_number=$1/
action application=log data=INFO ### service_type_id = '${service_type_id}' ##/
action application=log data=INFO ### dialed_number = '${dialed_number}' ##/
/condition
condition field=${service_type_id} expression=^1$ break=on-true
action application=log data=INFO KVIZ \n/
action application=execute_extension data=${dialed_number}_getVars_Kviz XML public/
/condition
/extension
extension name=getVars_Kviz
condition field=destination_number expression=(\d+)_getVars_Kviz$
action application=log data=INFO GetVars Kviz /
action application=set data=bNum=$1/
!--action application=getQuizServiceStatus_ch data=in $1, in ${caller_id_number}, out service_status1, out number_2_connect, out next_number_2_connect, out next_number_2_display/
action application=getServiceOutOfWorkingHoursPrompt_db data=in $1, out not_working_prompt/
action application=getServiceWinPrompt_db data=in $1, out service_win_prompt/
action application=getServiceLoosePrompt_db data=in $1, out service_loose_prompt/--
!--action application=sched_hangup data=+${cond(${regex($1|3856(\d)\d+|%1)} == 8 ? 120 : 3600)}/--
action
Re: [Freeswitch-users] Call Tracing
switch.conf.xml (btw: in console you can enable/disable logging on the fly - F8/F7) param name=loglevel value=debug/ your relevant sip profile: param name=sip-trace value=yes/ T. On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller klaus.tel...@gmx.net wrote: Hi, Say i have an inbound VoIP/SIP call that hits my FS box. Is it possible to to extract information about the intermediate hops that the call or the signaling went through? If so, what information can i get? Thanks, Gregoire. -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser ___ 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] Affordable GSM gateway for one cellphone?
hi, well, yes, it should be possible to crosscompile freeswitch on that platofrm... this is a totally different topic and to be honest i really don't see the point doing this. When i did it last time (porting stuff to Blackfin), it took several days of hard work. This is an external device/endpoint to freeswitch. You don't need any FXS ports... it is enough to have the GSM one (or two). Just send calls from FS to FX02 via SIP and that's it. T. On Sun, Sep 20, 2009 at 10:57 PM, Fred-145 codecompl...@free.fr wrote: Thanks Tihomir for the link. From what I read, it appears that EdgePBX's FX02G is a full-fledged Asterisk server with a GSM module and an FXS module. Did you reflash its NAND to run Freeswitch? At $300, I guess customers will rather take a subscription with a VoIP provided and use their GSM gateway, but I'm interested in knowing whether the FX02G can be used as a PSTN/GSM gateway, possibly with FreeSwitch running on that unit as well. Thank you. -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530130.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] Not able to make call using external profile
check this:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User
dial registered user: action application=bridge
data=sofia/external/$1%$${domain}/
dial external endpoint: action application=bridge
data=sofia/external/$...@$${domain}/
another issue you might have with RTP so check the wiki for NAT config as
well.
T.
On Sat, Sep 19, 2009 at 7:50 AM, pankaj anand pankajanan...@gmail.comwrote:
@Tihomir Culjaga
HI folks,
thanx for such a quick reply.
Q. what I want to achieve with FreeSwitch ?
A: I want to enable the outside users ( from internet) to have video
chat on peer2peer using freeSwitch for signaling. External Profile is being
used to for this. External profile is using 5080 port. That port is
forwarded on the NAT server. Users are able to connect using 5080 port.
They
get registered with no issues.
Q. where do you want to send calls ?
A. I want to send call from one extension to another extension ( both
extension exist on the are on public internet). Right now i m trying with
1000 and 1001 user available in the default directory.
1. What is 192.168.1.50 ?
Ans: well , this is my domain name which is by default the local-ip
address of the machine. My current setup is like this:
FreeSwitch ( 192.168.1.50)
NAT(122.162.153.224)--Internet(122.80.0.180)NAT--(192.168.1.15)1001(user)
2.
Where/how are you originating calls from ?
1. I am using X-lite, Phoner , LinPhone to make calls. All these phones
have stun server enabled .
For the public dial plan I have added these lines in the file
public.xml which is used by the external profile
extension name=public_extensions
condition field=destination_number
expression=^(10[01][0-9])$
action application=bridge data=sofia/external/$1@
$${domain}/
action application=echo/
/condition
/extension
extension name=echo
condition field=destination_number expression=^9996$
action application=answer/
action application=echo/
/condition
/extension
Now the echo calls works through the external profile. But when a call
is being made to some other user, for example if user 1000 makes a call to
the 1001 it reaches to the public_extensions but it generates the
error which I have already mentioned. For the gateway thing , not gateway
is
being used.
On Fri, Sep 18, 2009 at 7:41 PM, pankaj anand pankajanan...@gmail.comwrote:
I m using default configuration of freeswitch.. I m not using any gateway
for authentication.
in the $INSTALLDIR/conf/sip_profiles/external/ directory, there exist
only one file which example.xml , this files contains
include
!--gateway name=asterlink.com--
!--/// account username *required* ///--
!--param name=username value=cluecon/--
!--/// auth realm: *optional* same as gateway name, if blank ///--
!--param name=realm value=asterlink.com/--
!--/// username to use in from: *optional* same as username, if blank
///--
!--param name=from-user value=cluecon/--
!--/// domain to use in from: *optional* same as realm, if blank
///--
!--param name=from-domain value=asterlink.com/--
!--/// account password *required* ///--
!--param name=password value=2007/--
!--/// extension for inbound calls: *optional* same as username, if
blank ///--
!--param name=extension value=cluecon/--
!--/// proxy host: *optional* same as realm, if blank ///--
!--param name=proxy value=asterlink.com/--
!--/// send register to this proxy: *optional* same as proxy, if blank
///--
!--param name=register-proxy value=mysbc.com/--
!--/// expire in seconds: *optional* 3600, if blank ///--
!--param name=expire-seconds value=60/--
!--/// do not register ///--
!--param name=register value=false/--
!-- which transport to use for register --
!--param name=register-transport value=udp/--
!--How many seconds before a retry when a failure or timeout occurs --
!--param name=retry-seconds value=30/--
!--Use the callerid of an inbound call in the from field on outbound
calls via this gateway --
!--param name=caller-id-in-from value=false/--
!--extra sip params to send in the contact--
!--param name=contact-params value=tport=tcp/--
!--send an options ping every x seconds, failure will unregister and/or
mark it down--
!--param name=ping value=25/--
!--/gateway--
/include
as you can see, all the lines are commented. So i m not using any
gateways.
On Fri, Sep 18, 2009 at 3:43 PM, pankaj anand pankajanan...@gmail.comwrote:
hi folks, I m not able to make SIP calls using external profile.
i have added the following lines to the
$installdir/conf/dialplan/public.xml
extension name=echo
condition field=destination_number expression
Re: [Freeswitch-users] Affordable GSM gateway for one cellphone?
btw, you can check this GW: http://www.edgepbx.cn/shop/index.php?controller=productproduct_id=12 i have it on my desk and it works as a charm... T. On Sat, Sep 19, 2009 at 1:47 PM, Alberto Escudero aep.li...@it46.se wrote: If you can wait a few weeks, it will be one :) available and documented. /aep -- Stopping junk mailers is good for the environment Hello I'm selling a basic solution for SOHO customers (FS is installed on their work computer running Windows or Macs) to handle an analog phone line. When they're on the road, in addition or instead of getting a notification by e-mail when someone calls their office, some users might want to have the Freeswitch server actually ring their cellphone so they can take calls. Besides taking a subscription with a VoIP provider that the Freeswitch server will use to ring their cellphone, I'd like to know what my options are when it comes to setting up a GSM gateway on the customer's premises, in case they don't want to depend on the Internet. Are there Freeswitch-compatible, affordable solutions to handle a single GSM subscription? I guess all it takes is having them take a second subscription with their GSM provider and inserting the SIM chip inside the gateway to have Freeswitch ring their cellphone, but I've never used those things. Thank you. -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25520404.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] Not able to make call using external profile
in other works, what are you trying to achieve? where do you want send calls? what is 192.168.1.50? where/how are you originating calls from? basically can you please tell us what is your call flow scenario otherwise we can't help? T. On Fri, Sep 18, 2009 at 4:15 PM, Brian West br...@freeswitch.org wrote: OK pay attention this time. See this line: 2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway found You sent a call out the profile the far side sent you a challenge since you're not calling via a gateway we can't answer the challenge because we do not know HOW. What is the far end you're calling? /b On Sep 18, 2009, at 9:11 AM, pankaj anand wrote: I m using default configuration of freeswitch.. I m not using any gateway for authentication. in the $INSTALLDIR/conf/sip_profiles/external/ directory, there exist only one file which example.xml , this files contains ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Some Newbie questions about dialplan and local Sip registration
hi Filip,
for calling a user... please read this first:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User
for making a GW register into e.g. asterisk please use this:
include
gateway name=gw01
param name=username value=USERNAME_ON_ASTERISK/
param name=realm value=ASTERISK_IP_ADDRESS/
param name=password value=PASSWORD_ON_ASTERISK/
param name=register value=true/
param name=caller-id-in-from value=true/
/gateway
/include
this should be enough to register the GW... after that please read this:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways
in your case it will be something like this:
extension name=dialGW
condition field=destination_number
expression=^(NUMBER_TO_SEND_TO_ASTERISK)$
action application=bridge data=sofia/gateway/gw01/$1/
/condition
/extension
On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker lync...@lyth.de wrote:
Hi List,
for the first experiments with freeswitch I downloaded the Windows
installation.
Now Im trying to get my 2 Sipphones get connected to. Later I want
connect the freeswitch to my asterisk gateway.
I find the examples pretty complex therfore Im trying to build up a
simple solution to understand the functions from the scratch ..
my current problem is , that I cant route my local sips to each other (
registration seems to work now).
the next is , that freeshwitch is not able to connect to asterisk. but I
will describe this later.
I installed in the Directory a xml file ( called 22.xml) with the
following content :
include
domain name=$${domain}
user id=22 mailbox=22
params
param name=password value=Xk21%/param
param name=vm-password value=22/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=22/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
22/variable
variable name=effective_caller_id_number value=22/variable
/variables
/user
user id=24 mailbox=24
params
param name=password value=dudeldum/param
param name=vm-password value=24/param
param name=sip-port value=5060/param
/params
variables
variable name=accountcode value=24/variable
variable name=user_context value=default/variable
variable name=effective_caller_id_name value=Extension
24/variable
variable name=effective_caller_id_number value=24/variable
/variables
/user
/domain
/include
This seems to be ok now. Now I want to dial from 22 to 24 , wherefore I
configured this dialplan :
include
context name=any
condition field=destination_number expression=^(2[0-9])$
action application=bridge data=user/${dialed_extension}/
/condition
/include
wich doesnt work , mybe b/c the user/${dialed_extension} I dont know...
Freeswitch says:
[INFO] switch_core_state_machine.c:136 No Route, Aborting
[NOTICE] switch_core_state_machine.c:137 Hangup
sofia/internal/2...@192.168.1.34 [CS_ROUTING] [NO_ROUTE_DESTINATION]
[NOTICE] switch_core_session.c:1086 Session 17
(sofia/internal/2...@192.168.1.34) Ended
[NOTICE] switch_core_session.c:1088 Close Channel
sofia/internal/2...@192.168.1.34 [CS_DESTROY]
Im sure , for you guys this cant be a big deal;)
Next Point is my Asterisk registration , mybe you can help me out here
to .. :
In the sip-profiles/external I installed the my_asterisk.xml with that
content :
include
gateway name=asterisk
param name=username value=28/param
param name=password value=test/param
param name=realm value=28/param
param name=proxy value=192.168.1.119/param
param name=register value=true/param
/gateway
/include
Freeswitch allways complains a timeout error :
[ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request
Timeout [408]. failure #17
[WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry
to 540 seconds.
it seems that It cant connect ( I also tried out to explicit set the
port to 5060 b/c I read something about 5080 .. : param name=sip-port
value=5060/param but this didnt help)
In my Asterisk I set in the sip.conf the entry 28 with the pw test
If someone could help me with my first steps I would be verrry thankful ;))
cheers
Filip
--
_
Filip Lyncker, Dipl.-Inform. (FH)
Lyncker Theis GmbH
Wilhelmstr. 16
65185 Wiesbaden
Germany
Fon +49 611/9006951
Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden
Steuernummer: 4023897051
USt-IdNr.: DE255806399
Geschäftsführer:
Filip Lyncker,
Armin Theis
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Re: [Freeswitch-users] how to add new user for external profile
FS loads all users from $INSTALL_DIR/conf/directory/ and you did it correct.
freeswitch.xml:
section name=directory description=User Directory
X-PRE-PROCESS cmd=include data=directory/*.xml/
Than, you need to check sip profiles. By default FS will accept
registrations on internal profiles only... so you should enable it on the
external as well.
look at this portion of your adequate sip profile:
!-- this lets anything register --
!-- comment the next line and uncomment one or both of the other 2
lines for call authentication --
!-- param name=accept-blind-reg value=true/ --
!-- accept any authentication without actually checking (not a good
feature for most people) --
!-- param name=accept-blind-auth value=true/ --
!-- suppress CNG on this profile or per call with the 'suppress_cng'
variable --
!-- param name=suppress-cng value=true/ --
!--TTL for nonce in sip auth--
param name=nonce-ttl value=60/
!--Uncomment if you want to force the outbound leg of a bridge to only
offer the codec
that the originator is using--
!--param name=disable-transcoding value=true/--
!-- Used for when phones respond to a challenged ACK with method INVITE
in the hash --
!--param name=NDLB-broken-auth-hash value=true/--
!-- add a ;received=ip:port to the contact when replying to
register for nat handling --
!--param name=NDLB-received-in-nat-reg-contact value=true/--
param name=auth-calls value=$${internal_auth_calls}/
!-- Force the user and auth-user to match. --
param name=inbound-reg-force-matching-username value=true/
!-- on authed calls, authenticate *all* the packets not just invite --
param name=auth-all-packets value=false/
!-- param name=ext-rtp-ip value=$${external_rtp_ip}/ --
!-- param name=ext-sip-ip value=$${external_sip_ip}/ --
!-- rtp inactivity timeout --
param name=rtp-timeout-sec value=300/
param name=rtp-hold-timeout-sec value=1800/
!-- VAD choose one (out is a good choice); --
!-- param name=vad value=in/ --
!-- param name=vad value=out/ --
!-- param name=vad value=both/ --
!--param name=alias value=sip:10.0.1.251:/--
!--
These are enabled to make the default config work better out of the
box.
If you need more than ONE domain you'll need to not use these
options.
--
!--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}/
!--force suscription expires to a lower value than requested--
!--param name=force-subscription-expires value=60/--
!-- disable register and transfer which may be undesirable in a public
switch --
!--param name=disable-transfer value=true/--
!--param name=disable-register value=true/--
Just make sure you use correct IP_ADDRESS:PORT to match the correct profile
vars.xml:
!-- Internal SIP Profile --
X-PRE-PROCESS cmd=set data=internal_auth_calls=true/
X-PRE-PROCESS cmd=set data=internal_sip_port=5060/
X-PRE-PROCESS cmd=set data=internal_tls_port=5061/
!-- External SIP Profile --
X-PRE-PROCESS cmd=set data=external_auth_calls=false/
X-PRE-PROCESS cmd=set data=external_sip_port=5080/
X-PRE-PROCESS cmd=set data=external_tls_port=5081/
T.
On Wed, Sep 16, 2009 at 11:29 AM, pankaj anand pankajanan...@gmail.comwrote:
hi , i m very new to the FreeSwitch..
can any one tell me how to add a new user.
i have already tried creating a new user by creating a
$INSTALL_DIR/conf/directory/default/pankaj.xml :
include
user id=pankaj
params
param name=password value=pankaj/
param name=vm-password value=pankaj/
/params
variables
variable name=toll_allow value=domestic,international,local/
variable name=accountcode value=pankaj/
variable name=user_context value=default/
variable name=effective_caller_id_name value=Extension pankaj/
variable name=effective_caller_id_number value=pankaj/
variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
variable name=callgroup value=techsupport/
/variables
/user
/include
but when i try to connect it using , the softphone shows forbidden.
Can anyone tell me where i am making a mistake.
with regards
Pankaj anand
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Re: [Freeswitch-users] reloadxml question
perfect, thanks. T. On Wed, Sep 16, 2009 at 4:05 PM, Brian West br...@freeswitch.org wrote: Yes you're missing a switch_xml_free(xml); some place. /b On Sep 16, 2009, at 8:49 AM, Tihomir Culjaga wrote: hi, I've build a custom module for FS and everytihng work well except reloadxml command :P... m'I missing something in my module? ... i used mod_skeleton as a template when i started. When i start the FS without my module reloadxml works fine ... as soon as i include my module within modules.conf.xml and start FS .. it hangs. So, it is definitelly up to the custom module ... but what can it be? freeswi...@l01freeswitch1 freeswi...@l01freeswitch1 freeswi...@l01freeswitch1 reloadxml nothing happens ... i have to kill freeswitch (kill -9) to get the shell. T. ___ 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
