Re: [Freeswitch-users] ClueCon Presentations - Where?
Just seen Anthony presentation, very cool ;) Everyone, watch it! http://files.freeswitch.org/cluecon_2009/presentations/Day%2001%20Presentation%2002.Anthony%20Minessale.1500kbps.mp4 =D On Wed, Aug 12, 2009 at 5:07 PM, Michael Collins m...@freeswitch.org wrote: On Wed, Aug 12, 2009 at 2:25 PM, Brian West br...@freeswitch.org wrote: They are all getting gathered up and put online... files.freeswitch.org/cluecon_2009 just keep an eye there some of the videos are up also. /b FYI, I've uploaded the first batch and they should get synched up on files.freeswitch.org/cluecon_2009/presentations any time... -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] Confused about conferences
Bradley Brashier wrote:
I wrote:
This is a significant new fact for me. What you seem to be doing is
calling the commands referenced in the conference api here
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
by using application=conference and then the data string as the second
part of the command. Am I correct in the assumption that you can do this.
I agree that that's what it looks like. What I don't know is if it
works. I got this example from the page
http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did
exactly what you're trying, and never tried using the API in this fashion.
I just found this - which I think helps
http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
An API can be called from the dialplan but it is not recommended. Example:
extension name=Make API call from Dialplan
condition field=destination_number expression=^(999)$
!-- next line calls hupall, so be careful! --
action application=set
data=api_result=${hupall(normal_clearing)}/
/condition
/extension
Anyway - thanks for you help - I am going away to rethink that
particular interface again. Its getting so complicated that it might be
better to copy the Javascript approach in the examples.
--
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [Freeswitch-users] answer command
Hey Michael, Just wondering something, I have found that you added conference_set_auto_outcall on the dptools wiki, but I could not find that function in the mod_dptools.c, shouldn't that be part of the mod_conference wiki article? =D. Best regards, Diego On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris m...@jerris.com wrote: Sip does not support this functionality. The called device would have to support this via some other mechanism such as ctsa which I have seen recently someone was looking at for freeswitch. So the first issue you must resolve is the called device needs to support some way to do this. Mike On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov maxim.tsve...@gmail.com wrote: If I have two FS extensions A and B. I'm calling from A to B and want to answer from B-side in my CTI application and to make SIP phone to be synchronised to my CTI application. Is it possible to do it? Brian West-3 wrote: Well you can only truly answer an inbound call to FS... you can't force answer an outbound call. /b On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote: I will try to paraphrase my question. Is there any possibility to answer call from CTI application and synchronise answer with answer in SIP client?Maybe we can use SIP functions in our CTI application instead of FS api commands? I'm trying to find the way to make prototype of lineAnswer command in TAPI. ___ 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 -- View this message in context: http://www.nabble.com/answer-command-tp24912812p24941422.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
[Freeswitch-users] Grangstream Early Dial
Hello all. I want to use Grandstream Early Dial future. How i can enable support 484 response? I tried simply use action application=hangup data=484/ and action application=respond data=484/ on uncompleted extensions, but there is not work Thanks. Yuriy . ___ 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] Transporting SIP over TCP
Hi, Googeling about this shows that FS aims to support this (in fact it supports all 3: UDP/TCP/TLS). Yet I could not find the way to configure FS in order to support that. In fact, it does not work in my current install. I have TLS configured and work, but could not make TCP works thanks in advance /tzury ___ 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] answer command
It probably belongs there. It's a wiki, feel free to fix it. What does this have to do with this thread? On Aug 13, 2009, at 4:03 AM, Diego Viola diego.vi...@gmail.com wrote: Hey Michael, Just wondering something, I have found that you added conference_set_auto_outcall on the dptools wiki, but I could not find that function in the mod_dptools.c, shouldn't that be part of the mod_conference wiki article? =D. Best regards, Diego On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris m...@jerris.com wrote: Sip does not support this functionality. The called device would have to support this via some other mechanism such as ctsa which I have seen recently someone was looking at for freeswitch. So the first issue you must resolve is the called device needs to support some way to do this. Mike On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov maxim.tsve...@gmail.com wrote: If I have two FS extensions A and B. I'm calling from A to B and want to answer from B-side in my CTI application and to make SIP phone to be synchronised to my CTI application. Is it possible to do it? Brian West-3 wrote: Well you can only truly answer an inbound call to FS... you can't force answer an outbound call. /b On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote: I will try to paraphrase my question. Is there any possibility to answer call from CTI application and synchronise answer with answer in SIP client?Maybe we can use SIP functions in our CTI application instead of FS api commands? I'm trying to find the way to make prototype of lineAnswer command in TAPI. ___ 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 -- View this message in context: http://www.nabble.com/answer-command-tp24912812p24941422.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 ___ 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] calling through same gateway with multiple registrations
Hi, I am new to FreeSWITCH and need an advice. All calls to PSTN from our server will go through single gateway, which is a soft switch supporting SIP protocol. FreeSWITCH will need to register with soft switch, but soft switch permits only single active call (in either direction) per registration. So we will need 10 SIP accounts to allow 10 simultaneous connections. Question is - how should I configure FreeSWITCH for this scenario? I see two options: 1) Create 10 gateways with different registrations, use mod_limit to route only one outgoing call per gateway; 2) Create 10 gateways with different registrations, use event socket to route calls manually and monitor used lines (incoming and outgoing calls through soft switch). Are there any other possibilities? Corrections/ suggestions are very welcome. -- Timur Irmatov, xmpp:irma...@jabber.ru ___ 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] Transporting SIP over TCP
It just works... to force TCP you append ;transport=tcp In reality you should be using SRV records. /b On Aug 13, 2009, at 3:40 AM, Tzury Bar Yochay wrote: Hi, Googeling about this shows that FS aims to support this (in fact it supports all 3: UDP/TCP/TLS). Yet I could not find the way to configure FS in order to support that. In fact, it does not work in my current install. I have TLS configured and work, but could not make TCP works thanks in advance /tzury ___ 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] Grangstream Early Dial
I don't think we ever got this working correctly. Can you do a trace of it working vs not working? /b On Aug 13, 2009, at 3:14 AM, Yuriy Ivzhenko wrote: Hello all. I want to use Grandstream Early Dial future. How i can enable support 484 response? I tried simply use action application=hangup data=484/ and action application=respond data=484/ on uncompleted extensions, but there is not work Thanks. Yuriy . ___ 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] Transporting SIP over TCP
It just works... to force TCP you append ;transport=tcp Hi Brian In fact this is exactly what I did and I could not get it work. I am using a console based application supplied by pjsip.org and when trying to register i get some error messages saying 'invalid transport SIP/2.0/tcp' and 'REGISTER (30669) has invalid Via' using the very same client against iptel.org seems to work. In reality you should be using SRV records. can you please elaborate a bit more about this? I am dumping below the cli output. thanks in advance for your time and attention tport_wakeup(0x7fd82c2afaf0): events IN tport_recv_event(0x7fd82c2afaf0) tport_recv_iovec(0x7fd82c2afaf0) msg 0x7fd82c2a1830 from (tcp/80.74.97.189:42634) has 472 bytes, veclen = 1 tport_deliver(0x7fd82c2afaf0): msg 0x7fd82c2a1830 (472 bytes) from tcp/80.74.97.189:42634/sip next=(nil) nta: received REGISTER sip:cheerfulsanity.net;transport=tcp SIP/2.0 (CSeq 30669) nta: Via check: invalid transport SIP/2.0/tcp from 80.74.97.189:42634 nta: REGISTER (30669) has invalid Via tport(0x7fd82c2afaf0): reset timer tport(0x7fd82c2afaf0): set timer at 180 ms because idle since recv tport_wakeup_pri(0x713dd0): events IN tport_recv_event(0x713dd0) tport_recv_iovec(0x713dd0) msg 0x7fd82c2a1830 from (udp/67.23.5.142:5060) has 2 bytes, veclen = 1 tport_deliver(0x713dd0): bad msg 0x7fd82c2a1830 (2 bytes) from udp/199.245.214.130:5060/sip next=(nil) tport_wakeup_pri(0x713dd0): events IN tport_recv_event(0x713dd0) tport_recv_iovec(0x713dd0) msg 0x7fd82c2a1830 from (udp/67.23.5.142:5060) has 2 bytes, veclen = 1 tport_deliver(0x713dd0): bad msg 0x7fd82c2a1830 (2 bytes) from udp/199.245.214.130:5060/sip next=(nil) tport_wakeup(0x7fd84027d7d0): events IN tport_recv_event(0x7fd84027d7d0) ___ 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] Transporting SIP over TCP
I need to see the sip packet. TCP should be uppercase I'm pretty sure. /b On Aug 13, 2009, at 9:04 AM, Tzury Bar Yochay wrote: nta: Via check: invalid transport SIP/2.0/tcp from 80.74.97.189:42634 nta: REGISTER (30669) has invalid Via ___ 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] bind_server_ip issue
Hello! First of all, I would like to express my thanks to all the developers of Freeswitch. I am testing Freeswitch on a Debian machine with physical network interface with four virtual IP addresses. One of these IP addresses, aliased as eth0:3, has been created specifically for Freeswitch. I then set bind_server_ip with the IP addresses associated with eth0:3. To my surprise, however, tow things happen more or less randomly: 1) in certain cases, Freeswitch binds to eth0:2 instead (with a different IP address); and in another, although Freeswitch binds initially to eth0:3, after a few hours it changes its mind and rebinds to eth0:2. Is this an issue with bind_server_ip or am I missing some configuration detail? Thanks! Carlos ___ 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] bind_server_ip issue
If you read the latest vars.xml I have clarified this: !-- THIS IS ONLY USED FOR DINGALING bind_server_ip Can be an ip address, a dns name, or auto. This determines an ip address available on this host to bind. If you are separating RTP and SIP traffic, you will want to have use different addresses where this variable appears. Used by: dingaling.conf.xml -- X-PRE-PROCESS cmd=set data=bind_server_ip=auto/ So you'll need to open up the sip profile in sip_profiles and set the bind ip to exactly what you want. Thanks, Brian On Aug 13, 2009, at 9:14 AM, Carlos S. Antunes wrote: Hello! First of all, I would like to express my thanks to all the developers of Freeswitch. I am testing Freeswitch on a Debian machine with physical network interface with four virtual IP addresses. One of these IP addresses, aliased as eth0:3, has been created specifically for Freeswitch. I then set bind_server_ip with the IP addresses associated with eth0:3. To my surprise, however, tow things happen more or less randomly: 1) in certain cases, Freeswitch binds to eth0:2 instead (with a different IP address); and in another, although Freeswitch binds initially to eth0:3, after a few hours it changes its mind and rebinds to eth0:2. Is this an issue with bind_server_ip or am I missing some configuration detail? Thanks! Carlos ___ 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] Transporting SIP over TCP
On Aug 13, 2009, at 9:37 AM, Tzury Bar Yochay wrote: I need to see the sip packet. dumped below TCP should be uppercase I'm pretty sure. you mean the via should be Via: SIP/2.0/TCP right? Yep If so, then that would a bug in the client then. Some things might accept it but sofia is usually strict about some of this stuff. /b ___ 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] Setting max inbound for UA
Thanks Ken. I'll look at mod_limit The XLite softphone doesn't seem to have a switch for controlling it. -str On Aug 12, 2009, at 9:22 PM, Ken Rice wrote: Check out mod_limit... Other wise you'll have to look specifically at the UA you are trying to use, some like polycom and sipura offer a way to disable call waiting Remember with SIP there is no such thing as a line, its a SESSION and you can have as many sessions as the software allows (and most software doesn't put sane limits based on CPU/RAM/Bandwidth etc) From: String Larson strin...@gmail.com Reply-To: freeswitch-users@lists.freeswitch.org Date: Wed, 12 Aug 2009 19:42:02 -0500 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Setting max inbound for UA Is there a way to limit the number of calls a UA can receive in the FS configs? I'm doing some testing with XLite as the UA, and can not figure out how to keep line 2 from answering if line 1 is in use. THanks. -str ___ 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] Question about sharing conference between
Thank you Michael, I will tinker around with it and definitely follow-up with the results. - 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] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
On Wed, Aug 12, 2009 at 7:19 PM, Brian West br...@freeswitch.org wrote: Well you really can't ignore it... it happens with our ISDN stack too. Thats what the VETO handles. /b You lost me. What do you mean we can't ignore it? the way I see it, sure we can and we should. Currently that warning comes from the on_ringing() callback which blindly attempts to move the state of the zap channel to ZAP_CHANNEL_STATE_PROGRESS, even when the state may be already ZAP_CHANNEL_STATE_PROGRESS_MEDIA (which means on_proceed() was called first). As I see it, the VETO warning is more an aid to the programmer so you quickly realize your doing a useless state change, which should be fixed. In this case, the fix is simply checking the state of the channel before trying to move it to progress, and don't even try to move it if already in progress with media. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Grangstream Early Dial
On Thursday 13 August 2009 16:47:18 Brian West wrote: I don't think we ever got this working correctly. Can you do a trace of it working vs not working? I can't do working trace, only not working http://pastebin.freeswitch.org/9980 with dialplan action action application=hangup data=INVALID_NUMBER_FORMAT/ /b On Aug 13, 2009, at 3:14 AM, Yuriy Ivzhenko wrote: Hello all. I want to use Grandstream Early Dial future. How i can enable support 484 response? I tried simply use action application=hangup data=484/ and action application=respond data=484/ on uncompleted extensions, but there is not work Thanks. Yuriy . ___ 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] Confused about conferences
So it sounds like set can work. But you'd still have to parse it. And even
then it's not recommended.
I have another couple of possible methods for you:
1) modification of mod_conference.
2) event socket.
If you modify mod_conference, you can probably do what you want, but it
obviously requires using C and modifying existing code.
If you use the event socket, you've got a bigger learning curve, perhaps,
but you can use a variety of languages, your code is separate (and therefore
easier to maintain), and you then know how the event socket works in case
you need to do something else later.
Good luck with whatever you end up doing.
BB
On Thu, Aug 13, 2009 at 12:11 AM, Alan Chandler
a...@chandlerfamily.org.ukwrote:
Bradley Brashier wrote:
I wrote:
This is a significant new fact for me. What you seem to be doing is
calling the commands referenced in the conference api here
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
by using application=conference and then the data string as the
second
part of the command. Am I correct in the assumption that you can do
this.
I agree that that's what it looks like. What I don't know is if it
works. I got this example from the page
http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did
exactly what you're trying, and never tried using the API in this
fashion.
I just found this - which I think helps
http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
An API can be called from the dialplan but it is not recommended. Example:
extension name=Make API call from Dialplan
condition field=destination_number expression=^(999)$
!-- next line calls hupall, so be careful! --
action application=set
data=api_result=${hupall(normal_clearing)}/
/condition
/extension
Anyway - thanks for you help - I am going away to rethink that
particular interface again. Its getting so complicated that it might be
better to copy the Javascript approach in the examples.
--
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [Freeswitch-users] calling through same gateway with multiple registrations
On Thu, Aug 13, 2009 at 2:35 AM, Timur Irmatov irma...@gmail.com wrote: Hi, I am new to FreeSWITCH and need an advice. All calls to PSTN from our server will go through single gateway, which is a soft switch supporting SIP protocol. FreeSWITCH will need to register with soft switch, but soft switch permits only single active call (in either direction) per registration. So we will need 10 SIP accounts to allow 10 simultaneous connections. Are you going to have incoming calls as well? If so, how does the soft-switch handle two concurrent calls to the same number? Question is - how should I configure FreeSWITCH for this scenario? I see two options: 1) Create 10 gateways with different registrations, use mod_limit to route only one outgoing call per gateway; 2) Create 10 gateways with different registrations, use event socket to route calls manually and monitor used lines (incoming and outgoing calls through soft switch). Are there any other possibilities? Corrections/ suggestions are very welcome. This seems like a serious defect in the soft-switch. I can understand if it allows you to specify only one call per SIP registration, but to hard-code that limit seems pretty silly. Can you find out more about the soft-switch in question and see if that limitation is flexible? -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] Conference silence timeouts
Hi all. The solution to this one should be short. My conference hangs up when there's 2+ users and silence for 5 sec or so. I'm trying to find a parameter that changes that (I'd rather it be, say, 60 seconds). I didn't see a parameter like this specific to conferences, so I looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300 (the default), so I'm pretty sure that's not the problem. I also searched through the mod_conference.c code and didn't see it, though I was only skimming. I'm not 100% convinced that this is limited to conferences, but I don't currently have a way to test in a non-conference environment. Anybody know how to increase the conference silence-hangup timeout? BB ___ 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] Conference silence timeouts
Check out the 'waste' member flag. I think if at least one member has that set then RTP will get sent out even during silence. Let us know if that helps... -MC On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier bjbrash...@gmail.comwrote: Hi all. The solution to this one should be short. My conference hangs up when there's 2+ users and silence for 5 sec or so. I'm trying to find a parameter that changes that (I'd rather it be, say, 60 seconds). I didn't see a parameter like this specific to conferences, so I looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300 (the default), so I'm pretty sure that's not the problem. I also searched through the mod_conference.c code and didn't see it, though I was only skimming. I'm not 100% convinced that this is limited to conferences, but I don't currently have a way to test in a non-conference environment. Anybody know how to increase the conference silence-hangup timeout? BB ___ 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] Conference silence timeouts
I'm sure that would work, but I'm worried about it sucking up bandwidth, especially since you'd need it on every caller (since otherwise the one person who had it could hang up and you'd be back to square 1). Any other ideas, or should I hunt through the code to try to override the behavior manually? BB On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins m...@freeswitch.org wrote: Check out the 'waste' member flag. I think if at least one member has that set then RTP will get sent out even during silence. Let us know if that helps... -MC On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier bjbrash...@gmail.com wrote: Hi all. The solution to this one should be short. My conference hangs up when there's 2+ users and silence for 5 sec or so. I'm trying to find a parameter that changes that (I'd rather it be, say, 60 seconds). I didn't see a parameter like this specific to conferences, so I looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300 (the default), so I'm pretty sure that's not the problem. I also searched through the mod_conference.c code and didn't see it, though I was only skimming. I'm not 100% convinced that this is limited to conferences, but I don't currently have a way to test in a non-conference environment. Anybody know how to increase the conference silence-hangup timeout? BB ___ 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] Confused about conferences
I couldn't imagine managing a conference without a GUI. I need to see who is
making noise so I can boot/mute em ;)
If I were you I would dive into ESL and make a simple web app to frontend
the conferences. There will surely be something in contrib to get you
started.
On Thu, Aug 13, 2009 at 8:48 AM, Bradley Brashier bjbrash...@gmail.comwrote:
So it sounds like set can work. But you'd still have to parse it. And
even then it's not recommended.
I have another couple of possible methods for you:
1) modification of mod_conference.
2) event socket.
If you modify mod_conference, you can probably do what you want, but it
obviously requires using C and modifying existing code.
If you use the event socket, you've got a bigger learning curve, perhaps,
but you can use a variety of languages, your code is separate (and therefore
easier to maintain), and you then know how the event socket works in case
you need to do something else later.
Good luck with whatever you end up doing.
BB
On Thu, Aug 13, 2009 at 12:11 AM, Alan Chandler
a...@chandlerfamily.org.uk wrote:
Bradley Brashier wrote:
I wrote:
This is a significant new fact for me. What you seem to be doing is
calling the commands referenced in the conference api here
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
by using application=conference and then the data string as the
second
part of the command. Am I correct in the assumption that you can do
this.
I agree that that's what it looks like. What I don't know is if it
works. I got this example from the page
http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did
exactly what you're trying, and never tried using the API in this
fashion.
I just found this - which I think helps
http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
An API can be called from the dialplan but it is not recommended. Example:
extension name=Make API call from Dialplan
condition field=destination_number expression=^(999)$
!-- next line calls hupall, so be careful! --
action application=set
data=api_result=${hupall(normal_clearing)}/
/condition
/extension
Anyway - thanks for you help - I am going away to rethink that
particular interface again. Its getting so complicated that it might be
better to copy the Javascript approach in the examples.
--
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [Freeswitch-users] answer command
Err, I asked if that was wrong to fix it. On Thu, Aug 13, 2009 at 2:15 PM, Diego Viola diego.vi...@gmail.com wrote: I was talking with Michael about fixing stuff in the wiki, so I just asked to fix that also. On Thu, Aug 13, 2009 at 9:10 AM, Michael Jerris m...@jerris.com wrote: It probably belongs there. It's a wiki, feel free to fix it. What does this have to do with this thread? On Aug 13, 2009, at 4:03 AM, Diego Viola diego.vi...@gmail.com wrote: Hey Michael, Just wondering something, I have found that you added conference_set_auto_outcall on the dptools wiki, but I could not find that function in the mod_dptools.c, shouldn't that be part of the mod_conference wiki article? =D. Best regards, Diego On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris m...@jerris.com m...@jerris.com wrote: Sip does not support this functionality. The called device would have to support this via some other mechanism such as ctsa which I have seen recently someone was looking at for freeswitch. So the first issue you must resolve is the called device needs to support some way to do this. Mike On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov maxim.tsve...@gmail.com maxim.tsve...@gmail.com wrote: If I have two FS extensions A and B. I'm calling from A to B and want to answer from B-side in my CTI application and to make SIP phone to be synchronised to my CTI application. Is it possible to do it? Brian West-3 wrote: Well you can only truly answer an inbound call to FS... you can't force answer an outbound call. /b On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote: I will try to paraphrase my question. Is there any possibility to answer call from CTI application and synchronise answer with answer in SIP client?Maybe we can use SIP functions in our CTI application instead of FS api commands? I'm trying to find the way to make prototype of lineAnswer command in TAPI. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.orghttp://www.freeswitch.org -- View this message in context: http://www.nabble.com/answer-command-tp24912812p24941422.html http://www.nabble.com/answer-command-tp24912812p24941422.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.orghttp://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.orghttp://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://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] calling through same gateway with multiple registrations
Thanks for responding, Michael! On Thu, Aug 13, 2009 at 9:06 PM, Michael Collinsm...@freeswitch.org wrote: Are you going to have incoming calls as well? If so, how does the soft-switch handle two concurrent calls to the same number? Yes, we'll have incoming calls as well. I did not performed any tests myself yet, but a guy working with this soft-switch says it just rejects second call. So soft-switch treats each sip registration more or less like single phone line. Soft-switch is Huawei SoftX3000, if somebody experienced in it can prove me wrong, I'll be glad.. :) Question is - how should I configure FreeSWITCH for this scenario? I see two options: 1) Create 10 gateways with different registrations, use mod_limit to route only one outgoing call per gateway; 2) Create 10 gateways with different registrations, use event socket to route calls manually and monitor used lines (incoming and outgoing calls through soft switch). Are there any other possibilities? Corrections/ suggestions are very welcome. This seems like a serious defect in the soft-switch. I can understand if it allows you to specify only one call per SIP registration, but to hard-code that limit seems pretty silly. Can you find out more about the soft-switch in question and see if that limitation is flexible? Yes, that was my initial thought too - that this limit should be configurable. But a guy working with it says it is not configurable, and local support engineers from Huawei also confirm this. But, who knows, may be they are all wrong. -- Timur Irmatov, xmpp:irma...@jabber.ru ___ 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_managed users?
Hey Michael, I am a little late to the party I know - but just want to say thanks for your latest efforts. I updated my dev environment with the latest managed mod and swapped my app to the latest plugin architecture last night and all is working well. Love the dynamic loading of my dll into freeswitch - no more starting and stopping freeswitch! Also appreciate the f# example. Thanks again. Phillip Jones On Wed, Jul 29, 2009 at 8:04 PM, Michael Giagnocavom...@giagnocavo.net wrote: Which directory ends with ; ? I’m not following – if you want email me off list and we can work together on it . From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Diego Toro Sent: Wednesday, July 29, 2009 5:06 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? Hi, My dll's are loaded correctly, I have a trouble becouse the directory where are loaded it finish with ; (whitout ), i mean the default path assembly finish with ; Is possible remove the character ; ? thanks Diego --- On Wed, 7/29/09, Michael Giagnocavo m...@giagnocavo.net wrote: From: Michael Giagnocavo m...@giagnocavo.net Subject: Re: [Freeswitch-users] mod_managed users? To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org Date: Wednesday, July 29, 2009, 2:32 PM The error is that it just doesn’t find that alias to call the plugin. Assuming everything is spelled correctly, this probably means the DLL did not load. I just checked in a fix for dlls that don’t have both Api and App interfaces – it would not load them at all. Try with it now and see if that’s the problem. If it is, I apologize. If it still doesn’t, paste the full log of when it loads your file. Thanks, Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Diego Toro Sent: Wednesday, July 29, 2009 12:44 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? Hi Michael, I am working with lastest managed module, I have assemblies wich dependen of others assemblies (dll's), the past version it works fine, but I have now next the error message: EXECUTE sofia/internal/10...@192.168.27.10 managed(CIV_BPFSProcess) 2009-07-29 13:31:27.718750 [DEBUG] switch_cpp.cpp:1130 FreeSWITCH.Managed: attempting to run application 'CIV_BPFSProcess'. 2009-07-29 13:31:27.718750 [ERR] switch_cpp.cpp:1130 App plugin CIV_BPFSProcess not found. 2009-07-29 13:31:27.718750 [ERR] mod_managed.cpp:405 Application run failed for CIV_BPFSProcess (unknown module or exception). My declaration class is: public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin It has public method: public void Run(AppContext context). I have VS2008 on Windows Thanks Diego --- On Wed, 7/29/09, Michael Giagnocavo m...@giagnocavo.net wrote: From: Michael Giagnocavo m...@giagnocavo.net Subject: Re: [Freeswitch-users] mod_managed users? To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org Date: Wednesday, July 29, 2009, 1:46 AM Hi Łukasz, Would you please send me the DLL offlist and I'll figure it out? The new session you create is the b-leg. The parameter it takes in originate is the a-leg. So you'd do: var session = new ManagedSession(); session.Originate(context.Session, sofia/default/1000,10); As to non-blocking, I'm quite sure it's possible, but I don't recall offhand which functions. This should be the same as in any other language for FreeSWITCH -- these functions are just passthrough from the FS C++ API. -Michael -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lukasz Zwierko Sent: Wednesday, July 29, 2009 12:13 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? Hi Michael , thanks a lot for support on this. As to the main problem of your DLL not working, can you send me the full source code, or all the logging output from loading it? Try managedreload my.dll to reload the DLL and see how it is registering them. It should output something like Registering API FullName with Aliases fullname, shortname. I'm just using the Demo.cs example, I compile it to dll undef VC#, not mono, maybe that is the difference? The output from the log is just as you stated Registering API FullName with Aliases fullname, shortname. The difference between loading dll and csx is that, when loading csx all api and app classes are listed as registered, while with dll nothing is listed.. Anyway, I have another question regarding usage of the CoreSession and ManagedSession object. Basically in my script I want to start new session and originate a call. So what I do is
Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
Hi Michal, Just checking in to see if you've been able to take a stab at this. Thanks, Vladimir -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michal Bielicki Sent: Tuesday, August 11, 2009 5:33 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86 I'll retst it later today and give you a link with instructions Am 10.08.2009 um 20:14 schrieb vmorales: By ./compile I was referring to ./configure Vladimir -Original Message- From: vmorales [mailto:email.list.subscri...@gmail.com] Sent: Monday, August 10, 2009 11:49 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86 Thanks for the response(s): I ran the ./compile script with a set PREFIX. This took a few attempts with errors before it was able to complete error-free, as I had to install libtool. Since then, I have tried running 'make', 'gmake', and '/opt/gnu/bin/make', but each results with an error. This is the error when running 'make' or 'gmake': snip make: Fatal error: Command failed for target `all-recursive' Current working directory /home/vmorales/freeswitch-1.0.4 *** Error code 1 make: Fatal error: Command failed for target `all' /snip This is the error when running '/opt/gnu/bin/make': snip make[5]: *** [mod_amr.so] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_amr-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /opt/gnu/bin/make install + +--+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 /snip I re-untar'd before each compile/make attempt. Let me know if this is something that I can resolve. Vladimir -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Saturday, August 08, 2009 12:37 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86 This is not currently a supported platform, it only builds on 64 bit right now I think on solaris. Mike On Aug 6, 2009, at 6:03 PM, vmorales wrote: Hello, Does anyone have, or know where to get, a pre-compiled copy of FreeSwitch for Solaris 10/x86? ___ 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-use rs 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-use rs http://www.freeswitch.org Michal Bielicki Leiter der Niederlassung HaloKwadrat Sp. z o.o. Niederlassung Kleinmachnow Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P Ust.Id.: DE261885536 Geschaeftsfuehrer: Aleksander Wiercinski Meiereifeld 2b, 14532 Kleinmachnow t. +49 33203 263220 f. +49 33203 263229 sip. i...@halokwadrat.de e. michal.bieli...@halokwadrat.de | w. www.halokwadrat.de Hauptgeschäftsstelle: Halo Kwadrat Sp. z o.o. ul. Polna 46/14 00-644 Warszawa, Polen EIngetragen im HRB Warszawa, KRS 153539 ___ 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] Sangoma A102 Overrun Issue
I've been trying to bridge FreeSWITCH with a Toshiba CIX using qsig over a PRI. I have a Sangoma A102 card installed in a Dell PowerEdge with CentOS 5.3. The issue I am having is no packets are being transmitted back to FreeSWITCH. ifconfig w1g1 shows every frame received as an overrun. I've tried a different server with CentOS 5.3 with the same issue. I have a support ticket in with Sangoma, but was wondering if anybody had seen this before. The T1 shows connected so I think I have the Toshiba configured properly. From what I've read the overrun has to deal with the driver not reading the data in time so maybe this is a CentOS 5.3 specific issue. Any recommendations on alternative Linux distros known to work with the Sangoma A102 card? Thanks, Ryan [r...@voip ~]# ifconfig w1g1 w1g1 Link encap:Point-to-Point Protocol UP POINTOPOINT RUNNING NOARP MTU:80 Metric:1 RX packets:22298 errors:0 dropped:0 overruns:274 frame:274 TX packets:22298 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:100 RX bytes:1783840 (1.7 MiB) TX bytes:1783840 (1.7 MiB) Interrupt:169 Memory:f8e8-f8e81fff [r...@voip ~]# wanrouter status Devices currently active: wanpipe1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1| N/A | A101/1D/A102/2D/4/4D/8| 169 | 4 | 1 | N/A | 0 | Wanrouter Status: Device name | Protocol | Station | Status| wanpipe1| AFT TE1 | N/A | Connected | [r...@voip ~]# wanpipemon -i w1g1 -c Ta * w1g1: T1 Alarms (Framer) * ALOS: OFF | LOS: OFF RED:OFF | AIS: OFF RAI:OFF | OOF: OFF * w1g1: T1 Alarms (LIU) * Short Circuit: OFF Open Circuit: OFF Loss of Signal: OFF * w1g1: T1 Performance Monitoring Counters * Line Code Violation : 45 Bit Errors (CRC6/Ft/Fs) : 0 Out of Frame Errors : 0 Rx Level: -2.5db ___ 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] Conference silence timeouts
On Thu, Aug 13, 2009 at 12:47 PM, Bradley Brashier bjbrash...@gmail.comwrote: I'm sure that would work, but I'm worried about it sucking up bandwidth, especially since you'd need it on every caller (since otherwise the one person who had it could hang up and you'd be back to square 1). Any other ideas, or should I hunt through the code to try to override the behavior manually? BB Get a packet capture and debug trace of the symptom occurring. Put the PCAP where we can download it and pastebin the debug log. We need to confirm who the culprit is and what events precipitate the disconnect. I'd also be curious to know if anyone else can reproduce these symptoms. BTW, which rev of FS are you running? Thanks, 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
Re: [Freeswitch-users] Conference silence timeouts
My guess is that its the other end killing the call due to rtp timeouts, not us killing the call. If you can confirm this the best method would be to get them not to do rtp timeouts. On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote: I'm sure that would work, but I'm worried about it sucking up bandwidth, especially since you'd need it on every caller (since otherwise the one person who had it could hang up and you'd be back to square 1). Any other ideas, or should I hunt through the code to try to override the behavior manually? BB On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins m...@freeswitch.org wrote: Check out the 'waste' member flag. I think if at least one member has that set then RTP will get sent out even during silence. Let us know if that helps... -MC On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier bjbrash...@gmail.com wrote: Hi all. The solution to this one should be short. My conference hangs up when there's 2+ users and silence for 5 sec or so. I'm trying to find a parameter that changes that (I'd rather it be, say, 60 seconds). I didn't see a parameter like this specific to conferences, so I looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300 (the default), so I'm pretty sure that's not the problem. I also searched through the mod_conference.c code and didn't see it, though I was only skimming. I'm not 100% convinced that this is limited to conferences, but I don't currently have a way to test in a non-conference environment. Anybody know how to increase the conference silence-hangup timeout? BB _ ___ 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] Conference silence timeouts
I'm currently running current trunk synched up Tues morning, but it was happening in all of the versions I'd been using previous -- I first downloaded around the end of May. I'll look into getting you a PCap. I expected that this was a known thing with a parameter somewhere, so I haven't looked into it too terribly far myself, yet. I'm gonna try looking at the console outputs and logs myself, first. BB On Thu, Aug 13, 2009 at 12:44 PM, Michael Collins m...@freeswitch.orgwrote: On Thu, Aug 13, 2009 at 12:47 PM, Bradley Brashier bjbrash...@gmail.comwrote: I'm sure that would work, but I'm worried about it sucking up bandwidth, especially since you'd need it on every caller (since otherwise the one person who had it could hang up and you'd be back to square 1). Any other ideas, or should I hunt through the code to try to override the behavior manually? BB Get a packet capture and debug trace of the symptom occurring. Put the PCAP where we can download it and pastebin the debug log. We need to confirm who the culprit is and what events precipitate the disconnect. I'd also be curious to know if anyone else can reproduce these symptoms. BTW, which rev of FS are you running? Thanks, 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] Conference silence timeouts
I took a closer look at the SIP messages on the console. From it, I understand that it's not Freeswitch timing out, but rather FS is getting the BYE msg from somewhere else. I've tested phones and tested calling without going through the FS conference, though, and everything works fine. Then I saw something else odd inside the BYE msg: Reason: Q.850 ;cause=31 ;text=RTP Broken Connection So I Googled RTP Broken Connection and saw several sites talking about AudioCodes gateways and Asterisk -- and our gateway is an AudioCodes. From these sites it sounds like AudioCodes is rather aggressive in detecting RTP breaks, and is interpreting the silence from FS as a break. So now I'm looking into ways to maybe send I'm still here RTP packets from FS or to tune the gateway to be less aggressive. I can't stop and get a clean packet capture right now because I've got a bunch of testers working on it today. I'll do that sometime when the system is less busy. BB On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier bjbrash...@gmail.comwrote: I had just thought of the exact same thing. I'm trying to test that now. Thanks for your input. BB On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris m...@jerris.com wrote: My guess is that its the other end killing the call due to rtp timeouts, not us killing the call. If you can confirm this the best method would be to get them not to do rtp timeouts. On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote: I'm sure that would work, but I'm worried about it sucking up bandwidth, especially since you'd need it on every caller (since otherwise the one person who had it could hang up and you'd be back to square 1). Any other ideas, or should I hunt through the code to try to override the behavior manually? BB On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins m...@freeswitch.orgwrote: Check out the 'waste' member flag. I think if at least one member has that set then RTP will get sent out even during silence. Let us know if that helps... -MC On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier bjbrash...@gmail.com wrote: Hi all. The solution to this one should be short. My conference hangs up when there's 2+ users and silence for 5 sec or so. I'm trying to find a parameter that changes that (I'd rather it be, say, 60 seconds). I didn't see a parameter like this specific to conferences, so I looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300 (the default), so I'm pretty sure that's not the problem. I also searched through the mod_conference.c code and didn't see it, though I was only skimming. I'm not 100% convinced that this is limited to conferences, but I don't currently have a way to test in a non-conference environment. Anybody know how to increase the conference silence-hangup timeout? BB _ ___ 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] Conference silence timeouts
OK, I finally got a moment to do a packet capture and take a look at the streams. It became very clear very quickly that what happens is that during silence the gateway still sends RTP packets to Freeswitch, but Freeswitch doesn't send any back to the gateway. After 10s of this, the gateway says Oh, the RPT must be broken and it hangs up. We found a way to turn off this behavior in the gateway, and the good news is that it did indeed fix the problem. But we'd rather not rely on that as a long-term solution because then we can't detect and drop RTP streams that really are broken. So now I'm back to looking at Freeswitch to figure out how to send just a single packet every second or so during silence. If anyone knows of a way to do this, let me know, otherwise I'll get back to you if and when I find one. BB On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier bjbrash...@gmail.comwrote: I took a closer look at the SIP messages on the console. From it, I understand that it's not Freeswitch timing out, but rather FS is getting the BYE msg from somewhere else. I've tested phones and tested calling without going through the FS conference, though, and everything works fine. Then I saw something else odd inside the BYE msg: Reason: Q.850 ;cause=31 ;text=RTP Broken Connection So I Googled RTP Broken Connection and saw several sites talking about AudioCodes gateways and Asterisk -- and our gateway is an AudioCodes. From these sites it sounds like AudioCodes is rather aggressive in detecting RTP breaks, and is interpreting the silence from FS as a break. So now I'm looking into ways to maybe send I'm still here RTP packets from FS or to tune the gateway to be less aggressive. I can't stop and get a clean packet capture right now because I've got a bunch of testers working on it today. I'll do that sometime when the system is less busy. BB On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier bjbrash...@gmail.comwrote: I had just thought of the exact same thing. I'm trying to test that now. Thanks for your input. BB On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris m...@jerris.comwrote: My guess is that its the other end killing the call due to rtp timeouts, not us killing the call. If you can confirm this the best method would be to get them not to do rtp timeouts. On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote: I'm sure that would work, but I'm worried about it sucking up bandwidth, especially since you'd need it on every caller (since otherwise the one person who had it could hang up and you'd be back to square 1). Any other ideas, or should I hunt through the code to try to override the behavior manually? BB On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins m...@freeswitch.orgwrote: Check out the 'waste' member flag. I think if at least one member has that set then RTP will get sent out even during silence. Let us know if that helps... -MC On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier bjbrash...@gmail.com wrote: Hi all. The solution to this one should be short. My conference hangs up when there's 2+ users and silence for 5 sec or so. I'm trying to find a parameter that changes that (I'd rather it be, say, 60 seconds). I didn't see a parameter like this specific to conferences, so I looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300 (the default), so I'm pretty sure that's not the problem. I also searched through the mod_conference.c code and didn't see it, though I was only skimming. I'm not 100% convinced that this is limited to conferences, but I don't currently have a way to test in a non-conference environment. Anybody know how to increase the conference silence-hangup timeout? BB _ ___ 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] Loopback and bypass_media
Rupa / all, Just a quick follow up to this. This is appears to a timing issue. If I try and do a uuid_media off + uuid in api_after_bridge it fails with CHAN_NOT_IMPLEMENTED and the call is dropped. If appears to be trying to do a SIP reinvite on the loopback channel which is of course just about to / has disappear/ed. So I tried this, after the call is established, at the commend line, I do show calls and using the uuid shown, type uuid_media off uuid. The SIP REINVITE is issued and works. I think the switch_ivr_nomedia function in switch_ivr_c is getting the loopback uuid when it calls other_uuid = switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE) That's why the SIP REINVITE fails. So... in api_after_bridge I issue: sched_api, +3 none uuid_media off + uuid. This calls the switch_ivr_nomedia function 3 seconds after the calls bridge is established. And it works, Not nice - not scalable or production ready - but the SIP-REINVITE is successful and at least now I understand what is going on. Make sense? Thanks Phil On Wed, Aug 12, 2009 at 12:21 PM, Rupa Schomakerr...@rupa.com wrote: On Wed, Aug 12, 2009 at 10:22 AM, Phillip Jonespjinthe...@gmail.com wrote: Hi there, application=originate data=(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum) I agree. However, perhaps the ideal is not to specify the carriers at this level, as carriers are added and removed fairly often as costings change. So it would be nice to have some sort of proxy that resolves to a list of carriers: application=originate data=sofia/MyCarriers/bar,sofia/MyCarriers/yum MyCarriers action application=carrier data=sofia/foo/ action application=carrier data=sofia/baz/ action application=carrier data=sofia/etc/ /MyCarriers or something similar. This would achieve the same as loopback in this use case but without dangers of looping? Complicated stuff though. Well, that is all done by mod_lcr. I was simplifying to narrow down to just originate. First we need to see if this is worth pursuing over fixing (modifying, whatever) loopback to handle bypass media. If it is, then I'll modify mod_lcr to deal with the situation in question (comma or pipe sep list of numbers to call. mod_lcr would then group as appropriate). Right now, my bridge is setup in a small javascript script that builds the appropriate dialstring (using loopback for external calls, user/ for internal calls) and then when doing the loopback call to mod_lcr to get the dialstring with all providers in the right order. Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup? That's a good idea - I will look into that. Thanks again. Phillip Let us know how it works for you... -- -Rupa ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
It probably just VETO it so it avoid sending SWITCH_MESSAGE_INDICATE_PROGRESS again since the call is already making progress from the core's point of view? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 13-Aug-09, at 11:02 AM, Moises Silva wrote: On Wed, Aug 12, 2009 at 7:19 PM, Brian West br...@freeswitch.org wrote: Well you really can't ignore it... it happens with our ISDN stack too. Thats what the VETO handles. /b You lost me. What do you mean we can't ignore it? the way I see it, sure we can and we should. Currently that warning comes from the on_ringing() callback which blindly attempts to move the state of the zap channel to ZAP_CHANNEL_STATE_PROGRESS, even when the state may be already ZAP_CHANNEL_STATE_PROGRESS_MEDIA (which means on_proceed() was called first). As I see it, the VETO warning is more an aid to the programmer so you quickly realize your doing a useless state change, which should be fixed. In this case, the fix is simply checking the state of the channel before trying to move it to progress, and don't even try to move it if already in progress with media. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] Changing state on 1:1 from PROGRESS_MEDIA to PROGRESS
Yes, agreed, but there is no point in sending a WARNING since is a normal condition, therefore should not even try to change the state of the channel. On Thu, Aug 13, 2009 at 7:57 PM, Mathieu Rene mrene_li...@avgs.ca wrote: It probably just VETO it so it avoid sending SWITCH_MESSAGE_INDICATE_PROGRESS again since the call is already making progress from the core's point of view? Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 13-Aug-09, at 11:02 AM, Moises Silva wrote: On Wed, Aug 12, 2009 at 7:19 PM, Brian West br...@freeswitch.org wrote: Well you really can't ignore it... it happens with our ISDN stack too. Thats what the VETO handles. /b You lost me. What do you mean we can't ignore it? the way I see it, sure we can and we should. Currently that warning comes from the on_ringing() callback which blindly attempts to move the state of the zap channel to ZAP_CHANNEL_STATE_PROGRESS, even when the state may be already ZAP_CHANNEL_STATE_PROGRESS_MEDIA (which means on_proceed() was called first). As I see it, the VETO warning is more an aid to the programmer so you quickly realize your doing a useless state change, which should be fixed. In this case, the fix is simply checking the state of the channel before trying to move it to progress, and don't even try to move it if already in progress with media. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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 -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.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] OpenSolaris Compile Error [gcc]
64bit OpenSolaris w/ gcc-4.3.2 After a bootstrap and configure I get the following error when running make: ---snip--- Compiling src/switch_caller.c ... cc1: warnings being treated as errors src/switch_caller.c: In function 'switch_caller_profile_event_set_data': src/switch_caller.c:299: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t' src/switch_caller.c:301: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t' src/switch_caller.c:303: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t' src/switch_caller.c:305: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t' src/switch_caller.c:307: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t' src/switch_caller.c:309: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t' src/switch_caller.c:311: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t' make[2]: *** [libfreeswitch_la-switch_caller.lo] Error 1 ---snip--- I get this error in both the source for 1.0.4 and last nights snapshot. An suggestions or ideas? There are no apparent errors during the bootstrap or configure processes. Regards, Nicholas Lemberger Lakefield Communications ___ 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] Loopback and bypass_media
The reason it works when you wait 3 seconds is that mod_loopback bails out of the equation as soon as it detects a bridge. It'll call switch_ivr_uuid_bridge() and you'll end up with 2 sofia channels. Is there a hook that is fired when when that switch_ivr_uuid_bridge() successfully executes? So the uuid_media off is called on the appropriate sofia channels? Is api_after_bridge behaving correctly - should that only be called on the sofia channels and not the loopback? Is it being fired to early? On Thu, Aug 13, 2009 at 4:54 PM, Mathieu Renemrene_li...@avgs.ca wrote: Hi All, The reason it works when you wait 3 seconds is that mod_loopback bails out of the equation as soon as it detects a bridge. It'll call switch_ivr_uuid_bridge() and you'll end up with 2 sofia channels. Now the reason why you can't do uuid_media over a loopback channel is because it doesn't pass on SWITCH_MESSAGE_INDICATE_MEDIA and SWITCH_MESSAGE_INDICATE_NOMEDIA onto the underlying channel. The handler for those two events require accessing channel variables on the both channels to get the ip+port of where the audio should go through, so that mod_sofia can send a re-invite. Since mod_loopback is a completely different channel, it has its own channel variables, independent from mod_sofia (provided you have sofia channels on both side). That's why even sofialoopback won't do bypass media. On another note, mod_sofia will behave differently when it detects its being bridge with another sofia channel, providing optimizations when both call legs are SIP. My personal opinion is not to use mod_loopback unless absolutely necessary, FreeSWITCH's core is very flexible and there's often a (better) way than using mod_loopback. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 13-Aug-09, at 6:59 PM, Phillip Jones wrote: Rupa / all, Just a quick follow up to this. This is appears to a timing issue. If I try and do a uuid_media off + uuid in api_after_bridge it fails with CHAN_NOT_IMPLEMENTED and the call is dropped. If appears to be trying to do a SIP reinvite on the loopback channel which is of course just about to / has disappear/ed. So I tried this, after the call is established, at the commend line, I do show calls and using the uuid shown, type uuid_media off uuid. The SIP REINVITE is issued and works. I think the switch_ivr_nomedia function in switch_ivr_c is getting the loopback uuid when it calls other_uuid = switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE) That's why the SIP REINVITE fails. So... in api_after_bridge I issue: sched_api, +3 none uuid_media off + uuid. This calls the switch_ivr_nomedia function 3 seconds after the calls bridge is established. And it works, Not nice - not scalable or production ready - but the SIP-REINVITE is successful and at least now I understand what is going on. Make sense? Thanks Phil On Wed, Aug 12, 2009 at 12:21 PM, Rupa Schomakerr...@rupa.com wrote: On Wed, Aug 12, 2009 at 10:22 AM, Phillip Jonespjinthe...@gmail.com wrote: Hi there, application=originate data=(sofia/foo/bar|sofia/baz/bar), (sofia/foo/yum|sofia/baz/yum) I agree. However, perhaps the ideal is not to specify the carriers at this level, as carriers are added and removed fairly often as costings change. So it would be nice to have some sort of proxy that resolves to a list of carriers: application=originate data=sofia/MyCarriers/bar,sofia/ MyCarriers/yum MyCarriers action application=carrier data=sofia/foo/ action application=carrier data=sofia/baz/ action application=carrier data=sofia/etc/ /MyCarriers or something similar. This would achieve the same as loopback in this use case but without dangers of looping? Complicated stuff though. Well, that is all done by mod_lcr. I was simplifying to narrow down to just originate. First we need to see if this is worth pursuing over fixing (modifying, whatever) loopback to handle bypass media. If it is, then I'll modify mod_lcr to deal with the situation in question (comma or pipe sep list of numbers to call. mod_lcr would then group as appropriate). Right now, my bridge is setup in a small javascript script that builds the appropriate dialstring (using loopback for external calls, user/ for internal calls) and then when doing the loopback call to mod_lcr to get the dialstring with all providers in the right order. Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup? That's a good idea - I will look into that. Thanks again. Phillip Let us know how it works for you... -- -Rupa ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
