Re: [Freeswitch-users] Sangoma A500 - dial out from specific port group?
Just want to say that you were right. Updating to trunk solved the problem.
It seems that I updated/rebuilt my copy just before the patch was
applied on 4 September.
Thank you again!
On Thu, Sep 10, 2009 at 9:54 AM, Vassil Panayotov
panayotov...@gmail.com wrote:
Michael, Moises and Octavio thank you for your replies!
The server will be shipped to another site today and I can't test
thoroughly now.
When it is installed I will update this thread.
Best regards,
Vassil
On Thu, Sep 10, 2009 at 1:58 AM, Octavio Ruiz tac...@tacvbo.net wrote:
On Wed, Sep 9, 2009 at 01:20, Vassil Panayotov panayotov...@gmail.com
wrote:
Hi,
Is it possible to originate calls from specific A500 ports with FreeSWITCH?
I am using a A504 (8 BRI interfaces), and I want some outbound calls to be
made from specific BRI interfaces.
You can't define several spans in openzap.conf for boost, the
sangoma_brid config file is where you define groups, so your config
should look like this:
/// smg_bri.conf
..
group=1
spans=1
group=2
spans=2
group=3
spans=3
..
/// openzap.conf
[span wanpipe BoostBRI]
trunk_type = bri
b-channel = 1:1-2
b-channel = 2:1-2
b-channel = 3:1-2
b-channel = 4:1-2
b-channel = 5:1-2
b-channel = 6:1-2
b-channel = 7:1-2
b-channel = 8:1-2
/// openzap.conf.xml
boost_spans
span name=BoostBRI
param name=local-ip value=127.0.0.65/
param name=local-port value=53000/
param name=remote-ip value=127.0.0.66/
param name=remote-port value=53000/
param name=context value=default/
param name=dialplan value=XML/
param name=tonegroup value=uk/
/span
/boost_spans
Then, you can Dial to your span/group number 3 with:
freeswitch originate openzap/1/a/12...@g3
exten|application_name(app_args)
freeswitch originate openzap/1/a/12...@g3
exten|application_name(app_args)
freeswitch originate openzap/1/a/12...@r3
exten|application_name(app_args)
freeswitch originate openzap/1/a/12...@r3
exten|application_name(app_args)
If you are using FS 1.0.4, there is a bug, you can fix it with this
-already in trunk- patch.
Index: src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c
===
--- libs/openzap/src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c.orig
+++ libs/openzap/src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c
@@ -282,6 +282,8 @@
}
ss7bc_call_init(event, caller_data-cid_num.digits, ani, r);
+ //ss7_bc_call_init will clear the trunk_group val so we need to set
it again
+ event.trunk_group=tg;
if (gr *(gr+1)) {
Best regards,
--
Octavio H. Ruiz Cervera
Tel.: (+52 55) 8590-9000 Ext. 7016
Mobile: (+52 1 55) 4358-4565
Sent from Mexico City, DF, Mexico
___
FreeSWITCH-users mailing list
FreeSWITCH-users@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] FS 1.0.4 erl configure error
meftah, i disabled mod_erlang_event in modules.conf. unixodbc is installed already. still ... the same error message. tks for your input. /nandy On Sun, Sep 13, 2009 at 11:56 PM, Meftah Tayeb tayeb.mef...@gmail.comwrote: hello, i think you enabled mod_erlang_event in the modules.conf install unixodbc if is not installed thanks Nandy Dagondon a écrit : hi, i want to enable odbc support which is required in mod_lcr feature. however, i encounter ./configure problem after installing Erlang R13B01. this is the portion of the error messages: ... checking for erl... /usr/local/bin/erl checking erlang version... 5.7.2 checking erlang libdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib checking erlang incdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/include checking ei.h usability... yes checking ei.h presence... no configure: WARNING: ei.h: accepted by the compiler, rejected by the preprocessor! configure: WARNING: ei.h: proceeding with the compiler's result checking for ei.h... yes checking for ei_encode_version in -lei... yes checking for ei_link_unlink in -lei... no configure: Your erlang seems OK, do not forget to enable mod_erlang_event in modules.conf configure: creating ./config.status config.status: creating src/include/switch_version.h.in .infig.status: error: cannot find input file: Makefile END i set ERL_TOP environment variable to the source directory. has anyone encountered this problem? can anyone give me a hint what's wrong. i'm compiling FS 1.0.4. thank you, /nandy -- ___ FreeSWITCH-users mailing listfreeswitch-us...@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org __ Information from ESET NOD32 Antivirus, version of virus signature database 4421 (20090913) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __ Information from ESET NOD32 Antivirus, version of virus signature database 4421 (20090913) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.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] How to filter the allowed string
Hi, I'm newbie in FS. I want to know how to Filter the string to include only the allowed characters in FS? Kindly advice me. -- Regards, Ahmed Munir ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] freeswitch on blackfin + uclinux
Hi, Does anyone have any luck on porting freeswitch to blackfin + uclinux? Is this a feasible option? jb ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] compilation error with the latest codes
You have a merge conflict please svn revert sofia.c /b On Sep 14, 2009, at 3:46 AM, Jingwei Yang wrote: Hi Folks, I've got a compilation error with the latest codes (r14842) Making all in packages Creating mod_sofia_la-mod_sofia.lo Compiling mod_sofia.c ... Creating mod_sofia_la-sofia.lo Compiling sofia.c ... sofia.c: In function ‘sofia_handle_sip_r_invite’: sofia.c:3221: error: expected expression before ‘’ token make[5]: *** [mod_sofia_la-sofia.lo] Error 1 make[4]: *** [all] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 Does anyone have ideas about this? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Recording inbound call including DTMF - possible ?
Hi I need to measure DTM digits duration and interdigit delay for various phones in a two stage dialing scenario. I.e Phone dials DID and after answer then the second number My set-up is: Phone-PSTN network-DID(inband DTMF) -FS I ha ve FS to answer the call and record the call - all this is fine. However when i analyse the rdecording the Digits are being cut off down to 10 msec bursts - I trust its FS that cust the DTMF in order to avoid further propogation inband to second leg of the call. Is theer a way to avoid this ? I.e record the inbound call without DTMF processing ? Thx Morten ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Zaptel Analog FXO Outbound Dialing
_ My suspicion is that this is only for zaptel type cards. Our tests with Sangoma analog cards have all been pretty successful. But thanks for info! Anyone else using Rhino, Digium, or compatible analog cards?I am not experiencing an audio delay. My configuration is exactly as documented on the Zaptel Tutorial wiki page (http://wiki.freeswitch.org/wiki/Zaptel_Tutorial). I'm using a Digium TDM400P, Zaptel 1.4 revision 4630, and FreeSWITCH trunk revision 14842. If you want me to try anything for you, I'm 'Deeewayne' on IRC. -Dwayne.___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Asterisk 1.6 connecting to FS 1.4
Hi,
A client of ours is trying to connect his * to our FS, outgoing calls
work fine, unfortunately when we try to forward an incoming call to his
* it's not going through.
I see his registration in our internal profile which looks just fine.
We try to forward incoming calls using this in FS dialplan:
extension name=myext
condition field=destination_number expression=^5777$
action application=bridge
data=sofia/internal/4000...@$${domain}|sofia/internal/4000...@$${domain}/
/condition
/extension
Only abnormal things I can see in FS logs are:
2009-09-13 19:17:31.869158 [ERR] sofia_reg.c:1570 No Matching gateway found
2009-09-13 19:17:31.869158 [NOTICE] sofia_reg.c:1590 Hangup
sofia/internal/400[CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING]
Why would FS look for a gateway in this case? And what
MANDATORY_IE_MISSING would mean here? Call gets forwarded to VM as if
user was unavailable.Hangup is initiated by us in this case.
Client uses this configuration in *:
/etc/asterisk/sip.conf:
/etc/asterisk/sip.conf:
register=400:mysippassw...@versafon.com/400
[400]
type=friend
username=400
secret=mysippassword
host=versafon.com
canreinvite=no
fromuser=400
dtmfmode=rfc2833
context=versafon-incoming
___
FreeSWITCH-users mailing list
FreeSWITCH-users@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] problem with performance testing
Hi, I tried to performance test freeswitch with media proxy thur fs. With 400 cps, I start to see 2000 channels remaining in Freeswitch, and then no read codec error starts to pop up. With only 1875 channels, how come freeswitch is complaining about no read codec? Also, I am using media_proxy = true, whey should it need a codec anyway? freeswi...@mycom.com 2009-09-14 20:59:16.777675 [ERR] switch_core_io.c:118 sofia/external/12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:30.815547 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:30.815547 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: [] 2009-09-14 20:59:45.349181 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:45.350179 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: [] 2009-09-14 21:00:00.104559 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:04.495545 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:16.996438 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:25.905617 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. freeswi...@mycom.com show channels count API CALL [show(channels count)] output: 1875 total. freeswi...@mycom.com 2009-09-14 21:00:36.212767 [ERR] switch_core_io.c:118 sofia/external/12323...@192.168.1.116:5911 has no read codec. After I paused the traffic from sipp and when sipp finished, I still got a bunch of zombie channels that are in CONSUME_MEDIA stage. What is the cause of these zombies and how can I fix it? uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure 5f013ece-d8a0-4ee0-bce0-3a56c05dc225,outbound,2009-09-14 20:53:55,1252932835,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 9347ea12-7ffb-4e96-8ee9-b2ac1ee57752,outbound,2009-09-14 20:53:57,1252932837,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 5dc2f6f2-7d58-4582-b680-40bb3ed330ef,outbound,2009-09-14 20:54:02,1252932842,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 66361571-b022-42b9-9507-87d1bfc01b03,outbound,2009-09-14 20:54:03,1252932843,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 7a075ea5-7c01-4951-b826-cf982df03501,outbound,2009-09-14 20:54:03,1252932843,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, bd8c9905-dbab-4ab7-a9af-b7e2b87cc0b0,outbound,2009-09-14 20:54:05,1252932845,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 3af84a6c-02f2-44ae-8a4b-1c7940522005,outbound,2009-09-14 20:54:06,1252932846,sofia/external ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] A real puzzler for you guys... (perpetual registration causes denial of service.)
Swapping hardware... I've noticed other odd things... Things that shouldn't happen, do.. But not consistently The phrase, It's computing Jim, but not as we know it... pretty much describes the situation. Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Sep 13, 2009, at 6:30 PM, Karl Vesterling wrote: New development. Even though the initial registration succeeds, the subsequent registrations fail... ??Search me?? But that's just too weird for me... Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Sep 13, 2009, at 4:23 PM, Brian West wrote: I haven't seen this issue in 8.12 either... Maybe thats why 8.11 isn't on the website last I checked? /b On Sep 13, 2009, at 2:59 PM, Karl Vesterling wrote: RESOLVED!!! Folks, evidently this is a problem with Cisco Firmware P0S3-08-11-00 I forgot that (a long long time ago) I had dropped that firmware into that site. Phones hadn't been rebooted in (a while)... Oddly enough, once you get past (X) number of phones, the registration chatter created by the bug was too much for FS to keep up with. P0S3-08-8-00 works perfectly fine. Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] A real puzzler for you guys... (perpetual registration causes denial of service.)
The first hint was when the firmware rev began with the letters POS On Mon, Sep 14, 2009 at 8:15 AM, Karl Vesterling k...@ken-ton.com wrote: Swapping hardware... I've noticed other odd things... Things that shouldn't happen, do.. But not consistently The phrase, It's computing Jim, but not as we know it... pretty much describes the situation. Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Sep 13, 2009, at 6:30 PM, Karl Vesterling wrote: New development. Even though the initial registration succeeds, the subsequent registrations fail... ??Search me?? But that's just too weird for me... Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Sep 13, 2009, at 4:23 PM, Brian West wrote: I haven't seen this issue in 8.12 either... Maybe thats why 8.11 isn't on the website last I checked? /b On Sep 13, 2009, at 2:59 PM, Karl Vesterling wrote: RESOLVED!!! Folks, evidently this is a problem with Cisco Firmware P0S3-08-11-00 I forgot that (a long long time ago) I had dropped that firmware into that site. Phones hadn't been rebooted in (a while)... Oddly enough, once you get past (X) number of phones, the registration chatter created by the bug was too much for FS to keep up with. P0S3-08-8-00 works perfectly fine. Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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
Re: [Freeswitch-users] problem with performance testing
After I paused the traffic from sipp and when sipp finished, I still got a bunch of zombie channels that are in CONSUME_MEDIA stage. What is the cause of these zombies and how can I fix it? One way might be to not DDoS your box at 400cps? (You are out of rtp ports *and* you are pushing your machine too hard.) Only 1875 channels? hmm. *shrug* On Mon, Sep 14, 2009 at 8:08 AM, Woody Dickson woodydick...@gmail.comwrote: Hi, I tried to performance test freeswitch with media proxy thur fs. With 400 cps, I start to see 2000 channels remaining in Freeswitch, and then no read codec error starts to pop up. With only 1875 channels, how come freeswitch is complaining about no read codec? Also, I am using media_proxy = true, whey should it need a codec anyway? freeswi...@mycom.com 2009-09-14 20:59:16.777675 [ERR] switch_core_io.c:118 sofia/external/12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:30.815547 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:30.815547 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: [] 2009-09-14 20:59:45.349181 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:45.350179 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: [] 2009-09-14 21:00:00.104559 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:04.495545 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:16.996438 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:25.905617 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. freeswi...@mycom.com show channels count API CALL [show(channels count)] output: 1875 total. freeswi...@mycom.com 2009-09-14 21:00:36.212767 [ERR] switch_core_io.c:118 sofia/external/12323...@192.168.1.116:5911 has no read codec. After I paused the traffic from sipp and when sipp finished, I still got a bunch of zombie channels that are in CONSUME_MEDIA stage. What is the cause of these zombies and how can I fix it? uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure 5f013ece-d8a0-4ee0-bce0-3a56c05dc225,outbound,2009-09-14 20:53:55,1252932835,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 9347ea12-7ffb-4e96-8ee9-b2ac1ee57752,outbound,2009-09-14 20:53:57,1252932837,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 5dc2f6f2-7d58-4582-b680-40bb3ed330ef,outbound,2009-09-14 20:54:02,1252932842,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 66361571-b022-42b9-9507-87d1bfc01b03,outbound,2009-09-14 20:54:03,1252932843,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 7a075ea5-7c01-4951-b826-cf982df03501,outbound,2009-09-14 20:54:03,1252932843,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, bd8c9905-dbab-4ab7-a9af-b7e2b87cc0b0,outbound,2009-09-14 20:54:05,1252932845,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 3af84a6c-02f2-44ae-8a4b-1c7940522005,outbound,2009-09-14 20:54:06,1252932846,sofia/external ___ FreeSWITCH-users mailing list FreeSWITCH-users@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
Re: [Freeswitch-users] Asterisk 1.6 connecting to FS 1.4
This means the far end is sending you a challenge and we do not know how to answer it... please review how to setup a gateway on the Wiki so you can authenticate. /b On Sep 13, 2009, at 6:47 PM, paul.d...@gmail.com wrote: Only abnormal things I can see in FS logs are: 2009-09-13 19:17:31.869158 [ERR] sofia_reg.c:1570 No Matching gateway found 2009-09-13 19:17:31.869158 [NOTICE] sofia_reg.c:1590 Hangup sofia/internal/400[CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] A real puzzler for you guys... (perpetual registration causes denial of service.)
HAHA I couldn't have said this better! /b On Sep 14, 2009, at 8:17 AM, Anthony Minessale wrote: The first hint was when the firmware rev began with the letters POS ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Recording inbound call including DTMF - possible ?
On Sep 13, 2009, at 9:27 AM, Morten Henckel wrote: However when i analyse the rdecording the Digits are being cut off down to 10 msec bursts - I trust its FS that cust the DTMF in order to avoid further propogation inband to second leg of the call. Nope if its rfc2833 its not us cutting the dtmf if you have a TDM gateway in the mix that is prob. what is doing it. Is theer a way to avoid this ? I.e record the inbound call without DTMF processing ? No. The blib of DTMF you hear is not ours to remove its the remote gateways job... FreeSWTICH usually only does 2833 so it could be hearing the little bit of DTMF coming in or out from the endpoints. If you check the cdr hangup you'll have digit_log which will be a log of all digits dialed during the call. Thx Morten ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] problem with performance testing
Hi Woody, well, it is quite hard to answer you back with this logs... you didn't tell us: 1. what machine are you running (CPU/RAM) 2. what distro are you running - 32 or 64 bit (i had some lets say experience with a wrong selection :P) 3. what is your configuration (dialplan/sip_profiles) 4. did you disable all logging? 5. what modules are loaded (you should load minimal modules - at least disable conference) 6. if you moved the db files to a ram disk 7. how do you start the calls (slowly with 10 - 20 CPS or you are DDoS-ing it with 400 right away). 8. how long do you keep the calls going? 9. what is the the current CPU usage when stresstesting 10. what is the amount of read/writes to from/to your HDD ... i could go on and on with the list. Here Anthony has nothing to tell you except that you reached the maximum and you are currently killing the machine/application Also, what are you looking for? ... A machine that can do a lot of simultaneous calls or a machine that can do a lot of CPS? you should decide at some point. I have a lot of experience with commercial SoftSwitches and i can tell you that FreeSWITCH performance is something outstanding ... i'm getting a reliable 500 CPS on just one FS machine (dualcore xeon 2.33 GHz - bogomips 4670). just to compare: NetCentrex as a comercial SoftSwitch can do only 480 CPS (distribuited on 10 nodes)... and of course signaling only. And this is enough to run my 12.000 calls. Tihomir. On Mon, Sep 14, 2009 at 3:08 PM, Woody Dickson woodydick...@gmail.comwrote: Hi, I tried to performance test freeswitch with media proxy thur fs. With 400 cps, I start to see 2000 channels remaining in Freeswitch, and then no read codec error starts to pop up. With only 1875 channels, how come freeswitch is complaining about no read codec? Also, I am using media_proxy = true, whey should it need a codec anyway? freeswi...@mycom.com 2009-09-14 20:59:16.777675 [ERR] switch_core_io.c:118 sofia/external/12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:30.815547 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:30.815547 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: [] 2009-09-14 20:59:45.349181 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:45.350179 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: [] 2009-09-14 21:00:00.104559 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:04.495545 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:16.996438 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:25.905617 [ERR] switch_core_io.c:118 sofia/external/ 12323...@192.168.1.116:5911 has no read codec. freeswi...@mycom.com show channels count API CALL [show(channels count)] output: 1875 total. freeswi...@mycom.com 2009-09-14 21:00:36.212767 [ERR] switch_core_io.c:118 sofia/external/12323...@192.168.1.116:5911 has no read codec. After I paused the traffic from sipp and when sipp finished, I still got a bunch of zombie channels that are in CONSUME_MEDIA stage. What is the cause of these zombies and how can I fix it? uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure 5f013ece-d8a0-4ee0-bce0-3a56c05dc225,outbound,2009-09-14 20:53:55,1252932835,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 9347ea12-7ffb-4e96-8ee9-b2ac1ee57752,outbound,2009-09-14 20:53:57,1252932837,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 5dc2f6f2-7d58-4582-b680-40bb3ed330ef,outbound,2009-09-14 20:54:02,1252932842,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 66361571-b022-42b9-9507-87d1bfc01b03,outbound,2009-09-14 20:54:03,1252932843,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 7a075ea5-7c01-4951-b826-cf982df03501,outbound,2009-09-14 20:54:03,1252932843,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, bd8c9905-dbab-4ab7-a9af-b7e2b87cc0b0,outbound,2009-09-14 20:54:05,1252932845,sofia/external/12323...@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323...@192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000,
[Freeswitch-users] FS create directory
Hi,
i just have a maybe dummy question but it is still a question :P
*action application=record
data=${recordpath}/${service_instance}/${record_filename} 20 200/*
in my case ${service_instance} is something dynamic and has to be created on
the fly.
Is there any way FS can create a directory prior to dump the file there?
Tihomir.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Re: [Freeswitch-users] How to filter the allowed string
Ahmed, if you are talking about dial patterns then yes, freeswitch takes you a mile ahead and utilizes regular expressions for pattern matching, you could probably use something like this: ^([0-9]+)$ above simple regex will allow any digit from 0 to 9 and + indicates repetitive, so this regex is equal to following asterisk's pattern: _X. -gm João Mesquita wrote: Not sure I understand what you mean. Can you explain what you are trying to achieve a little bit better? jmesquita On Mon, Sep 14, 2009 at 4:18 AM, Ahmed Munir ahmedmunir...@gmail.com mailto:ahmedmunir...@gmail.com wrote: Hi, I'm newbie in FS. I want to know how to Filter the string to include only the allowed characters in FS? Kindly advice me. -- Regards, Ahmed Munir ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto:FreeSWITCH-users@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] Asterisk 1.6 connecting to FS 1.4
Thank you for the hint. But.. why would I need a gateway in this case? I am just trying to ring an FS extension, right? Anybody has a clue how to make * not to send the challenge? This means the far end is sending you a challenge and we do not know how to answer it... please review how to setup a gateway on the Wiki so you can authenticate. /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] FS create directory
yep,
just sow it in the meantime... thanks.
btw: can i use mod_shout to stream files to a server..
e.g. *action application=record data=shout://
server.domain.com/${recordpath}/${service_instance}/${record_filename} 20
200/*
can it work?
T.
On Mon, Sep 14, 2009 at 4:15 PM, Leon de Rooij l...@scarlet-internet.nlwrote:
Hi,
You could use a system call for that:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system
regards,
Leon
On Sep 14, 2009, at 3:58 PM, Tihomir Culjaga wrote:
Hi,
i just have a maybe dummy question but it is still a question :P
*action application=record
data=${recordpath}/${service_instance}/${record_filename} 20 200/*
in my case ${service_instance} is something dynamic and has to be created
on the fly.
Is there any way FS can create a directory prior to dump the file there?
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
___
FreeSWITCH-users mailing list
FreeSWITCH-users@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] Pastebin Username/Password Not Accepted
it's a rite of passage :) On Mon, Sep 14, 2009 at 11:29 AM, Jerry Richards jerry.richa...@teotech.com wrote: Aha... I have been notified that I failed the test. The username/password is given in the authentication pop-up itself. My bad... -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, September 14, 2009 8:13 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: Pastebin Username/Password Not Accepted What account do I need to create to post logs in the Pastebin? I tried my mailing list username/password, and also tried a jira.freeswitch.org username/password. Neither of these were accepted. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server database schema, etc. If you are good with a scripting language then we could definitely use help with rounding out the docs for your favorite language. We could also use code samples, so ask for a contrib folder if you have things you would like to share. Or how about this: you read something on the wiki, it doesn't quite work when you try, so you tinker until you figure it out. Now you're in a position to update the wiki for everyone else's benefit, too. As you can see, you don't have to be a FreeSWITCH expert before you can help the project. What we really need are people who care about the project and want to see it flourish. If you are such a person then please contact me off list. Tell me what you're good at or where you would like to help. Many thanks for all of your support! -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Pastebin Username/Password Not Accepted
What account do I need to create to post logs in the Pastebin? I tried my mailing list username/password, and also tried a jira.freeswitch.org username/password. Neither of these were accepted. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Pastebin Username/Password Not Accepted
Aha... I have been notified that I failed the test. The username/password is given in the authentication pop-up itself. My bad... -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, September 14, 2009 8:13 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: Pastebin Username/Password Not Accepted What account do I need to create to post logs in the Pastebin? I tried my mailing list username/password, and also tried a jira.freeswitch.org username/password. Neither of these were accepted. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS create directory
This works for me:
action application=mkdir data=${filebase_dir}/
You must set ' filebase_dir ' before.
- Original Message -
From: Tihomir Culjaga
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, September 14, 2009 4:58 PM
Subject: [Freeswitch-users] FS create directory
Hi,
i just have a maybe dummy question but it is still a question :P
action application=record
data=${recordpath}/${service_instance}/${record_filename} 20 200/
in my case ${service_instance} is something dynamic and has to be created on
the fly.
Is there any way FS can create a directory prior to dump the file there?
Tihomir.
--
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Re: [Freeswitch-users] FS create directory
nice ... thx.
T.
On Mon, Sep 14, 2009 at 4:41 PM, Evgeniy Zolotov zolo...@altron.ua wrote:
This works for me:
action application=mkdir data=${filebase_dir}/
You must set ' filebase_dir ' before.
- Original Message -
*From:* Tihomir Culjaga tculj...@gmail.com
*To:* freeswitch-users@lists.freeswitch.org
*Sent:* Monday, September 14, 2009 4:58 PM
*Subject:* [Freeswitch-users] FS create directory
Hi,
i just have a maybe dummy question but it is still a question :P
*action application=record
data=${recordpath}/${service_instance}/${record_filename} 20 200/*
in my case ${service_instance} is something dynamic and has to be created
on the fly.
Is there any way FS can create a directory prior to dump the file there?
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
___
FreeSWITCH-users mailing list
FreeSWITCH-users@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] Pastebin Username/Password Not Accepted
Try username pastebin with pasword freeswitch (without ) Jerry Richards wrote: What account do I need to create to post logs in the Pastebin? I tried my mailing list username/password, and also tried a jira.freeswitch.org username/password. Neither of these were accepted. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Asterisk 1.6 connecting to FS 1.4
There is no such things as FS 1.4, but 1.0.4 yes. On Mon, Sep 14, 2009 at 2:20 PM, paul.d...@gmail.com wrote: Thank you for the hint. But.. why would I need a gateway in this case? I am just trying to ring an FS extension, right? Anybody has a clue how to make * not to send the challenge? This means the far end is sending you a challenge and we do not know how to answer it... please review how to setup a gateway on the Wiki so you can authenticate. /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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] DTMF CSeq: 505 INFO vs RTP DTMF
Hi, I am using the function session.collectInput and session.streamFile to collect a number of DTMF digits. If the DTMF digits are sent in the RTP, i can collect several digits until timeout. No problem there! If the DTMFs are received as a sequence of SIP INFO packages, collectInput only receives the first one. Any ideas? -- Stopping junk mailers is good for the environment ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] 482 Request merged, in serial forking
We currently don't support forked dialogs.
Mike
On Sep 8, 2009, at 12:16 PM, Humberto Quintana wrote:
Hi Brian,
Thank you very much for your answer but both, Freeswitch and
Kamailio have public IPs, it's my NAT'd IP phone who has private IP
but this is fixed by Kamailio.
The problem is not the 1st call is failing ( the test is set that
way), the problem is FS answers back 482 when Kamailio tries a 2nd
route ( or 3rd ) for the same call...
Freeswitch is configured to use the Requested-URI sent by Kamailio:
action application=bridge data=sofia/external/${sip_req_uri}/
I noticed that there is no Log message in Freeswitch when receiving
the INVITE for the 2nd route.
The process in FS seems to be destroyed (11:46:21.396593) before the
2nd INVITE is received (11:46:21.401419
).
U 2009/09/08 11:46:21.395702 freeswitch:5060 - kamailio:5060
SIP/2.0 503 Service Unavailable.
Call-ID: ba748cd27cd16...@192.168.2.13
U 2009/09/08 11:46:21.395897 kamailio:5060 - freeswitch:5060
ACK sip:514...@gw1:5060 SIP/2.0.
Call-ID: ba748cd27cd16...@192.168.2.13
U 2009/09/08 11:46:21.401419 kamailio:5060 - freeswitch:5060
INVITE sip:1514...@gw2:5061 SIP/2.0.
Call-ID: ba748cd27cd16...@192.168.2.13
U 2009/09/08 11:46:21.401845 freeswitch:5060 - kamailio:5060
SIP/2.0 482 Request merged.
Call-ID: ba748cd27cd16...@192.168.2.13
2009-09-08 11:46:21.395503 [DEBUG] mod_sofia.c:417 Responding to
INVITE with: 503
2009-09-08 11:46:21.395503 [DEBUG] switch_core_state_machine.c:46
sofia/external/10092...@freeswitch Standard HANGUP, cause:
NORMAL_TEMPORARY_FAILURE
2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:434
(sofia/external/10092...@freeswitch) State HANGUP going to sleep
2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:476
(sofia/external/10092...@freeswitch) State Change CS_HANGUP -
CS_REPORTING
2009-09-08 11:46:21.396593 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/10092...@freeswitch [BREAK]
2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:398
(sofia/external/10092...@freeswitch) Running State Change CS_REPORTING
2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:612
(sofia/external/10092...@freeswitch) State REPORTING
2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:53
sofia/external/10092...@freeswitch Standard REPORTING, cause:
NORMAL_TEMPORARY_FAILURE
2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:612
(sofia/external/10092...@freeswitch) State REPORTING going to sleep
2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:411
(sofia/external/10092...@freeswitch) State Change CS_REPORTING -
CS_DESTROY
2009-09-08 11:46:21.396593 [DEBUG] switch_core_session.c:1068
Session 3 (sofia/external/10092...@freeswitch) Locked, Waiting on
external entities
2009-09-08 11:46:21.396593 [NOTICE] switch_core_session.c:1086
Session 3 (sofia/external/10092...@freeswitch) Ended
2009-09-08 11:46:21.396593 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/external/10092...@freeswitch [CS_DESTROY]
2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:564
(sofia/external/10092...@freeswitch) State DESTROY
2009-09-08 11:46:21.396593 [DEBUG] mod_sofia.c:255 sofia/external/
10092...@freeswitch SOFIA DESTROY
2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:60
sofia/external/10092...@freeswitch Standard DESTROY
2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:564
(sofia/external/10092...@freeswitch) State DESTROY going to sleep
Note: I'm using only the external sofia profile.
Thanks,
Humberto
==
Looks like FS is behind nat. You need to set local-network-acl and
the ext-rtp-ip and ext-sip-ip so FreeSWITCH properly puts in the right
IP's in the via headers and sdp.
Please refer to internal.xml in the latest SVN for an example of how
to do this.
/b
New! Open Hotmail faster on the new MSN homepage!
___
FreeSWITCH-users mailing list
FreeSWITCH-users@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] Inbound Gateway Call Not Working
Okay. I got the Grandstream Gateway's 1-stage dialing working with
Freeswitch (Thank You, Michael Collins and Thank All You Developers for
creating this really slick Softswitch/PBX).
Here are the changes/additions I made to the XML files:
conf/sip_profiles/exernal/grandstreamGXW4104.xml (added file):
include
gateway name=192.168.72.186
param name=username value=1000/
param name=password value=1234/
param name=proxy value=192.168.72.186/
param name=register value=false/
param name=extension value=1000/
/gateway
/include
conf/dialplan/default.xml (added to existing file):
extension name=GrandstreamTest
condition field=destination_number expression=^(9{0,1}\d{10})$
action application=bridge
data=sofia/gateway/192.168.72.186/$...@192.168.72.186/
/condition
/extension
conf/dialplan/public.xml (added to existing file):
extension name=GrandstreamTest
condition field=destination_number expression=^(5000)$
action application=transfer data=$1 XML default/
/condition
/extension
conf/autoload_configs/acl.conf.xml (added to existing file):
list name=lan default=allow
node type=allow cidr=192.168.72.186/32/
...
/list
...
list name=domains default=deny
node type=allow cidr=192.168.72.186/32/
...
/list
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, September 11, 2009 1:27 PM
To: 'freeswitch-users@lists.freeswitch.org'; 'Michael Collins'
Subject: RE: Inbound Gateway Call Not Working
Thanks. I added the node type=allow cidr=x.x.x.x/32/ to both the
lan list and domain list in the acl.conf.xml file and it does not try to
authenticate anymore.
However, now it replies to the INVITE with a 480 TEMPORARILY UNAVAILABLE.
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, September 11, 2009 10:57 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: RE: Inbound Gateway Call Not Working
By the way, the FS DEBUG console is saying the following when an inbound
call is made:
Rejected by acl domains. Falling back to Digest auth.
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, September 11, 2009 10:25 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: Inbound Gateway Call Not Working
I am trying to configure a Grandstream gateway to work with FS. I can make
outbound calls without a problem. However, inbound calls are getting a 403
Forbidden from FS in response to the INVITE from the gateway.
Now, the INVITE's from address is the caller's number (e.g. 111222),
which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy
Authentication Required and the gateway uses username Anonymous and the
uri sip:4...@192.168.72.38 (4000 is the destination for all calls from the
gateway).
Is there an example configuration for this scenario?
Thanks and Best Regards,
Jerry
___
FreeSWITCH-users mailing list
FreeSWITCH-users@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] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
You can assign two things to me. 1. libesl code documentation (partially done and Doxygened - needs cleaning) 2. Bug marshal. I am setting up the proper lab environment here to be able to test most stuff. Count me in for any questions I can answer and I am _always_ on IRC jmesquita On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins m...@freeswitch.org wrote: Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server database schema, etc. If you are good with a scripting language then we could definitely use help with rounding out the docs for your favorite language. We could also use code samples, so ask for a contrib folder if you have things you would like to share. Or how about this: you read something on the wiki, it doesn't quite work when you try, so you tinker until you figure it out. Now you're in a position to update the wiki for everyone else's benefit, too. As you can see, you don't have to be a FreeSWITCH expert before you can help the project. What we really need are people who care about the project and want to see it flourish. If you are such a person then please contact me off list. Tell me what you're good at or where you would like to help. Many thanks for all of your support! -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] ATTENTION BEHAVIOR CHANGE of sip_invite_params variable.
I just committed revision 14849 to make sip_invite_params only apply to the RURI, If you wish to modify the To param son the invite you MUST use sip_invite_to_params moving forward. you have sip_invite_contact_params and sip_invite_from_params to work with also which were already there just making sure you know that now. Thanks, Brian West ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] DAHDI Dial 9 Receiving Setup Acknowledge
Sorry this was meant for the Asterisk list. I wish FreeSWITCH had QSIG
support so I could go that route.
Ryan
On Mon, Sep 14, 2009 at 3:46 PM, Ryan Wagoner rswago...@gmail.com wrote:
I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make
calls from the Toshiba to Asterisk and internal calls from Asterisk to
the Toshiba. What I can't do is make an call with an outside
destination from Asterisk to the Toshiba. The Toshiba is looking for 9
to grab an outside line then it expects to see the 10 digits. In the
FreePBX dial plan I use 9|. which sends 9 plus the 10 digit number.
Using Wireshark to look at the QSIG commands coming from a Sangoma
wanpipemon trace I see the following for an Asterisk to Toshiba
internal call.
Asterisk - SETUP
Toshiba - CALL PROCESSING
Toshiba - CONNECT
Asterisk - CONNECT ACKNOWLEDGE
However when trying to dial 9 + number I received the following
Asterisk - SETUP
Toshiba - SETUP ACKNOWLEDGE
Looking at http://tools.ietf.org/html/rfc4497 I see the following
On receipt of a QSIG SETUP message containing no Sending complete
information element and a number in the Called party number
information element that the gateway cannot determine to be complete,
the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message, start
QSIG timer T302, and await further number digits.
Otherwise, the gateway SHALL wait for more digits
to arrive in QSIG INFORMATION messages.
Looking in the chan_dahdi.c code I see
case PRI_EVENT_SETUP_ACK:
chanpos = pri_find_principle(pri,
e-setup_ack.channel);
if (chanpos 0) {
ast_log(LOG_WARNING, Received
SETUP_ACKNOWLEDGE on unconfigured channel %d/%d span %d\n,
PRI_SPAN(e-setup_ack.channel), PRI_CHANNEL(e-setup_ack.channel),
pri-span);
} else {
chanpos =
pri_fixup_principle(pri, chanpos, e-setup_ack.call);
if (chanpos -1) {
ast_mutex_lock(pri-pvts[chanpos]-lock);
pri-pvts[chanpos]-setup_ack = 1;
/* Send any queued digits */
for (x = 0;x
strlen(pri-pvts[chanpos]-dialdest); x++) {
ast_debug(1,
Sending pending digit '%c'\n, pri-pvts[chanpos]-dialdest[x]);
pri_information(pri-pri, pri-pvts[chanpos]-call,
pri-pvts[chanpos]-dialdest[x]);
}
ast_mutex_unlock(pri-pvts[chanpos]-lock);
} else
ast_log(LOG_WARNING,
Unable to move channel %d!\n, e-setup_ack.channel);
}
break;
How do I get Asterisk to queue these digits so DAHDI can send them in
response to the SETUP ACKNOWLEDGE message. What should be happening is
Asterisk sends 9 via the SETUP message, waits for the SETUP
ACKNOWLEDGE, then send the 10 digits number via a INFORMATION message.
Ryan
___
FreeSWITCH-users mailing list
FreeSWITCH-users@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] DAHDI Dial 9 Receiving Setup Acknowledge
I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make
calls from the Toshiba to Asterisk and internal calls from Asterisk to
the Toshiba. What I can't do is make an call with an outside
destination from Asterisk to the Toshiba. The Toshiba is looking for 9
to grab an outside line then it expects to see the 10 digits. In the
FreePBX dial plan I use 9|. which sends 9 plus the 10 digit number.
Using Wireshark to look at the QSIG commands coming from a Sangoma
wanpipemon trace I see the following for an Asterisk to Toshiba
internal call.
Asterisk - SETUP
Toshiba - CALL PROCESSING
Toshiba - CONNECT
Asterisk - CONNECT ACKNOWLEDGE
However when trying to dial 9 + number I received the following
Asterisk - SETUP
Toshiba - SETUP ACKNOWLEDGE
Looking at http://tools.ietf.org/html/rfc4497 I see the following
On receipt of a QSIG SETUP message containing no Sending complete
information element and a number in the Called party number
information element that the gateway cannot determine to be complete,
the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message, start
QSIG timer T302, and await further number digits.
Otherwise, the gateway SHALL wait for more digits
to arrive in QSIG INFORMATION messages.
Looking in the chan_dahdi.c code I see
case PRI_EVENT_SETUP_ACK:
chanpos = pri_find_principle(pri,
e-setup_ack.channel);
if (chanpos 0) {
ast_log(LOG_WARNING, Received
SETUP_ACKNOWLEDGE on unconfigured channel %d/%d span %d\n,
PRI_SPAN(e-setup_ack.channel), PRI_CHANNEL(e-setup_ack.channel),
pri-span);
} else {
chanpos =
pri_fixup_principle(pri, chanpos, e-setup_ack.call);
if (chanpos -1) {
ast_mutex_lock(pri-pvts[chanpos]-lock);
pri-pvts[chanpos]-setup_ack = 1;
/* Send any queued digits */
for (x = 0;x
strlen(pri-pvts[chanpos]-dialdest); x++) {
ast_debug(1,
Sending pending digit '%c'\n, pri-pvts[chanpos]-dialdest[x]);
pri_information(pri-pri, pri-pvts[chanpos]-call,
pri-pvts[chanpos]-dialdest[x]);
}
ast_mutex_unlock(pri-pvts[chanpos]-lock);
} else
ast_log(LOG_WARNING,
Unable to move channel %d!\n, e-setup_ack.channel);
}
break;
How do I get Asterisk to queue these digits so DAHDI can send them in
response to the SETUP ACKNOWLEDGE message. What should be happening is
Asterisk sends 9 via the SETUP message, waits for the SETUP
ACKNOWLEDGE, then send the 10 digits number via a INFORMATION message.
Ryan
___
FreeSWITCH-users mailing list
FreeSWITCH-users@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] ERLang configuration callbacks
I seem to be missing something in implementing the ERLang callbacks
for Freeswitch. Our Freeswitch server is starting and getting
registered with ERLang, we're invoking the bind for configuration, but
I'm not seeing any of my callbacks fire. What am I missing?
Sample code follows:
-module(freeswitch_bind).
-behaviour(gen_server).
-record(st, {fsnode, pbxpid}).
-export([start/3, terminate/2, code_change/3, init/1,
handle_call/3, handle_cast/2, handle_info/2]).
%%
%% gen_server methods
start(Node, Section, Pid) -
gen_server:start(?MODULE, [Node, Section, Pid], []).
init([Node, Section, Pid]) -
io:format( freeswitch_bind:init( [Node=~w, Section=~w, Pid=~w])~n,
[Node, Section, Pid] ),
{api, Node} ! {bind, Section},
receive
ok -
{ok, #st{fsnode=Node, pbxpid=Pid}};
{error, Reason} -
{stop, {error, {freeswitch_error, Reason}}}
after 5000 -
{stop, {error, freeswitch_timeout}}
end.
terminate(_Reason, _State) -
ok.
code_change(_OldVsn, State, _Extra) -
{ok, State}.
%%
%%Configuration handler replies that the requested document section,
tag, and key are not
%%found.
%%
handle_call({fetch, configuration, Tag, Key, Value, Params}, _From,
State) -
io:format( freeswitch_fetch:handle_call( {fetch, configuration,
Tag=~w, Key=~w, Value=~w, Params=~w}, _From, State=~w)~n,
[Tag, Key, Value, Params, State]),
Xml =
document type=\freeswitch/xml\
section name=\result\
result status=\not found\ /
/section
/document,
{ reply, {ok, Xml }, State };
%%
%%Directory handler replies that the requested document section,
tag, and key are not
%%found.
%%
handle_call({fetch, directory, Tag, Key, Value, Params}, _From, State) -
io:format( freeswitch_fetch:handle_call( {fetch, directory, Tag=~w,
Key=~w, Value=~w, Params=~w}, _From, State=~w)~n,
[Tag, Key, Value, Params, State]),
Xml =
document type=\freeswitch/xml\
section name=\result\
result status=\not found\ /
/section
/document,
{ reply, {ok, Xml }, State };
%%
%%Dialplan handler replies that the requested document section, tag,
and key are not
%%found.
%%
handle_call({fetch, dialplan, Tag, Key, Value, Params}, _From, State) -
io:format( freeswitch_fetch:handle_call( {fetch, dialplan, Tag=~w,
Key=~w, Value=~w, Params=~w}, _From, State=~w)~n,
[Tag, Key, Value, Params, State]),
Xml =
document type=\freeswitch/xml\
section name=\result\
result status=\not found\ /
/section
/document,
{ reply, {ok, Xml }, State };
%%
%%Default handler replies that the requested document section, tag,
and key are not
%%found.
%%
handle_call({fetch, Section, Tag, Key, Value, Params}, _From, State) -
io:format( freeswitch_fetch:handle_call( {fetch, Section=~w,
Tag=~w, Key=~w, Value=~w, Params=~w}, _From, State=~w)~n,
[Section, Tag, Key, Value, Params, State]),
Xml =
document type=\freeswitch/xml\
section name=\result\
result status=\not found\ /
/section
/document,
{ reply, {ok, Xml }, State };
%%
%%If the request isn't recognized, just log it and do nothing.
%%
handle_call(Request, _From, State) -
io:format(freeswitch_bind:handle_call( ~w, _From, State)
unrecognized request~n,
[Request]),
{reply, {error, unrecognized_request}, State}.
handle_cast(Message, State) -
error_logger:error_msg(~p received unrecognized cast ~p~n,
[self(), Message]),
{noreply, State}.
handle_info({fetch, Section, Tag, Key, Value, FetchID, Params},
#st{fsnode=Node, pbxpid=Pid}=State) -
{ok, XML} = gen_server:call(Pid, {fetch, Section, Tag, Key, Value,
Params}),
{api, Node} ! {fetch_reply, FetchID, XML},
receive
ok -
{noreply, State};
{error, Reason} -
{stop, {error, Reason}, State}
end.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@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] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
Hi Michael, You can count with me for anything else, like documentation, coding/scripting, or any other FreeSWITCH related stuff. Regards, Diego 2009/9/14 João Mesquita jmesqu...@freeswitch.org You can assign two things to me. 1. libesl code documentation (partially done and Doxygened - needs cleaning) 2. Bug marshal. I am setting up the proper lab environment here to be able to test most stuff. Count me in for any questions I can answer and I am _always_ on IRC jmesquita On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins m...@freeswitch.orgwrote: Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server database schema, etc. If you are good with a scripting language then we could definitely use help with rounding out the docs for your favorite language. We could also use code samples, so ask for a contrib folder if you have things you would like to share. Or how about this: you read something on the wiki, it doesn't quite work when you try, so you tinker until you figure it out. Now you're in a position to update the wiki for everyone else's benefit, too. As you can see, you don't have to be a FreeSWITCH expert before you can help the project. What we really need are people who care about the project and want to see it flourish. If you are such a person then please contact me off list. Tell me what you're good at or where you would like to help. Many thanks for all of your support! -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
I'm good for coming up with some documentation (which I'm doing anyway for my guys at the office). Not that whats on the wiki isn't good and I'll likely steal; its all there if you read it. I'll submit this when I reach some measure of completeness. If its deemed good, great. If not, well, I probably need to know that anyway. Mike G. On Mon, Sep 14, 2009 at 4:32 PM, Diego Viola diego.vi...@gmail.com wrote: Hi Michael, You can count with me for anything else, like documentation, coding/scripting, or any other FreeSWITCH related stuff. Regards, Diego 2009/9/14 João Mesquita jmesqu...@freeswitch.org You can assign two things to me. 1. libesl code documentation (partially done and Doxygened - needs cleaning) 2. Bug marshal. I am setting up the proper lab environment here to be able to test most stuff. Count me in for any questions I can answer and I am _always_ on IRC jmesquita On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins m...@freeswitch.orgwrote: Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server database schema, etc. If you are good with a scripting language then we could definitely use help with rounding out the docs for your favorite language. We could also use code samples, so ask for a contrib folder if you have things you would like to share. Or how about this: you read something on the wiki, it doesn't quite work when you try, so you tinker until you figure it out. Now you're in a position to update the wiki for everyone else's benefit, too. As you can see, you don't have to be a FreeSWITCH expert before you can help the project. What we really need are people who care about the project and want to see it flourish. If you are such a person then please contact me off list. Tell me what you're good at or where you would like to help. Many thanks for all of your support! -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] compilation error with the latest codes
Thanks Brian. On Mon, Sep 14, 2009 at 8:55 PM, Brian West br...@freeswitch.org wrote: You have a merge conflict please svn revert sofia.c /b On Sep 14, 2009, at 3:46 AM, Jingwei Yang wrote: Hi Folks, I've got a compilation error with the latest codes (r14842) Making all in packages Creating mod_sofia_la-mod_sofia.lo Compiling mod_sofia.c ... Creating mod_sofia_la-sofia.lo Compiling sofia.c ... sofia.c: In function ‘sofia_handle_sip_r_invite’: sofia.c:3221: error: expected expression before ‘’ token make[5]: *** [mod_sofia_la-sofia.lo] Error 1 make[4]: *** [all] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 Does anyone have ideas about this? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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
