September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Asterisk 1.6 connecting to FS 1.4
by paul.degt＠gmail.com 14 Sep '09

14 Sep '09

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

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Re: [Freeswitch-users] Pastebin Username/Password Not Accepted
by Jerry Richards 14 Sep '09

14 Sep '09

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.richards@teotech.com] Sent: Monday, September 14, 2009 8:13 AM To: 'freeswitch-users(a)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

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Pastebin Username/Password Not Accepted
by Jerry Richards 14 Sep '09

14 Sep '09

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

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FS create directory
by Tihomir Culjaga 14 Sep '09

14 Sep '09

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.

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How to filter the allowed string
by Ahmed Munir 14 Sep '09

14 Sep '09

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

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problem with performance testing
by Woody Dickson 14 Sep '09

14 Sep '09

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? freeswitch(a)mycom.com> 2009-09-14 20:59:16.777675 [ERR] switch_core_io.c:118 sofia/external/12323232@192.168.1.116:5911 has no read codec. 2009-09-14 20:59:30.815547 [ERR] switch_core_io.c:118 sofia/external/ 12323232@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/ 12323232@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/ 12323232@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:04.495545 [ERR] switch_core_io.c:118 sofia/external/ 12323232@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:16.996438 [ERR] switch_core_io.c:118 sofia/external/ 12323232@192.168.1.116:5911 has no read codec. 2009-09-14 21:00:25.905617 [ERR] switch_core_io.c:118 sofia/external/ 12323232@192.168.1.116:5911 has no read codec. freeswitch(a)mycom.com> show channels count API CALL [show(channels count)] output: 1875 total. freeswitch(a)mycom.com> 2009-09-14 21:00:36.212767 [ERR] switch_core_io.c:118 sofia/external/12323232@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/12323232@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232@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/12323232@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232@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/12323232@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232@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/12323232@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232@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/12323232@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232@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/12323232@192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232@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

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Asterisk 1.6 connecting to FS 1.4
by paul.degt＠gmail.com 14 Sep '09

14 Sep '09

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="^555557777777$"> <action application="bridge" data="sofia/internal/4000000@$${domain}|sofia/internal/4000001@$${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/4000000[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=>4000000:mysippassword@versafon.com/4000000 >>>>>> >>>>>> [4000000] >>>>>> type=friend >>>>>> username=4000000 >>>>>> secret=mysippassword >>>>>> host=versafon.com >>>>>> canreinvite=no >>>>>> fromuser=4000000 >>>>>> dtmfmode=rfc2833 >>>>>> context=versafon-incoming

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freeswitch on blackfin + uclinux
by Juan Backson 14 Sep '09

14 Sep '09

Hi, Does anyone have any luck on porting freeswitch to blackfin + uclinux? Is this a feasible option? jb

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Sangoma A500 - dial out from specific port group?
by Vassil Panayotov 14 Sep '09

14 Sep '09

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. I tried to modify OpenZAP config as follows: conf/openzap.conf [span wanpipe boostbri1] 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 [span wanpipe boostbri2] trunk_type => bri b-channel => 7:1-2 b-channel => 8:1-2 conf/autoload_configs/openzap.conf.xml: <boost_spans> <span name="boostbri1">    <param name="dialplan" value="XML"/> <param name="context" value="isdn"/> </span> <span name="boostbri2"> <param name="dialplan" value="XML"/> <param name="context" value="isdn"/> </span> </boost_spans>

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voicemail problem
by Woody Dickson 13 Sep '09

13 Sep '09

Hi, While trying to record some sounds with the voicemail app, I keep getting message saying my record is below the minimal length even I was actually still speaking. Is it not detecting my voice? How can I configure it so that freeswitch's vm app can detect my speech? Thanks, woody

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