- FreeSWITCH-users - FreeSWITCH Mailing Lists

Wiki typo
by Dmitry Gromov 04 Nov '09

04 Nov '09

Hi! Was just reading wiki here: http://wiki.freeswitch.org/wiki/Home_PBX_Example It lists sample sofia.conf.xml which has this parameter:  I think it should read inbound-*bypass*-media and not inbound-*no*-media... I know, it says "outdated" but still, can be confusing. Anyone here who can edit wiki and correct? Thanks, Dmitry -- DG NJ

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3Com 3102 (3C10402B) Phone with FreeSwitch
by Dave Stevenson 04 Nov '09

04 Nov '09

Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension & password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave

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Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media
by Humberto Quintana 04 Nov '09

04 Nov '09

Hi, I tried r15332 and set in the sofia profile: a) bypass_media_after_bridge=true only b) bypass_media_after_bridge=true, param name="media-option" value="resume-media-on-hold"/> <param name="media-option" value="bypass-media-after-att-xfer"/> In both cases FS is hanging up the initial call (A to FS) after accepting the REFER to C: A <- reINVITE with FS' SDP <- FS A -> 200 -> FS A <- ACK <- FS A <- BYE <- FS The call to C is not even tried. I found this line is the logs that could give some idea: 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup sofia/external/514xxxxxx(a)a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] after sending the ACK for the reINVITE Regards, Humberto >please try r15326 >I think i have it working. > >I recommend for optimal results you set bypass_media_after_bridge=true >either as a global or in your DP in place of bypass_media=true > > >On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana <hjqlopez at hotmail.com>wrote: > >> Hi Mike, >> >> I re-tried with trunk rev 15319 but I got almost the same behavior: There >> is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But >> still there is no reINVITE for A (with C's SDP) after the call from FS to C >> is established. >> >> Anyway, we decided for now to do a different implementation but if you want >> to explore more in this issue count me in ;-) >> >> >> Thank you very much! >> >> Humberto _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://go.microsoft.com/?linkid=9691817

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Error checking for PMP [general error]
by Jerry Richards 03 Nov '09

03 Nov '09

When I start Freeswitch, I see an "Error checking for PMP [general error]" as shown below. Does anyone know what could cause this? [root@TeoProxy bin]# ./freeswitch Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch -waste. auto-adjusting stack size for optimal performance.... 2009-11-02 10:12:27.17579 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-02 10:12:27.18373 [DEBUG] switch_event.c:553 Create event dispatch thread 0 2009-11-02 10:12:27.428749 [INFO] switch_nat.c:392 Scanning for NAT 2009-11-02 10:12:27.428885 [DEBUG] switch_nat.c:152 Checking for PMP 1/5 2009-11-02 10:12:27.678480 [DEBUG] switch_nat.c:152 Checking for PMP 2/5 2009-11-02 10:12:27.679449 [DEBUG] switch_nat.c:152 Checking for PMP 3/5 2009-11-02 10:12:28.179388 [DEBUG] switch_nat.c:152 Checking for PMP 4/5 2009-11-02 10:12:29.179217 [DEBUG] switch_nat.c:152 Checking for PMP 5/5 2009-11-02 10:12:31.178879 [ERR] switch_nat.c:183 Error checking for PMP [general error] 2009-11-02 10:12:31.178902 [DEBUG] switch_nat.c:397 Checking for UPnP 2009-11-02 10:12:43.176881 [INFO] switch_nat.c:411 No PMP or UPnP NAT detected! 2009-11-02 10:12:43.210145 [INFO] switch_core_sqldb.c:538 Opening DB 2009-11-02 10:12:43.919804 [NOTICE] switch_scheduler.c:166 Starting task thread 2009-11-02 10:12:43.937881 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1257185563 2009-11-02 10:12:43.937980 [CONSOLE] switch_core.c:1449 Bringing up environment. 2009-11-02 10:12:43.937994 [CONSOLE] switch_core.c:1450 Loading Modules. 2009-11-02 10:12:43.938319 [INFO] switch_time.c:661 Timezone loaded 530 definitions 2009-11-02 10:12:43.938336 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_SOFTTIMER_MODULE] 2009-11-02 10:12:43.938351 [NOTICE] switch_loadable_module.c:228 Adding Timer 'soft' 2009-11-02 10:12:43.938413 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_PCM_MODULE] Best Regards, Jerry

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portaudio error
by Frank Carmickle 03 Nov '09

03 Nov '09

Hello Debian lenny with svn15321 freeswitch@internal> load mod_portaudio -ERR [module load file routine returned an error] 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input devicefreeswitch@internal> 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974 Cannot find an input device 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_portaudio.so **Module load routine returned an error** <configuration name="portaudio.conf" description="Soundcard Endpoint"> <settings>   <param name="indev" value="1"/>  <param name="outdev" value="1"/>  <param name="ringdev" value="1"/>        <param name="hold-file" value="$${hold_music}"/>    <param name="dialplan" value="XML"/> <param name="cid-name" value="$${outbound_caller_name}"/> <param name="cid-num" value="$${outbound_caller_id}"/>  <param name="sample-rate" value="48000"/> <param name="codec-ms" value="10"/> </settings> </configuration> Any help would be appreciated. --FC

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FreeSWITCH 1.0.5 Scheduled For November 10; 1.0.5pre5 Now Available
by Michael Collins 03 Nov '09

03 Nov '09

Greetings! The latest FreeSWITCH pre-release is now available: http://www.freeswitch.org/node/215 Please update and test as soon as possible. With the community's help we should be able to hit our target of releasing version 1.0.5 next Tuesday November 10. The FreeSWITCH developers appreciate all the hard work that the community does on behalf of the project. Like most open source projects, FreeSWITCH needs the community to "give back" a little and you all certainly do that. Please keep up the great work. -Michael

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How to get digitals and stop play when speak tts? Just like session:playAndGetDigits do
by Lei Tang 03 Nov '09

03 Nov '09

Hi all, I'm writing lua ivr scirpt, Does some known how to get digitals and stop play when speak tts? Just like session:playAndGetDigits do. Thanks lots! Best Regards! -- Lei.Tang lei.tlfly(a)gmail.com

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"Can not record session. Media not enabled on channel."
by Maciej Aniserowicz 03 Nov '09

03 Nov '09

Hello, I am using the same set of extensions for testing the system during development, they include XLite, Cisco sip phone and several extensions that just play some audio file. Sometimes, very rarely, this message "Can not record session. Media not enabled on channel." appears on FS console. Like I wrote before, I don't change any codec settings and always use the same set of devices/emulators. What can cause this message? Thanks, Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Can-not-record-session-Media-not-enabled-on-channel-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media
by Humberto Quintana 03 Nov '09

03 Nov '09

Hi Mike, I re-tried with trunk rev 15319 but I got almost the same behavior: There is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But still there is no reINVITE for A (with C's SDP) after the call from FS to C is established. Anyway, we decided for now to do a different implementation but if you want to explore more in this issue count me in ;-) Thank you very much! Humberto >Please re-try with latest svn trunk. > >Mike > >On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote: > >> >> Thanks for you answers guys, >> >> I test the parameters you suggested >> but still no audio due to the lack of reINVITE. By the way I'm using >> 1.0.4 but I also tried 1.0.5pre3. >> >> One particular condition is that there is no on-hold before the >> Blind Transfer. >> >> Regards, >> >> Humberto >> >>> <param name="media-option" value="resume-media-on-hold"/> >>> <param name="media-option" value="bypass-media-after-att-xfer"/> _________________________________________________________________ Lots of fantastic Windows 7 offers, in one convenient place. Get the perfect deal for you now. http://go.microsoft.com/?linkid=9691633

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SIP provider with extern rtp server
by Ivan C Myrvold 02 Nov '09

02 Nov '09

I have used a SIP provider for more than a year. A few days ago, he said he was moving to a new server, and asked me to reconfigure. I did, and everything seemed to work fine, until I did an outgoing call to an external telephone. I found out I had no audio, in neither direction. Incoming calls was working fine. My provider said that the rtp is not going through the sip server, as it did earlier, but now through several other IP's. Do I have to do some special configuration to handle that? Ivan

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