November 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

fs_cli Error
by tina＠a2unlimited.com 15 Nov '09

15 Nov '09

I'm trying to setup fs_cli on a server that is not running the FreeSWITCH server, and I keep getting the following error: "error while loading shared libraries: libedit.so.0: cannot open shared object file: No such file or directory" When I go to one of my FreeSWITCH servers, where fs_cli is working fine, I cannot find the existence of libedit.so.0 anywhere on the server, so I'm not sure what I'm missing... Any thoughts? - Tina

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Re: [Freeswitch-users] (no subject)
by Paul Thirumalai 15 Nov '09

15 Nov '09

Hi Brian, Thanks for your clarification. By uuid do you mean uuid of the channel? Can you tell me how I could determine the uuid of the channel Thanks Paul On Sat, Nov 14, 2009 at 6:36 PM, Paul Thirumalai <paul.thirumalai(a)gmail.com>wrote: > Paul, > The channel name is useless it means absolutely NOTHING in > FreeSWITCH... what matters in FreeSWITCH is the uuid. You can even > set the channel names to what ever you like with the include app in > > dptools. > > set_name,Name the channel,<name>,mod_dptools > > So to answer your question. No it shouldn't have the gateway name in > the channel name... As for the 500 error your provider responded 500 > > Internal Server Error. > > Pre-Answer is usually early media. > > /b > > On Nov 14, 2009, at 7:11 PM, Paul Thirumalai wrote: > > >* Hello All > *>* I am a freeswitch newbie. I have managed to get internal phones > *>* setup on different computers at home, using X-lite > *>* My issue is with making outbound calls. I'm trying to use voicepulse > *>* for that. My Freeswitch server IP is 11.111.123.23 > *>* > *>* My issue is as follows. I used extension 1000 (X-Lite softphone) to > *>* make an out going call my cell phone (555-123-1234) > *>* From the freeswitch.log logfile I see that the channel for 1000 goes > *>* from CS_NEW-> CS_INIT->CS_ROUTING . At this point it starts going > *>* through the dial plans and finds the correct dialplan. According to > *>* the dialplan the outgoing call needs to go to sofia/gateway/ > *>* voicepulse/5551231234 and I see the following lines in the logfile > *>* EXECUTE sofia/internal/1000 at 11.111.123.23 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> hash(insert/11.111.123.23- > *>* last_dial/global/115be3f6-d01c-11de-8360-976b377ef920) > *>* EXECUTE sofia/internal/1000 at 11.111.123.23 <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> bridge(sofia/gateway/ > *>* voicepulse/5551231234) > *>* > *>* > *>* This message makes sense to me and appears to be right. Freeswitch > *>* is trying to bridge a call from extension # 1000 to outbound number > *>* 5551231234 > *>* > *>* Now the very next line I see freeswitch attempting to create a > *>* channel to the outbound number. > *>* > *>* 2009-11-13 01:16:23.654519 [NOTICE] switch_channel.c:602 New Channel > *>* sofia/external/5551231234 [115c7c76-d01c-11de-8360-976b377ef920] > *>* > *>* Please correct me if I'm mistaken, but isnt freeswitch supposed to > *>* create a channel for sofia/gateway/voicepulse/5551231234 and not > *>* sjofia/external/5551231235 > *>* > *>* > *>* In any case the channel to sofia/external/5551231234 changes state > *>* from CS_NEW->CS_INIT->CS_ROUTING->CS_CONSUME_MEDIA . At this point > *>* the freeswitch gets an Remote SDP, I am assuming from voicepulse. > *>* > *>* In the end I see message which reads > *>* 2009-11-13 01:16:30.880417 [DEBUG] mod_sofia.c:306 sofia/external/ > *>* 5035440933 Overriding SIP cause 503 with 500 from the other leg > *>* > *>* Can someone please tell me what this error means. SIP error code 500 > *>* is what I get in X-Lite also. > *>* > *>* Also could someone please explain what Pre-Answer is. > *>* > *>* Thanks > *>* Paul > *>* > *>* _______________________________________________ > *>* FreeSWITCH-users mailing list > *>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > *>* users > *>* http://www.freeswitch.org > * > >

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Question about channel creation messages in logfile
by Paul Thirumalai 15 Nov '09

15 Nov '09

Hello All I am a freeswitch newbie. I have managed to get internal phones setup on different computers at home, using X-lite My issue is with making outbound calls. I'm trying to use voicepulse for that. My Freeswitch server IP is 11.111.123.23 My issue is as follows. I used extension 1000 (X-Lite softphone) to make an out going call my cell phone (555-123-1234) >From the freeswitch.log logfile I see that the channel for 1000 goes from CS_NEW-> CS_INIT->CS_ROUTING . At this point it starts going through the dial plans and finds the correct dialplan. According to the dialplan the outgoing call needs to go to sofia/gateway/voicepulse/5551231234 and I see the following lines in the logfile EXECUTE sofia/internal/1000@11.111.123.23hash(insert/11.111.123.23-last_dial/global/115be3f6-d01c-11de-8360-976b377ef920) EXECUTE sofia/internal/1000(a)11.111.123.23bridge(sofia/gateway/voicepulse/5551231234) * *This message makes sense to me and appears to be right. Freeswitch is trying to bridge a call from extension # 1000 to outbound number 5551231234 Now the very next line I see freeswitch attempting to create a channel to the outbound number. 2009-11-13 01:16:23.654519 [NOTICE] switch_channel.c:602 New Channel sofia/external/5551231234 [115c7c76-d01c-11de-8360-976b377ef920] Please correct me if I'm mistaken, but isnt freeswitch supposed to create a channel for sofia/gateway/voicepulse/5551231234 and not sjofia/external/5551231235 In any case the channel to sofia/external/5551231234 changes state from CS_NEW->CS_INIT->CS_ROUTING->CS_CONSUME_MEDIA . At this point the freeswitch gets an Remote SDP, I am assuming from voicepulse. In the end I see message which reads 2009-11-13 01:16:30.880417 [DEBUG] mod_sofia.c:306 sofia/external/5035440933 Overriding SIP cause 503 with 500 from the other leg Can someone please tell me what this error means. SIP error code 500 is what I get in X-Lite also. Also could someone please explain what Pre-Answer is. Thanks Paul

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Large number of destinations
by Robin Vleij 14 Nov '09

14 Nov '09

Hi all, I'm currently building a proof-of-concept box using Freeswitch. Coming from Asterisk/Kamalio/OpenSER it looks very cool so far, very complete. The plan is to make some sort of SIP router, some would call it an SBC I guess. There will be no PBX stuff, just gateways that talk to each other. PSTN Gateways or other operators or systems. If a system is locally connected (say a local voip platform or interconnected partner), traffic to those destinations should be routed directly to that system and not out to PSTN. I'm looking at a potentially large nr of destination nrs or ranges. Not all those destinations are in the local ENUM so I can't use that as a routing system. I'm thinking about mod_lcr, but it seems more suited for eh ... LCR routing, which is not what I want to do here. I just want to define which nrs or nr ranges are "directly" connected, so that when someone calls there from whatever way they come in (I'm running just one instance and thought about defining all gateways/systems as gateways in the SIP profile), they should end up there and not at PSTN. I think I have two ways of doing this: 1. Make a HUGE XML dialplan and use that to fall back to when internal ENUM lookup doesn't give a result back to where a nr is located 2. Use LCR and find out some kind of way to load all of these destinations into a LCR table and use it in the "wrong" way, ie no costs are involved, it should just be a way to know which nrs or ranges are to be sent to which gateway. Nr1 is probably best (anyone experience how many conditions one can have dialplan_xml?), but say that we would exchange traffic with an operator, it would really suck writing an XML dialplan with 5000 number ranges. :) Anyone experience with this or ideas how this can be solved? Since it's a proof-of-concept it's unclear how exactly those customers or systems are looking. They might be 6000 (un)ported individual nrs, or just a few large ranges. /Robin

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problem with mod_xml_odbc
by Woody Dickson 14 Nov '09

14 Nov '09

Hi, I am having problem trying to use mod_xml_odbc using freeswitch-1.0.5pre. Here is the error I am getting: 2009-11-15 00:17:23.571293 [INFO] mod_xml_odbc.c:647 XML ODBC module loading... 2009-11-15 00:17:23.571354 [NOTICE] mod_xml_odbc.c:563 Binding XML Search Function [directory] 2009-11-15 00:17:23.572299 [ERR] switch_odbc.c:188 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2009-11-15 00:17:23.572361 [CRIT] mod_xml_odbc.c:617 Cannot Open ODBC Database! 2009-11-15 00:17:23.572397 [ERR] mod_xml_odbc.c:650 Unable to load xml_odbc config file 2009-11-15 00:17:23.572424 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_xml_odbc.so **Module load routine returned an error** In my config, I have: <settings> <param name="binding" value="directory"/> <param name="odbc-dsn" value="myodbc:root:123456"/> <param name="debug" value="true"/> <param name="keep-files-around" value="true"/> </settings> [root@localhost autoload_configs]# isql myodbc -v +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> How can I fix this problem? thanks, woody

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FreeSWITCH Now Supports Broadvoice BV16, BV32 Voice Codecs
by Dome Charoenyost 14 Nov '09

14 Nov '09

http://freeswitch.org/node/217 Very fast develop :) one reason why i love FS. another is good performance Dome C.

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Freeswitch configure error using --srcdir option
by Robert Hadley 14 Nov '09

14 Nov '09

Hello All On CentOS 5.3, I am trying to build Freeswitch in a different directory and use the -srcdir= option. One reason I want to do this to have Debug and Release build targets from the same source. It doesn't work, the configure errors when it gets to the first library subdirectory lib/srtp and tries to configure in there. The steps I am doing are: 1. Building as root 2. Unzip freeswitch-1.0.4-tar.gz in /opt 3. cd into /opt/freeswitch-1.0.4 4. mkdir Debug 5. cd Debug 6. ../configure -srcdir=".." CFLAGS="-g -ggdb -O2" 7. After several seconds of configuring I get: === configuring in libs/srtp (/opt/freeswitch-1.0.4/Debug/libs/srtp) configure: running /bin/sh ../../../libs/srtp/configure.gnu --disable-option-checking '--prefix=/usr/local/freeswitch' 'CFLAGS=-g -ggdb -O2' --cache-file=/dev/null --srcdir=../../../libs/srtp ../../../libs/srtp/configure.gnu: line 2: ./configure: No such file or directory configure: error: ../../../libs/srtp/configure.gnu failed for libs/srtp [root@roberth-c53 Debug]# The file that's executing is this: [root@roberth-c53 srtp]# cd libs/srtp; cat ../../../libs/srtp/configure.gnu #! /bin/sh ./configure "$@" --disable-shared --with-pic Please tell me if I understood the -srcdir option correctly and if there is a way to do build in a different directory. Thanks, Robert

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Using play_and_get_digits (or IVR) without a final delay
by Robert L Mathews 14 Nov '09

14 Nov '09

Hi, I'm a new FreeSWITCH convert from asterisk. It's great software; thanks for making it. I'm trying to play a sound file while listening for possible digits dialed (although in most cases callers will not be dialing anything). If callers do start dialing an extension while the sound file is playing, I want them to be able to dial it slowly without any problems. So the inter-digit timeout should be, say, 2 seconds. However, if people don't start dialing anything while the sound file is playing, I don't want any delay at the end of it. I've tried play_and_get_digits with a 2000 timeout -- but that causes a 2 second "dead air" pause at the end of the sound file if callers don't dial anything. I've also tried using a trivial IVR menu to simulate this, but it has the same problem. Interestingly, it doesn't *look* like IVRs should have the problem, because they allow both "inter-digit-timeout" and "timeout" to be specified separately -- but at least in FreeSWITCH 1.0.4, a short "timeout" value always interrupts the dialing of digits, even if "inter-digit-timeout" is much longer. Is there any way to play a sound file from the dial plan with a long inter-digit delay, but without any final delay if no digits are dialed? Thanks for your time! -- Robert L Mathews, Tiger Technologies

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suggestions for hardware.
by Orien Love 13 Nov '09

13 Nov '09

First of all, Thanks to the help I received on my pfSense installation, especially to Michael. I have a basic test system up and running. I am still waiting on some hardware but the base system is working!!!! I am looking on advice on how to set up a simple office PBX, 20 phones and 4 outside lines.with 2 or 3 "operator" phones and the rest will be extensions. Here is my plan, please let me know if it does not make sense, or if I am going about it System Hardware 4 spa3000's to handle the outside lines. 2-3 polycom 601 phones with expansion modules (Operator phones) 18 polycom 330 or other phones for desks. 2-24 port cisco POE switches 1 pfSense server. System Design. Extension Numbers 2xx Outside line access 1xxxxxxxxxx groups 3xx auto-attendent ??? here are my questions #1 will a 1.6 Ghz Intel Atom 230 single core 533 Mhz FSB and 2 GB of memory handle this proposed system? (Here is the MB I am thing of using MSI 609-9832-010 http://www.logicsupply.com/products/ms_9832_010) #2 how do I pool my spa 3000 FXO lines so that the outgoing calls use the first available line? also how do insure that metro (non long distance) calls go to a specific line if available? I have learned a lot on how to set up Polycom 601 phones, I am planning on writing a how to document, is there any specific format? Thanks Orien

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How To Disable MD5 Authentication?
by Jerry Richards 13 Nov '09

13 Nov '09

How can I disable MD5 Authentication upon registration? Best Regards, Jerry

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