- FreeSWITCH-users - FreeSWITCH Mailing Lists

newbie dialplan question
by Pete Kay 18 Apr '08

18 Apr '08

Hi, I am working on some test on seeing how I can port my exist Asterisk stuff to Freeswitch. I am just getting started and I am hoping someone can give me some help to get started. I installed with all the default config and xml setting. Then, I bring up two SiP clients - one in the same machine as freeswitch (192.168.1.104) and the other one on another machine ( 192.168.1.102). When I dial an extension ( 1001, or 1002... etc ) from my SIP client on 192.168.1.102, I can make the call to the other client no problem. 2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1002@192.168.1.104:5060[9e6d6146-0c98-11dd-bb92-f1e303d528b1] 2008-04-18 00:09:12 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1002->81001@default 2008-04-18 00:09:12 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001@192.168.1.104:5061[9e93e564-0c98-11dd-bb92-f1e303d528b1] 2008-04-18 00:09:12 [NOTICE] sofia.c:1603 sofia_handle_sip_i_state() Ring-Ready sofia/default/1001@192.168.1.104:5061! However, when I dial extension from the other SIP client, the one on 192.168.1.104, the call can't be routed. 2008-04-18 00:10:34 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001@192.168.1.104:5061[cf710b58-0c98-11dd-bb92-f1e303d528b1] 2008-04-18 00:10:34 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1001->1002@public 2008-04-18 00:10:34 [NOTICE] switch_core_state_machine.c:198 switch_core_standard_on_execute() Hangup sofia/default/ 1001@192.168.1.104:5061 [CS_EXECUTE] [NORMAL_CLEARING] 2008-04-18 00:10:34 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 20 (sofia/default/ 1001@192.168.1.104:5061) Ended It seems like the the call is being routed to the wrong context. How come this happens? I am using the standard default config xml files. Can anyone please help me? With freeswitch, is there anyway to debug/trace the processing of the call so I can see which condition it is in, and where it is routed to? That way, I can debug the config easier? Thanks alot for all your inputs and help. Regards, Pete

4 19

multi-language voice
by Pete Kay 18 Apr '08

18 Apr '08

Hi, I have a questions about multi-language voice support in FS. How to I add a new module such as mod_fr.so? I can make the necessary wav files, but how to create the modules? Thank you very much for your inputs. Regards Pete

1 0

Build problem on Debian slice
by Anthony Knight 18 Apr '08

18 Apr '08

I'm having a problem building freeswitch on a xen debian etch slice (Details below). Everything from the standard modules.conf builds except for mod_spidermonkey. Here are the details of the build error. making all mod_spidermonkey cd config; make -j1 export cd pr; make -j1 export cd include; make export cd md; make export cd private; make export cd obsolete; make export cd src; make export cd io; make export cd malloc; make export cd md; make export cd unix; make export os_Linux_x86_64.s: Assembler messages: os_Linux_x86_64.s:46: Error: bad register name `%rdi)' os_Linux_x86_64.s:61: Error: bad register name `%rdi)' os_Linux_x86_64.s:75: Error: bad register name `%rdi)' os_Linux_x86_64.s:89: Error: bad register name `%rdi)' make[10]: *** [os_Linux_x86_64.o] Error 1 make[9]: *** [export] Error 2 make[8]: *** [export] Error 2 make[7]: *** [export] Error 2 make[6]: *** [export] Error 2 make[5]: *** [/usr/src/freeswitch-1.0/libs/js/libjs.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_spidermonkey-all] Error 1 make[2]: *** [all-recursive] Error 1 Details of my environment: ~:cat /proc/version Linux version 2.6.18-xenU (root(a)host277.rimuhosting.com) (gcc version 4.1.1 20070105 (Red Hat 4.1.1-52)) #4 SMP Wed Oct 24 01:50:19 UTC 2007 ~:uname -a Linux xyz.xyz.com 2.6.18-xenU #4 SMP Wed Oct 24 01:50:19 UTC 2007 x86_64 GNU/Linux ~:uname -m x86_64 ~:gcc --version gcc (GCC) 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) from googling about I think this is the same problem documented here ( http://trac.callcc.org/ticket/407) where the environment is reporting ARCH=x86_64, but should really be ARCH=x86. Any suggestions on how to overcome this problem? Thanks Tony

2 3

Segfault
by David Knell 17 Apr '08

17 Apr '08

Hi all - Just had an interesting FS segfault. switch_xml_free (called from mod_xml_cdr) recursed 3500+ times and, I guess, the thing ran out of stack space. Having a look at the core dump shows that it's freeing a valid XML CDR structure for a call which wasn't cleared for ages; FS sat in a loop playing a prompt. The net result would have been a CDR with a vast number of <application> lines, had the segfault not occurred. Fixes: (a) don't get in this situation in the first place; (b) rewrite switch_xml_free to eliminate recursion; (c) limit the number of children created in switch_ivr_generate_xml_cdr; - and probably some more. My favourite's (c) - any alternatives/objections? If not, I'll pop this with a patch in Jira. Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk

2 1

Need help with a gateway problem
by Pete Kay 17 Apr '08

17 Apr '08

Hi, I am still not yet able to get one sip phone to call the other due to the problem I posted. So, I used the working sip client to try to call an outside number by setting up a gateway. I tried to set up fastswitch to route call to my voip provider, I am getting channel error but can't know why. Can anyone please give some help? this is in my sofia.conf.xml <gateways> <gateway name="callwithus">  <param name="username" value="226"/>  <param name="realm" value="callwithus"/>  <param name="password" value="3336"/>  <param name="proxy" value="sip.callwithus.com"/>  <param name="expire-seconds" value="600"/> </gateway> </gateways> this is in my default dialplan <context name="home"> <extension name="PrintVars" continue="true"> <condition field="destination_number" expression="^[0-9]"> <action application="info"/> </condition> </extension> <extension name="PBX Extension"> <condition field="destination_number" expression="^(1[0-9]{2})$"> <!-- This will dial a registered phone at the ip or domain name you set in$ <!-- (The % indicates it is an internal extension) $ <action application="bridge" data="sofia/sip/$1%${ server-domain-name}"/> </condition> </extension> <extension name="Local Dial"> <condition field="destination_number" expression="^([0-9]{13})$"> <action application="set" data="effective_caller_id_name=John Freeswitch"/> <action application="set" data="effective_caller_id_number=1234567"/> <action application="log" data="before bridging [${1}]\n"/> <action application="bridge" data="sofia/gateway/callwithus/$1@callwithus"$ </condition> </extension> </context> Get error when I dial a 13 digits number: 2008-04-18 04:26:03 [ERR] mod_sofia.c:1692 sofia_outgoing_channel() Invalid Gateway 2008-04-18 04:26:03 [NOTICE] mod_sofia.c:1857 sofia_outgoing_channel() Close Channel N/A [CS_NEW] 2008-04-18 04:26:03 [ERR] switch_ivr_originate.c:813 switch_ivr_originate()Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2008-04-18 04:26:03 [INFO] mod_dptools.c:1536 audio_bridge_function()Originate Failed. Cause: INVALID_NUMBER_FORM AT Any suggestion will be greatly appreciated. Thanks Pete

3 5

Missing bootstrap
by Peder ＠ NetworkOblivion 17 Apr '08

17 Apr '08

FYI, rc2 and rc3 both appear to be missing bootstrap.sh. It is in rc1. I copied it from rc1 to rc3 and it squawked about missing docs/COPYING and docs/AUTHORS, but it appears to be running the rest of the setup so far. Peder

4 3

RC3 Callie & Music
by Peder ＠ NetworkOblivion 17 Apr '08

17 Apr '08

OK, I got through the make and install with rc3, but now I am trying "make sounds-install" and "make sounds-music-8000-install" and it is failing. It appears to be trying to download the following files which don't exist yet: http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.rc3.tar… http://files.freeswitch.org/freeswitch-sounds-music-8000-1.0.rc3.tar.gz Is there a way to copy the rc2 files locally so it will use those instead? Or can someone add the rc3 versions to files.freeswitch.org? I assume they rc1-rc3 sounds are the same, just different names. Peder

4 4

Re: [Freeswitch-users] Question about installing freeswitch - SOLVED
by Pete Kay 17 Apr '08

17 Apr '08

Hi, I made a silly mistake. The problem was solved. Please ignore. Thanks. Regards, Pete On Thu, Apr 17, 2008 at 12:38 PM, Pete Kay <petedao(a)gmail.com> wrote: > Hi, > > I followed the online installation but when I executed, make sure or > make moh, I am getting the following errors: > > ser:/usr/src/freeswitch-1.0.rc2/build# make moh-install > make: *** No rule to make target `moh-install'. Stop. > ser:/usr/src/freeswitch-1.0.rc2/build# > > What could be the problem? > > My installation procedure is exactly same as the freeswitch wiki. > > Thanks, > Pete >

1 0

Question about installing freeswitch
by Pete Kay 17 Apr '08

17 Apr '08

Hi, I followed the online installation but when I executed, make sure or make moh, I am getting the following errors: ser:/usr/src/freeswitch-1.0.rc2/build# make moh-install make: *** No rule to make target `moh-install'. Stop. ser:/usr/src/freeswitch-1.0.rc2/build# What could be the problem? My installation procedure is exactly same as the freeswitch wiki. Thanks, Pete

1 0

Freeswitch with Hylafax and Freeswitch with E1
by Pete Kay 17 Apr '08

17 Apr '08

Hi, Two more questions about Freeswitch. I have some application written to send and receive fax using Hylafax and IAXModem, will I be able to port that to Freeswitch? What is the best way to connect Freeswitch to an E1? Does anyone has experience with any easy-to-use, easy-to-maintain hardware with reasonable quality for voice and fax? Will a PCI gateway card be sufficient or do I need something like a Cisco voice gateway which is quite expensive to purchase and maintain? In fact, what is the key advantages and disadvantages between a PCI card option and a Cisco option? Thank you so much in advance for your inputs. Regards, Pete

1 0