Hey all, I'm trying freeswitch for the first time and I have a couple of questions: I have a SIP account from a provider that I use to make calls to cell phones and other, how can I use from freeswitch? Do I need to put the configurations (domain/user/password/proxy etc) in sofia.conf.xml or directory.xml and what's the difference? Once I get that configured, how can I transfer calls that are received to other numbers? When I use the console, I get plenty of messages such as: nta_agent: received garbage from udp/212.27.52.5:5060/sip is there a way to stop that? (it happened after I added my sip account to sofia.conf.xml) Is there a gui or someething to configure freeswitch? thanx in advance Pat
On Oct 20, 2007, at 10:13 PM, Patrick Aljord wrote:
Hey all,
I'm trying freeswitch for the first time and I have a couple of questions:
I have a SIP account from a provider that I use to make calls to cell phones and other, how can I use from freeswitch? Do I need to put the configurations (domain/user/password/proxy etc) in sofia.conf.xml or directory.xml and what's the difference?
I recommend putting them in directory.xml as that is the best place to centralize your configs for domains. But right now both are acceptable.
Once I get that configured, how can I transfer calls that are received to other numbers?
Via the tranfer app or via a sip transfer.
When I use the console, I get plenty of messages such as: nta_agent: received garbage from udp/212.27.52.5:5060/sip is there a way to stop that? (it happened after I added my sip account to sofia.conf.xml)
These are usually nat keep alive packets sent by the client and are nothing to be concerned with.
Is there a gui or someething to configure freeswitch?
http://www.wikipbx.org/ and your favorite text editor.
thanx in advance
Pat
ok thanx a lot, another couple of questions: what's the app to make a call? what are dialplan extensions? by the way freeswitch segfaults when I call my sip number from my mobile phone :/
Did you even look at the wiki? http://wiki.freeswitch.org/wiki/Getting_Started_Guide What platform are you on? what do the backtraces show? Are you running the latest code from right now? /b On Oct 20, 2007, at 10:48 PM, Patrick Aljord wrote:
ok thanx a lot, another couple of questions:
what's the app to make a call?
what are dialplan extensions?
by the way freeswitch segfaults when I call my sip number from my mobile phone :/
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On 10/21/07, Brian West <brian.west@mac.com> wrote:
Did you even look at the wiki?
Yes I did and yet I still don't get how to place calls using my providers and how to create conferences. Is it from the console or from a phone? The console is unusable because of the "nta_agent: received garbage from udp/212.27.52.5:5060/sip" message I get every second. I tried "pa call 1000" but I get "Unknown Command: pa" yet I did compile the module and it is listed in mod/mod_portaudio.so
What platform are you on? what do the backtraces show? Are you running the latest code from right now?
I'm using ubuntu gutsy with yesterday's svn trunk. I don't have backtraces for now sorry. Sorry for all the questions, it just feels like the wiki is full of xml configuration tips but I can seem to get how to do actual things (receive calls, create conference etc...) but I guess that's just me.
On 10/21/07, Patrick Aljord <patcito@gmail.com> wrote:
I tried "pa call 1000" but I get "Unknown Command: pa" yet I did compile the module and it is listed in mod/mod_portaudio.so
I was able to use portaudio by doing "load mod_portaudio" (I had to figure it out because I couldn't find it in the wiki). then I tried "pa call 1000" and I got: API CALL [pa(call 1000)] output: SUCCESS:3:d789457e-7fe1-11dc-a3ea-837c6a459406 is something else supposed to happen?
I don't have backtraces for now sorry.
here it is from gdb: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1261134960 (LWP 18180)] 0xb7ceef13 in strchr () from /lib/tls/i686/cmov/libc.so.6
You'll need to go into your src directory and open modules.conf and add portaudio to it.. ie uncomment it. Then make modules, make install again to get it.. but all the PA examples are removed from the default config. You need to get on IRC and ask questions because this email thing is too slow. /b On Oct 21, 2007, at 9:29 AM, Patrick Aljord wrote:
On 10/21/07, Patrick Aljord <patcito@gmail.com> wrote:
I tried "pa call 1000" but I get "Unknown Command: pa" yet I did compile the module and it is listed in mod/mod_portaudio.so
I was able to use portaudio by doing "load mod_portaudio" (I had to figure it out because I couldn't find it in the wiki). then I tried "pa call 1000" and I got: API CALL [pa(call 1000)] output: SUCCESS:3:d789457e-7fe1-11dc-a3ea-837c6a459406
is something else supposed to happen?
I don't have backtraces for now sorry.
here it is from gdb: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1261134960 (LWP 18180)] 0xb7ceef13 in strchr () from /lib/tls/i686/cmov/libc.so.6
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participants (2)
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Brian West
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Patrick Aljord