April 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

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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

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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

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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 >

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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

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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

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Re: [Freeswitch-users] Scheduled Hangup Issue
by Anthony Minessale 16 Apr '08

16 Apr '08

I added support in the latest trunk to set the variable execute_on_answer to an application of your choice <action application="set" data="execute_on_answer=sched_hangup +20 alloted_timeout"/> -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Asterisk vs. Freeswitch
by Pete Kay 16 Apr '08

16 Apr '08

Hi, This question may have come up a few times already. I am working on a application to provide IVR, voicemail, and tailored call routing services. The SIP registration will be handled by Openser, and Asterisk is only doing the media function. We are talking about over 100 users. Is Freeswitch better than Asterisk in terms of functionality, ease-of-maintain, and ease-of-use? Thanks alot in advance for your inputs. Pete

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What is Freeswitch equivalent of Asterisk AMI and Realtime
by Pete Kay 16 Apr '08

16 Apr '08

Hi, I have studied the Freeswitch doc and can't find any info related to setting up the sip users, queue, ans voicemail as a realtime DB, like Asterisk does. Is this feature available? If not, would it be a bit too much work to write to the XML all the times? Also, is there an Asterisk AMI-equivalent feature in Freeswitch? What is that called? Thanks for your input. Pete

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Re: [Freeswitch-users] Asterisk vs. Freeswitch - what about functionality
by Pete Kay 16 Apr '08

16 Apr '08

Hi, I have been seeing quite a few articles that mention Freeswitch is much more efficient than Asterisk. What about in terms of functionality? Can Freeswitch perform the same kind of functionality provided by Asterisk such as Queue, Conf call, MOH, Voicemail Main, etc? Can anyone give some inputs on this regards? Thanks, Pete

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