- FreeSWITCH-users - FreeSWITCH Mailing Lists

recommend PSTN / SIP trunking?
by Miles Keaton 13 Jan '09

13 Jan '09

I'm using Junction Networks (http://pstn.junctionnetworks.com/) for SIP trunking of phone numbers, and I'm happy with them. Their servers are in NYC. Now I'm trying to find a similar company whose servers are based in LA or San Diego. Any recommendations?

3 2

outbound call, new comer
by Will Smith 12 Jan '09

12 Jan '09

Hi, I am first time FS user, so it is a bit confused with all the setup. For inbound calls, I tried to add a voicepulse.xml in the sip_profiles/external with the following codes: <include> <gateway name="voicepulse"> <param name="username" value="3334445555"/> <param name="realm" value="my_sip_provider.com"/> <param name="password" value="3334445555"/> <param name="proxy" value="my_sip_provider.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> <gateway name="voicepulse-backup"> <param name="username" value="3334445555"/> <param name="realm" value="my_sip_provider.com"/> <param name="password" value="3334445555"/> <param name="extension" value="3334445555"/> <param name="proxy" value="my_sip_provider.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </include> ------------ and in the conf/dialplan/default.xml file I added:  <extension name="Long Distance - voicepulse"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/>   <action application="bridge" data="sofia/gateway/voicepulse/$1"/> </condition> </extension> ------------------ For inbound, I added <extension name="Voicepulse">  <condition field="destination_number" expression="3334445555">  <action application="transfer" data="1001 XML default"/> </condition> </extension> ---------- And, if I dial 3334445555 from a softphone registered with my_sip_provider, I got the message to the voice mail of 1001 - the 1001 extension does not ring. And if from 1001, I dial some real number like 18188892345, I got the error: Invalid Gateway ... Cannot create outgoing channel of type [fosia] cause: [Invalid_number_format] ... Would someone please give me some help to set this up. I am a bit confused with these. Thank you

3 5

mod_opal first unsuccessful test
by Apostolos Pantsiopoulos 12 Jan '09

12 Jan '09

Hi, I successfully compiled mod_opal using the latest svn for both opal and ptlib as Brian suggested. When I try to establish a call using h323 from my openphone client I get no audio although I can see RTP packets in both directions when I am doing a capture. Some details : I am using the 11094 revision of the FS trunk. I am using the PCMU codec. I tried dialing the default IVR (5000) and other testing extensions (freeswitch conference, echo test etc.) I tried using fast start on and off , h245 tunneling on and off, h245 in SETUP on and off. In my captures I have also noticed a strange behavior : FS sends to my client 2 "alerting" packets for no apparent reason. Could this be a cause of the problem? Had anyone any success with mod_opal lately? If yes, could you please reply quoting your config options (both on FS and on your client)? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs(a)kinetix.gr -------------------------------------------

2 1

how many telephones can I drive from a linksys SPA-2100 FXS?
by freeswitch＠davidnicol.otherinbox.com 12 Jan '09

12 Jan '09

2 1

Re: [Freeswitch-users] how many telephones can I drive from a linksys SPA-2100 FXS?
by freeswitch＠davidnicol.otherinbox.com 12 Jan '09

12 Jan '09

1 0

Skypiax, Skype compatible endpoint
by Giovanni Maruzzelli 12 Jan '09

12 Jan '09

Ciao FreeSWITCHers, mod_skypiax is now usable, for Skype calls and finding bugs :-). Its wiki page (http://wiki.freeswitch.org/wiki/Skypiax, at the moment it's just the README files concatenated) begins like that: WHAT IS SKYPIAX This software (Skypiax) uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype. Skypiax is an endpoint (channel driver) that use the Skype client as an interface to the Skype network, and allows incoming and outgoing Skype calls from/to FreeSWITCH (that can be bridged, originated, answered, etc. as in all other endpoints, eg sofia/SIP). Think at Skypiax as similar to OpenZAP for analog lines: for each channel you need an interface (a Skype client). So, for eg, for two concurrent calls, you will need two channels, two Skype clients running on server. If your server's Skype client(s) has got the Skype credits, Skypiax works for SkypeOut calls too. You can use it from the dialplan, eg with the provided modified "default.xml" dialplan, you can call "sip:skype/remote_skypename__OR__skypeout_phonenumber" for calling via the Skype network from a SIP softphone to remote_skypename or to a phone number via SkypeOut, or you can call the "2908" extension from any phone to be bridged to the Skype Test Call). With the provided skypiax.conf.xml all incoming Skype calls will be routed to the "5000" extension, the IVR in default FreeSWITCH installation. On Linux the Skype client uses a lot of CPU. To lower its CPU consumption, you can use the Xvfb "fake" X server and (more important) the snd-dummy ALSA "fake" sound driver. Scripts are provided for this. But for a low number of channels it would works with regular X servers and ALSA drivers. On a Linux machine with 3GB ram and a quad core intel6600, we got no problem with 20 concurrent calls, and plenty of room for adding more Skypiax channels (100? not tested). On Windows, no need to do anything special, the Skype client is lighter on CPU. Skypiax is now pre-beta, but usable for testing and finding bugs :-). You can download Skypiax source code with subversion with the command: svn co http://svn.freeswitch.org/svn/freeswitch/branches/gmaruzz/src/mod/endpoints… mod_skypiax then, follow the README file in the mod_skypiax directory. More info on skypiax: http://wiki.freeswitch.org/wiki/Skypiax http://www.celliax.org Sincerely, Giovanni Maruzzelli ========================================= Company : Celliax Website: www.celliax.org Address : via Pierlombardo 9, 20135 Milano Country/Territory : Italy Business Email: gmaruzz at celliax dot org Cell : 39-347-2665618 Fax : 39-02-87390039

4 3

Openzap doesn't work
by curly2009＠gmx.de 12 Jan '09

12 Jan '09

Hello, I'm a newbie in FS. I installed and configure Openzap and Zaptel. I have a PRI Digium Wildcard TE110P. When I call a number then I get an Error! 2009-01-12 10:20:01 [CRIT] zap_io.c:804 zap_channel_open_any() All circuits are busy. 2009-01-12 10:20:01 [ERR] mod_openzap.c:910 channel_outgoing_channel() No channels available 2009-01-12 10:20:01 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [openzap] cause: [DESTINATION_OUT_OF_ORDER] What's wrong? Do you need any configurations? Thanks! Best Regards, Curly -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a

4 5

loopback+intercept+bind_meta_app
by Tamas Cseke 12 Jan '09

12 Jan '09

Hello, could you please tell me if I can use bind_meta_app if 1, originate a call with loopback channel 2, push into fifo 3, originate another call to handle the previous call 4 and joining them with intercept. I have problems with it I quess the issue is that this isn't a simple bridge and there is not really a and b legs. Might not. Is it possible to use dtmf bindings with intercept? If it should work, then maybe it is an issue with loopback? Thanks, Tamas

1 0

XML lib curl - what is the best practice for directory binding?
by can_man＠gmx.de 12 Jan '09

12 Jan '09

Hello, I have been looking into the xml curl directory binding and I would like to update the wiki with the accepted best practice. I have listed the HTTP POST request I am getting and how I respond. If there is a better way please let me know and I will update the wiki accordingly. Btw, what I have done works - so no bug hunting this time ;-) I will make a pylons webserver available in the next few days, starting with dialplan and directory support. Thank you, Phil At boot: HTTP POST request 1 [('hostname', u'voip'), ('section', u'directory'), ('tag_name', u''), ('key_name', u''), ('key_value', u'')] my response: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="arbitrary stuff here"> </section> </document> I have left the response empty as I want to provide the users at runtime. ----------------------------------------------------------------------- At boot: HTTP POST request 2 [('hostname', u'voip'), ('section', u'directory'), ('tag_name', u''), ('key_name', u''), ('key_value', u'')] my response: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="arbitrary stuff here"> </section> </document> ----------------------------------------------------------------------- At boot: HTTP POST request 3 [('hostname', u'voip'), ('section', u'directory'), ('tag_name', u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'), ('domain', u'192.168.178.22'), ('purpose', u'network-list')] my response: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="arbitrary stuff here"> </section> </document> What is meant by network list here? If all the users should be loaded at boot time, is this the request which should get a response with the complete list? ---------------------------------------------------------------------- During runtime following this action: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="RE Dial Plan For FreeSwitch"> <context name="public"> <extension name="test10000"> <condition field="destination_number" expression="^(10000)$"> <action application="voicemail" data="default $${domain} 315"/> </condition> </extension> </context> </section> </document> HTTP POST request: ('hostname', u'voip'), ('section', u'directory'), ('tag_name', u'domain'), ('key_name', u'name'), ('key_value', u'192.168.178.22'), ('mailbox', u'315'), ('key', u'id'), ('user', u'315'), ('domain', u'192.168.178.22'), ('ip', u'217.10.79.9') my response: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="arbitrary stuff here"> <domain name="192.168.178.22"> //change to your domain <groups> <group name="default"> <users> <user id="315" mailbox="315"> <params> <param name="password" value="1234"/> <param name="vm-password" value="0000"/> </params> <variables> <variable name="accountcode" value="315"/> <variable name="user_context" value="default"/> <variable name="vm_extension" value="315"/> <variable name="max_calls" value="1"/> <variable name="fail_over" value="415"/> <variable name="cringback" value="us-ring"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

2 3

Freeswitch not passing more than two calls
by ahgindia 12 Jan '09

12 Jan '09

Hi all, I have a freeswitch setup for bridging calls between two gateways. i.e. Originator -> Gateway A -> freeswitch -> Gateeway B -> terminator user I have it running very well for last 43 days. But from last one week, I noticed that freeswitch was not able to pass more than two calls at once. All the other calls were failing with hang up cause 41 (NORMAL_TEMPORARY_FAILURE). The freeswitch was back to normal work only I restarted freeswitch. Can anybody suggest any issue or suggestion for this problem. -- View this message in context: http://www.nabble.com/Freeswitch-not-passing-more-than-two-calls-tp21355617… Sent from the Freeswitch-users mailing list archive at Nabble.com.

5 7