August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

T1 RBS Support Revisited
by John Wehle 14 Aug '08

14 Aug '08

Okay, I have a T1 up and running between our System 25 PBX and FreeSWITCH by simply configuring the Sangoma A104 as an analog span as previously mentioned on this list. It appears that this works when using Loopstart / Groundstart / Koolstart RBS protocols. I need to use Winkstart when doing certain things with this PBX in order to avoid spilling digits ... the zaptel.conf indicates this is supported by configuring the device as "e&m" and then implementing Winkstart in the userspace library. Is this currently supported by FreeSWITCH? If so, how is it configured? If not, what needs to happen in order to implement it? I may be up for contributing code if someone points me in the right direction. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Dialplan Bridge
by Robert Clayton 14 Aug '08

14 Aug '08

Is it possible to bridge to other extensions in the dial plan such as below where extension 100 would ring 9000, 9001 and 9002. <?xml version="1.0" encoding="utf-8"?>  <include> <context name="default"> <extension name="Main Line"> <condition field="destination_number" expression="^100$"> <action application="bridge" data="sofia/sip/9000@192.168.1.222,sofia/sip/9001@192.168.1.222,sofia/sip/9002@192.168.1.222"/> </condition> </extension> <extension name="DictationLines"> <condition field="destination_number" expression="^900[0-2]$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> </context> </include>

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./freeswitch -vvvvv Verbose mode
by Sheeju Alex 14 Aug '08

14 Aug '08

Dear All, I am actually asterisk user, now I am wetting my hand in freeswitch and so far it look great for me. Is there any way where I can use freeswitch CLI with verbose mode similar to asterisk where I could see what exactly is happening action by action similar to asterisk. Looking for your reply.. Cheers Alex

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Understanding Freeswitch
by Robert Clayton 13 Aug '08

13 Aug '08

In trying to learn the functionality of FreeSwitch does anyone have any suggestions for a hypothetical: To allow multiple users to call a single number and simultaneously and interactively leave recordings. I assume the line needs to bridge to other extensions which act as an ivr/recording connection. Initially it seems that Lua offers much functionality but would Lua answer the incoming line and branch to recording extensions in separate threads? Or is there other functionality (XML?) that should provide the branching and Lua answer the ivr/recording extensions? Does Lua even offer control over branching? Thanks in advance for the kick start. Bob

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Help with call forwarding
by Diego Viola 13 Aug '08

13 Aug '08

Hi everyone, I'm trying to do call forwarding in FS, this is what I have in my default dialplan: <extension name="10000"> <condition field="destination_number" expression="^10000$"> <action application="set" data="call_timeout=10"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER"/> <action application="bridge" data="sofia/internal/scot.clausing"/> <action application="bridge" data="sofia/gateway/teliax/blah"/> </condition> </extension> Whenever I dial 10000, I get a NO_ROUTE_DESTINATION error message because the UA is not registered. If I add NO_ROUTE_DESTINATION to the continue_on_fail line, it doesn't work too. I need it to pass to the next call even if the UA (scot.clausing) is not registered, or if there is no route to destination. Thanks, Diego

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Re: [Freeswitch-users] GUI
by Traun Leyden 13 Aug '08

13 Aug '08

> > > On Fri, Aug 1, 2008 at 8:02 AM, Lito Manansala > <lmanansala(a)tigercom.com.sg>wrote: > > > www.wikipbx.com > > > > Hi > > its asking Login, how can i download > > ram You can download from wikipbx.org The IVR's (for voicemail) are out of date due to recent mod_python changes, but the rest should work. I recently fixed a bug in the communication with freeswitch (event socket stuff).

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Installation: good, but some issues
by David Baggett 13 Aug '08

13 Aug '08

I've gotten FreeSWITCH running on this config: - Intel Atom D945GCLF in 1U rackmount - RedHat FC8 - Xorcom USB Astribank-8 (FXO) - Xorcom USB Astribank-8 (FXS) - Trunk FreeSWITCH code from a few days ago - zaptel-1.4.9.2.xpp.r5566 - Grandstream BT-100 SIP phones - Grandstream GXV-3000 SIP videophones - Various analog phone extensions First I tried installing on FC9. This didn't work because FC9 couldn't deal with the on-board NIC card on the D945GCLF. FC8 installed fine, though, so I just used that instead. Overall I am very impressed with FreeSWITCH. It is vastly easier to set up than Asterisk, and much cleaner. The XML config files are *so* much better than the ad hoc Asterisk config syntax. I like that I can make it run like a regular RedHat service, though I could not get it to run as user freeswitch -- it seems to want to run as root. I read through a bunch of the code and it is really well done. Thanks for a great open source project! Most things work: I set up the SIP phones and added the analog extensions, and can dial out on any phone over my POTS lines via Astribank FXO. I can receive incoming calls as well. I can videoconference between the videophones by using the code in the sample dialplan (default.xml) for "intercom" mode. Neat! Problems: Weirndess with FXS-originated calls ----------------------------------- If I call from an analog (FXS) extension to a BT-100, it rings normally. But if I then pick up the SIP phone, I can hear the ring on the analog extension get choppy. Soon after, the call is dropped. If I call from an analog (FXS) extension to a GXV-3000, it rings normally, but when I pick up the SIP phone, I get an immediate busy on the analog extension. If I call from an analog (FXS) extension to an X-Lite SoftPhone, it rings normally, but when I pick up the SIP phone, no audio comes through from the analog extension. If I call from either a BT-100, a GXV-3000, or an X-Lite SoftPhone *to* an analog (FXS) extension, it works fine. The dialplan is simple (note that I have 1- and 2-digit extensions):  <extension name="OpenZAP extensions"> <condition field="destination_number" expression="^(9|1[0-6])$"> <action application="set" data="dialed_ext=$1"/> <action application="bridge" data="OpenZAP/${dialed_ext}/1"/> </condition> </extension>  <extension name="local-extension"> <condition field="destination_number" expression="^([3-6]\d)$"> <action application="set" data="dialed_ext=$1"/> <action application="export"><![CDATA[sip_h_Call-Info=<sip:$${domain}>;answer-after=0]]></action> <action application="export" data="sip_invite_params=intercom=true"/> <action application="export" data="sip_auto_answer=true"/> <action application="bridge" data="user/${dialed_ext}@$${domain}"/> </condition> </extension> In general, I found it a bit strange that you can't (it seems) put analog extensions into the directory. Is it true that you have to define analog extensions manually in the dialplan? That's kind of counterintuitive, especially for Asterisk users; the Asterisk Now GUI makes analog and SIP extensions look the same for all practical purposes. Echo ---- I was getting massive distortion on analog connections involving any GXV-3000 before I changed echo_cancel_level in /etc/openzap/zt.conf. The default value was 64. I changed it to 0 and the distortion went away, but of course I get lots of annoying echo now. Other SIP phones don't seem to have this problem. I have read in earlier posts to this list that the GXV-3000 has known problems. But the interesting thing is that it was working fine with Asterisk and the Astribanks (though on a different CPU & M/B). The Astribank seems to have its own EC, but I can't figure out how to turn it on. And I have no idea what to set the value in zt.conf to; 32 seems to work better than either 16 or 64 (!). I tried fxotune but that didn't do anything: it ran for a long time and generated an /etc/fxotone that seemed to have all zero values. Basically, it would be great to have an "echo cancellation HOWTO" for OpenZap users since EC seems to be offered in several different layers. I also notice a bunch of different EC modes in the OpenZap code -- does anyone know how to try different ones out? FAX detection ------------- This just doesn't seem to work. Here's the relevant code from my diaplan. (It looks a little odd because I send all unanswered calls from my POTS lines to the ext. 31 voicemail box.)   <extension name="incoming-astribank"> <condition field="destination_number" expression="^([1-4])$"> <action application="answer" /> <action application="tone_detect" data="fax 1100 r +5000 transfer fax XML default" /> <action application="set" data="dialed_ext=31"/> <action application="export" data="dialed_ext=31"/> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="transfer_ringback=${us-ring}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <action application="set" data="call_timeout=20"/> <action application="sleep" data="1000"/> <action application="bridge" data="user/30@$${domain},user/31@$${domain},user/33@$${domain},user/36@$${domain},user/37@$${domain},OpenZAP/9/1,OpenZAP/10/1,OpenZAP/11/1"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension>  <extension name="fax"> <condition field="destination_number" expression="^fax$"> <action application="set" data="transfer_ringback=${us-ring}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="OpenZAP/12/1"/> </condition> </extension> Thanks for any help with these issues. Dave

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setting contact header
by Craig Guy 13 Aug '08

13 Aug '08

Hi, First, have been lurking for about a week and I feel Freeswitch is a great project, coming from a non-programmer background and heavy familiarity with Asterisk and Callweaver it's a bit of a learning curve to come to grips with the XML but I'm slowly getting there J I'm currently doing some interop testing with Freeswitch and the provider has come back to me stating that they require the Contact header to exhibit: a) e.164 format b) The From, RPID and Contact headers to match Try as I might I am seemingly unable to set the user portion of the Contact header for the outbound leg to the termination provider. The contact is always mod_sofia@<ip:port>. I have found the sip-force-contact setting however it seems to have no effect on the contact header (and I might be using it incorrectly in any case). The ideal behaviour for me is for the contact to take the form of callerid@<ip:port> For example if my sip profile is bound to 192.168.0.1:5060 and my handset callerid is +13035555555 I would like to set the contact to sip:+13035555555@192.168.0.1:5060 Is there an ability within Freeswitch to set the Contact header based on callerid? Craig

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Very high cpu utilization under load test of FS
by Erol Akarsu 13 Aug '08

13 Aug '08

I am trying a load on FS. I am making originate call between two FS boxes. Originator endpoint is playing one minute sound file with raw format, the other endpoint is sleeping during this time. I am using Xeon quad X5355, 2.66 Ghz box as originator, the other remote endpoint is hp laptop. The system can handle 500 simultaneous originate calls, successfully, without any NO_ANSWER event. But when I start more than 600 originate calls, the Xeon box is having high cpu utilization (almost 100%) and system almost freezes and getting a lot of no-answer calls. I thought this is due to RTP streams and I made Xeon to sleep instead of playing sound file. But I got same behavior, that means it did not help. I am thinking either there is some thread code block that eats CPU or something else. Could you please guide me so that I can achieve 1000 simultaneous originate calls? I appreciate your help, Best Regards Erol Akarsu

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Re: [Freeswitch-users] SIP/2.0 407 Proxy Authentication Required
by Shehzad Pankhawala 13 Aug '08

13 Aug '08

Hi all, I have solved the above problem of routing call from SIP client (Yate) to Freeswitch: All the configuration was same as my previous request, but instead of creating a local user and gateway, i only created one gateway in directory/default folder with all the settings like (used <include> <user id="4444"> <gateways> <gateway name="yate"> <param name="username" value="4444"/> <param name="realm" value="192.168.1.23"/> <param name="password" value="''your-password''"/> <param name="proxy" value="192.168.1.23"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </gateways> <params> <param name="password" value="1234"/> </params> </user> </include> Enjoy; Shehzad.

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