February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Any Known Dual-Registration Issue?
by Jerry Richards 11 Feb '10

11 Feb '10

I've noticed that sometimes my phones end up with two registrations with two Call-IDs at Freeswitch. Is there any known bug that would cause this? I've seen it on different phone models, so I'm thinking there is some timing issue with Freeswitch. Best Regards, Jerry

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Pause during dialing ?
by Eder Souza 10 Feb '10

10 Feb '10

Hi list!! How i can make calls with Pause in string outgoing ?? here examples in asterisk exten => _1201.,1,Dial(SIP/eder@eder,60,TtrD(ww891w${EXTEN:4})) OR exten => _8X.,1,Dial(Zap/g1/ww0w${EXTEN:1}) In Asterisk exist the flag "w", but im FS how make this ?? thx Eng Eder de Souza

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ACL question and js error
by Matthew Law 10 Feb '10

10 Feb '10

After some more experiments I have a working replacement for the asterisk box we were using before, which is great. I had problems getting incoming calls to work. Changing the entry in acl.conf.xml from: <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/> </list> to: <list name="domains" default="allow"> <node type="allow" domain="$${domain}"/> </list> and reloading xml works but this gets reverted every time FS starts up. I've scanned the wiki docs and can't see anything pertaining to that. Why/where is this happening and how do I make it the default? Actually, the question should probably be is it sensible to do that? - the box is out on the internet and I really only want to take incoming calls from voiptalk.org, but I can't find a list of IPs on their site which I could create an acl from... Second question: I have tried this example for an answer machine (mainly because it looked the shortest and simplest of the examples listed): http://wiki.freeswitch.org/wiki/Examples_answermachine I get this error: 2010-02-06 19:01:26.799118 [ERR] answermachine.js:135 ReferenceError: email is not defined Which is relating to this line: email(eMailFrom, eMailTo, "Subject: " + tmp_eMailSubject, eMailBody, tmp_Filename); The script says it requires mod_spidermonkey_etpan, but I can't find any reference to that anywhere (I'm using svn trunk). Where is it? Thanks, Matt - roughly 1% up the FreeSWITCH learning curve and climbing... PS: a nice FreeSWITCH 'cookbook' which starts out at the most simple example and goes on to add more juicy, *working* features would get my money any day. Hint, hint! :-)

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Question regarding testing IVRs
by Pablo Hernan Saro 10 Feb '10

10 Feb '10

Hi there, I was wondering if anyone knows about an automated tool for testing IVR systems... Or should I use SIPP for this purpose? I will really appreciate your inputs. Regards Pablo

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Re: [Freeswitch-users] UPnP Timeout
by Kim Culhan 10 Feb '10

10 Feb '10

On Mon, February 8, 2010 11:21 pm, Troy Anderson wrote: > I have been using pfSense (1.2.3) and FS. FS nicely uses UPnP to poke > holes in the firewall, but it seems that the holes close after a while. I > cannot find any documentation in FS nor in pfSense as to what the timeout > is. Is there a setting in FS to do some kind of keep-alive thing with > UPnP to keep, e.g. 5060, open? Or is it already doing that and pfSense is > the issue? FS has provisions for keep-alive, see the bottom of the page for ping time value: http://wiki.freeswitch.org/wiki/SIP_Provider_Examples To watch the pf firewall hole timing you can install pftop from FreeBSD ports/sysutils which displays the filter states 'and more'. -kim

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No reINVITE when bridging two sessions from JavaScript with bypass_media_after_bridge=true
by Тимур Валишев 10 Feb '10

10 Feb '10

Dear colleagues, The task is to start two sessions from JS script and then bridge them in no-media mode. Unfirtunately, FreeSwitch does not reINVITE the peers after bridging. Here is my script: SCRIPT #1 - 'callback-session.js' <<<<<<<<<<<<<<<<<<<<<<<<<< session = new Session("{bypass_media_after_bridge=true,ignore_early_media=true,hangup_after_bridge=true}user/1001"); session2 = new Session("{ignore_early_media=true}user/1001"); bridge(session, session2); >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Then I run this script from the freeswitch console: freeswitch@internal> jsrun callback-session.js And there is no re-invites between peers, peers get connected and the traffic goes through FS BUT!!! Here is another scipt: SCRIPT #2 - 'callback-bridge.js' <<<<<<<<<<<<<<<<<<<<<<<<<< session = new Session("{bypass_media_after_bridge=true,continue_on_fail=true,ignore_early_media=true}user/1001"); session.execute("bridge","{ignore_early_media=false,originate_timeout=90}user/1001"); >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> When I run this script, FreeSwitch successfuly sends reINVITES to both users after bridge, so they exchange media directly, not through FS. In my task, I need to have control when B-leg establishes (to start billing correctly), so I need to get the first scenario working. What I've already tried and did not succeed: 1) set bypass_media=true, on A leg only, on B leg only, on both legs 2) set bypass_media_after_bridge=true, on A leg only, on B leg only, on both legs Is there something I do wrong in the first script? What should I do to make FS reINVITE peers? Many thanks in advance! Best regards, Timur Valishev sip:tim@novion.ru

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Can't access event socket from 0.0.0.0
by Roly Maz 10 Feb '10

10 Feb '10

I am trying to execute commands over telnet from my XP box to my Debian FS box. I have modified the event_socket.conf.xml so that the Listen IP is 0.0.0.0, as per wiki When i try telnet i get a Connect Failed. If i run netstat on the FS box it shows 127.0.0.1 is listening on port 8021? It also shows my FS static IP 10.0.18.244 is listening on 5060, which is correct. I am using the default conf on Freeswitch version 1.0.trunk 16590M Am i missing a setting? Any pointers would be much appreciated Roly

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Way to pass max-members from dialplan into conference?
by Robert Hadley 10 Feb '10

10 Feb '10

I have setting max-members=10 in conference.conf.xml working. However, is there are way to pass in the max-members=10 from the dialplan/default.xml to mod_conference? I tried using action application="set" data="max-members=10" but it didn't work. Also tried action application="export" data="max-members=10" but it didn't work either. >From default.xml: <extension name="nb_conferences"> <condition field="destination_number" expression="^(30\d{2})$"> <action application="answer"/>  <action application="conference" data="$1-${domain_name}@default"/> Thanks, Robert

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Passing channel variables to mod_fifo and xml_cdr
by Adam Ford 10 Feb '10

10 Feb '10

When sending a call through mod_fifo I seem to be losing my custom channel variables that were assigned during prior processing of the call. In my example, I am trying to assign a unique identifier at the time the call enters my FreeSWITCH system in order to more easily tie the xml_cdr logs together. This works great, until a call is processed through mod_fifo, which drops my custom channel variable in the calls that it generates. Is it likely that I have something wrong with my config? Or does mod_fifo not support the passing of custom channel variables? The overall problem I am trying to solve is that mod_fifo generates a separate a-leg for every time it rings an agent. If the agent answers, the a-leg log gets tied to the associated b-leg log with the uuids and I am able to see the entire call in xml_cdr. However, if the agent rejects the call or doesn't answer, the a-leg is abandoned with seemingly no association back to a b-leg log. Anyone have a better suggestion for tying all these xml_cdr logs together for a full picture of the call? -Adam

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T.38 Fax mode?
by Lon Baker 10 Feb '10

10 Feb '10

Hi, Is 1.0.5 going to support T.38 fax mode? It looks like its partially implemented. The comment leads me to think its not /* Here goes the T.38 SpanDSP initializing functions T.38 will require a big effort as it needs a different approach but the pieces are already in place */ Lon

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