- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to retain the B-number in an Call forward
by Durk de Beer 20 Oct '09

20 Oct '09

Hello, I us a call forward on freeswitch to forward calls to my mobile phone. If now Freeswitch forwards a call, the number information presented to my mobile phone is that of the original caller, because that is what there is appearing in its display. What I want to see is that Freeswitch is calling me so I would know it is a business call and not an private call. Now I have no means to distinguishes between them. Kind regards, Durk de Beer

1 0

conference call
by Nikita Belov 20 Oct '09

20 Oct '09

HI all, I want to configure FS to make special conference call between three users (A, B, C). In this conference C will hear A and B, but A will hear only B. Can I make it using FS API commands? Does anybody know which approach is better to use? ___ Thanks, Nikita

6 13

CPU Usage Question
by DJB 20 Oct '09

20 Oct '09

We have about 1500 concurrent calls on the server. And, the CPU usage is shown below: 02:06:46 AM CPU %user %nice %sys %iowait %irq %soft %steal %idle intr/s 02:06:46 AM all 0.06 0.00 0.03 0.44 0.00 0.00 0.00 99.47 1028.25 02:06:46 AM 0 0.08 0.03 0.03 0.01 0.00 0.01 0.00 99.84 1007.07 02:06:46 AM 1 0.06 0.00 0.02 0.01 0.00 0.00 0.00 99.91 0.00 02:06:46 AM 2 0.07 0.00 0.08 3.49 0.02 0.00 0.00 96.34 21.18 02:06:46 AM 3 0.05 0.00 0.01 0.01 0.00 0.00 0.00 99.94 0.00 02:06:46 AM 4 0.04 0.00 0.02 0.00 0.00 0.00 0.00 99.93 0.00 02:06:46 AM 5 0.05 0.00 0.02 0.00 0.00 0.00 0.00 99.92 0.00 02:06:46 AM 6 0.05 0.00 0.02 0.02 0.00 0.00 0.00 99.91 0.00 02:06:46 AM 7 0.05 0.00 0.02 0.01 0.00 0.00 0.00 99.93 0.00 1) I am wondering whether it is normal that every time when i checked, only CPU 2 always have the highest %iowait (in this case 3.49%). 2) Shoudn't it spread out and balance them to utilize all CPUs evenly? Thank you.

1 0

Excellent FreeSWITCH Article For You To Read
by Michael Collins 20 Oct '09

20 Oct '09

Hello all, Just a heads up that there is a great article out there for you to read. Please visit the main freeswitch.org page here: http://www.freeswitch.org/node/211 Click the link and digg the article after you read it. Please spread the word! The twittersphere is starting to warm up already. Let's keep the momentum going! Thanks, Michael

2 1

validating dtmf digits received
by Mark Campbell-Smith 19 Oct '09

19 Oct '09

Hi! I simply want to validate the dtmf digits I read from a user. From the wiki, it appears I need to use inline="true" when setting the variable so it can be used directly within the same extension. What have I done wrong below? I have tried many different alternatives, but the second condition field, which is meant to match the dtmf digits received (in this case 0000) is never matched, and the anti-action is called instead. : <some code here> <action application="read" data="1 10 ivr/ivr-please_enter_pin_followed_by_pound.wav res 10000 9"/> <action application="phrase" data="spell,${res}"/> <action inline="true" application="set" data="code=${digits_dialed}"/>  </condition> <condition field="digits_dialed" expression="^0000$">   <some code here> <anti-action application="hangup"/> Thanks!

2 3

Re: [Freeswitch-users] Adding Leading Digits To CALLING Number of Outgoing Call
by Jerry Richards 19 Oct '09

19 Oct '09

Okay, I figured it out. I added the following line to the default.xml file, just prior to the "bridge" action: <action application="set" data="effective_caller_id_number=425349${caller_id_number}"/> Now, 425349 is prepended to the outgoing call's caller ID. Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Monday, October 19, 2009 9:14 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: Adding Leading Digits To CALLING Number of Outgoing Call How do I use the dial plan to add leading digits to an outgoing call through a gateway? My internal phone number is 5380, but when FS sends the call to the gateway I want the CALLING party to be 4253495380. Best Regards, Jerry

2 1

check to see if freeswitch is alive
by Christian Löschenkohl 19 Oct '09

19 Oct '09

hello we have the problem here that our freeswitch server "freezes" from time to time (no sip traffic is possible any more). has somebody here expirience or any idea how to monitor freeswitch to answer the questions - is the server alive - does he process sip messages (invites, registrations) and so on the monitoring should work via cron local and remote (with esl for example) a simple "fs_cli -x 'version'" wouldn't work - it's too simple, am i right? any ideas would be helpful br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

6 7

Adding Leading Digits To CALLING Number of Outgoing Call
by Jerry Richards 19 Oct '09

19 Oct '09

How do I use the dial plan to add leading digits to an outgoing call through a gateway? My internal phone number is 5380, but when FS sends the call to the gateway I want the CALLING party to be 4253495380. Best Regards, Jerry

1 0

Re: [Freeswitch-users] Freeswitch 1.0.4 problems with music on hold
by Michael Jerris 19 Oct '09

19 Oct '09

Try out trunk and see if this issue is resolved please. mike On Oct 19, 2009, at 3:11 AM, Durk de Beer wrote: > Hello, > This is something I came across on Freeswitch 1.0.4 > First let me explain what I'm trying to do. > I want Free-Switch to behave as a proxy so in the settings section > of Sofia.conf.xml I use > <param name="media-option" value="bypass-media-after-att-xfer"/> > and > <param name="inbound-bypass-media" value="true"/> > As fare as I am able to follow the RTP stream it is passing by > Freeswitch making it a proxy. > So far so good. Now I want to serv music on hold if a call is put on > hold. > The only way I am able to find for Freeswitch to serv music on hold > (MOH) is for it to be in media. > Ok we've got a command for this situation namely > <param name="media-option" value="resume-media-on-hold"/> > Ok if I put the last parram command into the Sofia.conf.xml > Freeswitch is starting to behave funny. > Let me try to explain this by an example. Let suppose Alice is > calling Bob without the last command, the “resume media on hold”. > If Bob puts Alice on hold al is silent on Bob's and Alice's phone. > If Bob takes Alice of hold there's two way audio. > Ok no I put the command in sofia.onf.xml, the “resume media on hold”. > Now Alice is calling Bob again and Bob is putting her on hold again. > You are expecting MOH on this moment but it isn't, all there is are > the sounds of silence. > Ok mistake made checking local streams, and they are there?. Yes it > is. > Ok other mistake, didn't put the “hold-music”command in > sofia.conf.xml. Noop it's in there so there must be MOH. Codecs then?. > Noop there in vars.xml. Ok for me it is time to throw in the towel, > I'll find it out later. > Ok Bob is now taking Alice of hold and surprise surprise there is > only one way audio, between Alice and Bob. > I didn't have the time to look with Wireshark where al the steams go > so sorry for that. > Did anyone came across the same problem with Freeswitch 1.0.4. I've > a 1.0.2 running and that is doing everything a ok.

1 0

How to enforce freeswitch replying to the source port instead of to the one specified in v/m header parametes (Symmetric NAT)
by Tzury Bar Yochay 19 Oct '09

19 Oct '09

Hi, I am struggling with a cellular operator which removes the rport from the SIP messages sent by the client. During the troubleshooting process I have been playing with the NDLB parameters I found mentioned at the fs-wiki <param name="NDLB-force-rport" value="true"/> <param name="NDLB-received-in-nat-reg-contact" value="true"/> as well as with sip-force-contact at the extension xml config file. <variables> <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> </variables> I Want to make freeswitch replying to the source port instead of the port supplied by the client, that is to port 12543 instead of to the 7608 as you can see below some pcap dumps of the case. I am not sure if that would solve thep roblem and am open to other suggestions as well. Thanks in advance, Tzury pcap dumps below: ### REGISTER ### User Datagram Protocol, Src Port: 12543 (12543), Dst Port: sip (5060) REGISTER sip:example.net SIP/2.0 v: SIP/2.0/UDP 212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167 Max-Forwards: 70 f: <sip:1013@example.net>;tag=9194adecbf8e8179530eb589e83631db t: <sip:1013@example.net> i: ebf2e3c254517938f90f33d8fd89326d CSeq: 17355 REGISTER m: <sip:1013@212.154.128.222:7608> Expires: 300 l: 0 ### 401 UNAUTHORIZED ### SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167 From: <sip:1013@example.net>;tag=9194adecbf8e8179530eb589e83631db To: <sip:1013@example.net>;tag=166HraN0B149g Call-ID: ebf2e3c254517938f90f33d8fd89326d CSeq: 17355 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="example.net", nonce="f2f1370c-bc7c-11de-b000-fbe81e221ab4", algorithm=MD5, qop="auth" Content-Length: 0

2 1