Subscribe to TXLAB feed TXLAB
Just another hacking blog
Updated: 1 hour 47 min ago

3G connectivity for PC Engines APU (MC8775)

Sat, 06/21/2014 - 02:21

PC Engines’ APU board has its mPCIe slot 2 wired to the SIM card socket, which allows using any standard mPCIe 3G modem. Most of modern modems are quite expensive, but there are plenty of Sierra Wireless MC8775 cards at aliexpress.com for around $20 apiece. This is a decent hardware, manufactured around 2007-2011. It doesn’t deliver the highest UMTS speeds possible, but still can be used in situations where speed is unimportant.

The cards that I bought came with firmware version 1_1_8_15, dated 2007/07/17. I didn’t test it fully, but there are some failure reports in the internet.

The firmware upgrade requires an adapter with a SIM card slot. I got mine from this eBay seller.

This page describes the firmware upgrade process. The links to istudioz.net are still valid, but you need to remove # (%23) from the URLs. The 3G watcher for the AirCard 875 is unavailable at its original place, but easy to find with Google. I got mine at this site. The upgrade requires a 32bit Windows machine, and takes about 20 minutes. I upgraded the firmware successfully with my old Vista laptop.

Also I bought the 3G antenna and the pigtail cable at aliexpress.

After inserting the 3G modem into mPCIe slot 2 and booting Debian Wheezy, the device was immediately visible as three serial USB interfaces (/dev/ttyUSB0  /dev/ttyUSB1  /dev/ttyUSB2). ttyUSB0 is used for data, and ttyUSB2 can be used for controlling the device with AT commands. The command “AT^CARDMODE” will tell if the SIM card is inserted, and “AT!GSTATUS?” displays the network status information. “AT+GMR” displays the current firmware version. Ctrl-a Ctrl-x sequence will finish the picocom session.

apt-get install -y wvdial picocom picocom -b 115200 /dev/ttyUSB2 AT^CARDMODE AT!GSTATUS? AT+GMR Ctrl-a Ctrl-x

The following /etc/wvdial.conf works with Sunrise.ch 3G network:

[Dialer Defaults] Modem = /dev/ttyUSB0 Baud = 460800 Init1 = ATZ Init2 = ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 Phone = *99# Username = '' Password = '' Ask Password = 0 Stupid Mode = 1 Compuserve = 0 Idle Seconds = 0 ISDN = 0 Auto DNS = 1

Execute “wvdial” comand from the command line, and it should immediately connect to the internet. The rest is easy: you can place wvdial into a startup script and execute it automatically at boot time.

Filed under: Networking Tagged: 3G, GSM, linux, networking, pcengines, UMTS

Simple performance test for FreeSWITCH conferencing

Thu, 05/08/2014 - 02:39

This is a simple test that gives you an estimation of audio conferencing scalability of FreeSWITCH on your hardware.

  1. You need one or two FreeSWITCH servers, and one of them should answer to sip:moh@IPADDR:5080. The fastest way is to install this FreeSWITCH configuration: https://github.com/xlab1/voip_qos_probe
  2. Edit vars.xml and remove G722 codec (or leave or replace it, if you want to test transcoding performance at the same time).
  3. Start FreeSWITCH: service freeswitch start
  4. Create conference participants by calling the MOH extension on the remote or the same server. This command will add a few dozens of participants in one go: timeout 2 sh -c "while true; do fs_cli -x 'conference xx dial sofia/internal/moh@IPADDR:5080'; done"
  5. check the number of participants: fs_cli -x 'show channels'
  6. run “top” or “mpstat -P ALL 1″ to see the CPU load, and add more batches of participants.

This test differs from real world because in a real conference, one speaks and others are listening. Here everyone speaks at the same time. FreeSWITCH evaluates the energy level to find the active speaker before replicating their voice, so I guess the real conference would take less CPU power (need to look into the source code).

Some test results: PC Engines APU platform with 50 conference participants had the CPU usage about 60%. A single core VPS at digitalocean.com was busy at around 50% during a test with 200 participants.

UPD1: (thanks bob bowles) Call out to yourself and monitor the sound quality with your own ear:

fs_cli -x 'conference human dial sofia/external/user@sip.domain.com'
Filed under: Networking Tagged: freeswitch, voip

End-to-end VoIP quality testing probes

Sun, 04/27/2014 - 02:50

This is a result of a project where we needed to measure voice QoS parameters (jitter and packet loss) in the customer network. I’ve set up small probe computers (old 10″ Intel Atom netbooks like Acer Aspire One) with FreeSWITCH and a few scripts for test automation. Each test consists of a 30-second call (producing approximately 1500 RTP packets in each direction), and tshark is measuring the received jitter and loss on each side.

Test details and the installation procedure are outlined on Github:



Filed under: Networking Tagged: freeswitch, network monitoring, sip, voip, xlab1

FreeSWITCH performance test on PC Engines APU

Sat, 04/19/2014 - 02:08

This test is analogous to the one I described for Intel Atom CPU.This time it’s the new APU board from PC Engines, the maker of famous ALIX and WRAP boards. APU is a fanless appliance board, with a dual-core 1GHz AMD G series CPU. The overall performance is comparable to that of Intel Atom.

In these tests, FreeSWITCH was forwarding the call to itself on request by pressing *1. Each such forwarding resulted in creating four new channels in G722 and G711, thus resulting in transcoding to G711 and back. For example, if “show channels” shows 5 channels, it’s equivalent to 2 simultaneous calls with transcoding.

Test result: 57 channels were running completely fine, 65 channels had slight distortions, and with 85 channels the speech was still recognizable, but with significant distortions. With Speex instead of G722, distortions were quite annoying at 25 channels. Thus, the APU platform can easily be used as a small-to-medium business PBX for  20-30 simultaneous calls if there’s not too much transcoding.

Test details follow.

Debian Wheezy was installed as described in my previous post. Then, GFreeSWITCH version 1.2.23 was installed from packages, as follows:

apt-get install -y curl git sysstat cat >/etc/apt/sources.list.d/freeswitch.list <<EOT deb http://files.freeswitch.org/repo/deb/debian/ wheezy main EOT curl http://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add - apt-get update apt-get install -y freeswitch-meta-all cd /etc git clone https://github.com/xlab1/freeswitch_conf_minimal.git freeswitch

Then, /etc/freeswitch/dialplan/public/05_test.xml was added as follows:

<include> <!-- Extension 100 accepts the initial call, plays echo, and on pressing *1 it transfers to 101 -->     <extension name="100">       <condition field="destination_number" expression="^100$">         <action application="answer"/>         <action application="bind_meta_app" data="1 a si transfer::101 XML ${context}"/>         <action application="delay_echo" data="1000"/>       </condition>     </extension>     <!-- Extension 101 plays a beep, then makes an outgoing SIP call from our internal profile to our own external profile and extension 200 -->     <extension name="101">       <condition field="destination_number" expression="^101$">         <action application="playback" data="tone_stream://%(100,100,1400,2060,2450,2600)"/>         <action application="unbind_meta_app" data=""/>         <action application="bridge"                 data="{absolute_codec_string=PCMA}sofia/internal/200@${sip_local_network_addr}:5080"/>       </condition>     </extension> <!-- Extension 200 returns the call to 100 as a new outgoing SIP call from our internal profile to our own external profile -->     <extension name="200">       <condition field="destination_number" expression="^200$">         <action application="answer"/>         <action application="bridge"                 data="{max_forwards=65}{absolute_codec_string=G722}sofia/internal/100@${sip_local_network_addr}:5080"/>       </condition>     </extension>     </include>

After sending the initial call from a SIP phone to extension 100 at our APU’s IP address and port 5080, after pressing *1 we get 2 new channels with transcoding. Below are results of “mpstat -P ALL 1″ command during the test:

# quite clear sound root@apu:/etc/freeswitch# fs_cli -x 'show channels' | grep total 57 total. 11:35:07 PM  CPU    %usr   %nice    %sys %iowait    %irq   %soft  %steal  %guest   %idle 11:35:08 PM  all   41.71    0.00    8.00    0.00    0.00    0.57    0.00    0.00   49.71 11:35:08 PM    0   43.68    0.00    5.75    0.00    0.00    1.15    0.00    0.00   49.43 11:35:08 PM    1   40.45    0.00   10.11    0.00    0.00    0.00    0.00    0.00   49.44 # slight distortions root@apu:/etc/freeswitch# fs_cli -x 'show channels' | grep total 65 total. 11:36:27 PM  CPU    %usr   %nice    %sys %iowait    %irq   %soft  %steal  %guest   %idle 11:36:28 PM  all   55.98    0.00    8.70    0.00    0.00    0.54    0.00    0.00   34.78 11:36:28 PM    0   55.91    0.00    7.53    0.00    0.00    2.15    0.00    0.00   34.41 11:36:28 PM    1   55.43    0.00    9.78    0.00    0.00    0.00    0.00    0.00   34.78 # significant distortions root@apu:/etc/freeswitch# fs_cli -x 'show channels' | grep total 85 total. 11:37:34 PM  CPU    %usr   %nice    %sys %iowait    %irq   %soft  %steal  %guest   %idle 11:37:35 PM  all   71.13    0.00    9.28    0.00    0.00    2.06    0.00    0.00   17.53 11:37:35 PM    0   71.72    0.00    9.09    0.00    0.00    2.02    0.00    0.00   17.17 11:37:35 PM    1   71.58    0.00    9.47    0.00    0.00    2.11    0.00    0.00   16.84

If G722 is replaced with Speex codec, the CPU load is significantly higher, and already with 25 channels the distortions are quite significant:

# speex 8kHz, distortions root@apu:/etc/freeswitch# fs_cli -x 'show channels' | grep total 25 total. 12:59:46 AM  CPU    %usr   %nice    %sys %iowait    %irq   %soft  %steal  %guest   %idle 12:59:47 AM  all   54.10    0.00    1.64    0.00    0.00    0.00    0.00    0.00   44.26 12:59:47 AM    0   53.85    0.00    2.20    0.00    0.00    0.00    0.00    0.00   43.96 12:59:47 AM    1   54.95    0.00    1.10    0.00    0.00    0.00    0.00    0.00   43.96
Filed under: Networking Tagged: freeswitch, pbx, pcengines, sip, voip

PC Engines APU board: installing Debian on mSATA drive

Sun, 04/06/2014 - 06:58

PC Engines started shipping its new APU board in 2014. It can boot from an SD card (slow on writes), and it can also have an mSATA drive and boot from it (fast read-write, and more write cycles). Voyage Linux is well optimized for SD card.

Here I started my scripts for building a Debian CD and installing it on APU’s mSATA drive: https://github.com/ssinyagin/pcengines-apu-debian-cd


Filed under: Networking Tagged: debian, linux, pcengines

Call forwarding/redirection in FreeSWITCH

Sun, 02/16/2014 - 06:57

Consider you have two different contexts in your dialplan for inbound and outbound calls: the “public” context transfers the calls into “XXX_inbound” (XXX being your organization name), and the user directory has “XXX_outbound” as “user_context” variable.

Having two contexts, you have more flexibility in defining the short dial strings and outbound destinations.

But there’s a little problem: if the SIP client redirects the ringing call, or if the user makes an attended transfer, FreeSWITCH would initiate a new outbound leg in the same context where the call was bridged toward the SIP client.

As a solution, you need to define a new extension in your XXX_inbound context which would match PSTN outbound numbers. The channel will already have all custom variables which were set before bridging toward the SIP client, so you can set an additional condition criteria to make sure that this is the redirected call. This example would be placed at the bottom of the inbound context, and “directory_ext” is the variable that was earlier in the same context before the call was bridged to the SIP client:

    <extension name="call_forward">       <condition field="destination_number" expression="^\d+$"/>       <condition field="${directory_ext}" expression="^70\d$">         <action application="set" data="hangup_after_bridge=true"/>         <action application="set" data="continue_on_fail=false"/>         <action application="bridge" data="${outgw}/${destination_number}"/>       </condition>             </extension>
Filed under: Networking Tagged: freeswitch, pbx, sip, voip

Reusing HTTP connections in client-server applications

Sat, 11/16/2013 - 02:16

I’m working on a clientserver application which uses HTTP as a transport protocol for API requests, and sometimes there are occasions with a need to execute a few hundreds requests, such as data import or synchronization.

With default Apache HTTP server settings and default LWP::UserAgent options, every new request would result in a new HTTP session, and each time a DNS query is sent out. So, a synchronization process with a thousand object floods the DNS service with the same requests for the HTTP server name. This results in delays, and some public DNS servers apply rate limits which cause DNS lookup failures (had this with a domain hosted at Godaddy name servers).

HTTP 1.1 protocol supports reusing of persistent connections, but it’s not enabled by default in Apache and in the client.

In Apache HTTP server, the following options need to be configured:

  KeepAlive On   MaxKeepAliveRequests 500

In the Perl client program, LWP::UserAgent needs the keep-alive option:

  my $ua = LWP::UserAgent->new(keep_alive => 1);

With these modifications, the DNS queries are only sent on every 500th API request, and the HTTP connection is reused between the requests, which saves CPU time on the server. This speeds up the whole process significantly, and also prevents the DNS failures caused by rate limiting.

Filed under: Networking, Programming Tagged: network api, perl

Minimal FreeSWITCH configuration

Wed, 11/13/2013 - 14:15

It’s always a bit of an effort to remove unneeded features from the default FreeSWITCH configuration. So, I made the minimal configuration which still allows to start the server, but does completely nothing. It’s now much easier to start a new server configuration for any new project.

The configuration is placed at Github. It’s very straightforward to use with FreeSWITCH Debian packages, and can also be used if you compile it from sources:

cd /etc git clone https://github.com/xlab1/freeswitch_conf_minimal.git freeswitch

The configuration contains a number of empty “stub.xml” files in order to make the XML pre-processor happy.

It also makes sense to start using Git for your own FreeSWITCH configurations :)

Filed under: Networking Tagged: freeswitch, pbx, sip, voip, xlab1

FreeSWITCH performance on Intel Atom CPU

Sun, 10/13/2013 - 00:22

There are multiple low-power, fanless  appliances on the market, and most of them are powered by Intel Atom processors. I needed an estimation how well an Atom would perform for a FreeSWITCH PBX application.

In this test, I use two Acer Aspire One notebooks with different processors:

  • atom01: Atom N2600 (2 cores, 4 virtual CPUs, 512KB cache and 600MHz per virtual CPU, 12768.02 BogoMIPS)
  • atom02: Atom N570 (2 cores, 4 virtual CPUs, 512KB cache and 1000MHz per virtual CPU, 13302.08 BogoMIPS)

Both notebooks are running 32-bit Debian 7 Wheezy (Kernel version 3.2.0-4-686-pae), and FreeSWITCH version 1.2.13 from pre-built Debian packages.

Test results summary

All calls in this test used transcoding between G.711alaw and G.722. The bottleneck in performance was always at the N2600 (atom01), because of slower CPU. In general, N570 can handle approximately 30% higher load than N2600.

With 10 concurrent calls (21 channels on atom01 and 20 channels on atom02), there is no voice distortion and new call processing does not disturb the ongoing calls. Each virtual CPU is busy at 20-25%

With 20 concurrent calls (41 channels on atom01 and 40 cannels on atom02), there is some minor voice distortion, especially during incoming calls, but quality s still acceptable.

With 27 concurrent calls (55 and 54 channels), voice distortions were too high and not acceptable. Every virtual CPU on atom01 was busy at around 50%, which means full load for the whole CPU.

With 20 concurrent calls without transcoding (PCMA only in all call legs), each CPU core on atom01 was utilized at around 9-10%. So, theoretically the platform can handle up to 40-50 simultaneous calls in non-transcoding mode.

Only the voice quality was tested. CPS was not tested, and it depends heavily on the complexity of the dialplan. But the overall response of the system was quite acceptable.

Testing details

I took my minimal FreeSWITCH configuration for the tests, and extended it as follows:


sip_profiles/external/itsp.xml registers at my vPBX, so that I could initiate the calls. Incoming calls are sent to extension 500.

sip_profiles/external/atom02.xml defines the gateway that points to atom02:

<include>   <gateway name="atom02">     <param name="register" value="false"/>     <param name="proxy" value=""/>     <param name="ping" value="27"/>   </gateway> </include>

dialplan/public/15_bulkmatch.xml consists of 100 identical conditions, like shown below. It does not do anything useful, and it’s only used to make FreeSWITCH  busy processing the dialplan:

<include>   <extension name="bmatch" continue="true">     <condition field="destination_number" expression="^(\d+)$">       <action application="set" data="xxxxx=$1"/>     </condition>   </extension>   <extension name="bmatch" continue="true">     <condition field="destination_number" expression="^(\d+)$">       <action application="set" data="xxxxx=$1"/>     </condition>   </extension> ......

dialplan/public/20_perftest.xmltakes the call at extension 500 and plays delayed echo. When I press *1, it makes a new call leg to atom02 in G711alaw codec, and atom02 makes a new call leg back to atom01 in G.722:

<include>   <extension name="500">     <condition field="destination_number" expression="^500$">       <action application="answer"/>       <action application="bind_meta_app" data="1 a si transfer::501 XML ${context}"/>       <action application="delay_echo" data="1000"/>     </condition>   </extension>   <extension name="501">     <condition field="destination_number" expression="^501$">       <action application="playback" data="tone_stream://%(100,100,1400,2060,2450,2600)"/>       <action application="unbind_meta_app" data=""/>       <action application="bridge" data="{absolute_codec_string=PCMA}sofia/gateway/atom02/600"/>     </condition>   </extension>  </include> atom02

sip_profiles/external/atom01.xml defines the gateway pointing to atom01:

<include>   <gateway name="atom01">     <param name="register" value="false"/>     <param name="proxy" value=""/>     <param name="ping" value="27"/>   </gateway> </include>

dialplan/public/15_bulkmatch.xml is identical to that on atom01.

dialplan/public/20_perftest.xml answers the call at extension 600 and places a new call to extension 500 at atom01:

<include>     <extension name="600">     <condition field="destination_number" expression="^600$">       <action application="answer"/>       <action application="bridge" data="{max_forwards=65}{absolute_codec_string=G722}sofia/gateway/atom01/500"/>     </condition>   </extension> </include>

When non-transcoding tests are made, the codec string is changed to PCMA.

Filed under: Networking Tagged: freeswitch, linux, pbx, sip, testing, voip, xlab1

How I bought Microsoft Visio Pro for Office 365

Tue, 10/08/2013 - 02:01

I needed to have MS Visio Professional on my new Win8 notebook, but paying $900 for the license was somewhat uncomfortable. Also relatively recently, Microsoft started offering Office 365 subscriptions where the software is offered as a monthly or yearly subscription instead of a one-off purchase.

It appears that MS Visio is also offered as subscription, but for some reason it’s not so easy to find: on the product page, you see only the full license for purchasing. But if you click to “Try or buy”, you have a “Learn more” link under the Visio Pro for Office 365 title.

Then, I could not find anywhere, which Office 365 subscription is needed to add Visio to it. So i thought, maybe I should just buy one, so I ordered the one which seemed the right one for my purposes, Office 365 Small Business Premium.

When I tried to add Visio Pro for Office 365 to my account, I got an error that this product is incompatible with my current subscription, and they tried to make me create a new subscription instead.

So, I opened a support request and they explained me that Visio can only be added to Office 365 Enterprise subscriptions!

I then asked them to cancel my subscription and promised to open an Enterprise one.

But: to say the truth, I actually had a valid Office 2007 license, and I only needed Visio. So, I made a new subscription for Visio only. Also funny, that even after deleting my first subscription, I could not use the same account name, and had to come up with a new name.

Actually that was not the end of the fun: after buying it, it took a while to find the installation link in the Office 365 Admin panel. Also when I clicked and downloaded something, it was not an installer in its traditional way. It was a self-extracting archive with a command line utility in it, and a sample XML file. Luckily this XML file had already an example for Visio, so I only needed to uncomment the relevant parts of it, and then use the command-line tool to download and set up the software. It was not difficult, but kind of surprising :)

So, finally I got Visio 2013 working, and Office 2007 has installed and activated smoothly. But I lost about hour or so because of:

  1. obscure product information on MS website
  2. strange incompatibility in subscription plans
  3. command-line installer with an XML that needed manual editing (!)


Filed under: Weird things Tagged: microsoft, office 365, visio

Out-of-business greeting with FreeSWITCH

Sat, 08/17/2013 - 23:22

After I got my US number at Callcentric, I got several wrong calls in the following days. The calls were quite late at night, and most of them dropped after few seconds, before I could take up the handset. And another ring around 3am was really long and loud, and it dropped anyway before I could come up and pick the call.

First, I needed to record my own greeting. In “default” dialplan, I added a new extension. It takes a recording and plays it back :

  <!-- 7396: Record a greeting -->   <extension name="app_7396">     <condition field="destination_number" expression="^7396$">       <action application="answer"/>       <action application="sleep" data="500"/>       <action application="playback" data="tone_stream://%(100,100,1400,2060,2450,2600)"/>       <action application="set" data="playback_terminators=#"/>       <action application="set" data="record_waste_resources=true"/>       <action application="set" data="recfilename=$${base_dir}/recordings/greeting_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>       <action application="record" data="${recfilename}"/>       <action application="sleep" data="700"/>       <action application="playback" data="tone_stream://%(100,100,1400,2060,2450,2600)"/>       <action application="playback" data="${recfilename}"/>       <action application="hangup"/>     </condition>   </extension>

As I’m using a Gigaset 610IP handset with G722 codec, the produced recording had 16KHz sampling rate, and needed to be resampled, because inbound external calls are G711 only:

sox recordings/greeting_2013-08-17-11-46-35.wav sounds/dvop/8000/ssinyagin_oob.wav rate 8000

Then my extension in “public” dialplan is modified to play the greeting unless the call is between 7am and 11pm. It also plays MOH for 5 seconds before bridging the call, and continues playing MOH while ringing my phone.  This gives the mistaken caller another chance to realize that something is wrong and drop the call.

  <extension name="pub_ssinyagin">     <condition field="destination_number" expression="^ssinyagin$" break="on-false">       <action application="set" data="timezone=Europe/Zurich" inline="true"/>     </condition>     <!-- 7:00 - 23:00 -->     <condition minute-of-day="420-1000" break="on-true">  <action application="answer"/> <action application="set" data="playback_timeout_sec=5"/> <action application="playback" data="$${hold_music}"/> <action application="set" data="ringback=$${hold_music}"/>       <action application="transfer" data="7110 XML default"/>     </condition>     <condition>       <action application="answer"/>       <action application="sleep" data="1000"/>       <action application="playback" data="$${sounds_dir}/dvop/ssinyagin_oob.wav"/>       <action application="hangup"/>     </condition>   </extension>
Filed under: Networking Tagged: freeswitch, pbx, voip

Free US number and Caller ID manipulation

Sat, 08/03/2013 - 17:46

Callcentric offers US numbers in NY area code, with zero recurring costs. This is very convenient if you want your USA customers to connect to your PBX. After ordering a free number, you create a SIP account and route the DID to it. The service allows to register multiple free numbers. Forwarding to SIP URI is not supported.

Upon receiving a call, the caller ID in From field will look like 16313335447, with the country code without any leading symbols. The following piece of FreeSWITCH configuration (in public context) alters the caller ID in order to look like a normal number in a European dial plan:

  <extension name="intl_normalize" continue="true">     <!-- remove Swiss country code -->     <condition field="${caller_id_number}" expression="^41(\d+)" break="on-true">       <action application="set" data="effective_caller_id_number=0$1"/>       <action application="set" data="effective_caller_id_name=0$1"/>     </condition>     <!-- add 00 in front of country code -->     <condition field="${caller_id_number}" expression="^[1-9]" break="on-true">       <action application="set" data="effective_caller_id_number=00${caller_id_number}"/>       <action application="set" data="effective_caller_id_name=00${caller_id_number}"/>     </condition>   </extension>

It is important to set both effective_caller_id_number and effective_caller_id_name variables. If only effective_caller_id_number is set, the effective_caller_id_name still keeps the original caller ID number, and if the call is bridged to a local extension, the SIP phone may want to use it for displaying the caller.



Filed under: Networking Tagged: freeswitch, pbx, sip, voip

voxserv.ch and Twitter Bootstrap site templates

Thu, 07/25/2013 - 23:48

Here’s a new website where I promote the VoIP integration services on Swiss market: http://www.voxserv.ch/

The website is built with the Twitter Bootstrap, and here are the templates for template-toolkit which separate the Bootstrap HTML from text content: https://github.com/ssinyagin/voxserv.ch/tree/master/builder

Filed under: Networking, Programming Tagged: linux, server management

Handling e164 numbers in FreeSWITCH

Sat, 07/06/2013 - 03:11

SIP clients installed on smartphones may pick up the destination number from the phone book, and it’s sometimes in e.164 format (+[countrycode][localdigits]).

The following piece of XML dialplan transforms such numbers into the standard form that is expected by most PSTN VoIP providers. This example assumes that the FreeSWITCH server is located in Switzerland and +41 is the e.164 prefix for in-land calls. It returns the call to the same context, making the switch traverse the whole context dialplan from the beginning. It makes sense to place this extension at the bottom of a context.

    <extension name="e164_pstn">       <condition field="destination_number" expression="^\+41(\d+)" break="on-true">         <action application="transfer" data="0$1 XML ${context}"/>       </condition>       <condition field="destination_number" expression="^\+(\d+)" break="on-true">         <action application="transfer" data="00$1 XML ${context}"/>       </condition>     </extension>
Filed under: Networking Tagged: freeswitch, pbx, voip

FreeSWITCH: Limiting the number of concurrent calls on multiple SIP accounts

Sun, 06/30/2013 - 02:52

The user has several SIP accounts on a vPBX, and he wants that maximum one call is possible at a time.

The limit application in FreeSWITCH allows to control the number of concurrent calls, but one should be careful with when this limit should be applied. The switch decrements the limit counter automatically when a channel is terminated. But if the limit is executed on a-leg, and b-leg is transferred, the limit counter decreases only when the a-leg finishes the call. As a result, the user may receive a call, transfer it to a new destination and hang up, but the new calls are not coming in because the limit counter is reset when the original call ends.

In order to reset the limit counter after the b-leg is transferred, the limit application needs to be executed on b-leg only. This is possible by exporting the execute_on_answer variable with nolocal modifier.

The example also shows how to retrieve user variables from the XML directory in the calls toward the user.

  <context name="moretti">     <extension name="common_variables" continue="true">       <condition>         <action inline="true" application="set" data="availability_username=moretti"/>       </condition>     </extension>     <extension name="pstn_out">             <condition field="destination_number" expression="^[01]" break="on-false">         <!-- For outbound calls, we only set the limit counters,              but do not limit the call -->         <action application="limit" data="hash ${domain_name} ${availability_username} -1"/>         <action application="set" data="hangup_after_bridge=true"/>         <action application="set" data="continue_on_fail=false"/>       </condition>       <condition>         <action application="bridge" data="${outgw}/${destination_number}"/>       </condition>     </extension>     <extension name="inbound_73x">       <condition field="destination_number" expression="^73(d)$" break="on-false">         <!-- retrieve variables from the user entry in the directory -->         <action application="set" data="directory_userid=70$1@${domain_name}"/>         <action application="set" data="call_timeout=${user_data(${directory_userid} var ring_timeout)}"/>       </condition>       <!-- check the limit -->       <condition field="${cond(${limit_usage(hash ${domain} ${availability_username})} > 0 ? true:false)}"                  expression="^true$" break="on-true">         <action application="hangup"/>       </condition>       <!-- group call to the SIP user and a mobile phone -->       <condition>         <action application="export" data="nolocal:execute_on_answer=limit hash ${domain} ${availability_username} -1"/>         <action application="set" data="ignore_early_media=true"/>         <action application="set" data="transfer_ringback=$${hold_music}"/>         <action application="set" data="hangup_after_bridge=true"/>         <action application="set" data="continue_on_fail=false"/>         <action application="bridge" data="user/${directory_userid},[leg_delay_start=10]${outgw}/0123456789"/>       </condition>     </extension>       </context>
Filed under: Networking Tagged: freeswitch, pbx, sip, voip


Using the greatness of Parallax

Phosfluorescently utilize future-proof scenarios whereas timely leadership skills. Seamlessly administrate maintainable quality vectors whereas proactive mindshare.

Dramatically plagiarize visionary internal or "organic" sources via process-centric. Compellingly exploit worldwide communities for high standards in growth strategies.

Get free trial

Wow, this most certainly is a great a theme.

John Smith
Company name

Startup Growth Lite is a free theme, contributed to the Drupal Community by More than Themes.