Calling on GSM/3G Networks

Posted on March 29, 2013 by Gernot

The Asterisk for Raspberry Pi project is continuously improving with new features and enhancements. The latest feature is particularly interesting, it allows direct calling on GSM/3G networks with USB modems from Huawei and the chan_dongle channel driver. A highly affordable GSM VoIP gateway can be built, using the USB modem as trunk in Asterisk.
You can find complete setup instructions in the documentation section.

A number of interesting applications are possible with this: Least cost routing of course (use the GSM network when it’s cheaper). In remote areas with weak cell phone reception, install the RPi in a location where reception is possible and connect the SIP phones in your house with the RPi. High-gain directional antennas can be connected to some of the USB modems and pointed towards the next cell phone tower, in order to receive weak signals that can’t be received by regular phones anymore. Finally the RPi also makes a good extension to an already existing PBX, enhancing it with GSM gateway capability. With a total power consumption of around 5W for both RPi and the modem, your electricity bill is not streched at all, even if you are only occasionally calling over GSM.

P.S.: If you are interested in this application, you may want to read about this RPi GSM gateway implementation.

83 thoughts on “Calling on GSM/3G Networks”

Dillo on November 12, 2013 at 2:05 pm said:

I’m planning to use Raspberry Pi, Asterisk and Huawei 3G USB modem to do this:

1 Receive a call – no need to answer it
2. If the call if from a configured caller ID then use GPIO to trigger a relay switch

How do I go about configuring #2?
- Gernot on November 13, 2013 at 6:11 am said:
  
  Have a look at this forum thread, it should give you some hints.
  - Dillo on December 22, 2013 at 5:50 pm said:
    
    Thanks
Matthew on October 19, 2013 at 10:00 am said:

Have been having some real issues the last few months with the dongle seem to keep disappearing giving an “all circuits busy now” recording.

I have pasted the details from CLI below – if anyone could help that’d be great!

== Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [01473123456@from-internal:1] Macro("SIP/1001-0000000a", "user-callerid,LIMIT,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/1001-0000000a", "AMPUSER=1001") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1001-0000000a", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1001-0000000a", "1?Set(REALCALLERIDNUM=1001)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/1001-0000000a", "AMPUSER=1001") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/1001-0000000a", "AMPUSERCIDNAME=1001") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1001-0000000a", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/1001-0000000a", "AMPUSERCID=1001") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/1001-0000000a", "CALLERID(all)="1001" ") in new stack -- Executing [s@macro-user-callerid:9] GotoIf("SIP/1001-0000000a", "0?limit") in new stack -- Executing [s@macro-user-callerid:10] ExecIf("SIP/1001-0000000a", "1?Set(GROUP(concurrency_limit)=1001)") in new stack -- Executing [s@macro-user-callerid:11] GotoIf("SIP/1001-0000000a", "1?continue") in new stack -- Goto (macro-user-callerid,s,24) -- Executing [s@macro-user-callerid:24] Set("SIP/1001-0000000a", "CALLERID(number)=1001") in new stack -- Executing [s@macro-user-callerid:25] Set("SIP/1001-0000000a", "CALLERID(name)=1001") in new stack -- Executing [s@macro-user-callerid:26] Set("SIP/1001-0000000a", "CHANNEL(language)=en") in new stack -- Executing [01473123456@from-internal:2] Set("SIP/1001-0000000a", "MOHCLASS=default") in new stack -- Executing [01473123456@from-internal:3] ExecIf("SIP/1001-0000000a", "1?Set(TRUNKCIDOVERRIDE=07867952292)") in new stack -- Executing [01473123456@from-internal:4] Set("SIP/1001-0000000a", "_NODEST=") in new stack -- Executing [01473123456@from-internal:5] Gosub("SIP/1001-0000000a", "sub-record-check,s,1(out,01473123456,)") in new stack -- Executing [s@sub-record-check:1] GotoIf("SIP/1001-0000000a", "1?check") in new stack -- Goto (sub-record-check,s,6) -- Executing [s@sub-record-check:6] Set("SIP/1001-0000000a", "__MON_FMT=wav") in new stack -- Executing [s@sub-record-check:7] GotoIf("SIP/1001-0000000a", "1?next") in new stack -- Goto (sub-record-check,s,10) -- Executing [s@sub-record-check:10] ExecIf("SIP/1001-0000000a", "0?Return()") in new stack -- Executing [s@sub-record-check:11] GotoIf("SIP/1001-0000000a", "0?out,1") in new stack -- Executing [s@sub-record-check:12] Set("SIP/1001-0000000a", "__REC_STATUS=INITIALIZED") in new stack -- Executing [s@sub-record-check:13] ExecIf("SIP/1001-0000000a", "0?Set(__REC_POLICY_MODE=)") in new stack -- Executing [s@sub-record-check:14] Set("SIP/1001-0000000a", "NOW=1382176632") in new stack -- Executing [s@sub-record-check:15] Set("SIP/1001-0000000a", "__DAY=19") in new stack -- Executing [s@sub-record-check:16] Set("SIP/1001-0000000a", "__MONTH=10") in new stack -- Executing [s@sub-record-check:17] Set("SIP/1001-0000000a", "__YEAR=2013") in new stack -- Executing [s@sub-record-check:18] Set("SIP/1001-0000000a", "__TIMESTR=20131019-105712") in new stack -- Executing [s@sub-record-check:19] Set("SIP/1001-0000000a", "__FROMEXTEN=1001") in new stack -- Executing [s@sub-record-check:20] Set("SIP/1001-0000000a", "__CALLFILENAME=out-01473123456-1001-20131019-105712-1382176632.10") in new stack -- Executing [s@sub-record-check:21] Goto("SIP/1001-0000000a", "out,1") in new stack -- Goto (sub-record-check,out,1) -- Executing [out@sub-record-check:1] ExecIf("SIP/1001-0000000a", "1?Set(__REC_POLICY_MODE=always)") in new stack -- Executing [out@sub-record-check:2] GosubIf("SIP/1001-0000000a", "1?record,1(exten,01473123456,1001)") in new stack -- Executing [record@sub-record-check:1] Set("SIP/1001-0000000a", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack -- Executing [record@sub-record-check:2] MixMonitor("SIP/1001-0000000a", "2013/10/19/out-01473123456-1001-20131019-105712-1382176632.10.wav,,") in new stack -- Executing [record@sub-record-check:3] Set("SIP/1001-0000000a", "__REC_STATUS=RECORDING") in new stack -- Executing [record@sub-record-check:4] Set("SIP/1001-0000000a", "CDR(recordingfile)=out-01473123456-1001-20131019-105712-1382176632.10.wav") in new stack -- Executing [record@sub-record-check:5] Return("SIP/1001-0000000a", "") in new stack == Begin MixMonitor Recording SIP/1001-0000000a -- Executing [out@sub-record-check:3] Return("SIP/1001-0000000a", "") in new stack -- Executing [01473123456@from-internal:6] Macro("SIP/1001-0000000a", "dialout-trunk,6,01473123456,") in new stack -- Executing [s@macro-dialout-trunk:1] Set("SIP/1001-0000000a", "DIAL_TRUNK=6") in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1001-0000000a", "0?sub-pincheck,s,1()") in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1001-0000000a", "0?disabletrunk,1") in new stack -- Executing [s@macro-dialout-trunk:4] Set("SIP/1001-0000000a", "DIAL_NUMBER=01473123456") in new stack -- Executing [s@macro-dialout-trunk:5] Set("SIP/1001-0000000a", "DIAL_TRUNK_OPTIONS=tm") in new stack -- Executing [s@macro-dialout-trunk:6] Set("SIP/1001-0000000a", "OUTBOUND_GROUP=OUT_6") in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1001-0000000a", "0?nomax") in new stack -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/1001-0000000a", "0?chanfull") in new stack -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1001-0000000a", "0?skipoutcid") in new stack -- Executing [s@macro-dialout-trunk:10] Set("SIP/1001-0000000a", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1001-0000000a", "outbound-callerid,6") in new stack -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1001-0000000a", "0?Set(CALLERPRES()=)") in new stack -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1001-0000000a", "0?Set(REALCALLERIDNUM=1001)") in new stack -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1001-0000000a", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [s@macro-outbound-callerid:6] Set("SIP/1001-0000000a", "USEROUTCID=") in new stack -- Executing [s@macro-outbound-callerid:7] Set("SIP/1001-0000000a", "EMERGENCYCID=") in new stack -- Executing [s@macro-outbound-callerid:8] Set("SIP/1001-0000000a", "TRUNKOUTCID=") in new stack -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1001-0000000a", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1001-0000000a", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/1001-0000000a", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/1001-0000000a", "1?Set(CALLERID(all)=07867952292)") in new stack -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/1001-0000000a", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/1001-0000000a", "0?sub-flp-6,s,1()") in new stack -- Executing [s@macro-dialout-trunk:13] Set("SIP/1001-0000000a", "OUTNUM=01473123456") in new stack -- Executing [s@macro-dialout-trunk:14] Set("SIP/1001-0000000a", "custom=AMP") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1001-0000000a", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/1001-0000000a", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack -- Executing [s@macro-dialout-trunk:17] Macro("SIP/1001-0000000a", "dialout-trunk-predial-hook,") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1001-0000000a", "") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1001-0000000a", "0?bypass,1") in new stack -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/1001-0000000a", "1?Set(CONNECTEDLINE(num,i)=01473123456)") in new stack -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/1001-0000000a", "1?Set(CONNECTEDLINE(name,i)=CID:07867952292)") in new stack -- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/1001-0000000a", "1?customtrunk") in new stack -- Goto (macro-dialout-trunk,s,25) -- Executing [s@macro-dialout-trunk:25] Set("SIP/1001-0000000a", "pre_num=AMP:dongle/dongle0/") in new stack -- Executing [s@macro-dialout-trunk:26] Set("SIP/1001-0000000a", "the_num=OUTNUM") in new stack -- Executing [s@macro-dialout-trunk:27] Set("SIP/1001-0000000a", "post_num=") in new stack -- Executing [s@macro-dialout-trunk:28] GotoIf("SIP/1001-0000000a", "1?outnum:skipoutnum") in new stack -- Goto (macro-dialout-trunk,s,29) -- Executing [s@macro-dialout-trunk:29] Set("SIP/1001-0000000a", "the_num=01473123456") in new stack -- Executing [s@macro-dialout-trunk:30] Dial("SIP/1001-0000000a", "dongle/dongle0/01473123456,300,") in new stack [2013-10-19 10:57:13] WARNING[3567][C-00000021]: channel.c:180 channel_request: [dongle0] Request to call on device which can not make call at this moment [2013-10-19 10:57:13] WARNING[3567][C-00000021]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'dongle' (cause 44 - Requested channel not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dialout-trunk:31] NoOp("SIP/1001-0000000a", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 44") in new stack -- Executing [s@macro-dialout-trunk:32] Goto("SIP/1001-0000000a", "s-CHANUNAVAIL,1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/1001-0000000a", "RC=44") in new stack -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/1001-0000000a", "44,1") in new stack -- Goto (macro-dialout-trunk,44,1) -- Executing [44@macro-dialout-trunk:1] Goto("SIP/1001-0000000a", "continue,1") in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/1001-0000000a", "1?noreport") in new stack -- Goto (macro-dialout-trunk,continue,3) -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/1001-0000000a", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 44 - failing through to other trunks") in new stack -- Executing [continue@macro-dialout-trunk:4] Set("SIP/1001-0000000a", "CALLERID(number)=1001") in new stack -- Executing [01473123456@from-internal:7] Macro("SIP/1001-0000000a", "outisbusy,") in new stack -- Executing [s@macro-outisbusy:1] Progress("SIP/1001-0000000a", "") in new stack -- Executing [s@macro-outisbusy:2] Playback("SIP/1001-0000000a", "all-circuits-busy-now,noanswer") in new stack -- Playing 'all-circuits-busy-now.ulaw' (language 'en') == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/1001-0000000a' in macro 'outisbusy' == Spawn extension (from-internal, 01473123456, 7) exited non-zero on 'SIP/1001-0000000a' -- Executing [h@from-internal:1] Hangup("SIP/1001-0000000a", "") in new stack == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-0000000a' == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/1001-0000000a [2013-10-19 10:57:27] WARNING[2924]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission 0009b781-3ebd000a-62559e5d-6d206cff@192.168.0.5 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 13121ms with no response
- Gernot on October 20, 2013 at 9:07 pm said:
  When issuing
```
dongle show devices
```
  on the Asterisk CLI, does it show the dongle is registered properly?
Dan on September 23, 2013 at 11:05 am said:

Would it be possible to use the same USB dongle to use 3G internet? The following article: http://shkspr.mobi/blog/2012/07/3g-internet-on-raspberry-pi-success/ suggests it works fine but I’m not sure if you could do voice and data at the same time (although its possible to do on a phone).
- Gernot on September 23, 2013 at 6:56 pm said:
  
  In theory yes, but it hasn’t been tried yet. If you can make it work please let us know!
  - Dan on September 24, 2013 at 12:28 am said:
    
    I’ve got the E169 ordered from ebay so I’ll go organise a SIM card too.
Mohammed Hasaballa on September 11, 2013 at 7:35 pm said:

can you please help me with the dongle installation.
i was using the following guide:
http://www.andrius.mobi/2013/03/asterisk-on-raspberry-pi-as-gsm-gateway.html
in the following step:
# download and install chan_dongle
wget https://github.com/jstasiak/asterisk-chan-dongle/archive/asterisk11.zip
unzip asterisk11.zip
cd asterisk-chan-dongle-asterisk11/
aclocal
autoconf
automake -a
./configure
make all
make install
cp etc/dongle.conf /etc/asterisk/

not able to move forward because of the make install command
- Gernot on September 12, 2013 at 4:54 pm said:
  
  You don’t need to compile this manually. Just follow the instructions in the documentation section.
Alex on August 21, 2013 at 3:28 am said:

Hi everybody.
I have a little problem with my 3G dongle Huawei E171.
It work’s fine with asterrisk 1.6 + chan_dongle in my ASUS WL-500Gp v2 router but now I try to configure my RaspberryPi for Asterisk.
So here is a problem – My RaspberryPi reboots when an incoming call. And I can’t see any problem in logs because it reboots (((((
Could somebody help with this?
- Gernot on August 21, 2013 at 6:53 pm said:
  
  This is a power problem. Please read the documentation here, section troubleshooting power problems. Your modem draws too much power during a call and the RPi’s voltage drops below a critical level.
  - Alex on August 27, 2013 at 12:36 pm said:
    
    Thank you! It’s really was a problem in USB-hub. I’m changed it to new one and now 3G-dongle working fine.
Matthew on July 16, 2013 at 9:20 am said:

I have this running now from my Pi using an E160 dongle that I re-flashed with the current firmware and then enabled voice on.

As part of our PBX, all calls need to go out as withheld (blocked, etc…). At the moment the call goes out with the mobile number. I have tried using calledid and it still shows the mobile number. I’ve also tried various combinations of blocked, unavailable, withheld, none, etc…

Any suggestions as I cannot use this fully until this has been sorted.

Cheers!
- Gernot on July 16, 2013 at 8:59 pm said:
  
  Have a look at this comment. You should also be able to set anonymous calling directly in the network.
  - Matthew on July 16, 2013 at 9:39 pm said:
    
    AMAZING! Perfect 🙂 Works a treat 😀
    - Matthew on July 17, 2013 at 9:27 pm said:
      
      I don’t suppose it is possible to use a did number as the caller id for the dongle? We have a few different did’s that come in. I’d like to set it so certain extensions call out with the outbound called id as 1 did and other extensions use another did number.
      - Gernot on July 18, 2013 at 11:56 am said:
        
        This is unfortunately not possible. The cell phone providers only allow sending the own phone number as caller id or blocking it (calling anonymously).
Ken on June 23, 2013 at 2:43 am said:

Can someone please help make head and tails of this error:

[2013-06-23 10:34:37] WARNING[7185][C-0000003a] pbx.c: Channel ‘Dongle/dongle0-0100000008’ sent to invalid extension but no invalid handler: context,exten,priority=from-trunk-dongle,+XXXXXXXXXXXX,1

I’ve checked the incoming routes and all extensions/misc destinations but I can’t seem to find the one that’s creating the error.
- Ken on June 23, 2013 at 11:07 am said:
  
  Got it…
  
  Been wondering what’s the purpose of these two lines…
  
  exten => _.,1,Set(CALLERID(name)=${CALLERID(num)})
  exten => _.,n,Goto(from-trunk,${EXTEN},1)
  
  Logs report a warning:
  
  [2013-06-23 17:36:28] WARNING[3646] pbx_config.c: The use of ‘_.’ for an extension is strongly discouraged and can have unexpected behavior. Please use ‘_X.’ instead at line 10 of /etc/asterisk/extensions_custom.conf
  [2013-06-23 17:36:28] VERBOSE[3646] pbx.c: — Added extension ‘_.’ priority 1 to from-trunk-dongle
  [2013-06-23 17:36:28] WARNING[3646] pbx_config.c: The use of ‘_.’ for an extension is strongly discouraged and can have unexpected behavior. Please use ‘_X.’ instead at line 11 of /etc/asterisk/extensions_custom.conf
  
  so I changed it accordingly… that was the source of the error.
  
  Remind me not to follow blindly all warnings… 🙂
  
  Thanks
  - Gernot on June 23, 2013 at 1:54 pm said:
    
    This is a feature requested by another user some time ago. 🙂 Seems like it doesn’t work when using _X. instead…
Ahmed on June 20, 2013 at 1:57 am said:

Can someone help me with installing 2 GSM dongles?

this is what my dongle.conf looks like
; dongle required settings
[dongle0]
audio=/dev/ttyUSB1 ; tty port for audio connection; no default value
data=/dev/ttyUSB2 ; tty port for AT commands; no default value

[dongle3]
audio=/dev/ttyUSB4 ; tty port for audio connection; no default value
data=/dev/ttyUSB5 ; tty port for AT commands; no default value

dongle3 seems to work fine, however dongle0 wont receive inbound calls,
- Gernot on June 20, 2013 at 7:33 pm said:
  
  Also here, please have a look at the logs as soon as the call comes in. You can also post your log in the forums.
  - Ahmed on June 22, 2013 at 1:51 am said:
    
    Gernot,
    Nevermind, I got myself another dongle that works just fine, i dont think that the E180 supports voice calls.
    Thanks
Dmtg on June 19, 2013 at 8:12 am said:

I found this script to another pbx, altered to make it work in rasppbx honestly do not know if opensource
Dmtg on June 18, 2013 at 6:47 pm said:

Script php to send message from chan_dongle http://speedy.sh/CX3uj/sms.rar
- Gernot on June 19, 2013 at 6:29 am said:
  
  Thank you very much for your script, it works perfect! Is this script open source, and if yes can it be included in RasPBX?
  - Dmtg on June 19, 2013 at 8:17 am said:
    
    I found this script to another pbx, altered to make it work in raspbx honestly do not know if opensource.
    For me it should be included in raspbx.
    - Dmtg on June 19, 2013 at 8:18 am said:
      
      * This product is solely a private production of the above named
      * author, and is neither endorsed nor supported by Eternal Works.
      * Although this product has been thuroughly tested, it is
      * distributed AS IS, and the author assumes no liability for any
      * damages this script may cause to your system. The author
      * has provided full source code and encourages you to review the
      * source code to determine any effects it may have on your system.
      *
      * (c) Copyright 2011, Troy A Nahrwold, Eternal Works, LLC.
      * All Rights Reserved.
    - Gernot on June 19, 2013 at 6:58 pm said:
      
      Ok great! I do want to include this with RasPBX, but I think some sort of password protection should be in there, otherwise everyone on the local network can reach it.
      - DMTG on June 20, 2013 at 7:17 am said:
        
        yes just put this folder inside the “modules” and put script from password protection
Stanlee on June 13, 2013 at 7:45 pm said:

I have a problem with DISA callthrough service. I can call the mobile inbound number, than i get dial tone, dial the destination number, it rings on the other side, and if the called phone just wants to decline the call, it is impossible, it will ring and ring, if i end the call on the calling mobile, the call will be ended. What is wrong?
Pingback: Calling on GSM/3G Networks | Asterisk for Raspb...
mohammed on May 26, 2013 at 10:30 pm said:

hi, with hawei dongle i used raspbx and followed instruction and it works out box, only have to run dongle install, now i want to transfer all incoming call to skype account, can anyone help me with step by step instruction for skype to work.
- Gernot on May 27, 2013 at 9:08 am said:
  
  It is not possible to directly integrate Skype with RasPBX. There is a binary module with Skype support available, but only for x86 platforms. But Skype itself offers SIP as well, it can be configured with Skype Connect. In the end you should be able to use Skype as a normal SIP trunk. For details read:
  http://www.skype.com/en/features/skype-connect/
- Ken on May 28, 2013 at 3:36 am said:
  
  Saw this in a forum (http://forums.whirlpool.net.au/archive/2039943). It works with some audio problems at times but I guess it’s more because of my crappy internet service or gsm provider. Bearable for home use not so much for business but it’s free.
Benjamin on May 23, 2013 at 4:04 pm said:

Hi there,

Does anyone already forge an SMS by replacing sender phone number by another number ? I would like to use this to do a “forward” of incoming SMS to my another mobile phone and display the original sender phone number and not my SIM number inside the dongle.

Is that possible ?

Thanks in advance,
Benjamin
- Gernot on May 23, 2013 at 8:00 pm said:
  
  This is not possible as the carrier is forcing the sender CID to the number of your SIM. You cannot override this number. The only possible way is to add the sender CID inside the SMS text. As long as the SMS text is shorter than 160-sendercid characters this should work. Requires some tweaking however in the SMS processing dialplan…
  - Benjamin on May 24, 2013 at 8:47 am said:
    
    That’s a really bad news for me.
    
    I just wonder how some companies find a way to provide such a service : a web interface with in particular a field to fill with the sender phone number.
    
    Thank you anyway for your answer.
Tk on May 17, 2013 at 3:48 pm said:

Another success story here:
http://takis.name/blog/asterisk-gsm-3g-voice-gateway-fct-using-raspberry-pi-and-huwaei-modem/

I wasn’t aware of asterisk for raspberry pi then, and I complied it from source.
- Gernot on May 17, 2013 at 5:17 pm said:
  
  Thanks for letting us know! I’ve read your blog before, you are probably one of the first who made chan_dongle work on the RPi. Congratulations! 🙂
Alex_Ukraine on May 16, 2013 at 7:31 pm said:

Hi all! i have Hackberry (https://www.miniand.com/products/Hackberry%20A10%20Developer%20Board) and i want do the same (install asterisk+dongle). I was install Asterisk 1.8.13.1~dfsg-3 from repo and compile dongle from source. compiling had several mistakes, but i am ignore that and when load module chan_dongle have warning WARNING[26056]: chan_dongle.c:232 opentty: tcgetattr() failed for /dev/usbdev1.1: Inappropriate ioctl for device.
who have any idea please write, thanks a lot.
P.S. sory, if i take mistakes in my post 😉
- Gernot on May 17, 2013 at 5:15 pm said:
  
  I’ve not been using Asterisk from the Raspbian repository, instead compiled it myself. On top, I’m using Asterisk 11 and a modified chan_dongle for Asterisk 11.
  Maybe you can get more help on the original chan_dongle forum.”
FD on May 6, 2013 at 7:47 am said:

Hi Guys, I managed to install all, please check a demo here:
http://youtu.be/hS5uUtUZpVI
Tingeling on May 5, 2013 at 8:08 am said:

Do I need apt-utils for any reason?
Can I install more applications of my own choice without disrupt the default settings for Astrisk?
- Gernot on May 5, 2013 at 11:29 am said:
  
  Apt-utils is not really required. You can use apt-get to install additional software in the same way as on the official Raspbian image. Installing additional software does not disrupt the Asterisk settings.
Ultrakiller on April 29, 2013 at 10:48 am said:

There is a fix who i want to talk about. Looking at the guides to make 3g dongles working on linux, most of times is suggest, and required, to push an AT Command who disable CDRom automated mount to get dongle recognized correctly and working on asterisk. Now, this is not so true, there is another less risky way.
I wrote a guide to make it possible:
# apt-get install usb-modeswitch usbutils
#nano /etc/usb_modeswitch.d/12d1:1446
and copy this configuration in: (i have an E1692, if u have another key get correct configuration here: )
# Huawei E1692
DefaultVendor= 0x12d1
DefaultProduct= 0x1446

TargetVendor= 0x12d1
TargetProduct= 0x140c

MessageContent=”55534243000000000000000000000011060000000000000000000000000000″

CheckSuccess=5

Now you have to add a rule in /etc/udev/rules.d/ creating 50-huawei-e1552.rules filename. Now add these text into that file:

SUBSYSTEM==”usb”,
ATTR{idVendor}==”12d1″,
ATTR{idProduct}==”1446″,
RUN+=”usb_modeswitch ‘%b/%k”

NOW,REBOOT THE WHOLE SYSTEM!!!!!
# reboot

On next reboot you can see by lsusb changing in a product ID, in fact, all Huawei keys are recognized as 12d1:1446 and now mine is recognized because of usb-modeswitch as 12d1:144c where 144c wich is modem ability, not CDRom issue anymore!! 😉
- Ultrakiller on April 29, 2013 at 10:51 am said:
  
  Known issue:
  Using that guide all ttyUSB are enabled
  ls -l /dev/ttyUSB*
  but asterisk have no rights to use that devices.
  
  Fix:
  execute a chmod 777 /dev/ttyUSB* and asterisk starts to working like a charm.
  But i have to run chmod for every reboot on a system.
  If someone know how to solve that, is welcome 😀
  - Ultrakiller on April 29, 2013 at 10:54 am said:
    
    Missed link in the guide:
    *(i have an E1692, if u have another key get correct configuration here: http://www.draisberghof.de/usb_modeswitch/device_reference.txt )
  - Gernot on April 29, 2013 at 5:17 pm said:
    
    Have you been using the installer script install-dongle? It is already taking care about permissions by creating the proper file in the rules.d directory.
  - dziny on June 11, 2013 at 9:14 pm said:
    
    This one is easy. Create file
    /etc/udev/rules.d/92-dongle.rules
    
    with content:
    KERNEL==”ttyUSB*”, MODE=”0666″, OWNER=”asterisk”, GROUP=”asterisk”
- Gernot on April 29, 2013 at 5:16 pm said:
  
  Are you absolutely sure it doesn’t work without usb_modeswitch? Most modems (like the E169) are working perfectly fine also without usb_modeswitch. If yours is an exception I can add this to the instructions.
  - Ultrakiller on April 30, 2013 at 11:50 am said:
    
    Abt 1° Comment:
    Yes, i used install-dongle script and asterisk have rights only with chmod 777 but as i said, logically, rights were lost on every reboots.
    i’ll try to add asterisk user to a right group, because as i saw (i’m not with rpi now) ttyUSB* are owned by another user (not root).
    
    I used install-dongle script either before or after my guide, in both cases results is the same!
    
    Abt 2° Comment:
    Yes, i’m absolutly sure, i don’t know if my dongle model is an exeption or not, anyway its common that kind of “trick” to switch usb dongles to get working as modem in *nixes distros.
    Gernot, feel free to use what i said there, if you want extend it to others dongles by parameters in the install-dongle script 🙂
    - Gernot on April 30, 2013 at 7:52 pm said:
      
      Thank you very much for all the detailed information! I will add this to the install-dongle script and the documentation soon. I will also have a look at the permission issue.
    - Gernot on May 1, 2013 at 3:44 pm said:
      
      The permissions issue can be solved by renaming the file /etc/udev/rules.d/99-permissions.rules to /etc/udev/rules.d/30-permissions.rules
      Installing usb_modeswitch disabled this file, that’s why you had the permissions problem.
      
      I also had a closer look at usb_modeswitch: From what I’ve learned it is not required to manually create the files /etc/usb_modeswitch.d/12d1:1446 and 50-huawei-e1552.rules because the right configuration is already contained in usb_modeswitch by default. Could you remove those 2 files, reboot and report if it is still working?
      - Ultrakiller on May 1, 2013 at 8:12 pm said:
        
        Thank you for the tip
        I’ll try ASAP (give me 2 days) both cases and i will let u know here 🙂
Edy on April 24, 2013 at 2:26 pm said:

Hi guys,
i have managed to establish successfull calls (both originating and terminating) with the above setup. Can you please guide me on what needs to be administered for USSD/SMS to work. Currently, an SMS or USSD reply are being received by my SIP soft client as voice calls.
- Gernot on April 24, 2013 at 8:51 pm said:
  To receive SMS make the following changes:
  Edit /etc/asterisk/dongle.conf and change the line context=from-trunk to
```
context=from-trunk-dongle
```
  Then edit the file /etc/asterisk/extensions_custom.conf and add these lines
```
[from-trunk-dongle]
exten => sms,1,Verbose(Incoming SMS from ${CALLERID(num)} ${BASE64_DECODE(${SMS_BASE64})})
exten => sms,n,System(echo ‘${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} – ${DONGLENAME} – ${CALLERID(num)}: ${BASE64_DECODE(${SMS_BASE64})}’ >> /var/log/asterisk/sms.txt)
exten => sms,n,Hangup()
exten => _.,1,Set(CALLERID(name)=${CALLERID(num)})
exten => _.,n,Goto(from-trunk,${EXTEN},1)
```
  Received SMSes will be stored in /var/log/asterisk/sms.txt
  It is of course also possible to forward the sms by email or to send it to a text message capable SIP phone.
  - Ken on June 11, 2013 at 2:01 am said:
    
    Hi Gernot… would it not be too much to ask if I’ll request for you to post a sample dialplan on how to forward received sms to another mobile? Seems that features such as this were being developed for chan_datacard before (zMessage, if I remember correctly) but there’s very few documentation and most links are now defunct.Thank you very much in advance.
    - Gernot on June 11, 2013 at 6:38 pm said:
      
      I will have a look at this and let you know. It will require a bit of time…
      - Ken on June 14, 2013 at 4:43 am said:
        
        Thanks as always, anyway, here’s what I’ve done so far….
        I added this as a second line to your dial plan as above…
        
        exten => sms,n,System(dongle sms dongle0 +XXXXXXXXXXXX From:${CALLERID(num)} Msg:${BASE64_DECODE(${SMS_BASE64})})
        
        but I get error saying:
        
        [2013-06-14 12:31:23] WARNING[31860][C-00000041] app_system.c: Unable to execute ‘dongle sms dongle0 +XXXXXXXXXX From:+XXXXXXXXXX Msg:testing 123’
        
        however, when I try to send:
        
        CLI > dongle sms dongle0 +XXXXXXXXXX From:+XXXXXXXXXX Msg:testing 123
        
        I receive the message…
      - Ken on June 14, 2013 at 6:01 am said:
        
        Just after posting I realized a mistake…
        
        the call to dongle sms has to be
        
        exten => sms,n,System(asterisk -rx “dongle sms dongle0 +XXXXXXXXXXXX From:${CALLERID(num)} Msg:${BASE64_DECODE(${SMS_BASE64})})”
        
        no errors but sms still not sent.
  - Ken on June 14, 2013 at 6:25 am said:
    
    sorry if this can be construed as flooding… just being excited… after further reading, I realized that the command to send sms within the dial plan is different… so I changed it to:
    
    exten => sms,n,DongleSendSMS(dongle0, destination #, message.
    
    changed it accordingly and received the message.
    
    will update what would be the result with long messages and what happens when a call on the same dongle is in progress…
    
    again, my apologies for flooding..
    - Gernot on June 14, 2013 at 5:57 pm said:
      
      No problem, you’re not flooding! 🙂 Great you’ve found it. I still didn’t get a chance to look at this myself, but hopefully will be able to do so soon.
    - Gernot on June 16, 2013 at 11:01 am said:
      
      I’ve check this myself and it works perfectly fine, exactly as you wrote it! Thanks a lot for your hint. You need to add this line
      
      exten => sms,n,DongleSendSMS(dongle0,+123456789,${BASE64_DECODE(${SMS_BASE64})} - from ${CALLERID(num)})
      
      To /etc/asterisk/extensions_custom.conf directly before the line
      
      exten => sms,n,Hangup()
      
      Replace +123456789 with your destination mobile number. Incoming SMSs will be forwarded to this number.
      - Ken on June 16, 2013 at 9:08 pm said:
        
        Glad you’ve verified it… the following caveat applies though:
        
        1. Messages longer than 160 characters are cut-off.
        2. When the dongle is being used, i.e. during a call, and sms comes in, it sending would still commence but would result in error. Dunno if it’s a protocol, telco or design limitation.
      - Ahmed on June 19, 2013 at 5:11 pm said:
        
        Im trying to set this up so the dongle can forward any received msg to my number, this is what i have in the extentions_custom.conf:
        exten => sms,n,DongleSendSMS(dongle0,+29391241030,${BASE64_DECODE(${SMS_BASE64}$
        exten => sms,n,Hangup()
        exten => _.,1,Set(CALLERID(name)=${CALLERID(num)})
        exten => _.,n,Goto(from-trunk,${EXTEN},1)
        
        I’ve sent a couple of msgs to my dongle but didnt receive anything. what am i doning wrong?
      - Gernot on June 19, 2013 at 6:55 pm said:
        
        Can you have a look at the logs to see what’s happening there? You can also post your log at pastebin or in the forums.
      - Ahmed on June 20, 2013 at 1:53 am said:
        
        What log would you like me to post?
      - Gernot on June 20, 2013 at 6:38 am said:
        
        /var/log/asterisk/full or run asterisk -rvvvvv
        But please don’t spill the comments, use pastebin or move over to the forums.
  - nom on July 16, 2013 at 2:51 pm said:
    
    Hi,
    Would like to request your assistance on how to forward Received SMSes stored in /var/log/asterisk/sms.txt
    to a text message capable SIP phone.
    
    thanks in advance
Richard on April 20, 2013 at 10:44 am said:

Hi

Does the image have iptables and things for security, i wouldnt mind changing our tower system over to a Pi but has we have been hacked once with our old system..
- Gernot on April 20, 2013 at 5:35 pm said:
  
  Iptables are installed of course. Fail2ban is available as well to effectively block intruders, see the documentation here. Besides this, the RPi should be kept behind a router with all ports closed. Port forwarding should only be used for RTP ports. An exception can be made for the SIP ports if remote extensions are required.
  If you follow the general security tips (see Razvan’s tutorial) you should be pretty safe.
Luiz on April 14, 2013 at 2:50 am said:

Is possible to do something like to this, but instead of redirecting directly the call, you call gsm gateway and make calls via sip? like using a local cellphone number to call the world using cheap voip calls
- Gernot on April 15, 2013 at 7:46 pm said:
  
  Yes this is possible. You can call the SIM card number, get a dialtone on the other side, and dial your destination with DTMF tones. The call can then be placed through any sip provider. (Check for the DISA module in FreePBX)
RealPBX on April 12, 2013 at 10:22 am said:

I appreciate this GSM Gateway very much?
guidocx84 on April 5, 2013 at 5:42 pm said:

This project is every day most wonderful! Thank you so much for your work!!!
Dan on April 4, 2013 at 12:19 pm said:

An alternative to uploading through the web GUI is to just copy the files over to the rPi using scp. WinSCP is reported to be a good client for Windows, or you can use scp at the command line on Mac or *nix. Once you’ve got the files on the Pi, copy them to /var/lib/asterisk/moh/ and you’ll be set.
nddn on April 3, 2013 at 9:40 am said:

Hi there,

I’m having problems uploading Hold music through freepbx. The error says that the filesize has exceeded the maximum file size in etc/php.ini. There isn’t any php.ini there but there is on at /etc/php5/apache2/php.ini and also in another in another folder in /etc/php5. I changed it from 2M to 20M but I can’t upload a file more than 2M.

Any ideas?

Thanks.
- Gernot on April 3, 2013 at 8:10 pm said:
  You have already found the right place. Edit the file /etc/php5/apache2/php.ini and set
```
upload_max_filesize = 20M
```
  and also
```
post_max_size = 20M
```
  After this you need to restart Apache:
```
service apache2 restart
```
  This has worked perfectly for me.
  - nddn on April 3, 2013 at 10:12 pm said:
    
    Hi,
    
    I changed the file size in /etc/php5/apache2/php.ini and /etc/php5/cli/php.ini to 20M but it still shows error: PHP Error Processing World Mix.wav! Check upload_max_filesize in /etc/php.ini
    
    Any ideas?
    
    Thanks
    - Gernot on April 4, 2013 at 8:41 pm said:
      
      Did you also set post_max_size to 20M? I have tried this and it definitely worked for me. Have a look at the Apache log file /var/log/apache2/error.log, maybe you can find a more detailed error message there.
      - nddn on April 5, 2013 at 1:52 pm said:
        
        Seem to have gotten it now. Changed it to 2000M, file was larger than 20M.
        
        Thanks again 🙂