Solved

asterisk chan_dongle error

Posted on 2014-01-23
4
2,280 Views
Last Modified: 2014-02-02
Hello, i am using chan_dongle on my Asterisk/FreePBX, so that I can forward some inbound SIP calls to a GSM cell, through a GSM gateway.

I have updated the dongle to the latest firmware, and I have confirmed it supports voice. through "mobile partner" software, I can make calls when the dongle is connected to PC or Mac.

However when in Asterisk/FreePBX, I get the below error, it says that the device can not make calls at this time.

I have also connected via minicom and confirmed that the modem supports C^VOICE

Here is the output. Any ideas what i'm doing wrong?

raspbx*CLI> dongle show devices
ID           Group State      RSSI Mode Submode Provider Name  Model      Firmware          IMEI             IMSI             Number        
dongle0      0     Free       18   5    4       CHN-UNICOM     E153       11.609.18.00.00   35237000000000  454030000000000  Unknown       
    -- Executing [+85200000000@from-trunk-dongle:1] Set("Dongle/dongle0-0100000000", "CALLERID(name)=+85211111111") in new stack
    -- Executing [+85200000000@from-trunk-dongle:2] Goto("Dongle/dongle0-0100000000", "from-trunk,+85200000000,1") in new stack
    -- Goto (from-trunk,+85200000000,1)
    -- Executing [+85200000000@from-trunk:1] Set("Dongle/dongle0-0100000000", "__FROM_DID=+85200000000") in new stack
    -- Executing [+85200000000@from-trunk:2] Set("Dongle/dongle0-0100000000", "CDR(did)=+85200000000") in new stack
    -- Executing [+85200000000@from-trunk:3] ExecIf("Dongle/dongle0-0100000000", "0 ?Set(CALLERID(name)=+85211111111)") in new stack
[2014-01-23 16:32:11] WARNING[4981][C-00000001]: func_callerid.c:905 callerpres_read: CALLERPRES is deprecated.  Use CALLERID(name-pres) or CALLERID(num-pres) instead.
    -- Executing [+85200000000@from-trunk:4] Set("Dongle/dongle0-0100000000", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [+85200000000@from-trunk:5] Set("Dongle/dongle0-0100000000", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [+85200000000@from-trunk:6] Goto("Dongle/dongle0-0100000000", "ext-trunk,3,1") in new stack
    -- Goto (ext-trunk,3,1)
    -- Executing [3@ext-trunk:1] Set("Dongle/dongle0-0100000000", "SS=$") in new stack
    -- Executing [3@ext-trunk:2] Set("Dongle/dongle0-0100000000", "TDIAL_STRING=dongle/dongle0/${OUTNUM}") in new stack
    -- Executing [3@ext-trunk:3] Set("Dongle/dongle0-0100000000", "DIAL_TRUNK=3") in new stack
    -- Executing [3@ext-trunk:4] Goto("Dongle/dongle0-0100000000", "ext-trunk,tcustom,1") in new stack
    -- Goto (ext-trunk,tcustom,1)
    -- Executing [tcustom@ext-trunk:1] Set("Dongle/dongle0-0100000000", "OUTBOUND_GROUP=OUT_3") in new stack
    -- Executing [tcustom@ext-trunk:2] GotoIf("Dongle/dongle0-0100000000", "0?nomax") in new stack
    -- Executing [tcustom@ext-trunk:3] GotoIf("Dongle/dongle0-0100000000", "0?hangit") in new stack
    -- Executing [tcustom@ext-trunk:4] ExecIf("Dongle/dongle0-0100000000", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack
    -- Executing [tcustom@ext-trunk:5] Set("Dongle/dongle0-0100000000", "DIAL_NUMBER=+85200000000") in new stack
    -- Executing [tcustom@ext-trunk:6] GosubIf("Dongle/dongle0-0100000000", "0?sub-flp-3,s,1()") in new stack
    -- Executing [tcustom@ext-trunk:7] Set("Dongle/dongle0-0100000000", "OUTNUM=+85200000000") in new stack
    -- Executing [tcustom@ext-trunk:8] Set("Dongle/dongle0-0100000000", "CALLERID(number)=+85211111111") in new stack
    -- Executing [tcustom@ext-trunk:9] Set("Dongle/dongle0-0100000000", "CALLERID(name)=+85211111111") in new stack
    -- Executing [tcustom@ext-trunk:10] Set("Dongle/dongle0-0100000000", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [tcustom@ext-trunk:11] Dial("Dongle/dongle0-0100000000", "dongle/dongle0/+85200000000,300,") in new stack
[b][2014-01-23 16:32:11] WARNING[4981][C-00000001]: channel.c:180 channel_request: [dongle0] Request to call on device which can not make call at this moment
[2014-01-23 16:32:11] WARNING[4981][C-00000001]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'dongle' (cause 44 - Requested channel not available)[/b]
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [tcustom@ext-trunk:12] Hangup("Dongle/dongle0-0100000000", "") in new stack
  == Spawn extension (ext-trunk, tcustom, 12) exited non-zero on 'Dongle/dongle0-0100000000'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5

Open in new window

0
Comment
Question by:sk391
  • 2
  • 2
4 Comments
 
LVL 15

Expert Comment

by:Phonebuff
ID: 39805556
I think you want Chan_mobile not Chan_dongle for this.

But many have found it some what problematic.  

http://www.voip-info.org/wiki/view/chan_mobile

If not what models of GSM Cell & Gateway are you trying to use ?

There are many "Gateways" that are straight SIP channels on Asterisk.  Here is a link to the GoIP product,  http://nerdvittles.com/?p=7581
0
 
LVL 1

Author Comment

by:sk391
ID: 39808378
Thank you, but I see everyone who has done a similar project is using chan_dongle, not chan_mobile (exactly as you point out chan_mobile is problematic)

This describes the setup I am trying to reproduce, a cheap custom raspberry pi GSM gateway:
http://blog.carrier-connect.com/raspberry-gsm-gateway/

UPDATE: I figured out it was a simple problem: in my outbound route, I didn't have the period "." in the patterns, so the calls weren't getting out. For testing purposes, I am now able to connect to my asterisk PBX and place calls from a SIP phone using asterisk's GSM trunk.

I only have two inbound SIP trunks registered on the PBX, and one SIP extension.  I wanted to ask what is the simplest way to achieve the below:

For any inbound route that comes in, I have set them to hit my "800" extension. How do I configure this extension to forward to a mobile number by dialing through the GSM trunk?

In case it's not clear: I only have one outbound route (out), and only one outgoing trunk (GSM)

I don't plan to use a SIP phone for this scenario though, I just want Asterisk to receive the inbound SIP calls and forward them to my cell through the GSM trunk.

Thank you!
0
 
LVL 1

Author Comment

by:sk391
ID: 39808481
I was able to setup this configuration by creating a ring group like below:

800
123456789# (123456789=my cell that I wanted to forward calls to)

then I set it to "ringall"

I set the ring group as destination for my two inbound routes,  so now when calls hit my asterisk pbx, they get routed to both my SIP phone and my cell phone (over the GSM trunk) and I can pickup from either of them.
0
 
LVL 15

Accepted Solution

by:
Phonebuff earned 500 total points
ID: 39808811
That will work, but it's over kill.  

You can create an extension type other and then just set the Dial value you want --
0

Featured Post

Do You Know the 4 Main Threat Actor Types?

Do you know the main threat actor types? Most attackers fall into one of four categories, each with their own favored tactics, techniques, and procedures.

Join & Write a Comment

This is a step by step guide on creating single number reach (mobility) for Cisco Call Manager.  After configuring this when someone calls your deskphone after 3-4 rings your cell phone will start to ring.  If you do not answer the call will go to y…
If your business is like most, chances are you still need to maintain a fax infrastructure for your staff. It’s hard to believe that a communication technology that was thriving in the mid-80s could still be an essential part of your team’s modern I…
Sending a Secure fax is easy with eFax Corporate (http://www.enterprise.efax.com). First, Just open a new email message.  In the To field, type your recipient's fax number @efaxsend.com. You can even send a secure international fax — just include t…
Internet Business Fax to Email Made Easy - With eFax Corporate (http://www.enterprise.efax.com), you'll receive a dedicated online fax number, which is used the same way as a typical analog fax number. You'll receive secure faxes in your email, fr…

757 members asked questions and received personalized solutions in the past 7 days.

Join the community of 500,000 technology professionals and ask your questions.

Join & Ask a Question

Need Help in Real-Time?

Connect with top rated Experts

22 Experts available now in Live!

Get 1:1 Help Now