Chan_dongle issues

[mark...@gmail.com]

I have 3 dongles and am trying to get more consistent behaviour from them

info on 3 dongles
[code]
#dongle show device state dongle0
-------------- Status -------------
  Device                  : dongle0
  State                   : Free
  Audio                   : /dev/ttyUSB1
  Data                    : /dev/ttyUSB2
  Voice                   : Yes
  SMS                     : Yes
  Manufacturer            : huawei
  Model                   : E303
  Firmware                : 21.158.13.00.864
  IMEI                    : 86907XXXXXXXXXX
  IMSI                    : 33402XXXXXXXXXX
  GSM Registration Status : Registered, home network
  RSSI                    : 22, -69 dBm
  Mode                    : No Service
  Submode                 : No service
  Provider Name           : TELCEL
  Location area code      : "36D2"
  Cell ID                 : "5CC7"
  Subscriber Number       : +521331006XXXX
  SMS Service Center      : +5294100001410
  Use UCS-2 encoding      : Yes
  USSD use 7 bit encoding : No
  USSD use UCS-2 decoding : Yes
  Tasks in queue          : 0
  Commands in queue       : 0
  Call Waiting            : Disabled
  Current device state    : start
  Desired device state    : start
  When change state       : now
  Calls/Channels          : 0
    Active                : 0
    Held                  : 0
    Dialing               : 0
    Alerting              : 0
    Incoming              : 0
    Waiting               : 0
    Releasing             : 0
    Initializing          : 0
__________________________________________________________________________

#dongle show device state dongle1
-------------- Status -------------
  Device                  : dongle1
  State                   : Free
  Audio                   : /dev/ttyUSB4
  Data                    : /dev/ttyUSB5
  Voice                   : Yes
  SMS                     : Yes
  Manufacturer            : huawei
  Model                   : E153
  Firmware                : 11.609.18.00.00
  IMEI                    : 35237XXXXXXXXXX
  IMSI                    : 3340202XXXXXXXX
  GSM Registration Status : Registered, home network
  RSSI                    : 25, -63 dBm
  Mode                    : WCDMA
  Submode                 : WCDMA
  Provider Name           : Mx Telcel GSM
  Location area code      : 36D2
  Cell ID                 : 5CC7
  Subscriber Number       : 322XXXXXXX
  SMS Service Center      : +5294100001410
  Use UCS-2 encoding      : Yes
  USSD use 7 bit encoding : Yes
  USSD use UCS-2 decoding : No
  Tasks in queue          : 0
  Commands in queue       : 0
  Call Waiting            : Disabled
  Current device state    : start
  Desired device state    : start
  When change state       : now
  Calls/Channels          : 0
    Active                : 0
    Held                  : 0
    Dialing               : 0
    Alerting              : 0
    Incoming              : 0
    Waiting               : 0
    Releasing             : 0
    Initializing          : 0


___________________________________________________________________________

dongle show device state dongle2
-------------- Status -------------
  Device                  : dongle2
  State                   : Free
  Audio                   : /dev/ttyUSB7
  Data                    : /dev/ttyUSB8
  Voice                   : Yes
  SMS                     : Yes
  Manufacturer            : huawei
  Model                   : E153
  Firmware                : 11.609.18.00.00
  IMEI                    : 35350XXXXXXXXXX
  IMSI                    : 33403XXXXXXXXXX
  GSM Registration Status : Registered, home network
  RSSI                    : 24, -65 dBm
  Mode                    : No Service
  Submode                 : No service
  Provider Name           : movistar
  Location area code      : 2CFB
  Cell ID                 : 8CCA
  Subscriber Number       : +52415123XXXX
  SMS Service Center      : +525512235100
  Use UCS-2 encoding      : Yes
  USSD use 7 bit encoding : Yes
  USSD use UCS-2 decoding : No
  Tasks in queue          : 0
  Commands in queue       : 0
  Call Waiting            : Disabled
  Current device state    : start
  Desired device state    : start
  When change state       : now
  Calls/Channels          : 0
    Active                : 0
    Held                  : 0
    Dialing               : 0
    Alerting              : 0
    Incoming              : 0
    Waiting               : 0
    Releasing             : 0
    Initializing          : 0
[/code]


1
 I see differences in the following section
  Use UCS-2 encoding      : Yes
  USSD use 7 bit encoding : No
  USSD use UCS-2 decoding : Yes

2
#dongle show devices
ID           Group State      RSSI Mode Submode Provider Name  Model      Firmware          IMEI             IMSI             Number       
dongle0      0     Free       22   0    0       TELCEL         E303       21.158.13.00.864  86907XXXXXXXXXX  33402XXXXXXXXXX  +52133XXXXXXXX
dongle1      0     Free       25   5    4       Mx Telcel GSM  E153       11.609.18.00.00   35237XXXXXXXXXX  33402XXXXXXXXXX  322XXXXXXX
dongle2      0     Free       24   0    0       movistar       E153       11.609.18.00.00   35350XXXXXXXXXX  33403XXXXXXXXXX  +52415XXXXXXX

I am concerned about the mode ands submode displayed above and not sure where this comes from

3
I run chan_dongle on a text mode asterisk box . That sends calls to my Elastix PBX over the internet.

It seems I am stuck using the slin codec to send calls to elastix or I miss the first several seconds of a call, get screeching noises or no audio at all . It helps also if I answer the call immediately in the asterisk dial plan then pass on to destination.

4
The e303 was a particular bugger to get voice activated on it required modification of drivers so it would load them once in download mode. I however can not disable the virtual CD nor the micoSD using at commands and in fact if I add the line u2diag=0 n d0ngle.conf it returns an error (or same command from console).

the e303 also will not load via imei in dongle.conf


THanks

Mark

[Alejandro Donato]

Hi:

Correct me if i missunderstand something.

You recieve calls in an asterisk and route them to another one.

The first test is to get the call in the original asterisk, that manage the dongles. This is, to discard issue comes from the channel driver.

If calls there works fine, maybe the problem is in the transcoding or the iax channel between both asterisks.

So, try registering an extension in the original asterisk and pick calls from there directly, to check channel performance. Comment the results.

The fact that some dongle dont accept some commands, is not relevant, if that one can manage voice calls.

Regards

Alejandro

El 15/11/15 a las 16:14, mark...@gmail.com escribió:

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[mark...@gmail.com]

Alejandro, Yes you are correct that there are 2 asterisks, however the codec/noise/Audio problem exists even on the same asterisk . This adterisk Gateway is the very place the translation seems to go wrong.  I orignally registered a softphone to the asterisk gateway.


Also I failed to mention some points

this gateway aside from the dongles has an active x100p clone card as well as dongles

This gateway sends the calls to Ther main elastix over SIP (I could not make IAX work probably due to no real timeing source on the elastix VPS???)

The gateway runs Ubuntu 14.04 asterisk 11.7.0~dfsg-1ubuntu1

[mark...@gmail.com]

Also

"The fact that some dongle dont accept some commands, is not relevant, if that one can manage voice calls."

This is YOUR OPINION, not mine nor reality. I need tio shut off the CD and MicroSD as this as I read elsewhere will often cause the dongle to autodetect

On Sunday, November 15, 2015 at 1:30:40 PM UTC-6, Alejandro Donato wrote:

[michaeldaly600]

the e303 also will not load via imei in dongle.conf

try running just the *one* dongle. Maybe your setup does not have enough power for the 3 dongles.
Also...do you know how to specify the interasterisk codec? Would recommend using g711. If not maybe try using the freepbx gui to interconnect, if you have sufficient cpu capacity
MD
 

[Alejandro Donato]

Its a fact that the use of the extra capabilities of the dongles, dont affect the voice management. Under Windows, ie, they work without deactivating any extra functions (like microsd reader or cd drive emulation) with no consecuences. Thats because this all is irrelevant to your issue. the mode change is, basically, to help device detection. You have no issue with that (finally you made it detect the dongle and driver looks like manages it correctly)

As MD says, maybe, is a power issue. Try only one dongle first, to discard this. Dongles need around 250ma at full throttle (during a call) and usb ports not always can provide it.

If problem continues, check transcodings. Dongles uses u-law or a-law as native codec. try not transcoding it. If locally you get a good call, maybe you have not enough CPU power to transcode.

The rest of the values look fine. Good signal, home registered.. 

El 16/11/15 a las 05:47, mark...@gmail.com escribió:

[mark...@gmail.com]

Michael

I was running one dongle previously with same issues and have since tested dongles individually

Also worth noting I have made sure that each dongle is on a USB bus to avoid power issues.

Also as I stated earlier registering to the gateway yields the same results with audio issues. I currently have to pass slin to the main PBX from the GSM gateway and answer the call immmediately. Changing either the codec or not answering the call immediately results in audio issues. The GSM Gateway and elastix PBX use SIP not IAX

Mark

[Michael Daly]

As discussed pls reconfigure to ulaw or alaw codec and retest. Also check that the rtp ports of both asterisk servers are matching.
MD

[mark...@gmail.com]

You are telling me to do exctly what I have already tested time and time again that DOES NOT WORK, and thus the main motivation for my post.

I can tell by your statement abouy "....matching rtp ports...." that clearly you are mistaken. In fact a SIP exchange the RTP ports are set in both directions, they DO NOT need to match. If this were the case I would need to match RTP ports for every ttrunk and extension on my PBX , which clearly I do not

If in fact  that this is NOT an issue with the FXO card in the same machine only the GSM channels

[Emiliano Vazquez]

El 16/11/15 a las 19:35, mark...@gmail.com escribió:

> You are telling me to do exctly what I have already tested time and
> time again that DOES NOT WORK, and thus the main motivation for my post.

This is not the best way to talk with another guy who is trying to help you.

Best regards.

Emiliano.

[mark...@gmail.com]

Emiliano,

I should tell him what?? Sure I will do it AGAIN? (which I have again and again)


 I mean seriously if the guy believes that you need matchiong RTP ports on asterisks he has already demonstrated the usefulness of his post. Just because he posts does not mean it has value. No value means wasting my time...Some people claim expertise by the number of posts they make not by the value of them, even when they are completely WRONG.

Had he actually read my post he would have known I tried ulaw already and it did not work thus I must use slin


You defending him here is no more useful than his post.


Mark

[michaeldaly600]

On Wednesday, 18 November 2015 17:03:26 UTC+10:30, mark...@gmail.com wrote:

Emiliano,

I should tell him what?? Sure I will do it AGAIN? (which I have again and again)

Hi Mark
There may be a misunderstanding here. I have read and reread your posts and cannot, I repeat I cannot, see where you have previously advised that you were already using alaw /ulaw. Perhaps if you could provide more detail of your setup here, then more useful suggestions or even a solution, may arise. In my experience, I have found that when I am required to explicate my setup, I often myself find the answer to the problem!
No hard feelings!
Just trying tohelp!
Michael
 

[mark...@gmail.com]

I said in original post

"3
I run chan_dongle on a text mode asterisk box . That sends calls to my Elastix PBX over the internet.

It seems I am stuck using the slin codec to send calls to elastix or I miss the first several seconds of a call, get screeching noises or no audio at all . It helps also if I answer the call immediately in the asterisk dial plan then pass on to destination."

Now using a LITT:E brain power one will extrapolate that I have attempted other codecs and this is the case. If this were not the case, then how would I know it only works with slin which is almost never used in cases like this.

[michaeldaly600]

I see...my only other thought is to replace the text based asterisk box, e.g with an Elastix Box. There is something bizarre happening, not only this audio issue that to some extent is codec related, but that  IAX trunking between the boxes does not work. IAX in my understanding and experience should be more reliable than SIP to set up. You said the other box is an Elastix VPS, so obviously you are familiar with it. Also, in my experience and understanding, although asterisk RTP ports have a default setting, I would advise making a check to ensure that the ports match on both boxes. Also look at rtp port forwarding on your switch/router...  that may be worth doing. Using a network packet analyser  may also yield more clues

Good luck! This is my last post to this thread!
MD

[dent...@yahoo.com]

hi.

how was this solved?

[Mark DM]

I moved all dongles to a Raspberry Pi

On Mon, Feb 25, 2019 at 7:21 AM dentys03 via dongle <chan_...@googlegroups.com> wrote:

hi.

how was this solved?

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