Askozia Forums
February 09, 2012, 05:53:04 am *
Welcome, Guest. Please login or register.

Login with username, password and session length
 
   Home   Help Search Login Register  
Pages: [1]
  Print  
Author Topic: outgoing caller id problem with version 2.0.rc1  (Read 2271 times)
mcnobrain
Newbie
*

Karma: 0
Posts: 14


View Profile
« on: February 20, 2010, 11:20:07 am »

Hello

On my old system 1.0.3 it was possible to send a "outgoing caller id" to my providers (voipbuster & voipcheap)...to send the caller id the "From User" field in the Advanced Option was used.

I set up a new box with 2.0.rc1 with the same configuration as in 1.0.3 but now the caller id (from user) doesnt work anymore....

Any idee ?

Greetz Mcnobrain
Logged
Michael
Askozia Staff
Hero Member
*

Karma: 50
Posts: 1020


View Profile
« Reply #1 on: February 20, 2010, 11:32:26 am »

I just checked and fromuser is saved and generated correctly in RC1. Do you mean it doesn't have the same effect as before?

Did you try using the Outbound Caller ID options in the provider account?
Logged
mcnobrain
Newbie
*

Karma: 0
Posts: 14


View Profile
« Reply #2 on: February 20, 2010, 11:37:33 am »

Hi

Yes i mean it has not the same effect as befor.....befor this field was used as my outgoing caller id..and it worked correctly...but now if i do a outgoing call...there is no number dsisplayed on the oder site.

If i enter my outgoing number in the "Outgoing Presentation" Field an set it to "Send string defined above" i will get the following error:

A valid Caller ID string must be specified
Logged
giovanni.v
Hero Member
*****

Karma: 51
Posts: 666


View Profile
« Reply #3 on: February 20, 2010, 11:45:08 am »

A valid Caller ID string must be specified

The correct format is CallerName <CallerNumber>, something like mcnobrain <00122334455>.

That issue was fixed some time ago (Ticket #44), you don't need to use any custom rule like in 1.0.3.
Logged
mcnobrain
Newbie
*

Karma: 0
Posts: 14


View Profile
« Reply #4 on: February 20, 2010, 12:14:24 pm »

 Does not work at all.... Huh now the caller id is accepted form askozia  ...but still not shown on the calling site..
Logged
mcnobrain
Newbie
*

Karma: 0
Posts: 14


View Profile
« Reply #5 on: February 20, 2010, 01:29:05 pm »

Now i found out some strange thing....

sometimes it works and sometimes it doesnt....that means sometimes the right number is displayd, sometimes no number is displayed and sometimes just the half of the number is displayed....strange thing !

is it a bug ?

Greetz Mcnobrain
« Last Edit: February 20, 2010, 03:42:44 pm by mcnobrain » Logged
mcnobrain
Newbie
*

Karma: 0
Posts: 14


View Profile
« Reply #6 on: February 23, 2010, 12:03:38 pm »

Hello

hm ..no one with the same error ?
Logged
Michael
Askozia Staff
Hero Member
*

Karma: 50
Posts: 1020


View Profile
« Reply #7 on: February 23, 2010, 03:02:23 pm »

Guess not. Try providing more debugging information (SIP traces similar to https://wush.net/trac/askozia/ticket/44)

That would reveal the cause instead of merely talking about the effect.
Logged
mcnobrain
Newbie
*

Karma: 0
Posts: 14


View Profile
« Reply #8 on: February 23, 2010, 03:38:19 pm »

Hello

I tryed to make some traces...but today it runs without any problems (did not change somthing) ...so it looks that it was a problem on the provieder site..i will observer this and keep you informed...

thanks for your help
Logged
ajitam
Newbie
*

Karma: 0
Posts: 3


View Profile
« Reply #9 on: February 28, 2010, 10:12:35 pm »

I use 2.0.rc1 on Generic built on Fri Feb 12 22:56:32 CET 2010.

My SIP provider told me that they do not override caller id or block caller id set by me. I am their contract based client, and as stated in contract, I can use any number that I own.

But I can not make calls when I set caller id via web GUI (usually I enter my mobile number: "Matija <0038640xxxxxx>"). I always get this:

 == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Executing [9040256416@SIP-PHONE-10732834754b89594f977f8:1] NoOp("SIP/101-0000000f", "Direct dialing via SIP uri : 9040256416@10.0.0.109") in new stack
    -- Executing [9040256416@SIP-PHONE-10732834754b89594f977f8:2] Dial("SIP/101-0000000f", "SIP/9040256416@10.0.0.109,,To") in new stack
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Called 9040256416@10.0.0.109
    -- Got SIP response 482 "Loop Detected" back from 10.0.0.109
    -- Now forwarding SIP/101-0000000f to 'Local/9040256416@public-direct-dial' (thanks to SIP/10.0.0.109-00000010)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/101-0000000f' status is 'CHANUNAVAIL'

Any idea?

All the best,

Matija
Logged
ajitam
Newbie
*

Karma: 0
Posts: 3


View Profile
« Reply #10 on: March 02, 2010, 12:32:39 pm »

I forgot to mention that if I don't set Caller ID, it works perfectly.

Matija
Logged
kobold
Newbie
*

Karma: 0
Posts: 6


View Profile
« Reply #11 on: March 03, 2010, 07:55:57 am »

I also don't get an Caller ID. My caller is set to: kobold <0031xxxxxxxxx>
I use voipcheap for outgoing calls and registered my phonenumber. When I use voipcheap's own tool it works but not with Askozia.
Logged
kobold
Newbie
*

Karma: 0
Posts: 6


View Profile
« Reply #12 on: March 05, 2010, 03:12:19 pm »

I did a factory default reset and now it works. However I had to enter the same callarID at the provider configuration as I did at the phone configuration.
Logged
Pages: [1]
  Print  
 
Jump to:  

Powered by MySQL Powered by PHP Powered by SMF 1.1.11 | SMF © 2006-2009, Simple Machines LLC Valid XHTML 1.0! Valid CSS!
Page created in 0.133 seconds with 22 queries.