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alexchengalan / csipsimple

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My display name (number) is not tranmeted #131

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago 
What steps will reproduce the problem?
1. Normal CALL to an other phone
2.
3.

What is the expected output? What do you see instead?

I hop seen ny Diplay name (number) on the called phone, but it's only "unknow 
number" or nothing !

What version of the product are you using? On what operating system?
CSipSimple v0.00-12
Android 2.2
HTC Desire

Please provide any additional information below.

the voici working perfectly, just without Caller ID
my Acount id is 0041********66<sip:sip.rynga.com> , With or without space )

The Caller ID works with other Voip android application

Original issue reported on code.google.com by florian....@gmail.com on 5 Aug 2010 at 6:35

GoogleCodeExporter commented 9 years ago 
Will be fixed in next release.
Was fixed with the issue 30 .
To be complete, you can already use the expert account to fix your issue (with 
0.00-12).
Just change your account into an expert account.
And change account id to "DisplayName <sip:0041********66@sip.rynga.com>" (the 
account id you mentioned is not a valid account id in SIP)

With this configuration the other phone will see "DisplayName" as callerId. 
Issue 30 add a new wizard that simplifies the way to enter the CallerID.
At any time you can switch the account wizard for an account by pressing menu > 
Choose wizard on the edit account view.

Original comment by r3gis...@gmail.com on 5 Aug 2010 at 8:22

GoogleCodeExporter commented 9 years ago 
Sorry ,
I was not clear, and my Account ip was not good in my description !
my Real Account id is 
**0041******66<sip:**mylogin-who-is-not-a-number**@sip.rynga.com>

But the other phone don't see "DisplayName" as caller ID, only "Anonymous"

Original comment by florian....@gmail.com on 5 Aug 2010 at 9:14

GoogleCodeExporter commented 9 years ago 
Ok, so I reopen the issue.
Strange. 
I assume your sip provider doesn't rewrite your contact id since it works with 
other sip clients...
Is that possible for you to test with the 0.00-12-07 available here, on 
googlecode and sending me logs using logcat?
It can help me to know what is sent in the SIP From header.

Original comment by r3gis...@gmail.com on 5 Aug 2010 at 10:11

GoogleCodeExporter commented 9 years ago 
oki, I have "record" the log, from the start of the application to the hang up.
The call works perfect, but one more time , without Call ID !

Yes If I use SIPDROID and put the **0041******66 into the good place, the Call 
ID is delivered.

Original comment by florian....@gmail.com on 5 Aug 2010 at 10:54

Attachments:

GoogleCodeExporter commented 9 years ago 
Sounds that the log has been recorded with the version from the market which is 
not verbose with sip packets.
Can you :
* Uninstall the version installed from the market
* Download the version available here :
http://code.google.com/p/csipsimple/downloads/detail?name=CSipSimple_0.00-12-07.
apk
* Install it (if you have never installed an application from an apk, you may 
enable unsinged app from settings > application > developement) (or if you have 
the sdk installed on your pc adb install path_to_the_apk.apk)

Then re-run a call and send me your logs :)
Thanks in advance.

(Just to be more precise about why I need your log : I don't reproduce on my 
accounts and callerID is well displayed on remote phone. There is probably 
something special I don't well manage in some configuration)

P.S. : you can directly send me your logs by mail, better for the privacy of 
your infos (even if logs doesn't show passwords, they show your account uri).

Original comment by r3gis...@gmail.com on 6 Aug 2010 at 7:08

GoogleCodeExporter commented 9 years ago 
Should be ok with latest version. Guess was something related to issue 30.
If not fixed in last release, please send me logs so that I can see how contact 
id is transmitted to your server.
Maybe we should support another header to get it works with your provider. 
However in most case the standard method (passing display name in contact 
header field) is the best one.

Original comment by r3gis...@gmail.com on 17 Oct 2010 at 10:57

GoogleCodeExporter commented 9 years ago 
I've got the same problem.

HTC Desire
CSipSimple 0.00-15
Voipbuster

I provided a caller-id (my mobile number), but is not received by the called 
person. Tried both with Advanced wizard and Expert wizard.
With Sipdroid it did work.

Original comment by matthias...@gmail.com on 28 Oct 2010 at 12:20

GoogleCodeExporter commented 9 years ago 
Ok, probably due to the fact the contact id is re-written by pjsip. I've to 
find a workaround to force pjsip to use the good account id.
Force contact could help, however that's not a good solution since it will not 
resolve your public ip correctly.

I'll also try to add a better way to change globally a default caller id (if 
not specified by a wizard).

Original comment by r3gis...@gmail.com on 29 Oct 2010 at 6:08

GoogleCodeExporter commented 9 years ago 
[deleted comment]
GoogleCodeExporter commented 9 years ago 
Hi

I had the same issue with Rynga using csipsimple. I am now using pbxes.org as 
intermediate "layer". This didn't solve the issue initially but when I changed 
the trunk settings to the following:

username: 0041xxxxxxxxx (my phone number verified with Rynga Desktop client)
password: password:username (Rynga username and password both in password 
field!)

...it worked! I didn't try it without pbxes.org as I use their dial plans as 
well.

Maybe that helps, this is probably the same issue with all Betamax providers.

Original comment by banc%swi...@gtempaccount.com on 1 Dec 2010 at 9:12

GoogleCodeExporter commented 9 years ago 
Please fix the caller id issue as soon as possible, cant wait to uninstall 
SIPDroid from my phone.

The clarity of csipsimple is so good.

Original comment by heehaw...@gmail.com on 5 Dec 2010 at 1:40

GoogleCodeExporter commented 9 years ago 
It may be a fix for the Betamax and clones which I just discovered today - 
sorry if it's not pertaining directly to CSipSimple.

I had two different problems - the caller ID was not displayed at all AND when 
I managed to get it displayed, it had two country prefixes (+420 for Czech 
Republic, my country of residence, and +40 for Romania, which was part of the 
initial Caller ID).

Do as in comment #10 and specify the Caller ID in the "username" field on 
PBXes.org; then put the "password:username" thing in the password field.
Then go to the VoIP site (12voip, rynga etc - I'm using 12voip), check your 
profile and make sure the proper country (matching the Caller ID) is set up.
Then start the PC dialer that the provider offered (the software you install on 
the PC) and _validate_ your VoIP number as a valid caller ID following their 
instructions - on 12voip, this means going to File - Your Personal Profile.. - 
bottom left corner; then you'll be called on the number you specify and be 
given a 4-digit PIN for validation).
Finally check the country listed in your profile _in the dialer_ is also 
correct, i.e. it matches _both_ the country listed in your web profile _and_ 
the country where your caller ID is originating from. 

You can also check in the "Call Monitor" section on PBXes.org if your Caller ID 
gets sent properly from the trunk or not. If it does, yet you don't see the 
right value, chances are you may have one or more of the above issues (wrong 
country in either place, unconfirmed number, invalid username/password setup).

Once you have checked all the above, make a test call. I didn't need to make 
any specific setting in the CSipSimple apart from username and password in the 
PBXes wizard and it is working fine. Just to be on the safe side, I've put the 
same Caller ID in the trunk and in the extension on PBXes.org.

Original comment by adrian.b...@gmail.com on 9 Dec 2010 at 1:03

GoogleCodeExporter commented 9 years ago 
The problem  persists with the new version 0.00-16 r410
and the "password" triks doesn't work without PBXes.org

Original comment by florian....@gmail.com on 10 Dec 2010 at 8:17

GoogleCodeExporter commented 9 years ago 
It seems that for Betamax providers, in the expert wizard you can set account 
id to "DisplayName <sip:0041********66@sip.rynga.com>" where you set 
DisplayName as you want and the '0041********66' that should be the sip login 
as the phone number.
I tried it with Smslisto and seems ok.

Original comment by Gianluca...@gmail.com on 10 Dec 2010 at 12:43

GoogleCodeExporter commented 9 years ago 
I try the comment n° 14 And it's working !

Thanks a lot !
But this is and it should be change.

Original comment by florian....@gmail.com on 10 Dec 2010 at 2:22

GoogleCodeExporter commented 9 years ago 
I had exactly the same problem... No Caller ID using voipraider.

But with sipdroid, the ID  worked ok.

To resolve this issue I've tried comment #14 and worked. (and my user name in 
voipraider is not a number!)

Original comment by moliv.m...@gmail.com on 16 Dec 2010 at 1:20

GoogleCodeExporter commented 9 years ago 
@Gianluca: could you detail what settings you used to configure your betamax 
account. I am having a hard time configuring the expert mode. I personaly use 
webcalldirect. I am not sure about the following settings: 
Registration URI: sip:sip.webcalldirect.com[:5060]
Realm: *
Then I did not change the parameters below Data (password)

Thanks for the help.

Original comment by christia...@gmail.com on 17 Dec 2010 at 9:09

GoogleCodeExporter commented 9 years ago 
The fancier way to select Caller Id will land in next version. As it's not easy 
to set for each account and can be confusing with "Display name" for the 
account (which is only used for display in csipsimple), it is deported in 
global settings as a default caller id option.

It will be possible to set it individually for each account, transforming your 
account into an advanced or an expert account. (In advanced there is a field 
done for that and for expert you have to follow Gianluca instruction in comment 
#14).

Original comment by r3gis...@gmail.com on 29 Jan 2011 at 9:41

GoogleCodeExporter commented 9 years ago 
Since last update, I am unable to place an outgoing call with wifi. It only 
works with 3G. 
And caller id is still not forwarded. 
I tried writing my number in the 'Caller ID' field in advanced mode -> still 
anonymous calls
I tried writing it in Call options, Caller ID -> still anonymous calls
So I am sorry to say that it is unfortunately not solved yet!

Original comment by christia...@gmail.com on 31 Jan 2011 at 1:59

GoogleCodeExporter commented 9 years ago 
Can you send me logs. See HowToCollectLogs.

Original comment by r3gis...@gmail.com on 31 Jan 2011 at 2:40

GoogleCodeExporter commented 9 years ago 
I have sent you the logs. 
I had very strange behaviours side of the app. 
First time with wifi: call did not work
Then it worked, but when I switched off the call, the receiving phone kept on 
ringing... 
Then it didn't work (this is the only part I have logs for so far)
Yet never was the caller id forwarded. 
I will try to collect more logs.

Original comment by christia...@gmail.com on 31 Jan 2011 at 3:08

GoogleCodeExporter commented 9 years ago 
I have sent you a 2nd log: 
with 3G call went through. Receiving phone correctly stopped ringing when I 
gave up the call on the calling phone. Caller ID not forwarded. 
I then switched to wifi. Call did not go through.

Original comment by christia...@gmail.com on 31 Jan 2011 at 3:27

GoogleCodeExporter commented 9 years ago

Original comment by r3gis...@gmail.com on 2 Feb 2011 at 10:20

GoogleCodeExporter commented 9 years ago 
Issue 641 has been merged into this issue.

Original comment by r3gis...@gmail.com on 2 Feb 2011 at 10:20

GoogleCodeExporter commented 9 years ago 
This issue was closed by revision r644.

Original comment by r3gis...@gmail.com on 15 Feb 2011 at 3:52

GoogleCodeExporter commented 9 years ago 
New wizard for betamax users is now included. Available in "World wide 
provider" section of wizards (just bellow the generics wizard section).

Features :
 * Automatically remove ICE cause betamax servers does not support it
 * Balance inquery (tested on 12voip, let me know how it goes with others betamax like providers).
 * Add the stun server of the betamax clone.
 * Allow to set caller ID (as betamax clones does not use the standard way to do that, it has to be done in the wizard ! ). (p.s. I didn't try with a + sign inside but I'd advise you to not use + sign in caller id).

If you don't find your betamax clone let me know (I need name, sip registrar 
and stun if available)

Available in revision 644 => http://nightlies.csipsimple.com/trunk/

Original comment by r3gis...@gmail.com on 15 Feb 2011 at 4:00

GoogleCodeExporter commented 9 years ago 
[deleted comment]
GoogleCodeExporter commented 9 years ago 
[deleted comment]
GoogleCodeExporter commented 9 years ago 
I'm having the same issue with Google Voice using pbxes.org. I updated the 
Gtalk trunk with the password:user trick and no dice. The issue does not happen 
with sipdroid or registering to pbxes.org with x-lite softphone.

Original comment by drae...@gmail.com on 18 Feb 2011 at 12:34

GoogleCodeExporter commented 9 years ago 
please add Voipzoom to betamax list in CSipSimple

Original comment by desantis...@gmail.com on 12 Jun 2012 at 2:56

GoogleCodeExporter commented 9 years ago 
Added in r1615. Will be produced tonight in nightly builds. Thanks for the 
report.

Original comment by r3gis...@gmail.com on 12 Jun 2012 at 3:44