No ringing when calling from text dialer

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What steps will reproduce the problem?
1. calling from text dialer

What is the expected output? What do you see instead?
The outgoing call is ok but on csipsimple I don't listen ringing

What version of the product are you using? On what operating system?
011 on HTC Legend 2.1

Please provide any additional information below.
Great improvements in the new version.
For me is very usefull in the availability profile the option "Only for 
outgoing"
No more freeze but sometimes the app is slow closing a call

Thanks, great job

Original issue reported on code.google.com by Gianluca...@gmail.com on 26 May 2010 at 5:30

[GoogleCodeExporter]

If you are calling another sip client, the callee should answer with Ringing 
state.
Once CSipSimple receive this response it starts ringing (I have to check but I 
think
it's the default behavior of the native stack pjsip on which csipsimple is 
based).

By the way, many sip client doesn't respond automatically with the ringing state
(that's also the case if there is no client registered to receive your call).

So can you precise the callee SIP client used and whether there is a client
registered on the other side?
Thanks in advance.

P.S. : for the "Only for outgoing option", it was already present before but 
was less
clear (it was in Network section). In one of the next release, at the first
application start, (before the add account screen) the "Easy configuration" 
screen
will be shown. Then, it will be still possible to tweak it using Network> Use 
WIFI/3G
for outgoing/incoming.

Original comment by r3gis...@gmail.com on 27 May 2010 at 11:22

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The other sip client, was Siemens Gigaset C450 IP registered at 
sip.smslisto.com as
was registered csipsimple.
I tried to call the C450 IP using others windows sip client or my nokia e65, the
client start ringing regularly.

Sometimes when I close a call, I think particularly when the call duration is 
longer
than 7 or 8 minutes, the app takes more than 6 seconds to close the call and to
change the sreen, It depends on my phone?

Thanks

Original comment by Gianluca...@gmail.com on 27 May 2010 at 12:50

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Ok for the first issue, i have to investigate it.

For the delay after 7 or 8 minutes I don't think it depends on your phone. I 
didn't
yet check if there is no leak (/it could also be link to the CPU locker or 
something
like that). I'll also check it.

Thanks for your tests.

Original comment by r3gis...@gmail.com on 28 May 2010 at 7:43

• Changed state: Accepted

[GoogleCodeExporter]

Ringback is implemented in latest release (0.00-12)

Can you confirm this version solve the ringback issue?

Original comment by r3gis...@gmail.com on 10 Jul 2010 at 4:41

• Changed state: Fixed

[GoogleCodeExporter]

I have the same problem with the new version, no ringback :-(
I don't use the expert configuration, maybe I should?

Thanks

Original comment by Gianluca...@gmail.com on 10 Jul 2010 at 10:41

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I don't think so. All wizards create the same account type in fact. It's just 
wrappers that simplify or not the configuration but it doesn't add core 
features.

I'll test on my side with other callee clients and sip provider to see if I can 
reproduce. I was confident in my fix cause I simply use a well tested 
implementation of the ringback for pjsip but there is probably something I miss 
in the android integration.

Original comment by r3gis...@gmail.com on 11 Jul 2010 at 7:09

• Changed state: Started

[GoogleCodeExporter]

Tested with internal extensions (softphone and Linksys ATA) as well as POTS 
calls via our SIP switch. In all cases, ringing was correct.

Can we close this?

Original comment by dc3de...@gmail.com on 6 Aug 2010 at 1:54

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There is probably something we miss with the configuration to be able to 
reproduce. (Maybe the behavior of the sip server).

As things has been improved on call management and has still to be improved we 
should left this bug as open until Gianluca say us it's ok with his 
configuration.

It's possible that a commit (already done or future), that makes things more 
stable on audio layer / call management, fixes this one. I hope Gianluca will 
say us when he will observe things are going better with his configuration. 

Original comment by r3gis...@gmail.com on 6 Aug 2010 at 2:02

[GoogleCodeExporter]

OK, I understand. It's strange since in the end, the numeric dialer produces a 
SIP address anyway!

Original comment by dc3de...@gmail.com on 6 Aug 2010 at 2:07

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Probably linked to the fact that text dialer is commonly use to make a call to 
another domain. And probably in this case sip call is routed differently and 
sip server of Gianluca replies with a different flow. 
Maybe not directly linked to text dialer.

Original comment by r3gis...@gmail.com on 6 Aug 2010 at 2:13

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I Tried the last version and I can reproduce the problem.
The sip call is on the same domain.
Maybe an issue with Betamax providers, I have tried voipdiscount and smslisto 
particularly.
I would like to see this issue closed, but I see a great job on the app.
Thanks

Original comment by Gianluca...@gmail.com on 7 Aug 2010 at 8:38

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I wasn't sure if this was a problem or not, but yeah, it is ringing I think, 
but I don't hear it.  It DOES say "ringing" on the screen, but I have no way of 
knowing for sure unless someone answers... I'll email you a log!

Original comment by jerald...@gmail.com on 27 Sep 2010 at 2:27

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I'm wondering if this issue is still reproducible. There were something I fixed 
about the way it keep the sip stack up and running when making an call. Maybe 
it fixed that.
Let me know if you still reproduce it with latest nightly build so that I'll 
reopen this issue.

Original comment by r3gis...@gmail.com on 7 May 2011 at 3:42

• Changed state: Obsolete

[GoogleCodeExporter]

I tried r386 and I can reproduce the same behavior with SMSListo, a betamax 
clone.
This behavior applies only to SIP addresses that do not represent geographic 
numbers.

Original comment by Gianluca...@gmail.com on 8 May 2011 at 7:58

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In your case I think that's more something linked to the provider as you said.
In fact, csipsimple automatically configure "proxy" for most sip accounts. 
It means that all sip invite will be transmitted to your smslisto server that 
should transmit the request to other servers (the one of the relevant domain in 
your sip address).
Sometimes, sip servers that are mainly registrar has really few features as 
proxy. Some does not transmit the Invite request that is not their own domain 
at all - in this case you can have a direct hangup-, others transmit but does 
not send the < 200 states (ringing is a temporary state : 180) so that the 
client is not aware about what is happening on the other side.

So I think that in this case it's more about the fact the sip server of your 
provider does not proxy correctly. 
To solve that there is mainly two solutions :
 * The first one is to remove proxy field from configuration on this account. Sometimes it's not hurting but sometimes it lead to have calls going to this domain not working add all.
 * The second one is simply to create a "local account" using the local wizard, and when you want to make a call to another domain, to choose this local account instead of the account which aim is to gateway with the pstn network. In this case the local account will directly contact the server that is after the "@" add send an invite to this server. (Here this sip server have to allow other domains to call their sip clients but if it does things should go fine :) )

Original comment by r3gis...@gmail.com on 8 May 2011 at 10:31