#38 No ringing when calling from text dialer

[Anonymous]

Originally created by: Gianluca...@gmail.com
Originally owned by: r3gis...@gmail.com

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

[Anonymous]

Originally posted by: r3gis...@gmail.com

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.

[Anonymous]

Originally posted by: Gianluca...@gmail.com

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

[Anonymous]

Originally posted by: r3gis...@gmail.com

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.

Owner: r3gis.3R
Status: Accepted

[Anonymous]

Originally posted by: r3gis...@gmail.com

Ringback is implemented in latest release (0.00-12)

Can you confirm this version solve the ringback issue?

Status: Fixed

[Anonymous]

Originally posted by: Gianluca...@gmail.com

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

Thanks

[Anonymous]

Originally posted by: r3gis...@gmail.com

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.

Status: Started

[Anonymous]

Originally posted by: dc3de...@gmail.com

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?

[Anonymous]

Originally posted by: r3gis...@gmail.com

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.

[Anonymous]

Originally posted by: dc3de...@gmail.com

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

[Anonymous]

Originally posted by: r3gis...@gmail.com

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.

[Anonymous]

Originally posted by: Gianluca...@gmail.com

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

[Anonymous]

Originally posted by: jerald...@gmail.com

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!

[Anonymous]

Originally posted by: r3gis...@gmail.com

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.

Status: Obsolete

[Anonymous]

Originally posted by: Gianluca...@gmail.com

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.

[Anonymous]

Originally posted by: r3gis...@gmail.com

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 :) )