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NG-Studio-Development / csipsimple

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Outgooing call dong work #1783

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago 
What steps will reproduce the problem?
1. Asterisk server NAT , TCP connect
2. Registration success
3. Trying call , but nothing happening , see "Calling ...", afret 32 seconds 
call end. In asterisk i don't see call request .  
If i call from another SIP to device(incoming call) work normaly. 

Outgoing call not work , incoming work :) 

What is the expected output? What do you see instead?
Nothing happening. Afrer 32 seconds call ends . Outgoing not work 

What version of the product are you using? On what operating system?
Last yesterday downloaded . Android , samsung s2 4.0.3

Please provide any additional information below.
3Cx Phone work correctly . 
if you need , i can send you SIP requests and all additionals information

Original issue reported on code.google.com by dim123di...@gmail.com on 14 Jun 2012 at 7:27

GoogleCodeExporter commented 9 years ago 
Yes I would be interested by application / sip logs.
For application logs see the HowToCollectLogs wiki page : 
http://code.google.com/p/csipsimple/wiki/HowToCollectLogs?wl=en

Also could you confirm that to use TCP you used the expert account wizard and 
choose the transport mode to TCP here ? Because using basic wizard it's by 
default UDP.

Original comment by r3gis...@gmail.com on 14 Jun 2012 at 11:41

GoogleCodeExporter commented 9 years ago 
ok , may be , you have a skype or msn ?
and i try test app on my samgung s2 , because i don't like 3CX  :)  and
think that you app is more better .

--Also could you confirm that to use TCP you used the expert account wizard
and choose the transport mode to TCP here ? Because using basic wizard it's
by default UDP.
i don't know why but my UDP connection not work , my Asterisk and local
provider is over NAT, may be local provider block some udp traffic ...

now i have samsung s2 installed 3CX and samsung tab (android 3,2) 3CX both
worked correctly
and nokia e52 - symbian work correctly

And you app , register to SIP server , and after outgoing call not work and
incoming works

Dima

Original comment by dim123di...@gmail.com on 14 Jun 2012 at 11:57

GoogleCodeExporter commented 9 years ago 
Well, 3CX application is based on the same sip stack than csipsimple, but they 
are just not opensource...

So it's probably something with configuration. 

About UDP, most of the time, it's something with packet size. Upper limit of 
UDP packets is sometimes low, which made big UDP packet not going through.

To send me the logs, you can use the tool described in the wiki page, it sends 
a mail to my email address by default. Just put the information of the issue 
number.

Also, another doubt I have... when you say last... is it latest from android 
market or real latest one => http://nightlies.csipsimple.com/trunk/ ?

Original comment by r3gis...@gmail.com on 14 Jun 2012 at 12:17

GoogleCodeExporter commented 9 years ago 
get last version from trunks
and

log

<--- SIP read from TCP:193.150.17.220:37049 --->
REGISTER sip:192.168.5.155:5060;transport=tcp;lr SIP/2.0
Via: SIP/2.0/TCP 193.150.17.220:37049
;rport;branch=z9hG4bKPjzwAGTyGjOoMJIIMNAD1TP.EVgNY79pQ6
Max-Forwards: 70
From: "200" <sip:200@192.168.5.155:5060

To: "200" <sip:200@192.168.5.155:5060>
Call-ID: .9wES0evdT4CM6GFYOdS7BouR1iGCdWd
CSeq: 59979 REGISTER
User-Agent: CSipSimple r1616 / GT-I9100-15
Contact: "200" <sip:200@192.168.1.103:38593;transport=TCP;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 193.150.17.220:37049 (NAT)

<--- Transmitting (NAT) to 193.150.17.220:37049 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 193.150.17.220:37049
;branch=z9hG4bKPjzwAGTyGjOoMJIIMNAD1TP.EVgNY79pQ6;received=193.150.17.220;rport=
37049
From: "200" <sip:200@192.168.5.155:5060

To: "200" <sip:200@192.168.5.155:5060>;tag=as3f465618
Call-ID: .9wES0evdT4CM6GFYOdS7BouR1iGCdWd
CSeq: 59979 REGISTER
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e7ac61b"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '.9wES0evdT4CM6GFYOdS7BouR1iGCdWd' in
32000 ms (Method: REGISTER)

<--- SIP read from TCP:193.150.17.220:37049 --->
REGISTER sip:192.168.5.155:5060;transport=tcp;lr SIP/2.0
Via: SIP/2.0/TCP 193.150.17.220:37049
;rport;branch=z9hG4bKPjy6BB7JFZE4WIOOuQ1MVBG6ndWEd0JT4q
Max-Forwards: 70
From: "200" <sip:200@192.168.5.155:5060

To: "200" <sip:200@192.168.5.155:5060>
Call-ID: .9wES0evdT4CM6GFYOdS7BouR1iGCdWd
CSeq: 59980 REGISTER
User-Agent: CSipSimple r1616 / GT-I9100-15
Contact: "200" <sip:200@192.168.1.103:38593;transport=TCP;ob>
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Authorization: Digest username="200", realm="asterisk", nonce="3e7ac61b",
uri="sip:217.159.130.82;transport=tcp;lr",
response="e0b19876cfa9ea2bf12216dc4acdfcfa", algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 193.150.17.220:37049 (NAT)

<--- Transmitting (NAT) to 193.150.17.220:37049 --->
SIP/2.0 200 OKTo: "200" <sip:200@192.168.5.155:5060>;tag=as3f465618
Call-ID: .9wES0evdT4CM6GFYOdS7BouR1iGCdWd
CSeq: 59980 REGISTER
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Expires: 900
Contact: <sip:200@192.168.1.103:38593;transport=TCP;ob>;expires=900
Date: Thu, 14 Jun 2012 13:45:32 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '.9wES0evdT4CM6GFYOdS7BouR1iGCdWd' in
32000 ms (Method: REGISTER)

<--- SIP read from TCP:193.150.17.220:37049 --->
REGISTER sip:192.168.5.155:5060;transport=tcp;lr SIP/2.0
Via: SIP/2.0/TCP 193.150.17.220:37049
;rport;branch=z9hG4bKPjZClSSHeMr9LucpK-EqxiuWz.e4wnaPW8
Max-Forwards: 70
From: "200" <sip:200@192.168.5.155:5060

To: "200" <sip:200@192.168.5.155:5060>
Call-ID: .9wES0evdT4CM6GFYOdS7BouR1iGCdWd
CSeq: 59981 REGISTER
Authorization: Digest username="200", realm="asterisk", nonce="3e7ac61b",
uri="sip:217.159.130.82;transport=tcp;lr",
response="e0b19876cfa9ea2bf12216dc4acdfcfa", algorithm=MD5
User-Agent: CSipSimple r1616 / GT-I9100-15
Contact: <sip:200@193.150.17.220:37049;transport=tcp;ob>
Contact: "200" <sip:200@192.168.1.103:38593;transport=TCP;ob>;expires=0
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 193.150.17.220:37049 (NAT)
[Jun 14 16:45:32] NOTICE[3099]: chan_sip.c:14440 check_auth: Correct auth,
but based on stale nonce received from '"200" <sip:200@192.168.5.155:5060

<--- Transmitting (NAT) to 193.150.17.220:37049 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 193.150.17.220:37049
;branch=z9hG4bKPjZClSSHeMr9LucpK-EqxiuWz.e4wnaPW8;received=193.150.17.220;rport=
37049
From: "200" <sip:200@192.168.5.155:5060

To: "200" <sip:200@192.168.5.155:5060>;tag=as3f465618
Call-ID: .9wES0evdT4CM6GFYOdS7BouR1iGCdWd
CSeq: 59981 REGISTER
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="258c8ff1",
stale=true
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '.9wES0evdT4CM6GFYOdS7BouR1iGCdWd' in
32000 ms (Method: REGISTER)

<--- SIP read from TCP:193.150.17.220:37049 --->
REGISTER sip:192.168.5.155:5060;transport=tcp;lr SIP/2.0
Via: SIP/2.0/TCP 193.150.17.220:37049
;rport;branch=z9hG4bKPj3lLfLMEufXcvlinbzOGUq-aNlqX-xcMO
Max-Forwards: 70
From: "200" <sip:200@192.168.5.155:5060

To: "200" <sip:200@192.168.5.155:5060>
Call-ID: .9wES0evdT4CM6GFYOdS7BouR1iGCdWd
CSeq: 59982 REGISTER
User-Agent: CSipSimple r1616 / GT-I91Contact: <sip:200@193.150.17.220:37049
;transport=tcp;ob>
Contact: "200" <sip:200@192.168.1.103:38593;transport=TCP;ob>;expires=0
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Authorization: Digest username="200", realm="asterisk", nonce="258c8ff1",
uri="sip:217.159.130.82;transport=tcp;lr",
response="83e868aba2ee403c10f86e715645780a", algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 193.150.17.220:37049 (NAT)

<--- Transmitting (NAT) to 193.150.17.220:37049 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 193.150.17.220:37049
;branch=z9hG4bKPj3lLfLMEufXcvlinbzOGUq-aNlqX-xcMO;received=193.150.17.220;rport=
37049
From: "200" <sip:200@192.168.5.155:5060

To: "200" <sip:200@192.168.5.155:5060>;tag=as3f465618
Call-ID: .9wES0evdT4CM6GFYOdS7BouR1iGCdWd
CSeq: 59982 REGISTER
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Expires: 900
Contact: <sip:200@193.150.17.220:37049;transport=tcp;ob>;expires=900
Date: Thu, 14 Jun 2012 13:45:33 GMT
Content-Length: 0

00-15

Via: SIP/2.0/TCP 193.150.17.220:37049
;branch=z9hG4bKPjy6BB7JFZE4WIOOuQ1MVBG6ndWEd0JT4q;received=193.150.17.220;rport=
37049
From: "200" <sip:200@192.168.5.155:5060

Original comment by dim123di...@gmail.com on 14 Jun 2012 at 1:50

GoogleCodeExporter commented 9 years ago 
Well at least registration works properly (there is even contact rewrite that 
does the work to change the registration port)... but no trace of the sip 
invite as you said.

Could you follow the instruction of the HowToCollectLogs wiki page 
(http://code.google.com/p/csipsimple/wiki/HowToCollectLogs?wl=en) so that I can 
see how it goes from sip client side? Maybe there is some clue on why the 
INVITE packet is not sent to the asterisk server.

Also, did you configured the "sip proxy" field in the expert account mode? 
Even if asterisk is not strictly a proxy, as used to place sip calls, it's good 
to set it as proxy for your outbound calls for this account. (with the same 
sip:xxx.xxx.xxx.xxx than the one set for the sip registrar).

Original comment by r3gis...@gmail.com on 14 Jun 2012 at 8:13

GoogleCodeExporter commented 9 years ago 
Yes :)
when i put proxi , outgoing call work , thx. :)
i hope in new version you fix this bug and make many people happy :)

Dima

Original comment by dim123di...@gmail.com on 15 Jun 2012 at 3:47

GoogleCodeExporter commented 9 years ago 
Lol, So ... it's not a bug ! ;)

It's just a wrong configuration.

If you use expert mode you should take care of settings. The basic wizard 
*automatically* set both proxy and registrar. So if you used the basic wizard 
first, and then switched to expert mode later (by long press the basic account 
row + choose wizard and choose the expert one), you never encounter this 
problem.

In your case, it's normal that the configuration in expert mode must contains 
the sip proxy field.
As it's not the case of all the configurations however, and that the expert 
mode allows to address *all* possible configurations, the proxy field can be 
leaved blank. And that's absolutely not a bug, but a possible configuration 
that address different needs than yours.
Even other people, could have a sip proxy that is not the asterisk in the 
middle and could address sip to sip calls leaving asterisk dedicated to gateway 
to pstn network.

Original comment by r3gis...@gmail.com on 15 Jun 2012 at 8:22

GoogleCodeExporter commented 9 years ago 
may be this is not bug ,  but  i have configured some apps like 3CX and
nokia e52 sip client . bout of them had same configuration .
in you issues tasks list is same task . i think for better usability you
must accept this task like bug :) ( nothing personal , i just try help you
to make app better  )
in expert mode help said that this option is OPTIONAL .

one thing , i am not expert off SIP . but i have one asterisk server in
Estonia and other one on Spain , just for my calls . In asterisk and other
apps i see ulaw , alaw , gsm names . may be you can put description of
codecs when press long to codec , for dump users like me :)

Dima

Original comment by dim123di...@gmail.com on 15 Jun 2012 at 10:57

GoogleCodeExporter commented 9 years ago 
But what would be the task?
The proxy is actually optional !

I don't know what I could fix here.
Some users has a configuration where proxy is not needed or differs from 
registrar.
Do you propose to break things for these users? 

That's not the aim of the project. I try to leave no users behind as an 
opensource project... but if you think the closed source product approach that 
don't really cares about users but about market share and how to address more 
efficiently mass, CSipSimple is not the right app to use. I believe in the fact 
leaving users free and spending time explaining / sharing how things works is 
part of the goals of the project. For a free as in speech project users are not 
consumers but people that like to learn new things and understand. Some could 
study and contribute source code, some could write docs,some could share help 
on forum,blogs etc.

For codecs I take the point.It's a good idea : I often get questions on ulaw 
(users don't know it is the same than pcmu / g711u). It could also give 
indication on the bandwidth it uses, quality and when to use each codec.
Thanks for the feedback.

Original comment by r3gis...@gmail.com on 16 Jun 2012 at 5:15

GoogleCodeExporter commented 9 years ago 
ok i understand about proxy , sorry may be my mistake, i tried some apps
and there is another mind on proxy settings like - if checked proxy field
use proxy configuration in you app settings as you said proxy field is not
optional.

One thing more, using and all thing is very ok , like calls and log , but
in your app is possible edit number in Log screen or some another way?

Dima

Original comment by dim123di...@gmail.com on 17 Jun 2012 at 1:26

GoogleCodeExporter commented 9 years ago 
Long press the log row and press the little dialpad icon on the bottom.
You'll switch the text dialer and you can edit here before call.

Original comment by r3gis...@gmail.com on 17 Jun 2012 at 1:43