Open GoogleCodeExporter opened 9 years ago
12:38:54.037 [MS105000] C:41978.1: No RTP packets were
received:remoteAddr=127.0.0.1:40670,extAddr=0.0.0.0:0,localAddr=127.0.0.1:7014
I think thats the cause.. is that the phone blocking the traffic? (maybe orange
locking it down)
Original comment by danceres...@googlemail.com
on 3 Aug 2009 at 11:40
I have the same issue.In wireshark is in log from my HTC hero
Proxy authentication required and i can not connect.But is runing perfectly
with IPHONE (any sip client)
Regards
Original comment by iphone.l...@gmail.com
on 20 Sep 2009 at 1:01
I am getting the same issue with G1. I have tried on both internal network
loops and
external loops the other phone rings and nothing. Also I cannot make a call to
my G1
from the other phone on the network. I am still going to monkey around with
3cx to
see if I can make this work.
Original comment by littlejo...@gmail.com
on 11 Oct 2009 at 4:02
I am not sure if this has anything to do with it but when I look at my System
logs on
3cx I notice the ip that is coming from my G1 is not the IP of wireless I am on
but
the Local-host address the phone is registering as 127.0.0.1:5060. Even when I
do
change to TCP It still registers as the local host not the IP it is connecting
from.
Does this seem right ?
Original comment by littlejo...@gmail.com
on 11 Oct 2009 at 5:51
Experimented with this more even when I connect from 3g it registers as
127.0.0.1:5060 so somehow 3cx is not see the true IP of the phone. I get my G1
to
work with Trixbox with no issues but with 3cx it is just not registering right.
Original comment by littlejo...@gmail.com
on 11 Oct 2009 at 6:22
I was hoping that this issue was going to go away once i updated my phone to the
latest MOD I am using but nope. Just to let you know I did try also with a
Un-moded
phone and it did not work either. Is there a way to force sipdroid to look at
the
twiland IP or RMNET IP. When I look at what is my IP it lists LO first and then
other
connections were trixbox just ignores LO it seems 3cx does not and I think that
is
where we are running into this issue. I have tried several registration
tricks like
xxx@xxx.xxx.xx.xxx in both the username field and the server field. when doing
that
it does not register at all. I think the only reason it is registering with
TCP is
3cx cannot see the other IP of the phone and falls back to having to register
with
TCP alone. I Hope to hear back from you all soon.
Original comment by littlejo...@gmail.com
on 14 Oct 2009 at 5:08
I have the same issue. Droid.
Original comment by paulh...@gmail.com
on 15 Nov 2009 at 11:18
I have not tested this on 1.18 yet to see if this issue has been resolved or
not but
I will run it tonight and update accordingly.
Original comment by littlejo...@gmail.com
on 16 Nov 2009 at 11:48
[deleted comment]
It seems to give the correct IP out now:
15:17:44.370 [CM503025]: Call(22309): Calling
Ext:Ext.259@[Dev:sip:259@10.0.0.33:5060]
15:17:44.370 [MS210004] C:22309.2:Offer provided. Connection(proxy mode):
10.0.0.100:7016(7017)
15:17:44.323 [CM503004]: Call(22309): Route 1:
Ext:Ext.259@[Dev:sip:259@10.0.0.33:5060]
15:17:44.323 [CM503010]: Making route(s) to <sip:259@10.0.0.100>
15:17:44.323 [MS210000] C:22309.1:Offer received. RTP connection:
127.0.0.1:21000(21001)
15:17:44.323 Remote SDP is set for legC:22309.1
15:17:44.323 [CM505001]: Ext.209: Device info: Device Not Identified: User
Agent not
matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent:
[Sipdroid/1.1.8 beta/HTC Hero] PBX contact:
[sip:209@10.0.0.100:5060;transport=TCP]
15:17:44.307 [CM503001]: Call(22309): Incoming call from Ext.209 to
<sip:259@10.0.0.100>
15:17:44.307 [CM500002]: Info on incoming INVITE:
INVITE sip:259@10.0.0.100 SIP/2.0
Via: SIP/2.0/TCP 127.0.0.1;rport=52527;branch=z9hG4bK95608;received=10.0.0.53
Max-Forwards: 70
Contact: <sip:209@127.0.0.1>
To: <sip:259@10.0.0.100>
From: <sip:209@10.0.0.100>;tag=z9hG4bK04387066
Call-ID: 114088798816@127.0.0.1
CSeq: 2 INVITE
Expires: 3600
Proxy-Authorization: Digest
username="209",realm="3CXPhoneSystem",nonce="414d535c01134f9610:8186f00efbe251fe
fd3f1c5447ac3b2a",uri="sip:259@10.0.0.100",algorithm=MD5,response="fcecf06db9f03
177251c921d42ded797"
User-Agent: Sipdroid/1.1.8 beta/HTC Hero
Content-Length: 0
No sound either side tho, Ill see if thats a 3cx thing..
Original comment by danceres...@googlemail.com
on 17 Nov 2009 at 3:20
This is a trace of a Software Sip Phone on the 3cx server:
15:21:07.259 [CM503007]: Call(22341): Device joined: sip:259@10.0.0.33:5060
15:21:07.259 [CM503007]: Call(22341): Device joined:
sip:209@10.0.0.130:54168;rinstance=c35f838b0077df24
15:21:07.259 [MS210007] C:22341.1:Answer provided. Connection(by pass mode):
10.0.0.33:16400(16401)
15:21:07.259 [MS210001] C:22341.2:Answer received. RTP connection[unsecure]:
10.0.0.33:16400(16401)
15:21:07.259 Remote SDP is set for legC:22341.2
15:21:07.259 [CM505001]: Ext.259: Device info: Device Not Identified: User
Agent not
matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent:
[Sipura/SPA921-4.1.10(b)] PBX contact: [sip:259@10.0.0.100:5060]
15:21:07.259 [CM503002]: Call(22341): Alerting sip:259@10.0.0.33:5060
15:21:05.009 [CM503025]: Call(22341): Calling
Ext:Ext.259@[Dev:sip:259@10.0.0.33:5060]
15:21:05.009 [MS210006] C:22341.2:Offer provided. Connection(by pass mode):
10.0.0.130:40006(40007)
15:21:04.962 [CM503004]: Call(22341): Route 1:
Ext:Ext.259@[Dev:sip:259@10.0.0.33:5060]
15:21:04.962 [CM503010]: Making route(s) to <sip:259@10.0.0.100:5060>
15:21:04.962 [MS210000] C:22341.1:Offer received. RTP connection:
10.0.0.130:40006(40007)
15:21:04.962 Remote SDP is set for legC:22341.1
15:21:04.962 [CM505001]: Ext.209: Device info: Device Not Identified: User
Agent not
matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent:
[3CXVoipPhone 4.0.9530.0] PBX contact: [sip:209@10.0.0.100:5060]
15:21:04.947 [CM503001]: Call(22341): Incoming call from Ext.209 to
<sip:259@10.0.0.100:5060>
15:21:04.947 [CM500002]: Info on incoming INVITE:
INVITE sip:259@10.0.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP
10.0.0.130:54168;branch=z9hG4bK-d8754z-0141ac556c63f666-1---d8754z-;rport=54168
Max-Forwards: 70
Contact: <sip:209@10.0.0.130:54168;rinstance=c35f838b0077df24>
To: <sip:259@10.0.0.100:5060>
From: "Dave"<sip:209@10.0.0.100:5060>;tag=8902a67e
Call-ID: YmE4ODY0M2UwYmU1OWNjODY5YzQ3ZTc5MTBhZDAxZmU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Proxy-Authorization: Digest
username="209",realm="3CXPhoneSystem",nonce="414d535c0113506083:aec68461b49454df
d4b18a382260ce96",uri="sip:259@10.0.0.100:5060",response="8b9040a84ff297b527c193
da89fc801f",algorithm=MD5
Supported: replaces
User-Agent: 3CXVoipPhone 4.0.9530.0
Content-Length: 0
Maybe of some help?
Original comment by danceres...@googlemail.com
on 17 Nov 2009 at 3:23
I need to reload my 3cx phone system which I will do tonight. I will comment
later
tonight if 1.18 fixes this issue.
Original comment by littlejo...@gmail.com
on 19 Nov 2009 at 1:16
Any Updates on this issue, I am having the same issue with 3cx server
Original comment by abedfa...@gmail.com
on 17 Dec 2009 at 3:03
As far as I can see from pieces of logs -- SipDroid do not provider correct IP
neither in INVITE's Contact header, nor in SDP. In case of INVITE 3CX is still
able
to answer, because 3CX SIP server sends a reply back to source address (see
rport/received params of Via), and thus to establish SIP session. But in case
of RTP
such trick doesn't work, because 3CX Media Server believes that 127.0.0.1 is a
real
destination and sends RTP to it.
Original comment by archie314@gmail.com
on 27 Jan 2010 at 2:16
Same issue for me as well. It will be great if this can be fixed.
Original comment by sheldon%...@gtempaccount.com
on 27 Jan 2010 at 8:52
I have a similar problem but not exactly the same. Nonetheless I am posting
here instead of creating a double :-)
SipDroid 1.5.2 beta HTC Incredible. Phone registers OK with 3CX using either
UDP or TCP over WLAN or 3G. Stun server selected stun.3cx.com/3478. Problem
occurs on WLAN behind a NAT router AND on 3G.
Call from another extension TO the SipDroid extension work FINE (WLAN and 3G).
Audio is fine, just perfect.
However, a call FROM SipDroid to another extension does not work: The call
completes, the other phone rings, and for about 2 seconds I near strange
squeaking sounds on SipDroid, then SipDroid goes back to the dial screen (with
the recently used list). Meanwhile, the Call In Progress notification continues
to appear at the top of the phone, and in the notification pulldown. It never
goes away. The only thing I can do is kill the SipDroid task.
Perhaps this is related, but in the 3CX admin console, SipDroid shows up
multiple times with various port numbers, where the other phones are always
using post 5060, 5061, 5062. The number of listed SIPDroid instances seems to
vary with usage.
Another possible clue: Ad reported by someone else, I'm getting this in the
Activity Log:
13:47:19.968 [MS105000] C:1.1: No RTP packets were
received:remoteAddr=70.167.219.234:21000,extAddr=0.0.0.0:0,localAddr=70.167.219.
233:7000
This was via WLAN with the Wireless router's WAN IP 70.167.219.234 and the 3CX
running on the machine at 70.167.219.233.
Original comment by dc3de...@gmail.com
on 23 Jun 2010 at 9:40
Attachments:
OK I got this working with SipDroid 1.5.2 (beta) and the 3CX V8 PABX/Proxy.
I've tested with the phone/SipDroid behind a WiFi router connected to our
public LAN (must NAT to get to the 3CX PABX), and via 3G.
The solution is in 3CX extension config: Turn off Supports Re-Invite on the
assigned extension (Edit Extension-Ext.xxx, Other tab, Extension Capabilities
section. I am no VoIP expert; I don't know WHY this works, but it does.
It is also necessary to use a STUN server.
The problem of SipDroid showing up multiple times on different ports is still
there (see my prev message) but it seems harmless.
Original comment by dc3de...@gmail.com
on 30 Jun 2010 at 2:23
Oh, I should mention that the above was done using UDP for both WiFi/NAT and 3G.
Original comment by dc3de...@gmail.com
on 30 Jun 2010 at 2:25
Original issue reported on code.google.com by
danceres...@googlemail.com
on 3 Aug 2009 at 9:06