PBXes - Print: RE: SipDroid 2.2 does not work after upgrading my account to SOHO

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--- RE: SipDroid 2.2 does not work after upgrading my account to SOHO (http://www1.pbxes.com/forum/threadid.php?threadid=1301992438)

Posted by jakube on 05.04.2011 at 10:33:

traurig SipDroid 2.2 does not work after upgrading my account to SOHO

Hi!

i'm having big troubles with my SipDroid client since i've upgraded my account from free account to SOHO.

When i dial a number (doesn't matter if it's a local extension or external number over a trunk) the call is terminated after few seconds. the called party rings one time.

i tried re-installing the app, changing settings, disabling codecs, ..everything. all other sip clients are working fine.

Any help is appreciated!

Trace log:
Apr 5 09:25:29 VERBOSE[111124] logger.c:
<-- SIP read
Apr 5 09:25:29 VERBOSE[111124] logger.c: --- (0 headers 0 lines)Apr 5 09:25:29 VERBOSE[111124] logger.c: --- (0 headers 0 lines) Nat keepalive Apr 5 09:25:29 VERBOSE[111124] logger.c: --- (0 headers 0 lines) Nat keepalive ---
Apr 5 09:25:34 VERBOSE[111124] logger.c:
<-- SIP read
REGISTER sip:pbxes.org SIP/2.0
Via: SIP/2.0/UDP 46.74.179.80:42503;rport;branch=z9hG4bK81445
Max-Forwards: 70
To: <sip:jakube-102@pbxes.org>
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK19595174
Call-ID: 598483377912@46.74.179.80
CSeq: 1 REGISTER
Contact: <sip:jakube-102@46.74.179.80:42503;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.2 beta/HTC Vision
Content-Length: 0
Apr 5 09:25:34 VERBOSE[111124] logger.c: --- (11 headers 0 lines)Apr 5 09:25:34 VERBOSE[111124] logger.c: --- (11 headers 0 lines)---
Apr 5 09:25:34 VERBOSE[111124] logger.c: Using latest REGISTER request as basis request
Apr 5 09:25:34 VERBOSE[111124] logger.c: Transmitting (NAT)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 46.74.179.80:42503;branch=z9hG4bK81445;received=88.198.69.250;rport=59345
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK19595174
To: <sip:jakube-102@pbxes.org>
Call-ID: 598483377912@46.74.179.80
CSeq: 1 REGISTER
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:jakube-102@88.198.69.250:30856>
Content-Length: 0
---
Apr 5 09:25:34 VERBOSE[111124] logger.c: Transmitting (NAT)
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 46.74.179.80:42503;branch=z9hG4bK81445;received=88.198.69.250;rport=59345
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK19595174
To: <sip:jakube-102@pbxes.org>;tag=as05ca3cfd
Call-ID: 598483377912@46.74.179.80
CSeq: 1 REGISTER
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:jakube-102@88.198.69.250:30856>
WWW-Authenticate: Digest realm="pbxes.org", nonce="044091b8260ccc4e25680a104c890a040cab3dd3"
Content-Length: 0
---
Apr 5 09:25:34 VERBOSE[111124] logger.c: Scheduling destruction of call '598483377912@46.74.179.80' in 15000 ms
Apr 5 09:25:34 VERBOSE[111124] logger.c:
<-- SIP read
INVITE sip:&800123456@pbxes.org SIP/2.0
Via: SIP/2.0/UDP 46.74.179.80:42503;rport;branch=z9hG4bK52008
Max-Forwards: 70
P-src-ip: 46.74.179.80
To: <sip:&800123456@pbxes.org>
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK36803835
Call-ID: 874620873041@46.74.179.80
CSeq: 1 INVITE
Contact: <sip:jakube-102@46.74.179.80:42503;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.2 beta/HTC Vision
Content-Length: 388
Content-Type: application/sdp
v=0
o=jakube-102@pbxes.org 0 0 IN IP4 46.74.179.80
s=Session SIP/SDP
c=IN IP4 46.74.179.80
t=0 0
m=audio 21000 RTP/AVP 9 8 0 97 3 106 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:106 BV16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 21070 RTP/AVP 103
a=rtpmap:103 h263-1998/90000
Apr 5 09:25:34 VERBOSE[111124] logger.c: --- (13 headers 16 lines)Apr 5 09:25:34 VERBOSE[111124] logger.c: --- (13 headers 16 lines)---
Apr 5 09:25:34 VERBOSE[111124] logger.c: Using INVITE request as basis request - 874620873041@46.74.179.80
Apr 5 09:25:34 VERBOSE[111124] logger.c: Reliably Transmitting (NAT)
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 46.74.179.80:42503;branch=z9hG4bK52008;received=88.198.69.250;rport=59345
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK36803835
To: <sip:&800123456@pbxes.org>;tag=as316270df
Call-ID: 874620873041@46.74.179.80
CSeq: 1 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:&800123456@88.198.69.250:30856>
Proxy-Authenticate: Digest realm="pbxes.org", nonce="6cac87c650bff0857da7d91b20c7b712617ced7e"
Content-Length: 0
---
Apr 5 09:25:34 VERBOSE[111124] logger.c: Scheduling destruction of call '874620873041@46.74.179.80' in 15000 ms
Apr 5 09:25:34 VERBOSE[111124] logger.c: Found user 'jakube-102'
Apr 5 09:25:35 VERBOSE[111124] logger.c:
<-- SIP read
REGISTER sip:pbxes.org SIP/2.0
Via: SIP/2.0/UDP 46.74.179.80:42503;rport;branch=z9hG4bK66136
Max-Forwards: 70
P-src-ip: 46.74.179.80
To: <sip:jakube-102@pbxes.org>
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK19595174
Call-ID: 598483377912@46.74.179.80
CSeq: 2 REGISTER
Contact: <sip:jakube-102@46.74.179.80:42503;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.2 beta/HTC Vision
Authorization: Digest username="jakube-102", realm="pbxes.org", nonce="044091b8260ccc4e25680a104c890a040cab3dd3", uri="sip:pbxes.org", response="3f219b69172417e2a68a9b3f378b74da"
Apr 5 09:25:35 VERBOSE[111124] logger.c: --- (13 headers 0 lines)Apr 5 09:25:35 VERBOSE[111124] logger.c: --- (13 headers 0 lines)---
Apr 5 09:25:35 VERBOSE[111124] logger.c: Using latest REGISTER request as basis request
Apr 5 09:25:35 VERBOSE[111124] logger.c: Transmitting (NAT)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 46.74.179.80:42503;branch=z9hG4bK66136;received=88.198.69.250;rport=59345
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK19595174
To: <sip:jakube-102@pbxes.org>
Call-ID: 598483377912@46.74.179.80
CSeq: 2 REGISTER
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:jakube-102@88.198.69.250:30856>
Content-Length: 0
---
Apr 5 09:25:35 VERBOSE[111124] logger.c: -- Registered SIP 'jakube-102' expires 3600
Apr 5 09:25:35 VERBOSE[111124] logger.c: Transmitting (NAT)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 46.74.179.80:42503;branch=z9hG4bK66136;received=88.198.69.250;rport=59345
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK19595174
To: <sip:jakube-102@pbxes.org>;tag=as05ca3cfd
Call-ID: 598483377912@46.74.179.80
CSeq: 2 REGISTER
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 3600
Contact: <sip:jakube-102@46.74.179.80:42503;transport=udp>;expires=3600
ate: Tue, 05 Apr 2011 07:25:35 GMT
Content-Length: 0
---
Apr 5 09:25:35 VERBOSE[111124] logger.c: Scheduling destruction of call '598483377912@46.74.179.80' in 15000 ms
Apr 5 09:25:35 VERBOSE[111124] logger.c: Retransmitting #1 (NAT)
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 46.74.179.80:42503;branch=z9hG4bK52008;received=88.198.69.250;rport=59345
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK36803835
To: <sip:&800123456@pbxes.org>;tag=as316270df
Call-ID: 874620873041@46.74.179.80
CSeq: 1 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:&800123456@88.198.69.250:30856>
Proxy-Authenticate: Digest realm="pbxes.org", nonce="6cac87c650bff0857da7d91b20c7b712617ced7e"
Content-Length: 0
---
Apr 5 09:25:35 VERBOSE[111124] logger.c:
<-- SIP read
ACK sip:&800123456@pbxes.org SIP/2.0
Via: SIP/2.0/UDP 46.74.179.80:42503;rport;branch=z9hG4bK52008
Max-Forwards: 70
P-src-ip: 46.74.179.80
To: <sip:&800123456@pbxes.org>;tag=as316270df
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK36803835
Call-ID: 874620873041@46.74.179.80
CSeq: 1 ACK
User-Agent: Sipdroid/2.2 beta/HTC Vision
Content-Length: 0
Apr 5 09:25:35 VERBOSE[111124] logger.c: --- (10 headers 0 lines)Apr 5 09:25:35 VERBOSE[111124] logger.c: --- (10 headers 0 lines)---
Apr 5 09:25:35 VERBOSE[111124] logger.c:
<-- SIP read
INVITE sip:&800123456@pbxes.org SIP/2.0
Via: SIP/2.0/UDP 46.74.179.80:42503;rport;branch=z9hG4bK63902
Max-Forwards: 70
P-src-ip: 46.74.179.80
To: <sip:&800123456@pbxes.org>
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK36803835
Call-ID: 874620873041@46.74.179.80
CSeq: 2 INVITE
Contact: <sip:jakube-102@46.74.179.80:42503;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.2 beta/HTC Vision
Proxy-Authorization: Digest username="jakube-102", realm="pbxes.org", nonce="6cac87c650bff0857da7d91b20c7b712617ced7e", uri="sip:&800123456@pbxes.org", response="5c11fd8c912f094c4beb39a11db494bb"
Content-Length: 388
Content-Type: application/sdp
v=0
o=jakube-102@pbxes.org 0 0 IN IP4 46.74.179.80
s=Session SIP/SDP
c=IN IP4 46.74.179.80
t=0 0
m=audio 21000 RTP/AVP 9 8 0 97 3 106 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:106 BV16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 21070 RTP/AVP 103
a=rtpmap:103 h263-1998/90000
Apr 5 09:25:35 VERBOSE[111124] logger.c: --- (14 headers 16 lines)Apr 5 09:25:35 VERBOSE[111124] logger.c: --- (14 headers 16 lines)---
Apr 5 09:25:35 VERBOSE[111124] logger.c: Using INVITE request as basis request - 874620873041@46.74.179.80
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found user 'jakube-102'
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found peer 'jakube-102'
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found RTP audio format 9
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found RTP audio format 8
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found RTP audio format 0
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found RTP audio format 97
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found RTP audio format 3
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found RTP audio format 106
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found RTP audio format 101
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found RTP video format 103
Apr 5 09:25:35 VERBOSE[111124] logger.c: Peer audio RTP is at port 46.74.179.80:21000
Apr 5 09:25:35 VERBOSE[111124] logger.c: Peer video RTP is at port 46.74.179.80:21070
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found description format G722
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found description format PCMA
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found description format PCMU
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found description format speex
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found description format GSM
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found description format BV16
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found description format telephone-event
Apr 5 09:25:35 VERBOSE[111124] logger.c: Found description format h263-1998
VERBOSE[111124] logger.c: Capabilities: us - 0x18161e (gsm|ulaw|alaw|g726|speex|ilbc|g722|h263|h263p), peer - audio=0x120e (gsm|ulaw|alaw|speex|g722)/video=0x100000 (h263p), combined - 0x10120e (gsm|ulaw|alaw|speex|g722|h263p)
Apr 5 09:25:35 VERBOSE[111124] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Apr 5 09:25:35 VERBOSE[111124] logger.c: Looking for &800123456 in from-internal (domain pbxes.org)
Apr 5 09:25:35 VERBOSE[111124] logger.c: list_route: hop: <sip:jakube-102@46.74.179.80:42503;transport=udp>
Apr 5 09:25:35 VERBOSE[111124] logger.c: Transmitting (NAT)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 46.74.179.80:42503;branch=z9hG4bK63902;received=88.198.69.250;rport=59345
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK36803835
To: <sip:&800123456@pbxes.org>
Call-ID: 874620873041@46.74.179.80
CSeq: 2 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:&800123456@88.198.69.250:30856>
Content-Length: 0
---
Apr 5 09:25:36 VERBOSE[111124] logger.c:
<-- SIP read
ACK sip:&800123456@pbxes.org SIP/2.0
Via: SIP/2.0/UDP 46.74.179.80:42503;rport;branch=z9hG4bK52008
Max-Forwards: 70
P-src-ip: 46.74.179.80
To: <sip:&800123456@pbxes.org>;tag=as316270df
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK36803835
Call-ID: 874620873041@46.74.179.80
CSeq: 1 ACK
User-Agent: Sipdroid/2.2 beta/HTC Vision
Content-Length: 0
Apr 5 09:25:36 VERBOSE[111124] logger.c: --- (10 headers 0 lines)Apr 5 09:25:36 VERBOSE[111124] logger.c: --- (10 headers 0 lines)---
Apr 5 09:25:36 VERBOSE[107460] logger.c: We're at 88.198.69.250 port 38088
Apr 5 09:25:36 VERBOSE[107460] logger.c: Video is at 88.198.69.250 port 44318
Apr 5 09:25:36 VERBOSE[107460] logger.c: Adding codec 0x1000 (g722) to SDP
Apr 5 09:25:36 VERBOSE[107460] logger.c: Adding codec 0x4 (ulaw) to SDP
Apr 5 09:25:36 VERBOSE[107460] logger.c: Adding codec 0x8 (alaw) to SDP
Apr 5 09:25:36 VERBOSE[107460] logger.c: Adding codec 0x200 (speex) to SDP
Apr 5 09:25:36 VERBOSE[107460] logger.c: Adding codec 0x2 (gsm) to SDP
Apr 5 09:25:36 VERBOSE[107460] logger.c: Adding codec 0x100000 (h263p) to SDP
Apr 5 09:25:36 VERBOSE[107460] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Apr 5 09:25:36 VERBOSE[107460] logger.c: Transmitting (NAT)
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 46.74.179.80:42503;branch=z9hG4bK63902;received=88.198.69.250;rport=59345
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK36803835
To: <sip:&800123456@pbxes.org>;tag=as5052d79c
Call-ID: 874620873041@46.74.179.80
CSeq: 2 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:&800123456@88.198.69.250:30856>
Content-Type: application/sdp
Content-Length: 348
v=0
o=root 111115 111115 IN IP4 88.198.69.250
s=session
c=IN IP4 88.198.69.250
t=0 0
m=audio 38088 RTP/AVP 9 0 8 97 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 44318 RTP/AVP 103
a=rtpmap:103 h263-1998/90000
---
Apr 5 09:25:36 VERBOSE[107460] logger.c: -- Called 800123456@from-internal-cont/n
Apr 5 09:25:36 NOTICE[107460] chan_local.c: No such extension/context @from-internal creating local channel
Apr 5 09:25:36 NOTICE[107460] app_dial.c: Unable to create channel of type 'Local' (cause 0 - Unknown)
Apr 5 09:25:36 NOTICE[107460] chan_local.c: No such extension/context @from-internal creating local channel
Apr 5 09:25:36 NOTICE[107460] app_dial.c: Unable to create channel of type 'Local' (cause 0 - Unknown)
Apr 5 09:25:37 VERBOSE[111124] logger.c:
<-- SIP read
Apr 5 09:25:37 VERBOSE[111124] logger.c: --- (0 headers 0 lines)Apr 5 09:25:37 VERBOSE[111124] logger.c: --- (0 headers 0 lines) Nat keepalive Apr 5 09:25:37 VERBOSE[111124] logger.c: --- (0 headers 0 lines) Nat keepalive ---
Apr 5 09:25:37 VERBOSE[111124] logger.c:
<-- SIP read
CANCEL sip:&800123456@pbxes.org SIP/2.0
Via: SIP/2.0/UDP 46.74.179.80:42503;rport;branch=z9hG4bK63902
Max-Forwards: 70
P-src-ip: 46.74.179.80
To: <sip:&800123456@pbxes.org>
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK36803835
Call-ID: 874620873041@46.74.179.80
CSeq: 2 CANCEL
Contact: <sip:jakube-102@46.74.179.80:42503;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.2 beta/HTC Vision
Content-Length: 0
Apr 5 09:25:37 VERBOSE[111124] logger.c: --- (12 headers 0 lines)Apr 5 09:25:37 VERBOSE[111124] logger.c: --- (12 headers 0 lines)---
Apr 5 09:25:37 VERBOSE[111124] logger.c: Reliably Transmitting (NAT)
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 46.74.179.80:42503;branch=z9hG4bK63902;received=88.198.69.250;rport=59345
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK36803835
To: <sip:&800123456@pbxes.org>;tag=as5052d79c
Call-ID: 874620873041@46.74.179.80
CSeq: 2 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:&800123456@88.198.69.250:30856>
Content-Length: 0
---
Apr 5 09:25:37 VERBOSE[111124] logger.c: Transmitting (NAT)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 46.74.179.80:42503;branch=z9hG4bK63902;received=88.198.69.250;rport=59345
From: <sip:jakube-102@pbxes.org>;tag=z9hG4bK36803835
To: <sip:&800123456@pbxes.org>;tag=as5052d79c
Call-ID: 874620873041@46.74.179.80
CSeq: 2 CANCEL
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:&800123456@88.198.69.250:30856>
Content-Length: 0
---

Posted by Diafora on 07.04.2011 at 11:19:

RE: SipDroid 2.2 does not work after upgrading my account to SOHO

After looking at your trace output, I am slightly confused, so lets try to concentrate on extension to extension calls first.

So please dial a call to another extension, and paste the log here.

Thanks.

Posted by jakube on 13.04.2011 at 20:27:

RE: SipDroid 2.2 does not work after upgrading my account to SOHO

thanks for your response! here is a log from an extension2extension call

Apr 13 20:17:46 VERBOSE[111124] logger.c: Found RTP audio format 9
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found RTP audio format 8
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found RTP audio format 0
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found RTP audio format 97
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found RTP audio format 3
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found RTP audio format 106
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found RTP audio format 101
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found RTP video format 103
Apr 13 20:17:46 VERBOSE[111124] logger.c: Peer audio RTP is at port 192.168.3.12:21000
Apr 13 20:17:46 VERBOSE[111124] logger.c: Peer video RTP is at port 192.168.3.12:21070
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found description format G722
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found description format PCMA
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found description format PCMU
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found description format speex
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found description format GSM
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found description format BV16
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found description format telephone-event
Apr 13 20:17:46 VERBOSE[111124] logger.c: Found description format h263-1998
Apr 13 20:17:46 VERBOSE[111124] logger.c: Capabilities: us - 0x18161e (gsm|ulaw|alaw|g726|speex|ilbc|g722|h263|h263p), peer - audio=0x120e (gsm|ulaw|alaw|speex|g722)/video=0x100000 (h263p), combined - 0x10120e (gsm|ulaw|alaw|speex|g722|h263p)
Apr 13 20:17:46 VERBOSE[111124] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Apr 13 20:17:47 VERBOSE[45935] logger.c: We're at 88.198.69.250 port 38070
Apr 13 20:17:47 VERBOSE[45935] logger.c: Video is at 88.198.69.250 port 44210
Apr 13 20:17:47 VERBOSE[45935] logger.c: Adding codec 0x1000 (g722) to SDP
Apr 13 20:17:47 VERBOSE[45935] logger.c: Adding codec 0x4 (ulaw) to SDP
Apr 13 20:17:47 VERBOSE[45935] logger.c: Adding codec 0x8 (alaw) to SDP
Apr 13 20:17:47 VERBOSE[45935] logger.c: Adding codec 0x200 (speex) to SDP
Apr 13 20:17:47 VERBOSE[45935] logger.c: Adding codec 0x2 (gsm) to SDP
Apr 13 20:17:47 VERBOSE[45935] logger.c: Adding codec 0x100000 (h263p) to SDP
Apr 13 20:17:47 VERBOSE[45935] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Apr 13 20:17:47 VERBOSE[45935] logger.c: -- Called 105@from-internal-cont/n
Apr 13 20:17:47 NOTICE[45935] chan_local.c: No such extension/context @from-internal creating local channel
Apr 13 20:17:47 NOTICE[45935] app_dial.c: Unable to create channel of type 'Local' (cause 0 - Unknown)
Apr 13 20:17:47 NOTICE[45935] chan_local.c: No such extension/context @from-internal creating local channel
Apr 13 20:17:47 NOTICE[45935] app_dial.c: Unable to create channel of type 'Local' (cause 0 - Unknown)
Apr 13 20:17:47 VERBOSE[45951] logger.c: We're at 88.198.69.250 port 42848
Apr 13 20:17:47 VERBOSE[45951] logger.c: Video is at 88.198.69.250 port 42518
Apr 13 20:17:47 VERBOSE[45951] logger.c: Adding codec 0x1000 (g722) to SDP
Apr 13 20:17:47 VERBOSE[45951] logger.c: Adding codec 0x4 (ulaw) to SDP
Apr 13 20:17:47 VERBOSE[45951] logger.c: Adding codec 0x8 (alaw) to SDP
Apr 13 20:17:47 VERBOSE[45951] logger.c: Adding codec 0x10 (g726) to SDP
Apr 13 20:17:47 VERBOSE[45951] logger.c: Adding codec 0x400 (ilbc) to SDP
Apr 13 20:17:47 VERBOSE[45951] logger.c: Adding codec 0x200 (speex) to SDP
Apr 13 20:17:47 VERBOSE[45951] logger.c: Adding codec 0x2 (gsm) to SDP
Apr 13 20:17:47 VERBOSE[45951] logger.c: Adding codec 0x100000 (h263p) to SDP
Apr 13 20:17:47 VERBOSE[45951] logger.c: Adding codec 0x80000 (h263) to SDP
Apr 13 20:17:47 VERBOSE[45951] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Apr 13 20:17:47 VERBOSE[45951] logger.c: -- Called jakube-105
Apr 13 20:17:47 VERBOSE[111124] chan_sip.c: SIP response 100 to standard invite
Apr 13 20:17:48 VERBOSE[45951] chan_sip.c: Hangup call SIP/jakube-105-b06f, SIP callid 649192c93b8d38775749a32100f10198@88.198.69.250
Apr 13 20:17:48 VERBOSE[45935] chan_sip.c: Hangup call SIP/jakube-102-9a86, SIP callid 520639595317@192.168.3.12
Apr 13 20:17:48 VERBOSE[111124] chan_sip.c: SIP response 180 to standard invite

the interesting thing is that sometimes it starts working again for a some time and it stops working again. sometimes it helps to restart the phone. the problem is the same on all wifi and UMTS networks. it seems like a bug in Sipdroid to me, because other clients are working fine. unfortunately sipdroid is on my main phone now.