Telephone.app os x

Hello,

Have any of you got Telephone on os x to work with pbxes.com? The older version 0.14 worked, but the latest 0.15 does not.

Any help to debug this would be great, it's my favorite softphone.

Cheers,
Will

RE: Telephone.app os x

Hey Will,

It is highly probable the latest update, causes some kind of incompatibility with PBXes. In order to troubleshoot this, we will need some kind of trace with the older working version and with the latest one.

When you say it doesn't work, do you mean that it does not even register on PBXes, or that it does but can't complete any phone calls?

RE: Telephone.app os x

Here are the traces




============================ Telephone 0.14.13==========================

17:23:22.311 sip_endpoint.c Message Request msg NOTIFY/cseq=102 (rdata0x894c64) from 76.191.104.53:36999 was dropped/unhandled by any modules
17:23:23.491 pjsua_core.c RX 804 bytes Response msg 183/INVITE/cseq=24485 (rdata0x894c64) from UDP 76.191.104.53:36999:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.1.97:56758;branch=z9hG4bKPjgrWaS4Z9qwaS1KkQAMqh9sF4t7J6ODMa
From: "Will Prater - pbxes.com" <sip:wprater-203@pbxes.com>;tag=vMDnorvHFdxwPwnzkst3XaBPl03dtdtp
To: <sip:5551112222@pbxes.com>;tag=as6aec3747
Call-ID: trE7iPGIAHQ9KnHwZ7oGqTkvn.ZNkSy7
CSeq: 24485 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:5551112222@76.191.104.53:36999>
Content-Type: application/sdp
Content-Length: 297

v=0
o=root 114151 114151 IN IP4 76.191.104.53
s=session
c=IN IP4 76.191.104.53
t=0 0
m=audio 42248 RTP/AVP 0 8 117 102 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:117 iLBC/8000
a=rtpmap:102 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

--end msg--
17:23:23.491 AKTelephoneCal Call 0 state changed to EARLY (183 Session Progress)
17:23:23.492 strm0x1141574 Encoder stream started
17:23:23.492 strm0x1141574 Decoder stream started
17:23:23.492 pjsua_media.c Media updates, stream #0: PCMU (sendrecv)
17:23:23.492 conference.c Port 2 (sip:5551112222@pbxes.com) transmitting to port 0 (Built-in Microphone)
17:23:23.492 conference.c Port 0 (Built-in Microphone) transmitting to port 2 (sip:5551112222@pbxes.com)
17:23:23.492 AKTelephoneCal Media for call 0 is active
17:23:23.499 Master/sound Underflow, buf_cnt=0, will generate 1 frame
17:23:24.310 pjsua_core.c RX 559 bytes Request msg NOTIFY/cseq=102 (rdata0x894c64) from UDP 76.191.104.53:36999:
NOTIFY sip:wprater-203@10.0.1.97:56758 SIP/2.0
Via: SIP/2.0/UDP 76.191.104.53:36999;branch=z9hG4bK69efed41;rport
From: "Unknown" <sip:Unknown@76.191.104.53:28724>;tag=as3a08f868
To: <sip:wprater-203@10.0.1.97:56758>
Contact: <sip:Unknown@76.191.104.53:36999>
Call-ID: 020428c50b200cd918cb8c9362847611@76.191.104.53
CSeq: 102 NOTIFY
User-Agent: PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 89

Messages-Waiting: yes
Message-Account: sip:*97@76.191.104.53
Voice-Message: 5/1 (0/0)

--end msg--
17:23:24.310 sip_endpoint.c Message Request msg NOTIFY/cseq=102 (rdata0x894c64) from 76.191.104.53:36999 was dropped/unhandled by any modules
17:23:28.309 pjsua_core.c RX 559 bytes Request msg NOTIFY/cseq=102 (rdata0x894c64) from UDP 76.191.104.53:36999:
NOTIFY sip:wprater-203@10.0.1.97:56758 SIP/2.0
Via: SIP/2.0/UDP 76.191.104.53:36999;branch=z9hG4bK69efed41;rport
From: "Unknown" <sip:Unknown@76.191.104.53:28724>;tag=as3a08f868
To: <sip:wprater-203@10.0.1.97:56758>
Contact: <sip:Unknown@76.191.104.53:36999>
Call-ID: 020428c50b200cd918cb8c9362847611@76.191.104.53
CSeq: 102 NOTIFY
User-Agent: PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 89

Messages-Waiting: yes
Message-Account: sip:*97@76.191.104.53
Voice-Message: 5/1 (0/0)

--end msg--
17:23:28.309 sip_endpoint.c Message Request msg NOTIFY/cseq=102 (rdata0x894c64) from 76.191.104.53:36999 was dropped/unhandled by any modules
17:23:32.309 pjsua_core.c RX 559 bytes Request msg NOTIFY/cseq=102 (rdata0x894c64) from UDP 76.191.104.53:36999:
NOTIFY sip:wprater-203@10.0.1.97:56758 SIP/2.0
Via: SIP/2.0/UDP 76.191.104.53:36999;branch=z9hG4bK69efed41;rport
From: "Unknown" <sip:Unknown@76.191.104.53:28724>;tag=as3a08f868
To: <sip:wprater-203@10.0.1.97:56758>
Contact: <sip:Unknown@76.191.104.53:36999>
Call-ID: 020428c50b200cd918cb8c9362847611@76.191.104.53
CSeq: 102 NOTIFY
User-Agent: PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 89

Messages-Waiting: yes
Message-Account: sip:*97@76.191.104.53
Voice-Message: 5/1 (0/0)

--end msg--
17:23:32.309 sip_endpoint.c Message Request msg NOTIFY/cseq=102 (rdata0x894c64) from 76.191.104.53:36999 was dropped/unhandled by any modules
17:23:33.350 pjsua_core.c RX 790 bytes Response msg 200/INVITE/cseq=24485 (rdata0x894c64) from UDP 76.191.104.53:36999:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.97:56758;branch=z9hG4bKPjgrWaS4Z9qwaS1KkQAMqh9sF4t7J6ODMa
From: "Will Prater - pbxes.com" <sip:wprater-203@pbxes.com>;tag=vMDnorvHFdxwPwnzkst3XaBPl03dtdtp
To: <sip:5551112222@pbxes.com>;tag=as6aec3747
Call-ID: trE7iPGIAHQ9KnHwZ7oGqTkvn.ZNkSy7
CSeq: 24485 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:5551112222@76.191.104.53:36999>
Content-Type: application/sdp
Content-Length: 297

v=0
o=root 114151 114152 IN IP4 76.191.104.53
s=session
c=IN IP4 76.191.104.53
t=0 0
m=audio 42248 RTP/AVP 0 8 117 102 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:117 iLBC/8000
a=rtpmap:102 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

--end msg--
17:23:33.350 AKTelephoneCal Call 0 state changed to CONNECTING
17:23:33.350 inv0x113ba64 SDP negotiation done, message body is ignored
17:23:33.350 pjsua_core.c TX 382 bytes Request msg ACK/cseq=24485 (tdta0x10d6c00) to UDP 76.191.104.53:36999:
ACK sip:5551112222@76.191.104.53:36999 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.97:56758;rport;branch=z9hG4bKPjOUlkXJBkeSQNgAxxvGyUiFcrs.UxLLbH
Max-Forwards: 70
From: "Will Prater - pbxes.com" <sip:wprater-203@pbxes.com>;tag=vMDnorvHFdxwPwnzkst3XaBPl03dtdtp
To: <sip:5551112222@pbxes.


============================= Telephone 0.15.0 ===============================================


17:20:46.485 pjsua_core.c TX 322 bytes Response msg 200/NOTIFY/cseq=102 (tdta0x101079000) to UDP 76.191.104.53:36999:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 76.191.104.53:36999;rport=36999;received=76.191.104.53;branch=z9hG4bK30590ef1
Call-ID: 3f09c5164943058f4b61fd3a5a810a60@76.191.104.53
From: "Unknown" <sip:Unknown@76.191.104.53>;tag=as70ab9977
To: <sip:wprater-203@10.0.1.97>;tag=z9hG4bK30590ef1
CSeq: 102 NOTIFY
Content-Length: 0


--end msg--
17:20:49.702 pjsua_media.c Opening sound device PCM@16000/1/20ms
17:20:49.711 ec0x115f47c10 AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=20 ms
17:20:49.904 pjsua_call.c Making call with acc #1 to <sip:5551112222@pbxes.com>
17:20:49.905 pjsua_media.c Media index 0 selected for call 0
17:20:49.905 pjsua_core.c TX 1155 bytes Request msg INVITE/cseq=24762 (tdta0x1010f6000) to UDP 76.191.104.53:36999:
INVITE sip:5551112222@pbxes.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.97:53508;rport;branch=z9hG4bKPjGJU49dPeNzoEa9DGC6wy5lTq9YTJbCXo
Max-Forwards: 70
From: "Will Prater - pbxes.com" <sip:wprater-203@pbxes.com>;tag=y5NASoQpXB1wnkh3VbDEi83fSAydtW-b
To: <sip:5551112222@pbxes.com>
Contact: "Will Prater - pbxes.com" <sip:wprater-203@10.0.1.97:53508>
Call-ID: KSQHgJ0Z42ERCwW.aOKFCqRi90SINDb-
CSeq: 24762 INVITE
Route: <sip:pbxes.com:36999;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 0.15.0
Content-Type: application/sdp
Content-Length: 450

v=0
o=- 3495226849 3495226849 IN IP4 10.0.1.97
s=pjmedia
c=IN IP4 10.0.1.97
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4001 IN IP4 10.0.1.97
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--end msg--
17:20:49.906 AKSIPUserAgent Call 0 state changed to CALLING
17:20:49.911 pjsua_media.c Closing Soundflower (2ch) sound playback device and Soundflower (2ch) sound capture device
17:20:49.966 pjsua_core.c RX 583 bytes Response msg 407/INVITE/cseq=24762 (rdata0x1009026a from UDP 76.191.104.53:36999:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.1.97:53508;branch=z9hG4bKPjGJU49dPeNzoEa9DGC6wy5lTq9YTJbCXo
From: "Will Prater - pbxes.com" <sip:wprater-203@pbxes.com>;tag=y5NASoQpXB1wnkh3VbDEi83fSAydtW-b
To: <sip:5551112222@pbxes.com>;tag=as6e6682b6
Call-ID: KSQHgJ0Z42ERCwW.aOKFCqRi90SINDb-
CSeq: 24762 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:5551112222@76.191.104.53:36999>
Proxy-Authenticate: Digest realm="pbxes.org", nonce="7cae32c23fa6d8d021ef43473dd5d22a6393cb11"
Content-Length: 0


--end msg--
17:20:49.966 pjsua_core.c TX 405 bytes Request msg ACK/cseq=24762 (tdta0x101062000) to UDP 76.191.104.53:36999:
ACK sip:5551112222@pbxes.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.97:53508;rport;branch=z9hG4bKPjGJU49dPeNzoEa9DGC6wy5lTq9YTJbCXo
Max-Forwards: 70
From: "Will Prater - pbxes.com" <sip:wprater-203@pbxes.com>;tag=y5NASoQpXB1wnkh3VbDEi83fSAydtW-b
To: <sip:5551112222@pbxes.com>;tag=as6e6682b6
Call-ID: KSQHgJ0Z42ERCwW.aOKFCqRi90SINDb-
CSeq: 24762 ACK
Route: <sip:pbxes.com:36999;lr>
Content-Length: 0


--end msg--
17:20:49.966 tcpc0x10108f42 TCP client transport created
17:20:49.966 tcpc0x10108f42 TCP transport 10.0.1.97:64094 is connecting to 76.191.104.53:36999...
17:20:49.966 pjsua_core.c TX 1353 bytes Request msg INVITE/cseq=24763 (tdta0x1010f6000) to tcp 76.191.104.53:36999:
INVITE sip:5551112222@pbxes.com SIP/2.0
Via: SIP/2.0/TCP 10.0.1.97:64094;rport;branch=z9hG4bKPj2YIrRgp032wXB3uPkJQ6kl4VNRr..H8q
Max-Forwards: 70
From: "Will Prater - pbxes.com" <sip:wprater-203@pbxes.com>;tag=y5NASoQpXB1wnkh3VbDEi83fSAydtW-b
To: <sip:5551112222@pbxes.com>
Contact: "Will Prater - pbxes.com" <sip:wprater-203@10.0.1.97:53508>
Call-ID: KSQHgJ0Z42ERCwW.aOKFCqRi90SINDb-
CSeq: 24763 INVITE
Route: <sip:pbxes.com:36999;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 0.15.0
Proxy-Authorization: Digest username="wprater-203", realm="pbxes.org", nonce="7cae32c23fa6d8d021ef43473dd5d22a6393cb11", uri="sip:5551112222@pbxes.com", response="e2ae1f76351398b72d38f63873eaaef5"
Content-Type: appli

This post has been edited 1 time(s), it was last edited by wpr on 05.10.2010 at 02:29.

RE: Telephone.app os x

Hey Will,

I have looked through the traces, but they don't provide any clues as to why the 0.14.13 works but the 0.15.0 does not.

Having said this, can you explain what doesn't work on the 0.15.0 version? Does it register properly on PBXes, or not? Will it ring on an inbound call but is not able to answer it?

Regarding the SIP traces, try to start the trace when the application starts, especially on the 0.15.0 version, since we wish to see whether it even registers on PBXes. Then try to capture a complete trace of a successful phone call in the 0.14.13 version. Calls start with an INVITE message while they end with an ACK following the BYE message.

Let me know if you need help getting the traces, and consider reaching the application's developer as well, to inquire what changed significantly between these two versions.