making a call with different accounts

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Hi,

I have created two different accounts. When I make a loop call (to local uri where pjsip is running) for test purposes with one of the created accounts, the default account is matched with the call. Is this specific to loop calls or am I making a mistake? 

account 1
status = pjsua_acc_add_local(udp_transport_id,0,&app_acc1_id);
account 2

status = pjsua_acc_add_local(udp_transport_id,0,&app_acc2_id);

I specify the account when making a call, but only the default account accepts calls.

Thank you for your help.




--- On Sun, 5/2/10, pjsip-request at lists.pjsip.org <pjsip-request at lists.pjsip.org> wrote:

From: pjsip-request@xxxxxxxxxxxxxxx <pjsip-request at lists.pjsip.org>
Subject: pjsip Digest, Vol 33, Issue 1
To: pjsip at lists.pjsip.org
Received: Sunday, May 2, 2010, 10:00 AM

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Today's Topics:

???1. [patch] Incoming RFC 2833 DTMF detection (Bram Kuijvenhoven)
???2. PJMEDIA_EAUD_NODEFDEV on chumby (tintar)


----------------------------------------------------------------------

Message: 1
Date: Sat, 01 May 2010 23:44:56 +0200
From: Bram Kuijvenhoven <mailinglists.bram@xxxxxxxxxxxxxxxx>
To: pjsip list <pjsip at lists.pjsip.org>
Subject: [patch] Incoming RFC 2833 DTMF detection
Message-ID: <4BDCA0D8.2040800 at kuijvenhoven.net>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"

Hi!

When testing my pjsip app with another SIP gateway I noticed not all DTMF events were coming through. I particular of subsequent events with the *same* digit only the first was recognized; all subsequent events did not trigger a callback to my application.

A patch is attached to this message; the following is an explanation.

=====================================================================

Inspection of the RTP packets (with Wireshark) and the code in the on_rx_rtp and handle_incoming_dtmf functions (in pjmedia/src/pjmedia/stream.c) reveals that:
- The SIP gateway sends several RTP packets for one DTMF event, but all of these always have the *same* value for the duration field: 800. Other gateways incrementally increase the duration in each subsequent packet of one event (see sections 3.4 - 3.6 of RFC 2833, http://www.ietf.org/rfc/rfc2833.txt).
- PJSIP considers any subsequent DTMF packet with the same digit and an *equal* or larger duration as pertaining to the same event as the previous packet.
So this explains why subsequent events for the same digit are discarded; the durations of the packets are always equal.

There are various ways a new event could be detected:
- all packets belonging to the same event have the same timestamp field (in the RTP header)
- first packet has market bit set (in the RTP header)
- last packet of an event has an 'end' bit set (in the RFC 2833 payload); note: usually at least three duplicates of this packet are sent
- the first packet of a new event may have a smaller duration than that of the last packet

The attached patch considers all subsequent packets with the same timestamp (and digit) as one event.

The only case that it would not handle correctly is if the packets for two events would somehow overlap; this may be fixed by changing
??? pj_ntohl(hdr->ts) == stream->last_dtmf_ts
in the patch to
??? pj_ntohl(hdr->ts) <= stream->last_dtmf_ts
but then we'd have to rely on proper ordering of the timestamps.

Regards,

Bram
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------------------------------

Message: 2
Date: Sun, 02 May 2010 07:17:38 -0400
From: tintar <tintar@xxxxxxxxx>
To: pjsip at lists.pjsip.org
Subject: PJMEDIA_EAUD_NODEFDEV on chumby
Message-ID: <4BDD5F52.7010304 at gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

1.5.5 as well as 1.6. building on x64 system, using arm-2008q3 
toolchain. the only configure flags I used were --build=i386-linux 
--host=arm-linux if that makes any difference.

(also, just to be sure I wasn't crazy or missing some dependency libs, I 
built the same source on the same build machine except target x86 i.e. 
the build machine itself. works perfectly.)

I did try the legacy audiodev patch against those versions, having read 
the previous threads about it vs. the iphone. still no dice.

could never get a straight answer from the chumby developers what 
version they used, but obviously it would have been older (early 2008 at 
the latest)... tried 1.3, same result. and also 0.9, basically the same 
thing: PJMEDIA_ENOSNDPLAY.

sndinfo from 0.9.0 gave me this:
? 07:11:03.632 os_core_unix.c pjlib 0.9.0-release for POSIX initialized
? 07:11:03.668? ? ? pasound.c PortAudio sound library initialized, status=0
? 07:11:03.669? ? ? pasound.c PortAudio host api count=1
? 07:11:03.670? ? ? pasound.c Sound device count=1
? 07:11:03.673? ? ? ? ? pjlib select() I/O Queue created (0x4933c)
Device #00:
???Name? ? ? ? ? ? ? ? : /dev/dsp
???# of input channels : 16
???# of output channels: 0
???Default clock rate? : 44100 Hz

yeah, I guess that confirms it.

any advice on how to trace this would be appreciated...

thanks!



and just for completeness:

? 06:21:14.596? pjsua_media.c? Opening sound device PCM at 16000/1/20ms
? 06:21:14.598? pjsua_media.c? Opening sound device PCM at 44100/1/20ms
? 06:21:14.598? pjsua_media.c? Opening sound device PCM at 48000/1/20ms
? 06:21:14.599? pjsua_media.c? Opening sound device PCM at 32000/1/20ms
? 06:21:14.599? pjsua_media.c? Opening sound device PCM at 16000/1/20ms
? 06:21:14.600? pjsua_media.c? Opening sound device PCM at 8000/1/20ms
? 06:21:14.600? pjsua_media.c? Unable to open sound device: Unable to 
find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=420006]



------------------------------

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