How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this: Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 That's funny, I thought Inband DTMF was ONLY supported on G.711 u-law So in the sip.conf file, for the hell of it, I changed dtmfmode to rfc2833 and then placed a call with FireFly. This time the error didn't show, however, I noticed that the GSM codec was being used, as could be heard and seen with "sips how channels". I have GSM unchecked in FireFly, and yet it chose it anyways. In my [general] section of sip.conf disallow=all ; First we Deny Everything allow = ulaw ; Then we set our preferred codec allow = gsm ; Then our backup codec allow = g729 ; And our last resort codec and my account entry used to be: [user_test] context=user_testing type=friend username=user_test secret=hidden qualify=yes host=dynamic canreinvite=no dtmfmode=inband nat=yes mailbox=1012@home_users callerid="Test" <1234567890> accountcode=1 amaflags=omit but dtmfmode=rfc2833 now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041103/7643bc1d/attachment.htm
Ok. For [user_test] I specifically disabled all codecs and ONLY allowed Ulaw. This seems to have forced firefly into using Ulaw, regardless of what's checked in the program. Also, I left dtmfmode as rfc2833 in sip.conf as well as set rfc2833 in FireFly, and it appears to work. I didn't think rfc2833 could be used on ULaw, but then again I know little about dtmf, my general rule was "Inband for Ulaw, RFC2833 for everything else, and stay away from info as I have no idea what it does" Also, I had to do "qualify=no" as FireFly apparently doesn't respond right, I keep seeing "Unreachable" in the "sip show peers" commands for my firefly clients, even though they can place and receive calls just fine. I just did qualify=no and now it just says "Unmonitored" or something. This may be a bug, I saw the "poke" thing was fixed in IAX, maybe they accidentally broke it in SIP? _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Rodan Sent: Wednesday, November 03, 2004 1:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] FireFly Problems How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this: Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 That's funny, I thought Inband DTMF was ONLY supported on G.711 u-law So in the sip.conf file, for the hell of it, I changed dtmfmode to rfc2833 and then placed a call with FireFly. This time the error didn't show, however, I noticed that the GSM codec was being used, as could be heard and seen with "sips how channels". I have GSM unchecked in FireFly, and yet it chose it anyways. In my [general] section of sip.conf disallow=all ; First we Deny Everything allow = ulaw ; Then we set our preferred codec allow = gsm ; Then our backup codec allow = g729 ; And our last resort codec and my account entry used to be: [user_test] context=user_testing type=friend username=user_test secret=hidden qualify=yes host=dynamic canreinvite=no dtmfmode=inband nat=yes mailbox=1012@home_users callerid="Test" <1234567890> accountcode=1 amaflags=omit but dtmfmode=rfc2833 now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041103/85cb163f/attachment.htm
Hello, I have firefly installed and it is somewhat working. It is registering with my Asterisk server and I can call out, but I receive no audio coming into Firefly. From the Asterisk end, everything looks OK with the call, just no audio is being received on the Firefly end. I am using 1.9.6 Any ideas? Here is the info from my iax.conf ... [firefly1] type=friend accountcode=iaxy host=dynamic secret=xxxx context=home disallow=all allow=ilbc allow=gsm auth=md5 trunk=no qualify=no Thanks much, Chris -- ------------------------ Chris Olson chrisolson@webbeams.com 708-261-4770 www.webbeams.com ------------------------ -- ------------------------ Chris Olson chrisolson@webbeams.com 708-261-4770 www.webbeams.com ------------------------
Chris Olson wrote:> Hello, > > I have firefly installed and it is somewhat working. It is registering > with my Asterisk server and I can call out, but I receive no audio > coming into Firefly. From the Asterisk end, everything looks OK with > the call, just no audio is being received on the Firefly end. I am > using 1.9.6 > > Any ideas? >a fix for this will be out tommorrow - you can temporarily fix it by inserting the r option into your dial cmd cheers, Adam
> Chris Olson wrote: > >>> Hello, >>> >>> I have firefly installed and it is somewhat working. It is registering >>> with my Asterisk server and I can call out, but I receive no audio >>> coming into Firefly. From the Asterisk end, everything looks OK with >>> the call, just no audio is being received on the Firefly end. I am >>> using 1.9.6 >>> >>> Any ideas? >>> > > > a fix for this will be out tommorrow - you can temporarily fix it by > inserting the r option into your dial cmd > > cheers, > > AdamThanks Adam. Can you let us know when the fix is available and where we can download the fixed 3rd-party from? A little more info ... this is actually a one-way audio problem as audio passes from Firefly to Asterisk, but not from Asterisk to Firefly.
> Chris Olson wrote: > > >>>>> Chris Olson wrote: >>>>> >>> >>>>>>>>> Hello, >>>>>>>>> >>>>>>>>> I have firefly installed and it is somewhat working. It is registering >>>>>>>>> with my Asterisk server and I can call out, but I receive no audio >>>>>>>>> coming into Firefly. From the Asterisk end, everything looks OK with >>>>>>>>> the call, just no audio is being received on the Firefly end. I am >>>>>>>>> using 1.9.6 >>>>>>>>> >>>>>>>>> Any ideas? >>>>>>>>> >>> >>>>> >>>>> >>>>> a fix for this will be out tommorrow - you can temporarily fix it by >>>>> inserting the r option into your dial cmd >>>>> >>>>> cheers, >>>>> >>>>> Adam >> >>> >>> >>> >>> Thanks Adam. Can you let us know when the fix is available and where we >>> can download the fixed 3rd-party from? >>> >>> A little more info ... this is actually a one-way audio problem as audio >>> passes from Firefly to Asterisk, but not from Asterisk to Firefly. >>> >>> > > > You can grab the new one from http://www.virbiage.com/ now, if anyone > wants the old version, it's at > http://www.virbiage.com/firefly/firefly-thirdparty196.exeThanks Adam. The new version is working very well. I really like this softphone. Since you are doing such a great job on this :), I have 1 more question. (1) Is there a way to pre-configure the phone for some clients? By this I mean by having some extensions pre-programmed and also having the IAX option pre-programmed? I have some people I'd like to do beta testing with and was hoping I could just send them the installation file, and it would be preconfigured for them and ready to go. Thanks, Chris -- ------------------------ Chris Olson chrisolson@webbeams.com 708-261-4770 www.webbeams.com ------------------------