Re: [asterisk-users] Two Asterisks behind NAT and need to link themusing IAX trunk
It is possible to run openVPN in TCP mode over an SSH tunnel. Don't turn compression on on both though - I'd just switch it on the openVPN if you have to. You will probably find the speech is rather choppy due to the delays and fragmentation, but I have done this. Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: 18 January 2008 12:21 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Two Asterisks behind NAT and need to link themusing IAX trunk Hi; Via OpenVPN or port forwarding is known for me, but via SSH is new for me, how I can do it and what is the difference by SSH and OpenVPN? Regards Bilal - Good question. I have never tried tunneling IAX over SSH but it seems like it should work just like anything else. How about a port opened up for OpenVPN. You know you can run IAX on any port you wish, port 80 may work for you if you have some extra external IPs not being used for HTTP. The same is true for OpenVPN. Thanks, Steve Totaro On Jan 17, 2008 8:09 PM, John Constalgie <[EMAIL PROTECTED]> wrote: > > Hi there > > this is an interesting topic that I see here and a problem that I am > trying to solve too. > > But I was wondering if the forwarding solution will work for my case. > > So I have two Asterisk boxes A and B. > > A is behind a corporate NAT such that A can SSH to B, but not vice versa( > "One-way SSH" ) . The UDP port 5060 of the corporate NAT is blocked off and > I will not be able to have it unblocked for security reasons. > > Hence, is my only choice using an SSH tunnel between A and B for the IAX > connection to work? Will it work though with that "One-way SSH" factor > mentioned before? > > Thanks > John Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of "slin" as a codec
Partially answering my own question, it looks like "slin" is a 128 kbps codec. Peter From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Whisker, Peter Sent: 05 December 2007 16:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Use of "slin" as a codec Where bandwidth is not an issue but good call quality is, is there any theoretical quality improvement to be had by using "slin" as the codec over an inter-Asterisk IAX trunk rather than a-law (or u-law in the US). Does anyone know what the slin bandwidth is (compared to 64 kbps a-law). Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Use of "slin" as a codec
Where bandwidth is not an issue but good call quality is, is there any theoretical quality improvement to be had by using "slin" as the codec over an inter-Asterisk IAX trunk rather than a-law (or u-law in the US). Does anyone know what the slin bandwidth is (compared to 64 kbps a-law). Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What web GUI are people happy with?
I used DEC's EDT for almost 20 years on PDP-11 and find jed with the EDT interface useful! You can't teach an old dog new tricks! Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym Sent: 16 October 2007 17:10 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] What web GUI are people happy with? So nano just makes things too easy for you? -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Monday, October 15, 2007 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What web GUI are people happy with? I use vi. Not sure if it has a web interface yet. PaulH On Tue, 2007-10-16 at 00:51 +0200, Dovid B wrote: > None. Asterisk vanilla is the best IMHO. > - Original Message - > From: Anciso, Roy > To: asterisk-users@lists.digium.com > Sent: Monday, October 15, 2007 7:28 PM > Subject: [asterisk-users] What web GUI are people happy with? > > > Just wondering what web GUI people like for asterisk. I > installed asterisk from source and I was looking at possibly > installing web GUI for system management. So far freepbx.org > looks promising anybody else have any suggestions. > > Thanks > > > > Roy Anciso > > Director of Technology > > Manistee Intermediate School District > > 1710 Merkey Road > > Manistee, MI 49660 > > Ph: 231-723-4264 > > Fx: 231-723-1690 > > [EMAIL PROTECTED] > > > > > > __ > > ___ > --Bandwidth and Colocation Provided by > http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cell phone that can be connected to standadphone switch network
On Wed, 18 Apr 2007, Joseph wrote: > Are there any cell phone (gadgets) that can be connected to standard > switch phone network? (ability to check email would be a plus). > > Digium adapter S101i can be connected to any network and it allow a > standard phone to act as your local extension over the Internet (by > registering to asterisk), it works "almost" perfectly. > So it would be handy to have a cellular phone that can be connected to > standard switched phone network, are there any toys like this? Dock'n'Talk. Check the archives. Although let's hope you're not in the UK - I eventually managed to get in-touch with the person D&T passed me on to who was their UK disty, but after an initial reply they haven't bothered to email me back advising me of stock, prices or avalability. Gordon --- Some of the new Nokia mobile phones now support SIP over WiFi (802.11g) as well as GSM. Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with 3-way calls from a Sipura ATA
I have an Asterisk servers (recent SVN version 1.2) and two Sipura ATAs (one 2000 and one 1001). I have Three-way Conf Serv and Three-way Call Serv enabled on both ATAs. When I make a SIP call from phone 1 to phone 2 on my Asterisk box, it works fine, then when I press the hookflash on phone 1, the caller on phone 2 is placed on hold (Asterisk MoH plays). I get the "second dial tone" on the Sipura and dial another extension (phone 3) from phone 1 and it rings and I am connected. Now, if I press hookflash again on phone 1, it switches the connection to phone 2, but phone 3 is placed on hold rather than getting conferenced in. Anyone got any idea why the three-way call might not be working in these circumstances? The message type for setting of Hookflash (none, AVT, INFO) seems to make no difference. I normally set it to "none" as it should be handled locally rather than get sent to Asterisk. Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Skype<>SIP gateway
I managed to get it to work and make a test call from Asterisk to Skype. Pity it's not implemented on Linux and needs Skype to be running on the PC also. Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten Sent: 05 April 2006 16:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] New Skype<>SIP gateway Shad Mortazavi wrote: > Message: 24 > Date: Mon, 03 Apr 2006 19:21:57 -0500 > From: "Michael Graves" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] New Skype<>SIP gateway > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Anyone seen or tried this yet? > > http://www.voip-weblog.com/50226711/uplink_connects_sip_skype.php > > Michael > > - > > I have tried to register with both Asterisk and SER; Unfortunately > this does not seem to work. > > Great idea. Guess we need to wait for the next version. > > I'll post some comments to the nch website. > > Shad > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users dam.. i was hoping for something server side, not some windoze client.. oh well.. guess it's back to more waiting for * <-> skype This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADPCM - vs - G.726
The G.726 codec is the current Asterisk 1.2 version (revision 7221). I am using G.711a (alaw) between a Sipura ATA and Asterisk at each end of the link and am testing alternative codecs on an IAX link (not in trunk mode) between the two Asterisk servers. (Yes, I know that the Sipura can do G.726-32 also). When I have G.726 switched in as the codec, it seems to add a little gain as well as the clipping clicks. If I use G.711 on the Asterisk-Asterisk link there is no problem. ADPCM seems to behave better than G.726 but is percepibly "fuzzy". GSM as I expected sounds like an old pre-EFR mobile phone. And the rest are worse or like G.729 or Speex, eat processor power. I have G.723 and G.729 compiled from the Intel distro. Peter Steve Underwood wrote: Whisker, Peter wrote: I have been looking at the medium-rate codecs in Asterisk - ADPCM and G.726. Both of these are adaptive PCM codecs - the G.726 one is a little more expensive in processing power, however both are 32k bit-rate. I am experiencing problems using G.726 where the audio level is high. It produces loud clicks as if clipping. For quiet audio however, it seems fine. ADPCM (Digilogic VOX?) seems to be better in this respect without the clicks but has slight artifacts audible. Anybody had the same problems with G.726? I can't find any comparisons out there between the two codecs - does anyone have a link? Thanks Peter If it does strange things at high volume the G.726 codec is buggy. G.726 has a number of bells and whistles, such that DTMF and some lower speed modems will pass through OK, and so tandem operation does not become progressively worse. The OKI/Dialogic ADPCM doesn't have these, and as a result cause far less CPU loading. However, for typical speech only use they should offer comparable quality. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADPCM - vs - G.726
I have been looking at the medium-rate codecs in Asterisk - ADPCM and G.726. Both of these are adaptive PCM codecs - the G.726 one is a little more expensive in processing power, however both are 32k bit-rate. I am experiencing problems using G.726 where the audio level is high. It produces loud clicks as if clipping. For quiet audio however, it seems fine. ADPCM (Digilogic VOX?) seems to be better in this respect without the clicks but has slight artifacts audible. Anybody had the same problems with G.726? I can't find any comparisons out there between the two codecs - does anyone have a link? Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.726 codec - can we select bandwidth?
Can someone please tell me if it's possible to select the G726 codec bandwidth for an IAX trunk between two Asterisk 1.2.5 servers? I can select "disallow=all / allow=g726" but I think it defaults to the g726-32 variant. Is there any way of forcing Asterisk to use g726-24 for such a trunk example? Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax softphones
Dante's DIAX is pretty good IMHO. Peter Hector medina wrote: can anyone recomend a good iax softphone?? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip not 100% RFC3665 compliant - re-REGISTERsfail.
There are issues with Asterisk chan_sip. Have a look at bug 759 at bugs.digium.com. Comments in the feature report and source code like those below probably go a way to explain your problems. I don't know how much of this test version has been back-ported to chan_sip, however the chan_sip2.c with a November 2004 CVS seems to work quite well. Olle Johansson has suspended work on this for now due to workload and it probably won't compile any more against the latest CVS due to changes elsewhere. Peter "* Added support for WWW-auth for registrations (according to SIP RFC). " "* + WARNING: This version changes a lot of functionality in regards *to authentication, we use the digest auth username to check *credentials for INVITES, not the username@ in the From: URI *INVITEs are authenticated this new way, not REGISTER/SUBSCRIBE *yet " -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vahan Yerkanian Sent: 07 March 2005 13:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip not 100% RFC3665 compliant - re-REGISTERsfail. Greetings, For the past 2 months I've been struggling with registration problems with asterisk+external FXS/FXO gateways (www.addpac.com) that use RFC3665 re-registration procedure. This problem occured for devices with more than one FXS port with a set non-empty password. Those gateway attempt to re-register after the initial register timeout period expires fully compliant with RFC3665, clause 2.2 (http://www.zvon.org/tmRFC/RFC3665/Output/chapter2.html#sub2), but asterisk fails to authenticate them. The 1st FXS port of the device always registers successfuly (although still uses same RFC3665, clause 2.2 procedure), but the remainder fail miserably. Using an account/username with an empty password for the affected ports fixes the problem - so this is something with www-digest method (?). I've spent 2 weeks debugging this with addpac development team, and the same device authenticates flawlessly with Sonus Proxy Server, SNOM Proxy Server, LongBoard Proxy Server, Nortel Proxy so this seems to be a problem with chan_sip. I'm hesitant to post the long sip debug outputs to the mailing list to conserve the bandwidth. More info and sip debugs are available at http://bugs.digium.com/bug_view_page.php?bug_id=0003726 Is there anyone else with the same problem? regards, Vahan This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extra sounds (Weather)
Hi
This is my script for my local forecast for SE England. I have had
problems getting festival to work integrated so I have cron run this
script every 3 hours and use Playback to play it in Asterisk:
Script
--
#!/bin/sh
cd /var/lib/asterisk/sounds
curl
"http://www.bbc.co.uk/weather/ukweather/printables/print_regional_outloo
k.shtml?pmslondon" 2>/dev/null \
| (sed -n '/print area open/,/print area close/ { s/<.*>//;s/deg
C/Celsius/;s/deg F/Fahrenheit/;s/ deg$/ /;s/^C /Celsius
/;s/^F)/Fahrenheit)/;p }' && date +'B B C forecast, %A %e %B at %l %p')
\
| /usr/local/bin/text2wave -f 8000 - -o wx.tmp.wav
sox wx.tmp.wav -r 8000 -c 1 wx.tmp.gsm
mv wx.tmp.gsm wx.gsm;rm -rf wx.tmp.wav
Extensions.conf
---
;Weather forecast for SE England 0_0 WX (199)
exten => 199,1,Answer
exten => 199,2,Playback(wx)
exten => 199,3,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Liaan vd
Merwe
Sent: 16 February 2005 11:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Extra sounds (Weather)
Hi Trevor
This i know
I just send you a other script doing the same task this will give you a
guideline to make you own
- Original Message -
From: "Trevor G. Hammonds" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Wednesday, February 16, 2005 1:50 PM
Subject: RE: [Asterisk-Users] Extra sounds (Weather)
> Liaan vd Merwe wrote on Wednesday, 16 February 2005
2:53 AM:
>
>> This is the example script (extracted from that
link) you will need
>> to find a weather page for your region an then
change the urls and
>> grep statements chow L
>
> Once again, this is NOT the script mentioned at Eric
Wieling's former
> site,
> http://www.fnords.org/~eric/asterisk/, referenced it
the message in the
> archives at
>
http://lists.digium.com/pipermail/asterisk-users/2003-November/025983.ht
ml.
>
>
> Sincerely,
> Trevor Hammonds
>
> ___
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>
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>
http://lists.digium.com/mailman/listinfo/asterisk-users
>
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RE: [Asterisk-Users] DIAX
GSM Codec is 13k plus overhead. That may work? Peter -Original Message- From: Bilal Ghayad [mailto:[EMAIL PROTECTED] Sent: 15 January 2005 07:07 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DIAX Dear Dan; Thanks alot for your kindly reply. Well, what u advise us to use if the bandwidth is about 22kbps (dial up connection in very old countries)? Another thing: u have idea if it is working on Microsoft Windows OS? As most of clients here are using Microsoft and not linux. Regards Bilal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DIAX 0.9.9g more features and higher stabili ty
I have had the same problem when calling across Asterisk from Diax to a SIP phone. If Asterisk "Answers" the call before the "Dial" to the SIP phone there is no delay. Otherwise there is a 10-20 second delay in the Voice path! Peter -Original Message- From: Dan [mailto:[EMAIL PROTECTED] Sent: 14 January 2005 15:57 To: Denis Galvão - iSolve; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability Hi Denis, > Doesn't have effect, the problem continue. >The strange thing is that the delay is aprox 10-20 seconds!!! Too high!! >Is there another thing to do!? > >I really want to use DIAX because it supprts my USB Phone and IAX protocol. I don't think your problem is DIAX related. Can you provide more info about your environment? In the Control Panel Sound Configuration have you selected MIC as only input for the used soundcard and wave out as the output? Best regards, Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX users
SIP is a XML-like control channel and is used to negotiate a separate RTP channel which carries the audio. It is complicated to set-up in cases of firewalls and NAT, but is an open standard. IAX2 is a candidate open standard and merges all traffic onto a single UDP stream - control and audio data. It has two modes, trunk and non-trunk. Trunk mode is highly efficient for transmitting multiple calls on a single UDP bearer and has minimal overhead. Standard IAX2 is easier to set-up than SIP. In terms of user experience, there should be little difference in call handling and audio quality - in general all of the same codecs and features are supported. IAX2 is a native protocol of Digium's Asterisk switch and I believe stands for Inter-Asterisk-eXchange version 2. To answer the query below, IAX (ie IAX11) was the precursor of IAX2. It is obsolete and should no longer be used. I use IAX when referring to IAX2, but obviously not all do! HTH Peter -Original Message- From: Serge Schumacher [mailto:[EMAIL PROTECTED] Sent: 31 December 2004 15:41 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IAX users Sorry ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: vendredi 31 décembre 2004 16:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX users IAX2 - Original Message - From: "Serge Schumacher" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Friday, December 31, 2004 8:00 AM Subject: [Asterisk-Users] IAX users > Hi, > > I do not understand the difference between SIP and IAX, is it only two > different protocols or something more special. > > The problem I have is that I've created two users > > > Aix.conf > > register => users1:passwd1 > register => user2:passwd2 > > [user1] > type=user > context=default > secret=passwd1 > host=dynamic > > > [user2] > type=user > context=default > secret=passwd2 > host=dynamic > > extensions.conf > > exten => 550,1(Dial,IAX/user1); > exten => 551,1(Dial,IAX/user2); > > and the error I get : > > > Dec 31 15:03:16 WARNING[2885]: pbx.c:1280 pbx_extension_helper: No > application 'IAX/user1)' for extension (default, 550, 1) > == Spawn extension (default, 550, 1) exited non-zero on > 'IAX2/[EMAIL PROTECTED]:1059/1' > -- Hungup 'IAX2/[EMAIL PROTECTED]:1059/1' > > Can someone help me how to get both users connected ? > > Thank you, > > > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g711 ulaw vs alaw
Partly is is down to the fact that G.711u (mu-law) is primarily used in the USA and G.711a (a-law) is used in Europe. Like you, I am not sure if the exact differences - they have the same bitrate and audio, although there are minor differences in the format. Peter -Original Message- From: Eric Bishop [mailto:[EMAIL PROTECTED] Sent: 16 December 2004 10:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] g711 ulaw vs alaw Hi All, Can someone explain to me the difference between g711's ulaw and alaw codecs? Is it just different header info or is the actual payload in each encoded differently? I have thus far noe been able to find any difinative information onthe matter. All I've managed to find out that they are "similar", they sound the same and that it doesn't matter which you use. Could someone knowledgable please enlightmen me? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK available SIP phone?
I picked up a Sipura SPA-2000 (new) on e-Bay for ~£70. The voice quality is excellent. Peter -Original Message- From: Mike Dent [mailto:[EMAIL PROTECTED] Sent: 29 November 2004 14:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK available SIP phone? I should reply back now and say that I managed to get the Tecom IP2005 from solwise.co.uk working with Asterisk. I did however buy a Grandstream Budge Tone when I had almost given up on the Tecom one, so now I have both! :) The external quality of rhe grandstream is not as nice as the Tecom but they it is a bit cheaper. I'm still in the very early stages of Asterisk so I'm not sure what features the Tecom one does/not support yet. Thanks to those who replied. Regs Mike (now if only I could get IAX/FWD/SIP or something to work I'd be happy) On Mon, 29 Nov 2004 13:48:12 +, Jon Lawrence <[EMAIL PROTECTED]> wrote: > On Sunday 21 November 2004 12:03, Clive Carter wrote: > > > > >Hi, > > > Anybody here from the UK using Asterisk at home? > > >I'm looking for a SIP phone which will work with Asterisk and > > >not leave me broke! > > > > > >I got one of the Tecom ones from Solwise but it refuses to > > >login to Asterisk server for some reason. May have to send it back. > > > > > >What are the other options please? > > > > > >Thanks > > >Mike > > > > I use Grandstream Budge Tones. They are cheap, and some people say they > > look it, but they work ! > > I have also got ipDialogs SipTone II. They are twice the price, and > > although I have got the basic functions working, for some reason they > > just will not connect to VoiceMail > > > > I have used both sipura 2k's and ata286 - both worked perfectly with my dect > phones. > Currently 2K is connected to DECT, ata286 to fax and a 7960G for main use. > I have in the past had budgetones and yes, they do look cheap - but so what if > it's only for home use. > I work from home, hence the 7960G (I simply needed more lines). But imho > you'll struggle to beat a sipura 2K with a good quality DECT phone - although > that works out a similar price as my 7960G. > > HTH > Jon > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Huge ten second audio delay on SIP channel
I have two Asterisk servers interconnected with IAX (non-trunk). I place a call on Server B (using DIAX) which goes to an extension on Server A and terminates with a Dial to a local SIP phone (Sipura SPA 2200). The SIP phone rings immediately but when it is answered there is a delay of about 10-15 seconds on the line. However, if I put an Answer() just before the final Dial(SIP/..) command on Server A's extension, there is the normal tens of millisecond delay in the Audio. Clearly undesirable for billing reasons but does anyone know what on earth is going on!? I am using a recent CVS. Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] processing power / codecs
This is my codec translation timing table (500MHz PIII). The fastest codec is G.711 (ulaw/alaw) which is uncompressed, next is GSM. The slinr->codec row gives the amount of time (and probably relates to processing power) in coding and the codec->slinr columns are the decoding time. Speex is variable (OGG based) but is the slowest in general with ILBC and G729A pretty slow too. Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - - - - - - - - - - - GSM - - 4 412 4 320508559 ULAW - 9 - 110 2 118488357 ALAW - 9 1 -10 2 118488357 G726 -16 9 9 - 9 825559064 ADPCM - 9 2 210 - 118488357 SLINR - 8 1 1 9 1 -17478256 LPC10 -1710101810 9 -569165 G729A -19121220121128 -9367 SPEEX -1710101810 92656 -65 ILBC -181111191110275792 - Peter -Original Message- From: Kristian Kielhofner [mailto:[EMAIL PROTECTED] Sent: 10 November 2004 04:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] processing power / codecs TELUX wrote: > is there a lot of processing power difference between ulaw an G711 when > in a meet-me? > > Thanks > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users Telux, G.711 is a standard that defines Ulaw and Alaw, commonly called Ulaw and Alaw. But last I checked Meetme transcodes all codecs to Ulaw for the purposes of the conference. So, I suppose G.711u would be your best bet for low processor overhead in Meetme. Perhaps you meant G.729, G.723, or G.726? G.729 - High processor usage, and a license G.723 - No * support for transcoding G.726 - No presonal experience from me. -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transcoding - when and when not?
I am a little confused about voice data transcoding in Asterisk. I can make a call between two u-law-only phones over an IAX GSM-codec link and the two Asterisk servers handle the transcoding ulaw-GSM...GSM-ulaw fine. However, over a SIP channel, this doesn't seem to work. Asterisk appears to be reluctant to transcode and keeps complaining about the wrong codec type. When does and when doesn't this work!? Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 audio problems but SIP OK?
>One is on a T1 connection and the other is on 576k/288k ADSL. The Ping time >is about 30ms between the two servers, 90% of which is the ADSL delay. > >When I interconnect them with IAX2, I get rather choppy audio - with what >sounds like dropped packets and jitter. > >However when I interconnect with SIP is it clean and with no dropouts. The >network path and timings are identical for both protocols and there is >little noticeable difference when I play with the jitterbuffer setting in >iax.conf. > >Does anyone have any idea why IAX protocol is causing this kind of problem? >My ADSL is PPPoE which has an MTU of 1492 I think. Could this be causing the >problems? > > No, you shouldn't be sending any IAX2 packets which approach this size. 1) What codecs are you using in each case? 2) Do you have other traffic on the link? It's possible that somewhere along your path, the RTP audio traffic from SIP is getting some kind of helpful QoS benefit, while the IAX2 traffic is not? = No. I test when the link is quiet. I have tried Ulaw and GSM - in fact bizarrely the GSM is worse. The processors on the Asterisk boxes are only 5% loaded - they are 500MHz Pentiums so this should not be the problem. I don't know about the QoS but have never noticed any difference in the various settings. Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OpenSource Proxies ?.
Oops. Sorry about this post. Pressed the send button my mistake! -Original Message- From: Whisker, Peter [mailto:[EMAIL PROTECTED] Sent: 02 November 2004 14:18 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OpenSource Proxies ?. I have an * switch at home and one in the office. Both similar new CVS head versions and both with chan_sip2 built in: Asterisk CVS-HEAD-10/12/04-17:43:26 Asterisk CVS-HEAD-10/13/04-12:53:52 One is on a T1 connection and the other is on 576k/288k ADSL. The Ping time is about 30ms between the two servers, 90% of which is the ADSL delay. When I interconnect them with IAX2, I get rather choppy audio - with what sounds like dropped packets and jitter. However when I interconnect with SIP is it clean and with no dropouts. The network path and timings are identical for both protocols and there is little noticeable difference when I play with the jitterbuffer setting in iax.conf. Does anyone have any idea why IAX protocol is causing this kind of problem? My ADSL is PPPoE which has an MTU of 1492 I think. Could this be causing the problems? Peter -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: 02 November 2004 13:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OpenSource Proxies ?. SER - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Tuesday, November 02, 2004 8:47 AM Subject: [Asterisk-Users] OpenSource Proxies ?. What are my alternatives when it comes to OPENSOURCE SIP proxies ?. /Hitete ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 audio problems but SIP OK?
[sorry about previous mis-post] I have an * switch at home and one in the office. Both similar new CVS head versions and both with chan_sip2 built in: Asterisk CVS-HEAD-10/12/04-17:43:26 Asterisk CVS-HEAD-10/13/04-12:53:52 One is on a T1 connection and the other is on 576k/288k ADSL. The Ping time is about 30ms between the two servers, 90% of which is the ADSL delay. When I interconnect them with IAX2, I get rather choppy audio - with what sounds like dropped packets and jitter. However when I interconnect with SIP is it clean and with no dropouts. The network path and timings are identical for both protocols and there is little noticeable difference when I play with the jitterbuffer setting in iax.conf. Does anyone have any idea why IAX protocol is causing this kind of problem? My ADSL is PPPoE which has an MTU of 1492 I think. Could this be causing the problems? Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OpenSource Proxies ?.
I have an * switch at home and one in the office. Both similar new CVS head versions and both with chan_sip2 built in: Asterisk CVS-HEAD-10/12/04-17:43:26 Asterisk CVS-HEAD-10/13/04-12:53:52 One is on a T1 connection and the other is on 576k/288k ADSL. The Ping time is about 30ms between the two servers, 90% of which is the ADSL delay. When I interconnect them with IAX2, I get rather choppy audio - with what sounds like dropped packets and jitter. However when I interconnect with SIP is it clean and with no dropouts. The network path and timings are identical for both protocols and there is little noticeable difference when I play with the jitterbuffer setting in iax.conf. Does anyone have any idea why IAX protocol is causing this kind of problem? My ADSL is PPPoE which has an MTU of 1492 I think. Could this be causing the problems? Peter -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: 02 November 2004 13:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OpenSource Proxies ?. SER - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Tuesday, November 02, 2004 8:47 AM Subject: [Asterisk-Users] OpenSource Proxies ?. What are my alternatives when it comes to OPENSOURCE SIP proxies ?. /Hitete ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: call progress - what are the sticking po ints?
It looks for tones (currently hardwired as US). I have updated to include UK tones but is hard to get it to reliably recognise. For example the tones in the switch here at work are 5-10% off frequency. Correcting for this, and doing a lot of fiddling it did recognise the tones but was unreliable. I have a problem in that our office switch clears to dialtone rather than busy if the other end hangs up. I would like a way of recognising unexpected dialtone and hanging-up. So far, this has not been easy. I have changed the busydetect to clear if it gets continupus tone for 8 seconds but this does false hangups and would be useless for a fax machine. Peter -Original Message- From: Steve Underwood [mailto:[EMAIL PROTECTED] Sent: 28 October 2004 14:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: call progress - what are the sticking points? Stephen David wrote: >i don't have a specific bug in mind, i was just wondering WHY call progress doesn't work so well -- in particular, on analog lines. ie. is it a hardware or software problem (or both). with more info, i'd like to help to work out the kinks, for myself and everyone. :) > > Back in the days of Stowger exchanges you knew when the called party answered, by a reversal of the DC voltage on your analogue line. With digital exchanges that stopped, and no solid feedback is given to the caller on ordinary analogue lines. You have to infer that someone has answered, and the reliability of that can be poor. Digital lines, like ISDNand SS7, and protocols like MFC/R2 tell you positively that someone has answered. >>I have the same problem. >>callprogress is very experimental and buggy now. >>and i've lost the .call files feature of asterisk. >>what do you think about submitting a bug on bugs.digium.com? >> >> >> > >not sure what you mean by 'lost the .call files feature', but if you have a specific bug to post, i think it would be great if you posted it. > > > >> regards, >> shabanip >> >> > Hello, >> > >> > I've been experimenting with the call progress analysis features of *, >> > with mixed success on Zap as well as IAX channels. I've read all the >> > posts about it, including (but not limited to) >> > http://www.voip-info.org/wiki-Asterisk+auto-dial+out and the pages it >> > references. >> > >> > My question is, what's the current state -- is there any work in progress >> > right now to improve the reliability of * call progress detection? last I >> > saw it was still listed as 'experimental'. >> > >> > What are the "problems" that are preventing a more robust implementation >> > of call progress detection? Would this work better with different >> > hardware (ie. I've had success in the past using Dialogic telephony >> > boards)? Or is this primarily a software issue with *? >> >> If you had good results with Dialogic it was merely luck. Because they have to infer the phone has been answered, their detection only works if the calls follow their model of how someone answers the phone. Depending on your circumstances, and the nature of the calls you make, it can be hopelessly unreliable. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] windows messenger
The SIP client in Windows Messenger 5.0 seems to work fine with Asterisk though. Peter -Original Message- From: Robert Rozman [mailto:[EMAIL PROTECTED] Sent: 11 October 2004 22:08 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] windows messenger I'm not sure if I remember right, but I think that 4.7v of Windows Messenger did work in presence sense, but as I told I'm not sure. Regards, Robert. - Original Message - From: "Bill Seddon" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Sent: Monday, October 11, 2004 6:13 PM Subject: RE: [Asterisk-Users] windows messenger > Asterisk doesn't support MSN9 the protocol Windows Messenger (and MSN > Messenger) uses to communicate with a messenger server such as MSN or > Windows 2003 running the Live Conferencing server. > > It should be possible to write an MSN9 server independently of Asterisk > since the information needed by such a server is available via the Manager > API. > > Bill Seddon > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of shabanip > Sent: October 11, 2004 4:55 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] windows messenger > > is it possible to windows messenger clients of an asterisk server to chat > (text chat) with each other? > what about the status presence? is it possible to each windows messenger > client of an asterisk server to see the presence on other clients? > if not, what is missing in asterisk? > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FireFly w/ SIP
Adam On UK keyboards ,I have to type a "£" to get a "#" into Firefly. The proper "#" key does nothing. If you are updating the code, perhaps you might look at this? Many thanks Peter -Original Message- From: Adam Hart [mailto:[EMAIL PROTECTED] Sent: 16 October 2004 07:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FireFly w/ SIP The best way for me or yourself to debug it is using ethereal (google for it) and debugview from www.sysinternals.com. I'm happy to help, so send the logs, the native transfer might be the issue. -Adam Willem de Groot wrote: > Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk? > > It works in IAX mode, but in SIP mode I am unable to hear anything (no > dialtone, no voice). I am able to setup a conversation with another SIP > phone though (Xlite, Grandstream) and the other side can hear the > FireFly user just fine (both sides using g711u). > > I tried different PC's with different audio hardware. They all work fine > using FireFly in IAX mode and using other softphones, so I guess it must > be related so FireFly in SIP mode. > > This is my SIP config: > > [201] > type=friend > host=dynamic > dtmfmode=rfc2833 > context=sip > canreinvite=yes > > FireFly is also configured for rfc2833 dtmf. > > Thanks for any suggestions! > Willem > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DIAX 0.9.9b - now multi codec support
Thanks Dan. It seems to crash less too. Is there any way to enter DTMF tones? Thanks Peter -Original Message- From: Dan [mailto:[EMAIL PROTECTED] Sent: 18 October 2004 14:48 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DIAX 0.9.9b - now multi codec support Hi David, >- Original Message - >From: "David J Carter" <[EMAIL PROTECTED]> > Can I import all the settings from a previous version into 0.9.9b to save > re-inputting all the info? > For the moment you can only do that by direct editing of the diax.cfg file You just need to copy the section [REGISTRATION] The Phonebook (diax.pb) and Calls list (diax.cl) are the same as in the previous version. Just replace them with the old ones. If you need further assistance, please drop me a mail. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: {SPAM?} [Asterisk-Users] Asterisk VIA SSH Tunnels
You would need a TCP version of IAX to use SSH as I don't think it supports
UDP.
Asterisk does work (tunelling IAX) through Zebedee (an SSH-like TCP & UDP
tunnel).
Peter
-Original Message-
From: Tom Neville [mailto:[EMAIL PROTECTED]
Sent: 13 October 2004 16:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: {SPAM?} [Asterisk-Users] Asterisk VIA SSH Tunnels
I've been running ssh tunnels for a couple of years now. For years,
they've worked well. However, now that I've got asterisk up I do
notice problems. Biggest indication of this is if I'm on a call and
run a program in another window that scrolls and scrolls call quality
drops off significantly. (I'm using FreeBSD on all the tunnel machines
going back to the office. I work at an ISP, so I have a machine here
at the office and use the tunnels across my DSL lines.)
Based on advice from David McNett, I'm looking at moving to OpenBSD for
the tunnel machines. With that, I'll be able to use pf+altq
(http://slacker.com/~nugget/asterisk4.php) on the tunnel interfaces.
Hopefully, that will take care of the only issue I've had with the
tunnels since installing them.
On Oct 12, 2004, at 9:42 PM, Christopher Jacob wrote:
> Anyone ever set up Asterisk to use SSH Tunneling? Anyone know the pros
> &
> cons?
>
> Thanks,
>
> Chris
>
> ___
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RE: [Asterisk-Users] Prerelease of DIAX 0.9.9a
Same for me. After a few minutes the program crashes. Any chance of support for ULAW / ALAW which is mandatory for FWD IAX? Thanks Peter -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: 14 October 2004 09:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Prerelease of DIAX 0.9.9a I like it but it always "generates errors" and closes on my win2k box. - Original Message - From: "Brian" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, October 14, 2004 3:15 AM Subject: Re: [Asterisk-Users] Prerelease of DIAX 0.9.9a > Any chance you could include support for placing a call on hold? > > And like BKW said, It'd be awesome if we could talk ya into releasing > the source... ;) > > -Brian > > > Brian West wrote: > > Anyway we could talk you into releasing the source? I would love to see > > wider codec support. And the ability to launch the URL sent with the IAX > > call. > > > > bkw > > > > > >>-Original Message- > >>From: [EMAIL PROTECTED] [mailto:asterisk-users- > >>[EMAIL PROTECTED] On Behalf Of Dan > >>Sent: Wednesday, October 13, 2004 10:02 AM > >>To: Jon Bebeau; Asterisk Users Mailing List - Non-Commercial Discussion > >>Subject: Re: [Asterisk-Users] Prerelease of DIAX 0.9.9a > >> > >>Hi Jon, > The application is distributed as a freeware, source code not included. > >> > >>Sorry for the inconvenience. > >>If you need some specific help, please send me a mail directly. > >> > >>Best regards, > >>Dan > >> > >>>- Original Message - > >>>From: "Jon Bebeau" <[EMAIL PROTECTED]> > >>>Sent: Wednesday, October 13, 2004 5:52 PM > >> > >> > >>>Hi Dan, > >>> > >>>Did you release the source for DIAX? I'm trying to build a drop-on > >>>component for MS .NET (2005) and I've been looking for a good starting > >>>place. I spent some time with IAXClient and a few other from wiki, but > >>>most > >>>are Linux specific..then there's X10, but it's commercial. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: bt communicator`
Then you use just "username". I don't think the MD5 is critical - the auth=username:[EMAIL PROTECTED] is what chan_sip2 uses. You must not use port=5060 in sip.conf - it has to be port=5052. You also have to do funny things with the externip - point it to the *internal* address and use "nat=yes" as per my example. It seems to need to think it is coming via NAT. Otherwise it may register and fail the second proxy_auth_required. It is all very strange. If you change the username.brz in my example to your username (as seen by ethereal) and normal password and give that a try. Also note the fact that btinternet.com and sip.btcommunicator.net can not be used interchangeably and it is fussy which is used where. I put a "81.144.106.4btinternet.com" line into my /etc/hosts file. This is the address of sip.btcommunicator.bt.net. But if you put host=sip.btcommunicator.bt.net into the sip.conf file, authentication fails. Put a host=btinternet.com in sip.conf which will then connect to the right machine using the wrong name and this works! The implementaion of domains and servers seems a bit odd. Peter -Original Message- From: Robert Boardman [mailto:[EMAIL PROTECTED] Sent: 11 October 2004 18:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: bt communicator` Hi Peter, just a couple of quick questions if the my ethereal trace only shows [EMAIL PROTECTED] do I need the .brz? is the MD5sum in the ethereal trace, as I have compared all combinations of MD5sum with the ethereal trace and cannot see it any where? Still cannot register, any advice would be greatly appreciated Regards Robb Whisker, Peter wrote: >Hi Robert; > >First, you have to use the SIP2 channel code (chan_sip2.c) from >http://bugs.digium.com/bug_view_page.php?bug_id=759 as this does the >proxy-authenticate properly. > >Get the module, follow the build instructions, and add "noload=chan_sip.so" >to stop the old code loading. It will autoload the new one. > >You need to know the username that the Yahoo Communicator uses. Ethereal or >similar will trace SIP for you. The username I type into communicator has >".brz" appended by the Communicator for some reason. The password is the one >you type into communicator but I had to "MD5" it. Comment below password >shows how. > >[general] >;port = 5060; Port to bind to >port = 5052 ; change to 5052 as 5060 will not >authorise on BTCommunicator >; Note if you want local SIP on 5060, you need to use siproxd or similar to >redirect (unless anyone knows otherwise) >pedantic=no >disallow=all; Disallow all codecs >allow=alaw ; Allow codecs in order of preference ; BT >uses a-law >allow=ulaw >allow=gsm >;allow=ilbc >defaultexpirey=1200 ; Change for BT as it objects to 3600 - >note deliberate spelling error > >register => >[EMAIL PROTECTED]:[EMAIL PROTECTED] > >; Need to state externip as the internal address otherwise BT won't work - >something to do with NAT >;externip = 195.13x.xx.xx >externip = 192.168.10.250 >localnet = 192.168.10.0/255.255.255.0 >;. >;. >;. > >[bt] >type=friend >nat=yes >disallow=all >allow=alaw >canreinvite=no >username=[username].brz >authuser=[username].brz >fromdomain=btinternet.com >fromuser=[username].brz >auth=[username]:[EMAIL PROTECTED] ; I didn't have the >.brz here and it works? >md5secret=6eb36df5f5d94381973b6090b30e0f59 >host=btinternet.com >;outboundproxy=sip.btcommunicator.bt.net ;not needed >;outboundproxyport=5060;not needed >;MD5 >;alambil:/etc/asterisk# echo -n >"[EMAIL PROTECTED]:btinternet.com:[password]" | md5sum >;6eb36fd5f5d94381973b6090b30e0f59 - > >Once this worked, I didn't change it. There are probably unneeded lines >above. > >Regards >Peter > > >-Original Message- >From: Robert Boardman [mailto:[EMAIL PROTECTED] >Sent: 09 October 2004 21:40 >To: Whisker, Peter >Subject: bt communicator` > > >Hi Peter > >I have been following your post but didn't see the other emails about >getting it working until now!! > >Could you please send me the details for the chan_sip2 method > >Thanks >Robb > >This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all co
[Asterisk-Users] RE: bt communicator`
Hi Robert; First, you have to use the SIP2 channel code (chan_sip2.c) from http://bugs.digium.com/bug_view_page.php?bug_id=759 as this does the proxy-authenticate properly. Get the module, follow the build instructions, and add "noload=chan_sip.so" to stop the old code loading. It will autoload the new one. You need to know the username that the Yahoo Communicator uses. Ethereal or similar will trace SIP for you. The username I type into communicator has ".brz" appended by the Communicator for some reason. The password is the one you type into communicator but I had to "MD5" it. Comment below password shows how. [general] ;port = 5060; Port to bind to port = 5052 ; change to 5052 as 5060 will not authorise on BTCommunicator ; Note if you want local SIP on 5060, you need to use siproxd or similar to redirect (unless anyone knows otherwise) pedantic=no disallow=all; Disallow all codecs allow=alaw ; Allow codecs in order of preference ; BT uses a-law allow=ulaw allow=gsm ;allow=ilbc defaultexpirey=1200 ; Change for BT as it objects to 3600 - note deliberate spelling error register => [EMAIL PROTECTED]:[EMAIL PROTECTED] ; Need to state externip as the internal address otherwise BT won't work - something to do with NAT ;externip = 195.13x.xx.xx externip = 192.168.10.250 localnet = 192.168.10.0/255.255.255.0 ;. ;. ;. [bt] type=friend nat=yes disallow=all allow=alaw canreinvite=no username=[username].brz authuser=[username].brz fromdomain=btinternet.com fromuser=[username].brz auth=[username]:[EMAIL PROTECTED] ; I didn't have the .brz here and it works? md5secret=6eb36df5f5d94381973b6090b30e0f59 host=btinternet.com ;outboundproxy=sip.btcommunicator.bt.net;not needed ;outboundproxyport=5060 ;not needed ;MD5 ;alambil:/etc/asterisk# echo -n "[EMAIL PROTECTED]:btinternet.com:[password]" | md5sum ;6eb36fd5f5d94381973b6090b30e0f59 - Once this worked, I didn't change it. There are probably unneeded lines above. Regards Peter -Original Message- From: Robert Boardman [mailto:[EMAIL PROTECTED] Sent: 09 October 2004 21:40 To: Whisker, Peter Subject: bt communicator` Hi Peter I have been following your post but didn't see the other emails about getting it working until now!! Could you please send me the details for the chan_sip2 method Thanks Robb This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Failed to authenticate on INVITE
For info The new chan_sip2.c and recent CVS (yesterday) fix this and I can now use Asterisk to make calls on the sip.btcommunicator.bt.net service. If anyone wants help with the settings, e-mail me off list. :) Peter -Original Message-From: Whisker, Peter [mailto:[EMAIL PROTECTED]Sent: 17 September 2004 14:40To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Failed to authenticate on INVITE I am getting this also. I am trying to get Asterisk to talk similarly to BT Communicator to the BT server. I can register but then the INVITE fails. BT are mixed up with their domains (in fact in the INVITE their software has a To: header with @domain1 and an auth URI referencing @domain2. The realm is domain1.) This can't be done in Asterisk where it is consistent about the URI. I had been blaming this, but if you are having problems too... I get the standard 407 header requesting Proxy Auth for the call. Asterisk submits the INVITE with auth and after the usual "Trying" I just get another 407. I have traces of Asterisk and the client which works and they seem so similar in what they do. I have made all the port ranges the same too. BT Communicator fails if you use port 5060 for the SIP client - they use 5052. Peter -Original Message-From: Stig Thune [mailto:[EMAIL PROTECTED]Sent: 17 September 2004 12:55To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Failed to authenticate on INVITE NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429' sip.conf register => 1234:[EMAIL PROTECTED] extension.conf -- ;; Own extensions;exten => 0852509516,1,Goto(resepsjon-own,s,1) ;[resepsjon-own]; exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,Background(own/choose) ; Meny, 1 for support, 2 for support, 3 for wx3 exten => s,6,Wait(1) exten => s,7,Background(own/choosenumber) ; dialer pushes a # ,and being sent to.. ; ip-phone must be picked up in ,2ms,tr or hangup exten => 1,1,Goto(privatanslutningar,s,1) exten => 2,1,Goto(foretagsanslutningar,s,1) ; #=hangup exten => #,1,Playback(custom/no-key-registered) exten => #,2,Hangup exten => t,1,Goto(#,1) ; If they take too long, give up exten => i,1,Playback(invalid) ; "That's not valid, try again" inmenu] ;[privatanslutningar]; exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,Background(own/privatanslutningar) ; Meny, 1 for support, 2 for support, 3 for wx3 ; dialer pushes a # ,and being sent to.. exten => 1,1,Answer exten => 1,2,Queue(help-privatanslutningar-queue) exten => 2,1,Answer exten => 2,2,Queue(order-privatanslutningar-queue) exten => 3,1,Answer exten => 3,2,Queue(info-privatanslutningar-queue) ; #=hangup ;exten => #,1,Playback(custom/no-key-registered) ;exten => #,2,Hangup exten => t,1,Queue(general-privatanslutningar-queue) ; If they take too long, give up exten => i,1,Playback(invalid) ; "That's not valid, try again" inmenu] ;[foretagsanslutningar]; exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,Background(own/foretagsanslutningar) ; Meny, 1 for support, 2 for support, 3 for wx3 ; dialer pushes a # ,and being sent to.. ; ip-phone must be picked up in ,2ms,tr or hangup exten => 1,1,Answer exten => 1,2,Queue(info-bedriftsanslutningar-queue) exten => 2,1,Answer exten => 2,2,Queue(help-bedriftsanslutningar-queue) exten => 3,1,Answer exten => 3,2,Queue(error-bedriftsanslutningar-queue) ; #=hangup ;exten => #,1,Playback(custom/no-key-registered) ;exten => #,2,Hangup exten => t,1,Queue(general-bedriftsanslutningar-queue) ; If they take too long, give up exten => i,1,Playback(invalid) ; "That's not valid, try again" inmenu] -- The call gets into queue, then... the other phone rings.. and when I pick up - I get this message: NOTICE[98310]: chan_sip.c:6638 handle_response: Failed
RE: [Asterisk-Users] Failed to authenticate on INVITE
I am getting this also. I am trying to get Asterisk to talk similarly to BT Communicator to the BT server. I can register but then the INVITE fails. BT are mixed up with their domains (in fact in the INVITE their software has a To: header with @domain1 and an auth URI referencing @domain2. The realm is domain1.) This can't be done in Asterisk where it is consistent about the URI. I had been blaming this, but if you are having problems too... I get the standard 407 header requesting Proxy Auth for the call. Asterisk submits the INVITE with auth and after the usual "Trying" I just get another 407. I have traces of Asterisk and the client which works and they seem so similar in what they do. I have made all the port ranges the same too. BT Communicator fails if you use port 5060 for the SIP client - they use 5052. Peter -Original Message-From: Stig Thune [mailto:[EMAIL PROTECTED]Sent: 17 September 2004 12:55To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Failed to authenticate on INVITE NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429' sip.conf register => 1234:[EMAIL PROTECTED] extension.conf -- ;; Own extensions;exten => 0852509516,1,Goto(resepsjon-own,s,1) ;[resepsjon-own]; exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,Background(own/choose) ; Meny, 1 for support, 2 for support, 3 for wx3 exten => s,6,Wait(1) exten => s,7,Background(own/choosenumber) ; dialer pushes a # ,and being sent to.. ; ip-phone must be picked up in ,2ms,tr or hangup exten => 1,1,Goto(privatanslutningar,s,1) exten => 2,1,Goto(foretagsanslutningar,s,1) ; #=hangup exten => #,1,Playback(custom/no-key-registered) exten => #,2,Hangup exten => t,1,Goto(#,1) ; If they take too long, give up exten => i,1,Playback(invalid) ; "That's not valid, try again" inmenu] ;[privatanslutningar]; exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,Background(own/privatanslutningar) ; Meny, 1 for support, 2 for support, 3 for wx3 ; dialer pushes a # ,and being sent to.. exten => 1,1,Answer exten => 1,2,Queue(help-privatanslutningar-queue) exten => 2,1,Answer exten => 2,2,Queue(order-privatanslutningar-queue) exten => 3,1,Answer exten => 3,2,Queue(info-privatanslutningar-queue) ; #=hangup ;exten => #,1,Playback(custom/no-key-registered) ;exten => #,2,Hangup exten => t,1,Queue(general-privatanslutningar-queue) ; If they take too long, give up exten => i,1,Playback(invalid) ; "That's not valid, try again" inmenu] ;[foretagsanslutningar]; exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,Background(own/foretagsanslutningar) ; Meny, 1 for support, 2 for support, 3 for wx3 ; dialer pushes a # ,and being sent to.. ; ip-phone must be picked up in ,2ms,tr or hangup exten => 1,1,Answer exten => 1,2,Queue(info-bedriftsanslutningar-queue) exten => 2,1,Answer exten => 2,2,Queue(help-bedriftsanslutningar-queue) exten => 3,1,Answer exten => 3,2,Queue(error-bedriftsanslutningar-queue) ; #=hangup ;exten => #,1,Playback(custom/no-key-registered) ;exten => #,2,Hangup exten => t,1,Queue(general-bedriftsanslutningar-queue) ; If they take too long, give up exten => i,1,Playback(invalid) ; "That's not valid, try again" inmenu] -- The call gets into queue, then... the other phone rings.. and when I pick up - I get this message: NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429' I know that the register => works.. I have checked with my SIP-provider, and they say that it is logged in. What else can be wrong ? / Stig Henning This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___
RE: [Asterisk-Users] BT Communicator (SIP???) and Asterisk
MMm Got it authenticated at last. Generating the right MD5 hash was a help! But it is only working on one copy of asterisk (this morning's CVS) so I will rebuild the other. Below are my sip.conf settings if it helps anyone. You may need to trace the login from BT Communicator to see if you need the .brz or something else. Now to try to get it to work! Peter sip.conf register => [EMAIL PROTECTED]:[EMAIL PROTECTED]/sip.btcommuni cator.bt.net [sip.btcommunicator.bt.net] type=peer canreinvite=no externip=213.86.115.71 username=username.brz authuser=username.brz fromdomain=btinternet.com fromuser=username.brz md5secret= host=sip.btcommunicator.bt.net ;/etc/asterisk# echo -n "[EMAIL PROTECTED]:btinternet.com:password" | md5sum ; - Peter -Original Message----- From: Whisker, Peter [mailto:[EMAIL PROTECTED] Sent: 06 September 2004 09:18 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] BT Communicator (SIP???) and Asterisk I have tried to get this working, but can not get it to authorise: I created my Communicator logon from a Yahoo account (not a btinternet account). Assuming my Yahoo username is "", BT Communicator software logs on to the SIP proxy as "[EMAIL PROTECTED]" according to the trace which seems a little odd. What is the ".brz" bit for? I have tried the password I set up in the BT Communicator phone but it is being rejected with Authorization failure. I have also tried the Yahoo password and that is also being rejected. I can only assume that they also do some transformation on the password. Any ideas? Thanks Peter -Original Message- From: gARetH baBB [mailto:[EMAIL PROTECTED] Sent: 28 August 2004 12:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BT Communicator (SIP???) and Asterisk On Mon, 23 Aug 2004, Robert Boardman wrote: > Heartened by your that you have got x-lite working, I have been trying, > but failing to now get x-lite working, don suppose you could send me a > quick screen shot of you x-lite settings? Not really, but it's not hard to get going. Presuming you have an account [EMAIL PROTECTED], in System Settings->SIP Proxy->Default you put: Display name: username Username: username Authorizarion User: username Password: [password] Domain/Realm: btinternet.com SIP Proxy: sip.btcommunicator.bt.net Out Bound Proxy: sip.btcommunicator.bt.net I must get round to playing with the simple Outbound Proxy stuff in Asterisk CVS - though I think it's a global Outbound Proxy so not really useful to use in earnest. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BT Communicator (SIP???) and Asterisk
I have tried to get this working, but can not get it to authorise: I created my Communicator logon from a Yahoo account (not a btinternet account). Assuming my Yahoo username is "", BT Communicator software logs on to the SIP proxy as "[EMAIL PROTECTED]" according to the trace which seems a little odd. What is the ".brz" bit for? I have tried the password I set up in the BT Communicator phone but it is being rejected with Authorization failure. I have also tried the Yahoo password and that is also being rejected. I can only assume that they also do some transformation on the password. Any ideas? Thanks Peter -Original Message- From: gARetH baBB [mailto:[EMAIL PROTECTED] Sent: 28 August 2004 12:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BT Communicator (SIP???) and Asterisk On Mon, 23 Aug 2004, Robert Boardman wrote: > Heartened by your that you have got x-lite working, I have been trying, > but failing to now get x-lite working, don suppose you could send me a > quick screen shot of you x-lite settings? Not really, but it's not hard to get going. Presuming you have an account [EMAIL PROTECTED], in System Settings->SIP Proxy->Default you put: Display name: username Username: username Authorizarion User: username Password: [password] Domain/Realm: btinternet.com SIP Proxy: sip.btcommunicator.bt.net Out Bound Proxy: sip.btcommunicator.bt.net I must get round to playing with the simple Outbound Proxy stuff in Asterisk CVS - though I think it's a global Outbound Proxy so not really useful to use in earnest. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap X100P oscillation
What is particularly weird is that if I connect a call to Echo on my local server, I don;t usually get this oscillation, however if I connect it to Echo on another Asterisk server (over IAX), it does happen. The main difference will be the delay in the echo, as the software versions are within the last two days. The Echo application is the same on both, and the call should be entirely digital so there should be no gain differences and there is no echo suppression on the IAX link, so it must be the local Zap echo canceller fighting with the apparently "longer" line? Peter -Original Message- From: Mike Benoit [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 21:59 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zap X100P oscillation I wonder if your issue and mine are related somehow. I have a asterisk server with 4 FXO cards in it, and when a call comes in one ZAP channel, then dials out another, I hear what could be described as a steam engine starting up. It starts off kinda slower/ quiet, then quickly (in about 2-4 seconds) completely over powers the line. The only way I could stop it was by adjusting the gains. rxgain=-8.5 txgain=4 Seemed to do the trick. As did: rxgain=-6.5 txgain=1 An rxgain of even -8.0 or -6.0 in either case would result in this "steam engine" sound. -8.5 or -6.5 would make it go away completely. I'm using a CVS checkout from yesterday, and I tried with both echotraining=800 and turning echo cancellation off completely. Neither made any difference. It would be really nice to be able to use a positive rxgain value. I haven't tried with the echo app, but using just one FXO card works fine with almost any rx/txgain value. As soon as the call utilizes two FXO card at the same time, the "steam engine" sound occurs. On Mon, 2004-06-28 at 16:26 +0100, Whisker, Peter wrote: > Has anyone seen this problem before? > > I have a server with a single X100P card. The audio level is a low, but if I > raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo > test. Not at a high frequency but with a noise that is best described as a > steam engine starting up. It then starts to clip and crackle. If I bring the > gain down to Rx=-2.0 and Tx=0.0 or lower then it settles down but it is very > very quiet. > > I have tried the latest CVS Head with echotraining=800 set and also complied > with the aggressive echo cancelling, but nothing seems to help. > > Ideas welcome! > > Many thanks > Peter Whisker > > This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compile error with CVS HEAD zaptel
I get a compile warning when building zaptel (current CVS head) against 2.4.18 kernel (Debian stable dist) zaptel.c: In function `zt_net_close': zaptel.c:1238: warning: implicit declaration of function `hdlc_close' It completes but fails to install with modprobe finding unresolved references. This error was introduced with the introduction of the second clause of the statement below in zconfig.h #if defined(CONFIG_HDLC) || defined(CONFIG_HDLC_MODULE) Clearly there is a problem building against the kernel modules in 2.14.18 as hdlc_close does not exist. There is no problem if I comment out the second part from and including the || above. It builds ok against 2.4.23 but I need to keep 2.4.18 for cnxadsl compatibility. Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playing the invalid extension input
Maybe it is trying to say "i" as a digit?
You could have an [invalid] context with
[invalid]
exten => _.,1,Saydigits(${EXTEN})
and then include it at the very end of the [default] context (or wherever
you want to use it). That would then pick up anything that drops through. If
you do it any other way it will get sorted to the top and you will have
trouble!
Peter
-Original Message-
From: Isamar Maia [mailto:[EMAIL PROTECTED]
Sent: 29 June 2004 15:54
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Playing the invalid extension input
I'm trying to do the following:
exten => i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
Any suggestions?
Thanks,
Isamar
___
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RE: [Asterisk-Users] Zap X100P oscillation
I find that if I drop the RX gain too much I start to lose DTMF decoding. The Asterisk calls lose at least 3-6db end-to-end compared with a "normal" call. If I bring the gain up, the symptoms sound exactly like yours. The gain I am using is more like Rx=-2, Tx=0 but this is still quite quiet. I guess that line impedance mismatch between US and European standards accounts for some of the gain loss. Peter -Original Message- From: Mike Benoit [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 21:59 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zap X100P oscillation I wonder if your issue and mine are related somehow. I have a asterisk server with 4 FXO cards in it, and when a call comes in one ZAP channel, then dials out another, I hear what could be described as a steam engine starting up. It starts off kinda slower/ quiet, then quickly (in about 2-4 seconds) completely over powers the line. The only way I could stop it was by adjusting the gains. rxgain=-8.5 txgain=4 Seemed to do the trick. As did: rxgain=-6.5 txgain=1 An rxgain of even -8.0 or -6.0 in either case would result in this "steam engine" sound. -8.5 or -6.5 would make it go away completely. I'm using a CVS checkout from yesterday, and I tried with both echotraining=800 and turning echo cancellation off completely. Neither made any difference. It would be really nice to be able to use a positive rxgain value. I haven't tried with the echo app, but using just one FXO card works fine with almost any rx/txgain value. As soon as the call utilizes two FXO card at the same time, the "steam engine" sound occurs. On Mon, 2004-06-28 at 16:26 +0100, Whisker, Peter wrote: > Has anyone seen this problem before? > > I have a server with a single X100P card. The audio level is a low, but if I > raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo > test. Not at a high frequency but with a noise that is best described as a > steam engine starting up. It then starts to clip and crackle. If I bring the > gain down to Rx=-2.0 and Tx=0.0 or lower then it settles down but it is very > very quiet. > > I have tried the latest CVS Head with echotraining=800 set and also complied > with the aggressive echo cancelling, but nothing seems to help. > > Ideas welcome! > > Many thanks > Peter Whisker > > This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap X100P oscillation
Thanks for the responses. I have tried it with aggressive cancellation both on and off. I think that "on" helps a tiny bit. I'm glad that Mike Benoit as seen something similar, but of course sorry that he is suffering like me! It is worse when I have a "Phone-switch-X100P-IAX-Internet-IAX-X100P-switch-Phone" link set up. The other thing which may have helped a bit is using a large set of IAX Jitter buffers. It may be the latency which is helping though, rather than the anti-jitter aspects. Peter -Original Message- From: Brian McSpadden [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 22:57 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zap X100P oscillation Try recompiling your zaptel package without the aggressive echo cancellation enabled. I have aggressive cancellation help before, I but I have also seen it hurt things before. Brian On Mon, 28 Jun 2004 16:26:32 +0100, Whisker, Peter <[EMAIL PROTECTED]> wrote: > I have tried the latest CVS Head with echotraining=800 set and also complied > with the aggressive echo cancelling, but nothing seems to help. > > Ideas welcome! > > Many thanks > Peter Whisker > > This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap X100P oscillation
Has anyone seen this problem before? I have a server with a single X100P card. The audio level is a low, but if I raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo test. Not at a high frequency but with a noise that is best described as a steam engine starting up. It then starts to clip and crackle. If I bring the gain down to Rx=-2.0 and Tx=0.0 or lower then it settles down but it is very very quiet. I have tried the latest CVS Head with echotraining=800 set and also complied with the aggressive echo cancelling, but nothing seems to help. Ideas welcome! Many thanks Peter Whisker This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
BT do occasionally tweak up line gain a bit if you keep complaining that you have a modem and are getting a very slow speed. I have had a 40k V90 come up to 48k after this was done on my line at home (System X switch). You have to get a sympathetic engineer though - frequently they will tell you that it can't be done. Peter -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: 25 June 2004 14:33 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X101P on a UK BT line txgain issue > > I do get echo, lots of it, I am waiting until the new patch they > > are all on about on the list gets into a stable release, then I > > will upgrade and see if that does the trick. Not likely the patch will get applied to the Stable release since its been stated several times that's all but dead. > The patch didn't seem to work for me. > > > I am told that some of the echo may be to do with a mismatch in the > > impedance with the BT line. There are at least several sources of echo, which have been noted several times in the past six months or so: a. echo can in * not functioning correctly in some circumstances (patch) b. mismatched x100p -> pstn line c. pstn line problems (eg, imbalance between tip/ring and ground) d. 2-wire to 4-wire conversion along the end-to-end voice path Any single pstn line could have one or more of those happening. > Problem is do we really want BT messing with gain there end and impedance > cos it might mess our ADSL lines up =) I know im on the limits. The telco isn't going to mess with changing impedance on their stuff as that would require re-engineering their entire outside plant (cables), line interfaces within the CO, etc. You couldn't pay them enough money to do it. Its also highly unlikely they have any technician adjustable transmission level adjustments on ordinary pstn CO line interfaces, as those are engineered to 'standards', and manufacturing engineers typically don't want support technicians to muck with those for lots of very valid reasons. I'm in the US and don't have any real clue what the UK standards are for impedance, transmission levels, etc. (It would be somewhat interesting to here from someone who knows for sure what those are.) The x100p card (from digium) uses the Silicon Labs ( www.silabs.com ) 3012 chip to interface with the pstn line, the 3021 chip to interface the 3012 (analog-to-digital converter) to the Tigerjet PCI controller. The 3012 is responsible for matching pstn line impedance. The spec sheets at their site tend to suggest the 3012 was built to interface to 600 ohm pstn lines and is not adjustable/setable to other values. If the UK pstn lines are not 600 ohm impedance, then its unlikely the x100p is going to properly match up with UK lines from an impedance matching perspective. However, imedance mismatches have to be rather dramatic to cause a lot of echo. The tdm fxo module uses the 3019 and 3050 chipset, where the 3019 pstn line interface chip has many different pstn line impedance settings including 600, 900, 270, 220, 370, 320, 275, 120, 350, etc, ohms. Have no clue which countries use which settings, but obviously Silicon Labs intended this chip set to operate in different countries, whereas the 3012 spec sheet doesn't seem to support those objectives. So, backing into exactly what is causing the echo in the three UK cases noted yesterday on this list... - not likely to be "d" (2-wire to 4-wire conversion along the end-to-end voice path) as that would impact all telco users, not just * users. - item "c" (pstn line problems) can contribute to echo depending upon how bad the pstn line actually happens to be. Most telco's have the equipment to measure line quality, however most will stop at the cable entrance to your home/business, leaving you to guess at what's happening inside. - item "b" (mismatched x100p -> pstn line) can contribute, but without knowing the exact specs (and probably more info from Silicon Labs), its impossible to guess at this one. - item "a" (echo can in *) is still a very real possibility, and Mark is about the only person I know of that has the knowledge of the spec's and * to weight in on this one. One of the methods that I used to help determine whether my tdm echo was a pstn line or * issue was to eval echo on three different pstn lines using the exact same physical * port (x100p). If the lines are clean from an analog phone perspective (eg, no hum, no noise) and the lines cause equal echo when used with *, then its highly likely the issue is either "a" or "b". If the answers to "b" rule out impedance mismatches, then "a" is likely. Why? Your not likely to have multiple pstn lines with exactly the same fault. (Could happen but not likely.) Second, if your pstn line is also a DSL line with appropriate filters, the DSL line is far more critical of pstn imperfections then is the * interface to the analog pstn line. In the majority o
RE: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
I get a problem with what appears to be a slow oscillation on the line if the rxgain + txgain adds up to more than -1db. If I use rxgain=-1.0 and txgain=0.0, it doesn't oscillate but the levels are far too low. The card is an X100P. The oscillation (even on the standard built-in Asterisk echo test) comes over as a loud hiss and crackle at about 1-2 per second making the line unusable. I have tried the settings below and it is dreadful. Any ideas? I am using yesterday's CVS Head. Peter -Original Message- From: Chris Bond [mailto:[EMAIL PROTECTED] Sent: 24 June 2004 18:03 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X101P on a UK BT line txgain issue > I am finding that I have to increase the txgain in zapata.conf to 8 when > my X101P is connected to my BT phone line, otherwise people can hardly > hear me. This then gives echo issues. Im having the same issue so far im on rxgain=2.0 and txgain=6.0. Seems to work perfectly apart from the echo issue. Im just about to checkout the latest cvs and apply the echotraining=800 Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: -- [Asterisk-Users] Serious issues with current CVS?
I had a compile problem with the CVS I downloaded on 21 June. I have a Debian box with 2.4.18 kernel (version needed for support of Conexant ADSL). There is a difficulty with Zaptel build regarding HDLC detection. It tries to build it in and then results in unresolved kernel symbols and fails to load. I have had to comment out the entire HDLC defines in zconfig.h to get a driver to install at all (ie the stuff below:) The previous cvs download from a few weeks ago (June 9th) compiled and loaded fine. /* We now use the linux kernel config to detect which options to use */ /* You can still override them below */ /* #if defined(CONFIG_HDLC) || defined(CONFIG_HDLC_MODULE) #define CONFIG_ZAPATA_NET #if LINUX_VERSION_CODE <= KERNEL_VERSION(2,4,20) #define CONFIG_OLD_HDLC_API #else #if LINUX_VERSION_CODE >= KERNEL_VERSION(2,6,3) #define HDLC_MAINTAINERS_ARE_MORE_STUPID_THAN_I_THOUGHT #endif #endif #endif #ifdef CONFIG_PPP #define CONFIG_ZAPATA_PPP #endif */ They also seem to have broken the SayUnixTime app - it can't cope with a 'digits/at' bit in the middle of the string any more. I have had to change exten => 123,1,SayUnixTime(||AdBY 'digits/at' IM) to exten => 123,1,SayUnixTime(||AdBY) exten => 123,2,Playback(digits/at) exten => 123,3,SayUnixTime(||IM) Peter -Original Message- From: Dave Cotton [mailto:[EMAIL PROTECTED] Sent: 24 June 2004 06:45 To: Asterisk List Subject: Re: -- [Asterisk-Users] Serious issues with current CVS? For me chan_capi set up the call but no sound available. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
