Re: [asterisk-users] Loudness of recorded wav-audio
I tried to ajust the tx- and rxgain for the sip peer in sip.conf. And restarted the asterisk. But it takes no effect. Any suggestion? 2011/3/4 Danny Nicholas da...@debsinc.com Defaults are 0.0 (leave volume unchanged) +values make volume louder, - softer. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Friday, March 04, 2011 8:55 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Loudness of recorded wav-audio Could yoz tell me the default value of rxgain or txgain, if there is no rxgain or txgain in conf-data defined? Von meinem iPad gesendet Am 04.03.2011 um 15:34 schrieb Danny Nicholas da...@debsinc.com: In sip.conf, add rxgain=-4.0 to the peer. This (feel free to correct) should reduce the incoming volume by 4 decibels. You’ll have to do a “sip reload” for this to take effect. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Friday, March 04, 2011 8:33 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Loudness of recorded wav-audio Thank you! How can I reduce the RXgain? Am 04.03.2011 um 15:21 schrieb Danny Nicholas da...@debsinc.com da...@debsinc.com: -- *From:* asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Friday, March 04, 2011 2:31 AM *To:* asterisk-users@lists.digium.com asterisk-users@lists.digium.com asterisk-users@lists.digium.com *Subject:* [asterisk-users] Loudness of recorded wav-audio Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix two options are: 1. reduce RXgain – assuming your are using Record() command 2. use sox to reduce the volume; something like sox –v .8 file1.wav file2.wav -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hellohttp://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usershttp://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loudness of recorded wav-audio
This settings are for ISDN configurations I think. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Monday, March 07, 2011 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudness of recorded wav-audio I tried to ajust the tx- and rxgain for the sip peer in sip.conf. And restarted the asterisk. But it takes no effect. Any suggestion? 2011/3/4 Danny Nicholas da...@debsinc.com Defaults are 0.0 (leave volume unchanged) +values make volume louder, - softer. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Friday, March 04, 2011 8:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudness of recorded wav-audio Could yoz tell me the default value of rxgain or txgain, if there is no rxgain or txgain in conf-data defined? Von meinem iPad gesendet Am 04.03.2011 um 15:34 schrieb Danny Nicholas da...@debsinc.com: In sip.conf, add rxgain=-4.0 to the peer. This (feel free to correct) should reduce the incoming volume by 4 decibels. You’ll have to do a “sip reload” for this to take effect. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Friday, March 04, 2011 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudness of recorded wav-audio Thank you! How can I reduce the RXgain? Am 04.03.2011 um 15:21 schrieb Danny Nicholas da...@debsinc.com: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Friday, March 04, 2011 2:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Loudness of recorded wav-audio Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix two options are: 1. reduce RXgain – assuming your are using Record() command 2. use sox to reduce the volume; something like sox –v .8 file1.wav file2.wav -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loudness of recorded wav-audio
it should work for sip channel too. I recorded the downlink channel in wav-format. Does the rx or txgain ajusting only work with alaw or ulaw? 2011/3/7 Faisal Hanif fai...@vopium.com This settings are for ISDN configurations I think. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Monday, March 07, 2011 6:07 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Loudness of recorded wav-audio I tried to ajust the tx- and rxgain for the sip peer in sip.conf. And restarted the asterisk. But it takes no effect. Any suggestion? 2011/3/4 Danny Nicholas da...@debsinc.com Defaults are 0.0 (leave volume unchanged) +values make volume louder, - softer. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Friday, March 04, 2011 8:55 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Loudness of recorded wav-audio Could yoz tell me the default value of rxgain or txgain, if there is no rxgain or txgain in conf-data defined? Von meinem iPad gesendet Am 04.03.2011 um 15:34 schrieb Danny Nicholas da...@debsinc.com: In sip.conf, add rxgain=-4.0 to the peer. This (feel free to correct) should reduce the incoming volume by 4 decibels. You’ll have to do a “sip reload” for this to take effect. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Friday, March 04, 2011 8:33 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Loudness of recorded wav-audio Thank you! How can I reduce the RXgain? Am 04.03.2011 um 15:21 schrieb Danny Nicholas da...@debsinc.com: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Felix Dong *Sent:* Friday, March 04, 2011 2:31 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Loudness of recorded wav-audio Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix two options are: 1. reduce RXgain – assuming your are using Record() command 2. use sox to reduce the volume; something like sox –v .8 file1.wav file2.wav -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Lync / Call Center Transfer / Refer
This is a problem in chan_sip.c After REFER asterisk does not notify dialplan or AGI of REFER. I've tried to convince asterisk developers this is a problem but they only offered me 3 solutions: 1. Fix it yourself 2. Pay someone to fix it 3. Try to convince enough people that this is a problem and it may get fixed. BTW this is not a simple fix, it would require architectural changes in asterisk. On Sun, Mar 6, 2011 at 9:32 PM, Louis Carreiro carreir...@gmail.com wrote: So does anyone have any other thoughts about this? I've done some searching through the bug tracker for Asterisk but haven't seen anything related to refer's failing. Does anyone know of a specific issue number for this? If not, is this a valid bug to submit? Also, does anyone remember an Asterisk version that this worked on? Thanks all! On Fri, Mar 4, 2011 at 1:35 PM, Louis Carreiro carreir...@gmail.comwrote: Ha! Thanks Vip! Sorry about not including my version numbers too. On my production box I'm using 1.8.3 (that's the debug from the original email). On my demo box I just build I'm using 1.8 SVN-trunk-r309404 and that's what generated these logs. I'm not sure if this is a chan_sip.c problem or if this is a dial plan problem. So digging in a bit deeper, Asterisk is receving the real REFER message. The REFER-TO: sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad2 787?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d% 3Bto-tag%3D8be38bb187 is accurate and in chan_sip.c it knows how to manipulate it. It does grab the from-tag and to-tag and parses the data. On one of the lines below you can see it says Looking for Call ID: 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 (Checking From) --From tag 15826bef52 --To-tag as41bacc0b. Then it moves on to bridging the peers/channels together. It's not until later that I get the final SIP/2.0 481 Call leg/transaction does not exist which doesn't make sense to me. Also, the Lync client says Call was not transferred because [Original Extension] cannot be reached and may be offline. SNIP - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loudness of recorded wav-audio
You could always just use sox to adjust the levels -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Forcing Asterisk/Zaptel to wait until calleeanswers?
On Thu, 3 Mar 2011 08:42:36 -0600, Danny Nicholas
da...@debsinc.com wrote:
Having traversed this rabbit-hole the answer is that it depends on your
carrier. If they offer call-supervision, asterisk can wait for pickup on
the other side. The resolution I came up with for my offering:
I was going through the script this afternoon, and have a couple of
questions:
1. Why use instead of = to compare the extension with SIP?
exten = s,n,Gotoif($[${EXTEN} SIP]?start)
2. According to www.voip-info.org/wiki/view/Asterisk+cmd+Wait, Wait
only takes one parameter, while WaitExten() can take m, which plays
music on hold: Is the wiki out of date and you are using a more recent
version of Asterisk than the 1.4 I have?
exten = s,n,Wait(9,m)
Thank you.
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Re: [asterisk-users] Mirrors in Australia?
On Mon, Mar 7, 2011 at 2:11 AM, Stuart Longland redhat...@gentoo.orgwrote: http://mirror.aarnet.edu.au/pub/gentoo/distfiles/asterisk-1.8.3.tar.gz I haven't checked that URL, but it should be correct. That, and that mirror should be unmetered if you're on a university network. Thanks mate, it works :) Although we were in a bit of a hurry so we bit the bullet and downloaded it from downloads.asterisk.org and it had blazing speed. Downloaded at about 1.93 MB/s But will use the one you suggested in the future. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help on incoming
Hi, I am using IAXmodem + hylafax to do outgoing and incoming fax with asterisk. I wonder how to write a dialplan to differentiate incoming call or fax. I am sharing a line for both voice and fax. CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Forcing Asterisk/Zaptel to wait untilcalleeanswers?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Monday, March 07, 2011 8:14 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [1.4] Forcing Asterisk/Zaptel to wait
untilcalleeanswers?
On Thu, 3 Mar 2011 08:42:36 -0600, Danny Nicholas
da...@debsinc.com wrote:
Having traversed this rabbit-hole the answer is that it depends on your
carrier. If they offer call-supervision, asterisk can wait for pickup on
the other side. The resolution I came up with for my offering:
I was going through the script this afternoon, and have a couple of
questions:
1. Why use instead of = to compare the extension with SIP?
exten = s,n,Gotoif($[${EXTEN} SIP]?start)
2. According to www.voip-info.org/wiki/view/Asterisk+cmd+Wait, Wait
only takes one parameter, while WaitExten() can take m, which plays
music on hold: Is the wiki out of date and you are using a more recent
version of Asterisk than the 1.4 I have?
exten = s,n,Wait(9,m)
Thank you.
#1 is Lazy notation to say ${EXTEN} starts with SIP (as opposed to Local
or DAHDI)
#2 Might just be a typo on my part. I frequently switch usage between Wait()
and WaitExten().
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Re: [asterisk-users] Asterisk - Lync / Call Center Transfer / Refer
On 03/04/2011 12:35 PM, Louis Carreiro wrote: Ha! Thanks Vip! Sorry about not including my version numbers too. On my production box I'm using 1.8.3 (that's the debug from the original email). On my demo box I just build I'm using 1.8 SVN-trunk-r309404 and that's what generated these logs. I'm not sure if this is a chan_sip.c problem or if this is a dial plan problem. If your version string is 'SVN-trunk-r309404', you are not using 1.8, you are using 'trunk'. If you want to follow the 1.8 Subversion branch, you need to checkout that branch, not trunk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good tutorials for setting up Asterisk SNMP and Cacti for remote monitoring?
Thanks. This comes really close. My asterisk currently has snmp setup properly and I can see it shows the output when I do snmpwalk command. I am stuck at Cacti end. Wondering what to do to setup the asterisk remote end. The tutorial you provided is for Nagios (which I tend to stay away due to it's install complications) Thanks again, On Mon, Mar 7, 2011 at 2:14 AM, Faisal Hanif fai...@vopium.com wrote: http://www.danielaliaman.com/blog/files/AsteriskSNMPtutorial.pdf *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* Sunday, March 06, 2011 10:59 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Any good tutorials for setting up Asterisk SNMP and Cacti for remote monitoring? Hi Everyone, I have been searching the web and I don't know if SNMP is just that complex to setup or that not many people use SNMP to monitor Asterisk but the information is scattered all over. I have got to the point to configure SNMP with Asterisk and then it's all confusing from there on to actually see the graphs in Cacti. I would appreciate it if you can post your steps or point me to a good guide posted somewhere on the web. I have followed this but it's not complete: http://www.voipphreak.ca/2008/10/28/asterisk-snmp-with-cacti-howto-upgraded-for-asterisk-16-and-ubuntu/ ***Please don't post any smart-aleck comments like google it. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gosub and 'h' (again?)
Thanks for your reply - but I did it a slightly different way:
Nevermind - I've re-written my dialplan so that all subs are in one
context. Now I only need 1 more line of code.
Thanks anyway :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faisal
Hanif
Sent: 06 March 2011 01:54
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Gosub and 'h' (again?)
Well a solution for you to put original context name in variable and
then use that variable in goto statement on h.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew
Thomas
Sent: Friday, March 04, 2011 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Gosub and 'h' (again?)
Problem as follows:
[default]
exten = 777,1,Gosub(sub,1,1)
exten = 777,n,Hangup()
exten = h,1,NoOp(hung up in 'default' context)
[sub]
exten = 1,1,NoOp(in sub)
exten = 1,n,Playback(tt-monkeys)
exten = 1,n,Return()
exten = h,1,NoOp(hung up in 'sub' context)
This works fine if the caller listens to all the 'tt-monkeys' and let's
the system hangup. You get the hang up in the 'default' context.
But, if the caller hangs up BEFORE the end of 'tt-monkeys' - the hang up
occurs in the 'sub' context. This means that I have to force each sub
routine to go to the main contexts 'h' extension ('exten =
h,1,Goto(default,h,1)' in this case).
Is there a way to tell * to use the default 'h' extension on a hang up -
rather than having to put a 'h' in to every separate sub routine?
I know Tilghman said ...Gosub, on the other hand, isn't really even
executing at that point, so there isn't a code path that exists whereby
the Gosub can empty the return stack and return to the original
place [see
http://lists.digium.com/pipermail/asterisk-dev/2008-May/033153.html].
But what does that mean in English ;)?
Thanks
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Re: [asterisk-users] Asterisk - Lync / Call Center Transfer / Refer
Kevin, I had no clue! Thanks for the note! I'll be checking out the 1.8 SVN branch here shortly then for testing! Thanks again! Louis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Monday, March 07, 2011 9:34 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Lync / Call Center Transfer / Refer On 03/04/2011 12:35 PM, Louis Carreiro wrote: Ha! Thanks Vip! Sorry about not including my version numbers too. On my production box I'm using 1.8.3 (that's the debug from the original email). On my demo box I just build I'm using 1.8 SVN-trunk-r309404 and that's what generated these logs. I'm not sure if this is a chan_sip.c problem or if this is a dial plan problem. If your version string is 'SVN-trunk-r309404', you are not using 1.8, you are using 'trunk'. If you want to follow the 1.8 Subversion branch, you need to checkout that branch, not trunk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Forcing Asterisk/Zaptel to wait untilcalleeanswers?
On Mon, 7 Mar 2011 08:20:26 -0600, Danny Nicholas
da...@debsinc.com wrote:
#1 is Lazy notation to say ${EXTEN} starts with SIP (as opposed to Local
or DAHDI)
#2 Might just be a typo on my part. I frequently switch usage between Wait()
and WaitExten().
Thanks for the clarification.
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Re: [asterisk-users] Configuration for Multiple PRI cards
On Sat, Mar 05, 2011 at 08:14:37PM +0200, Elliot Murdock wrote: Hello All, How does one go about creating a dahdi configuration file for multiple PRI cards? 1. vi 2. dahdi_genconf handles the common case quite well and will normally be a good start. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Forcing Asterisk/Zaptel to wait untilcalleeanswers?
On Monday 07 March 2011 08:20:26 Danny Nicholas wrote:
On Monday 07 March 2011 08:14:27 Gilles wrote:
1. Why use instead of = to compare the extension with SIP?
exten = s,n,Gotoif($[${EXTEN} SIP]?start)
#1 is Lazy notation to say ${EXTEN} starts with SIP (as opposed to
Local or DAHDI)
Then you probably want ${CHANNEL}, not ${EXTEN}. ${EXTEN} is always
going to be s, which is always greater than SIP.
--
Tilghman
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Re: [asterisk-users] [1.4] Forcing Asterisk/Zaptel to waituntilcalleeanswers?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Monday, March 07, 2011 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [1.4] Forcing Asterisk/Zaptel to
waituntilcalleeanswers?
On Monday 07 March 2011 08:20:26 Danny Nicholas wrote:
On Monday 07 March 2011 08:14:27 Gilles wrote:
1. Why use instead of = to compare the extension with SIP?
exten = s,n,Gotoif($[${EXTEN} SIP]?start)
#1 is Lazy notation to say ${EXTEN} starts with SIP (as opposed to
Local or DAHDI)
Then you probably want ${CHANNEL}, not ${EXTEN}. ${EXTEN} is always
going to be s, which is always greater than SIP.
--
Tilghman
I said it was Lazy, and I did this back on 1.4.21 under Zaptel. The ${EXTEN}
was sufficient for my purposes, though looking back I see it is not
syntactically correct.
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Re: [asterisk-users] Configuration for Multiple PRI cards
Basically each PRI card will be configured as g0, g1 and so on. Try this link http://www.voip-info.org/wiki/view/Asterisk+PRI http://www.voip-info.org/wiki/view/Asterisk+PRIif you are using sangoma cards then try http://wiki.sangoma.com On Mon, Mar 7, 2011 at 10:00 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Sat, Mar 05, 2011 at 08:14:37PM +0200, Elliot Murdock wrote: Hello All, How does one go about creating a dahdi configuration file for multiple PRI cards? 1. vi 2. dahdi_genconf handles the common case quite well and will normally be a good start. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuration for Multiple PRI cards
Un-top-posting... On Sat, Mar 05, 2011 at 08:14:37PM +0200, Elliot Murdock wrote: How does one go about creating a dahdi configuration file for multiple PRI cards? On Mon, Mar 7, 2011 at 10:00 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: 1. vi 2. dahdi_genconf handles the common case quite well and will normally be a good start. On Mon, 7 Mar 2011, Gopalakrishnan A.N wrote: Basically each PRI card will be configured as g0, g1 and so on. Group is not bound by card or span. It is applied to a range of channels. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail2ban + asterisk
On Sat, Mar 5, 2011 at 8:54 PM, Pezhman Lali l...@lopl.net wrote: Dear this note is only for fresh administrators don't think about asterisk security. Do you know where you go to 'un-ban' an IP if they made some mistake? Using webmin I was not able to find the IP address that was was banned. Thanks, Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error loading module 'res_fax_digium.so'
Hi, I installed Free FAX for Asterisk on my home Asterisk 1.8(Centos5.5) server. Everything seems fine but I just saw this WARNING shows up in the log every time I start the asterisk: /[2011-03-07 10:50:41] WARNING[13429] loader.c: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_unregister/ And in later in the log file, I also saw: /[2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Digium FAX technology module version 1.8.0_1.3.0, Copyright (C) 2008-2009 Digium, Inc. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: This module is supplied under a commercial license granted by Digium, Inc. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Please see the full license text supplied by the accompanying [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: register utility, or ask for a copy from Digium. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: This product includes software developed by the OpenSSL Project [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Copyright (C) 1998-2008 The OpenSSL Project/ How can I fix this WARNING error? Thanks. Jian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.8.3 - IAX - echo - jitterbuffer
I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On the office side, they hear an echo of _their_ speech, not mine. The office uses sip-providers generally without any echo problem. Where do I start to figure this out? How do I narrow it down? Can I figure out if it is an iaxagent problem? Could using jitterbuffer cause this? Thanks for any help. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error loading module 'res_fax_digium.so'
On 03/07/2011 12:58 PM, Jian Gao wrote: Hi, I installed Free FAX for Asterisk on my home Asterisk 1.8(Centos5.5) server. Everything seems fine but I just saw this WARNING shows up in the log every time I start the asterisk: /[2011-03-07 10:50:41] WARNING[13429] loader.c: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_unregister/ And in later in the log file, I also saw: /[2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Digium FAX technology module version 1.8.0_1.3.0, Copyright (C) 2008-2009 Digium, Inc. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: This module is supplied under a commercial license granted by Digium, Inc. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Please see the full license text supplied by the accompanying [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: register utility, or ask for a copy from Digium. [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: This product includes software developed by the OpenSSL Project [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [2011-03-07 10:50:43] NOTICE[13429] res_fax_digium.c: Copyright (C) 1998-2008 The OpenSSL Project/ How can I fix this WARNING error? You can follow the instructions with the product and ensure that res_fax.so is loaded before res_fax_digium.so. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add
On 03/07/2011 03:35 PM, RR wrote: Hello all, mmm a bit embarrassing about not having a clue as to why we're getting this error on make of 1.8.3 [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o - libdb1.a [LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a - asterisk astobj2.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' astobj2.o:/usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: more undefined references to `__sync_fetch_and_add_4' follow utils.o: In function `ast_atomic_dec_and_test': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:635: undefined reference to `__sync_sub_and_fetch_4' utils.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking Have you specified any '-march' or '-mcpu' options to the compiler? This sort of thing can occur if you are building for a plain-jane i386 processor or something similar. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8.3 - IAX - echo - jitterbuffer
On 03/07/2011 04:15 PM, sean darcy wrote: I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On the office side, they hear an echo of _their_ speech, not mine. The office uses sip-providers generally without any echo problem. Where do I start to figure this out? How do I narrow it down? Can I figure out if it is an iaxagent problem? Could using jitterbuffer cause this? This is probably acoustic echo from your phone. The jitterbuffer has nothing to do with this. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add
On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 03/07/2011 03:35 PM, RR wrote: Hello all, mmm a bit embarrassing about not having a clue as to why we're getting this error on make of 1.8.3 [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o - libdb1.a [LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a - asterisk astobj2.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' astobj2.o:/usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: more undefined references to `__sync_fetch_and_add_4' follow utils.o: In function `ast_atomic_dec_and_test': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:635: undefined reference to `__sync_sub_and_fetch_4' utils.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking Have you specified any '-march' or '-mcpu' options to the compiler? This sort of thing can occur if you are building for a plain-jane i386 processor or something similar. Hey Kevin, nope, nothing...just doing the standard ./configure; make menuselect; make this is a Sun Sparc v240 machine running Debian 6.0 Squeeze sparc64-smp kernel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add
On 03/07/2011 04:31 PM, RR wrote: On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: Please do not reply directly to posters on the mailing list unless they request it. On 03/07/2011 03:35 PM, RR wrote: Hello all, mmm a bit embarrassing about not having a clue as to why we're getting this error on make of 1.8.3 [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o - libdb1.a [LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a - asterisk astobj2.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' astobj2.o:/usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: more undefined references to `__sync_fetch_and_add_4' follow utils.o: In function `ast_atomic_dec_and_test': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:635: undefined reference to `__sync_sub_and_fetch_4' utils.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking Have you specified any '-march' or '-mcpu' options to the compiler? This sort of thing can occur if you are building for a plain-jane i386 processor or something similar. Hey Kevin, nope, nothing...just doing the standard ./configure; make menuselect; make this is a Sun Sparc v240 machine running Debian 6.0 Squeeze sparc64-smp kernel Someone with SPARC experience will have to chime in then... for some reason the configure script has determined that your compiler provides atomic instructions, but they aren't being found at link time. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add
On Mon, Mar 7, 2011 at 5:34 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 03/07/2011 04:31 PM, RR wrote: On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: Please do not reply directly to posters on the mailing list unless they request it. Sorry, the default on my gmail is Reply All and usually I remove relevant parties from the To/Cc: headers, guess missed it this time. Wasn't intentional. On 03/07/2011 03:35 PM, RR wrote: Hello all, mmm a bit embarrassing about not having a clue as to why we're getting this error on make of 1.8.3 [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o - libdb1.a [LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a - asterisk astobj2.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' astobj2.o:/usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: more undefined references to `__sync_fetch_and_add_4' follow utils.o: In function `ast_atomic_dec_and_test': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:635: undefined reference to `__sync_sub_and_fetch_4' utils.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking Have you specified any '-march' or '-mcpu' options to the compiler? This sort of thing can occur if you are building for a plain-jane i386 processor or something similar. Hey Kevin, nope, nothing...just doing the standard ./configure; make menuselect; make this is a Sun Sparc v240 machine running Debian 6.0 Squeeze sparc64-smp kernel Someone with SPARC experience will have to chime in then... for some reason the configure script has determined that your compiler provides atomic instructions, but they aren't being found at link time. Ok...thanks. Is there no way for me to tell the compiler or provide flags in ./configure that can tell it to not do that? Conversely can I use -march and/or -mcpu kind of options to make this compile for my platform? If so, then what would the value be of these options or are there no values for them and one just specifies them? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add
On 03/07/2011 04:41 PM, RR wrote: Someone with SPARC experience will have to chime in then... for some reason the configure script has determined that your compiler provides atomic instructions, but they aren't being found at link time. Ok...thanks. Is there no way for me to tell the compiler or provide flags in ./configure that can tell it to not do that? Conversely can I use -march and/or -mcpu kind of options to make this compile for my platform? If so, then what would the value be of these options or are there no values for them and one just specifies them? The answer to all of those questions is probably 'yes', but that's why I said someone with SPARC experience would have to chime in. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
Hello all, Figured I'd repost this with an edited subject line, to attract attention of people with Debian On Sparc experience. Apologies in advance if this kind of thing is frowned upon :) [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o - libdb1.a [LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a - asterisk astobj2.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' astobj2.o:/usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: more undefined references to `__sync_fetch_and_add_4' follow utils.o: In function `ast_atomic_dec_and_test': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:635: undefined reference to `__sync_sub_and_fetch_4' utils.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
Okay, so here's the configuration I have for MySQL Realtime (Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers = mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 And here's the error messages I get: voip2*CLI realtime mysql status localhost configured for mya2billing@localhost, port 3306 with username a2billinguser. mya2billing configured for mya2billing@localhost, port 3306 with username a2billinguser. [Mar 7 14:38:59] ERROR[15943]: res_config_mysql.c:1575 mysql_reconnect: MySQL RealTime: Failed to connect database server mya2billing on localhost (err 2002). Check debug for more info. [Mar 7 14:38:59] ERROR[15943]: res_config_mysql.c:1575 mysql_reconnect: MySQL RealTime: Failed to connect database server mya2billing on localhost (err 2002). Check debug for more info. This doesn't make any sense. res_mysql.conf contains working mysql credentials that I can verify with running mysql from the command line. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
On Mon, Mar 7, 2011 at 5:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote: Okay, so here's the configuration I have for MySQL Realtime (Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers = mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: Don't know what res_mysql.conf is, I think it should be res_config_mysql.conf? Sorry it's been a LONG time since I configured/used realtime and that also was with ODBC and TDS but I know that the file res_config_mysql.conf should definitely be there HTH \R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 Try adding dbsock = /var/lib/mysql/mysql.sock to the end of this config stanza and see if that helps (or whatever is the actual location of your mysql.sock file). -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
On 03/08/11 08:49, RR wrote: Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking No SPARC expert, but I seem to recall the lowest-common-denominator SPARCs lack things like hardware multiply in the instruction set. Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Another place to ask might be the Debian-SPARC mailing list? -- Stuart Longland (aka Redhatter, VK4MSL) .'''. Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : . . . . . . . . . . . . . . . . . . . . . . .'.' http://dev.gentoo.org/~redhatter :.' I haven't lost my mind... ...it's backed up on a tape somewhere. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
Hello Stuart On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.orgwrote: On 03/08/11 08:49, RR wrote: Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking No SPARC expert, but I seem to recall the lowest-common-denominator SPARCs lack things like hardware multiply in the instruction set. Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Ok, where would I put this -mcpu=v9 in the configure line? I tried ./configure CFLAGS=-mcpu=v9? BTW, at the end of the configure script, it's already detecting the host cpu as sparc64. If that helps. Maybe -march needs to be specified somewhere? Another place to ask might be the Debian-SPARC mailing list? haha funny, I was just writing an email to that list when your email hit my inbox :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuration for Multiple PRI cards
On Mon, Mar 07, 2011 at 10:16:52AM -0800, Steve Edwards wrote: Un-top-posting... On Sat, Mar 05, 2011 at 08:14:37PM +0200, Elliot Murdock wrote: How does one go about creating a dahdi configuration file for multiple PRI cards? On Mon, Mar 7, 2011 at 10:00 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: 1. vi 2. dahdi_genconf handles the common case quite well and will normally be a good start. On Mon, 7 Mar 2011, Gopalakrishnan A.N wrote: Basically each PRI card will be configured as g0, g1 and so on. Group is not bound by card or span. It is applied to a range of channels. And this is actually what dahdi_genconf generates (group=0,11 for the first, group=0,12 for the second, etc.) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
On 03/08/11 09:21, RR wrote: Hello Stuart On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org mailto:redhat...@gentoo.org wrote: Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Ok, where would I put this -mcpu=v9 in the configure line? I tried ./configure CFLAGS=-mcpu=v9? Normally it's specified in the environment; so maybe CFLAGS=-mcpu=v9 ./configure… BTW, at the end of the configure script, it's already detecting the host cpu as sparc64. If that helps. Maybe -march needs to be specified somewhere? Maybe, the fact that it detected 'sparc64' probably is more a case of telling the build system that the system is big-endian, requires that data structures be 64-bit aligned, etc. Use of features that weren't in the first SPARC is an optional extra. Another place to ask might be the Debian-SPARC mailing list? haha funny, I was just writing an email to that list when your email hit my inbox :) Telepathy; seems we think alike. :-D Must be due to me being from the same part of the world. -- Stuart Longland (aka Redhatter, VK4MSL) .'''. Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : . . . . . . . . . . . . . . . . . . . . . . .'.' http://dev.gentoo.org/~redhatter :.' I haven't lost my mind... ...it's backed up on a tape somewhere. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
On Mon, Mar 7, 2011 at 4:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 Try adding dbsock = /var/lib/mysql/mysql.sock to the end of this config stanza and see if that helps (or whatever is the actual location of your mysql.sock file). Hmm. This appears to have fixed the problem, even though I swear I've done this already. (and for anyone reading this, on Debian the file is mysqld.sock) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
On Mon, Mar 7, 2011 at 6:31 PM, Stuart Longland redhat...@gentoo.orgwrote: On 03/08/11 09:21, RR wrote: Hello Stuart On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org mailto:redhat...@gentoo.org wrote: Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Ok, where would I put this -mcpu=v9 in the configure line? I tried ./configure CFLAGS=-mcpu=v9? Normally it's specified in the environment; so maybe CFLAGS=-mcpu=v9 ./configure… I tried both ways, my way and yours i.e. setting them as env variables and it still gets that error. Also found some other stuff on the net related to that in different context but none of those work for me. Some where in some old debian archives there's some mention of the Boost libraries and the flag that must be used on Sparc with Boost libraries. Although it also says that it was fixed in some later release which was back in 2008, so am assuming that fix is still in place in Squeeze. BTW, at the end of the configure script, it's already detecting the host cpu as sparc64. If that helps. Maybe -march needs to be specified somewhere? Maybe, the fact that it detected 'sparc64' probably is more a case of telling the build system that the system is big-endian, requires that data structures be 64-bit aligned, etc. Use of features that weren't in the first SPARC is an optional extra. Ok, if that doesn't help then another interesting insight is that in config.log, it says that the response to 'arch' and 'arch -k' commands is 'unknown'. Don't know if that helps. Another place to ask might be the Debian-SPARC mailing list? haha funny, I was just writing an email to that list when your email hit my inbox :) Telepathy; seems we think alike. :-D Must be due to me being from the same part of the world. Possibly :) although I have found that there's not a lot of activity in that list on a regular basis. So not sure if my problem will get resolved there or not :( -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8.3 - IAX - echo - jitterbuffer
I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On the office side, they hear an echo of _their_ speech, not mine. The office uses sip-providers generally without any echo problem. Where do I start to figure this out? How do I narrow it down? Can I figure out if it is an iaxagent problem? Could using jitterbuffer cause this? One thing you must consider, is that this echo they're hearing may be entirely an acoustic one, within (or around) the Droid itself. It's very possible for the microphone in a handset to pick up sound being emitted by the handset's speaker. This acoustic feedback can occur within the handset itself (sound from the speaker leaks through the chassis of the handset and reaches the microphone from behind), via mechanical coupling through the handset body, or by the mic picking up the sound from the outside (after it has come out of the speaker into the air). The best way to determine whether this is the case, is probably to shut down the speaker and isolate the mic... plug in an earphone which has a separate mic on its cord, and see if the callers still report the echo. If they do, it's due to electronic or digital goofs somewhere, but if they do not, it's due to acoustic feedback at the handset. There are (in principle) three ways to reduce or eliminate the echo: - Damp it out physically - block the acoustic feedback pathways. In a small USB phone handset I have, I found it necessary to stuff the open spaces inside the handset with cotton and foam, to block the back-wave from the speaker before it reached the microphone. - Use software which has some sort of VOX (voice-operated switch) detection or squelching... so that when the sound level at the microphone is less than you'd get by speaking into the mic, the handset cuts off the mic audio pathway entirely, and sends only silence (or sends nothing at all, although Asterisk doesn't always deal gracefully with this). - Use a better handset. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip/google
http://www.disruptivetelephony.com/2011/03/google-voice-now-offers-sip-addresses-for-calling-directly-over-ip.htmlNice ;) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8.3 - IAX - echo - jitterbuffer
On 03/07/2011 05:26 PM, Kevin P. Fleming wrote: On 03/07/2011 04:15 PM, sean darcy wrote: I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On the office side, they hear an echo of _their_ speech, not mine. The office uses sip-providers generally without any echo problem. Where do I start to figure this out? How do I narrow it down? Can I figure out if it is an iaxagent problem? Could using jitterbuffer cause this? This is probably acoustic echo from your phone. The jitterbuffer has nothing to do with this. Yup. Turning down the volume on the call reduces the echo. Of course, now I can barely hear the office! I can keep the volume up on standard calls from the Droid X, which suggests that Android has some echo cancelling on phone calls. I'll try to see if the developer of iaxagent can do anything. BTW, if you haven't, try iaxagent on your phone. It's a very clever use of the iax protocol and leverages iax's strengths. iax makes a lot of sense on mobiles, dealing with the NAT issues from inconsistent access points easily. Thanks for the help. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM410P dahdi driver == no lights?
Hello, I have just installed an Asterisk server with a Digium TDM410P card with 3 FXO modules (no module in the 4th slot). It's lived on two different machines (a test machine, which had Linux kernel 2.6.28, and a new dedicated machine which has Linux kernel 2.6.32). On the test machine (2.6.28), I used the Zaptel drivers. Once the kernel modules were loaded, the lights on the TDM410P came on green for the installed FXO modules. On the new server, the Zaptel drivers wouldn't build so I switched over to dahdi. Everything seems to be working, EXCEPT there are no lights on the TDM410P! I guess I can ignore that the lights aren't lit up, because it seems to be functioning as expected (I can dial out and receive incoming calls)...but it's disconcerting that the lights aren't on. Yes, the Molex power connector is connected (although I think that's only needed by FXS modules). I've tried google searches but haven't found anything mentioning this odd behavior. Is this expected? Many thanks, ~Brian Henning -- Brian Henning, Software Engineer /\Pine Research Instrumentation //\\ 5908 Triangle Drive ///\\\ Raleigh, NC 27617 USA || ||phone: 919.782.8320 fax: 919.782.8323 email: bhenn...@pineinst.com -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add
On Mon, Mar 7, 2011 at 5:45 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 03/07/2011 04:41 PM, RR wrote: Someone with SPARC experience will have to chime in then... for some reason the configure script has determined that your compiler provides atomic instructions, but they aren't being found at link time. Ok...thanks. Is there no way for me to tell the compiler or provide flags in ./configure that can tell it to not do that? Conversely can I use -march and/or -mcpu kind of options to make this compile for my platform? If so, then what would the value be of these options or are there no values for them and one just specifies them? The answer to all of those questions is probably 'yes', but that's why I said someone with SPARC experience would have to chime in. Ok, so this is solved! The culprit was the the line mcpu=v8 in the Makefile. Comment that out, and it makes properly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail2ban + asterisk
On Mon, Mar 7, 2011 at 9:15 AM, Jamie A. Stapleton jstaple...@computer-business.com wrote: iptables -L -v will give you the IP address that was banned -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thanks Jamie, I will look around to see the steps to clear an IP. Do you know if you can do this through webmin? I know there is an iptables plug-in. -Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Solved] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
On Mon, Mar 7, 2011 at 6:48 PM, RR ranjt...@gmail.com wrote: On Mon, Mar 7, 2011 at 6:31 PM, Stuart Longland redhat...@gentoo.orgwrote: On 03/08/11 09:21, RR wrote: Hello Stuart On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org mailto:redhat...@gentoo.org wrote: Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Ok, where would I put this -mcpu=v9 in the configure line? I tried ./configure CFLAGS=-mcpu=v9? Normally it's specified in the environment; so maybe CFLAGS=-mcpu=v9 ./configure… I tried both ways, my way and yours i.e. setting them as env variables and it still gets that error. Also found some other stuff on the net related to that in different context but none of those work for me. Some where in some old debian archives there's some mention of the Boost libraries and the flag that must be used on Sparc with Boost libraries. Although it also says that it was fixed in some later release which was back in 2008, so am assuming that fix is still in place in Squeeze. BTW, at the end of the configure script, it's already detecting the host cpu as sparc64. If that helps. Maybe -march needs to be specified somewhere? Maybe, the fact that it detected 'sparc64' probably is more a case of telling the build system that the system is big-endian, requires that data structures be 64-bit aligned, etc. Use of features that weren't in the first SPARC is an optional extra. Ok, if that doesn't help then another interesting insight is that in config.log, it says that the response to 'arch' and 'arch -k' commands is 'unknown'. Don't know if that helps. Another place to ask might be the Debian-SPARC mailing list? haha funny, I was just writing an email to that list when your email hit my inbox :) Telepathy; seems we think alike. :-D Must be due to me being from the same part of the world. Possibly :) although I have found that there's not a lot of activity in that list on a regular basis. So not sure if my problem will get resolved there or not :( Ok, so this is solved! The culprit was the the line mcpu=v8 in the Makefile. Comment that out, and it makes properly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip/google
Dean, Thank you for great news. Let us see how the second SIP GV incarnation survives. -Vladimir On 3/7/2011 6:51 PM, Dean Collins wrote: http://www.disruptivetelephony.com/2011/03/google-voice-now-offers-sip-addresses-for-calling-directly-over-ip.html Nice ;) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip/google
On Tue, Mar 8, 2011 at 1:51 AM, Dean Collins d...@cognation.net wrote: http://www.disruptivetelephony.com/2011/03/google-voice-now-offers-sip-addresses-for-calling-directly-over-ip.html Nice ;) Hi Dean, What I'm waiting for is when you can send GV calls to a SIP URI without all the gymnastics needed today. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail2ban + asterisk
On Mon, 7 Mar 2011 08:50:27 -1000, Matt Darnell mattdarn...@gmail.com wrote: On Sat, Mar 5, 2011 at 8:54 PM, Pezhman Lali l...@lopl.net wrote: Dear this note is only for fresh administrators don't think about asterisk security. Do you know where you go to 'un-ban' an IP if they made some mistake? Using webmin I was not able to find the IP address that was was banned. I'm no expert but ISTR that Webmin has its own set of Iptables rules, so I'm not sure that it'll show the Asterisk chain? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
