[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 -- _ -- 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] Sangoma PCI vs PCI Express card
Am 03.03.2011 16:02, schrieb satish patel: Hey Guy, I have quick question. I am purchasing Sangoma A102D card but i am confused between PCI and PCI Express. Which card would be good for me. Definitely PCI Express is advance but i just want to know is there any major difference, like quality, performance etc.. As far as I know you should prefer PCI Express. There should be less problems with IRQ-Sharing and IRQ-Overruns. We use a A104D (PCIe) and have no problems with the current driver set. -- _ -- 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] mySQL connection testing
Danny - Thanks, but that wouldn't work either - as I am fetching
multiple rows (not in that example - but I do in a production
environment).
Steve - If mySQL in the dialplan is so bad - why did Digium include it
in the first place? JFYI - I use mySQL in the dialplan all the time -
and it always works a treat - first time, every time. I do use AGI for
'other' things (eg. I've completely re-written the AgentCallbackLogin
feature in php) and that also works a treat. Each to their own I guess.
Anyway - back to the question (repeated in case it got lost amongst all
this) Is there a way to check if a specific MYSQL connection id is
connected or not?.
BTW - using a 'disconnect {connid}' twice doesn't actually break
anything - it just causes an error on the console. So I can live with a
'no' answer.
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: 03 March 2011 17:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing
On Thu, 3 Mar 2011, Andrew Thomas wrote:
Gentlemen, can we please not turn this in to an Asterisk and DB
commands
bashing thread?
I'm just suggesting that maybe you are 'swimming upstream' trying to use
MySQL within the dialplan.
Much the same as if you were proposing an office system using a 'tin
cans
and string' mesh with carrier pigeons for out of band call signaling and
having a problem with poop buildup on the endpoints -- I might propose
using Asterisk :)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
PST
Newline Fax:
+1-760-731-3000
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Re: [asterisk-users] mySQL connection testing
On Thursday 03 March 2011 08:42:42 Andrew Thomas wrote: Does anybody know of a way to test whether a mySQL connection invoked from the dialplan is current or not? There is no way to test it. If you want this, you should track the information yourself or don't disconnect anywhere but in the h extension. BTW, the disconnect is not strictly needed in all versions of the addons since 1.4.9. Due to the possibility of a memory leak, the connections are tracked and deleted when the channel is destroyed. See this issue (and the patch) for more information: https://issues.asterisk.org/view.php?id=14757 -- Tilghman -- _ -- 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] mySQL connection testing
On Friday 04 March 2011 02:47:56 Andrew Thomas wrote: If mySQL in the dialplan is so bad - why did Digium include it in the first place? Digium is not responsible for everything that appears in Asterisk. This is a community project, and community volunteers have written large swaths of Asterisk, including the MYSQL command. -- Tilghman -- _ -- 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] Converting MP3 files to wav for Asterisk
On Thu, 2011-03-03 at 08:19 -0800, Steve Edwards wrote:
Try something 'simpler'
mpg123 -q -w ${TEMP} ${INPUT}
sox ${TEMP} -c 1 -s -w -r 8000 ${OUTPUT}
and see if that helps. Otherwise, how do the 'intermediate' files in
your
process sound? Can you hear when things fall apart?
I had been having the same issue and this above method has really
improved the quality of my wav files (I had previously been using
sox -V ${INPUT} -r 8000 -c 1 -s ${OUTPUT} resample -ql)
Thanks for that
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] mySQL connection testing
Thanks Tilghman - this is exactly what I wanted to hear. As for the 'inclusion' bit - true, but it's still infused in to the addons package at the Digium end (isn't it?). Anyway, I'll go create a mysql.conf file now :) Cheers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: 04 March 2011 08:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mySQL connection testing On Thursday 03 March 2011 08:42:42 Andrew Thomas wrote: Does anybody know of a way to test whether a mySQL connection invoked from the dialplan is current or not? There is no way to test it. If you want this, you should track the information yourself or don't disconnect anywhere but in the h extension. BTW, the disconnect is not strictly needed in all versions of the addons since 1.4.9. Due to the possibility of a memory leak, the connections are tracked and deleted when the channel is destroyed. See this issue (and the patch) for more information: https://issues.asterisk.org/view.php?id=14757 -- Tilghman -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] mySQL connection testing
On Friday 04 March 2011 03:03:41 Andrew Thomas wrote: Thanks Tilghman - this is exactly what I wanted to hear. As for the 'inclusion' bit - true, but it's still infused in to the addons package at the Digium end (isn't it?). While Digium hosts the repository and the project head (Russell) is a Digium employee, what winds up in the repository is largely up to the Asterisk community, including many non-Digium developers with commit access. While Digium does contribute a great deal to the releases, suggesting that Digium is responsible for everything that ends up in a release is reductionist and diminutive of the many contributions made by the community. -- Tilghman -- _ -- 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] Testing from where number is...
On Thursday 03 Mar 2011, Piotr Górski wrote: As free I mean no subscription. I can write AGI that will query numberingplans.com - that's not a problem... but I can query site only 20 times a day without a subscription... So it's not free. Well, free is as free does :) For the time being, keep making your 20 free queries per IP address per day, and build up a local MySQL database. Populate it also from any other data sources you have available (maybe you have letters with addresses and phone numbers? .) Then have your AGI script always look in the local database first. If what you need is not in there, and you still have some free queries remaining today (even this information can be held within the database), query numberingplans.com and save the result in your database. If you have run out of free queries, then you'll have to return something less precise (just a country, perhaps; this information at least should be in your phone directory). I can tell you now for free that 44 is the code for the UK; and UK numbers beginning with (0)7 are mobiles, (0)20 is London and (0)28 is Northern Ireland :) -- AJS Answers come *after* questions. -- _ -- 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] 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
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] Gosub and 'h' (again?)
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
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew
Thomas
Sent: 04 March 2011 11:38
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
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents of
this message, and any attachments, are the property of DataVox, and are
intended for the confidential use of the named recipient only. If you
are not the intended recipient, employee or agent responsible for
delivery of this message to the intended recipient, take note that any
dissemination, distribution or copying of this communication and its
attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable. Every effort has been made
to ensure that this e-mail or any attachments are free from viruses.
While the company has taken every reasonable precaution to minimise this
risk, neither company, nor the sender can accept liability for any
damage which you sustain as a result of viruses. It is recommended that
you should carry out your own virus checks before opening any
attachments.
Registered in England. No. 27459085.
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[asterisk-users] Asterisk - Lync / Call Center Transfer / Refer
Hey all, Alright. So we decided to not go with Avaya for our next PBX and we are now full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP gateway and call center and Lync is our internal UC and IP-PBX server. I've already got Asterisk tied with our Nortel/Merridian Option 11 with QSig and all is beautiful (except for the Opt11 not receiving names from * but that's another topic). So, my problem now is with the call center. This setup may be a bit convoluted at first but it'll make sense I hope. I've created the queues in Asterisk via FreePBX. I then created a ring group for each Lync extension so we get the Confirm Calls option and dodge the voice mail problem. The agents the login via their Lync phone with the Ring Group extension as their Agent ID. It kind of looks like this: Queue 2001 Agent 4001 Agent 4002 Agent 4003 Ring Group 4001 - Lync Extention 5001 Ring Group 4002 - Lync Extention 5002 Ring Group 4003 - Lync Extention 5003 This all works beautifuly! The problem I have is on transfers. If Lync extension 5001 trasnfers to Lync extension 5010, Asterisk is unaware of the transfer and shows that 5001 is still active with the call. We're using OrderlyStats to monitor the queue so I watch the Talking counter just keep counting instead of being aware the transfer took place. Now to me, that says to me that the transfer took place within Lync so Asterisk is unaware of the transfer. So my next step was to enable Refer support in Lync so Lync sends the refer message back to Asterisk to transfer the call so Asterisk is fully aware of what's going on. It seems like the refer message is trying to work and Lync is sending it and Asterisk is receiving it but the Refer-To is changing between the two so I'm at a loss. (Logs are below signature) Lync says it's sending the following message with a Refer-to: sip:us...@domainname.com Asterisk is seeing the following and the refer-to changed, it's now REFER-TO: sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad2787?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d%3Bto-tag%3D8be38bb187. At first it seems like Lync is sending a true SIP URI so I need to get Asterisk to know how to handle that SIP URI and then secondly, it seems like Asterisk doesn't even receive the same REFER-TO message that Lync sent. Is this because Asterisk doesn't know how to handle the SIP URI? So I guess I'm left with wondering if fixing the REFER message stuff is going to fix my problem even? The end goal is for Asterisk to be aware that a call was transferred to another extension in Lync. Thanks in advance everyone! Louis = Begin Lync SIP message TL_INFO(TF_PROTOCOL) [0]0B10.1E88::03/04/2011-13:21:17.501.0004fcd9 (SIPStack,SIPAdminLog::TraceProtocolRecord:SIPAdminLog.cpp(125))$$begin_record Trace-Correlation-Id: 215606761 Instance-Id: 00011F02 Direction: outgoing Peer: lyncserver.internal.domain:5070 Message-Type: request Start-Line: REFER sip:lyncserver.internal.domain:5070;grid=ed392a6bc0344a30b0841cd69be137ed SIP/2.0 From: sip:1173;phone-context=defaultprof...@domainname.com;user=phone;epid=e9688aa93e;tag=8be38bb187 To: sip:500;phone-context=defaultprof...@domainname.com;user=phone;epid=B3E26C1E76;tag=9227b8a39d CSeq: 2 REFER Call-ID: aa6f8871-4151-4149-ad5a-29ab941bf4d0 Via: SIP/2.0/TLS 20.20.20.20:54166;branch=z9hG4bKEB39D72C.F05E7E34CF9EF4FD;branched=FALSE Max-Forwards: 69 Via: SIP/2.0/TLS 172.16.2.29:53851;ms-received-port=53851;ms-received-cid=400 User-Agent: CPE/4.0.7577.107 OCPhone/4.0.7577.107 (Microsoft Lync 2010 Phone Edition) Supported: ms-dialog-route-set-update Refer-to: sip:us...@domainname.com Referred-By: sip:us...@domainname.com;ms-referee-uri=sip:500;phone-context=enterpr...@domainname.com;user=phone;ms-identity=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:Fri, 04 Mar 2011 13:21:17 GMT;ms-identity-info=sip:Lyncserver.internal.domain:5061;transport=tls;ms-identity-alg=rsa-sha1 Content-Length: 0 P-Asserted-Identity: sip:us...@domainname.com Privacy: id Message-Body: - $$end_record = End Lync SIP message = Begin Asterisk Debug [Mar 4 08:21:05]
Re: [asterisk-users] Loudness of recorded wav-audio
_ 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
Re: [asterisk-users] Asterisk - Lync / Call Center Transfer / Refer
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro Sent: Friday, March 04, 2011 8:07 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Lync / Call Center Transfer / Refer Hey all, Alright. So we decided to not go with Avaya for our next PBX and we are now full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP gateway and call center and Lync is our internal UC and IP-PBX server. I've already got Asterisk tied with our Nortel/Merridian Option 11 with QSig and all is beautiful (except for the Opt11 not receiving names from * but that's another topic). So, my problem now is with the call center. This setup may be a bit convoluted at first but it'll make sense I hope. I've created the queues in Asterisk via FreePBX. I then created a ring group for each Lync extension so we get the Confirm Calls option and dodge the voice mail problem. The agents the login via their Lync phone with the Ring Group extension as their Agent ID. It kind of looks like this: Queue 2001 Agent 4001 Agent 4002 Agent 4003 Ring Group 4001 - Lync Extention 5001 Ring Group 4002 - Lync Extention 5002 Ring Group 4003 - Lync Extention 5003 This all works beautifuly! The problem I have is on transfers. If Lync extension 5001 trasnfers to Lync extension 5010, Asterisk is unaware of the transfer and shows that 5001 is still active with the call. We're using OrderlyStats to monitor the queue so I watch the Talking counter just keep counting instead of being aware the transfer took place. Now to me, that says to me that the transfer took place within Lync so Asterisk is unaware of the transfer. So my next step was to enable Refer support in Lync so Lync sends the refer message back to Asterisk to transfer the call so Asterisk is fully aware of what's going on. It seems like the refer message is trying to work and Lync is sending it and Asterisk is receiving it but the Refer-To is changing between the two so I'm at a loss. (Logs are below signature) Lync says it's sending the following message with a Refer-to: sip:us...@domainname.com Asterisk is seeing the following and the refer-to changed, it's now REFER-TO: sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad278 7?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d%3Bto -tag%3D8be38bb187. At first it seems like Lync is sending a true SIP URI so I need to get Asterisk to know how to handle that SIP URI and then secondly, it seems like Asterisk doesn't even receive the same REFER-TO message that Lync sent. Is this because Asterisk doesn't know how to handle the SIP URI? So I guess I'm left with wondering if fixing the REFER message stuff is going to fix my problem even? The end goal is for Asterisk to be aware that a call was transferred to another extension in Lync. Thanks in advance everyone! Louis snip First of all, I assume you are using 1.8.X. Regardless, Queueing and referring have some known issues. If you look at chan_sip.c, you'll see that REFER is considered broken at this time (I know this to be the case in 1.4.37 and at least 1 flavor of 1.8). So my suggestion is that you either devise some workaround for this or set up multiple queues so you can feed calls to these phantom-busy folks. My Expertise (such as it is) is at the AGI level; I only fool with the portions of the actual tree code that are patently obvious (usually tweaks to patches). -- _ -- 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
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: reduce RXgain – assuming your are using Record() command 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
Re: [asterisk-users] Loudness of recorded wav-audio
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: mailto: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 -- 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
[asterisk-users] GXW4004 - lines get stuck
Hi, I have an issue with a GWX4004 used a as a VoIP trunk to PSTN lines converter. In some instances, lines get stuck (both parties hang up, but the GXW4004 status shows off hook for the lines). It stays like this until reboot. Is there a specific setting I should be looking for? I couldn't find anything about that specifically. Mike -- _ -- 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
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: reduce RXgain – assuming your are using Record() command 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
Re: [asterisk-users] Loudness of recorded wav-audio
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 mailto:da...@debsinc.com da...@debsinc.com: _ From: mailto: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: mailto:asterisk-users@lists.digium.com mailto: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/hello 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 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
Re: [asterisk-users] Sangoma PCI vs PCI Express card
PCI Express is always good also it take less power and faster when compare to PCI in interupts . On Fri, Mar 4, 2011 at 2:13 PM, Thorsten Göllner t...@ovm-group.com wrote: Am 03.03.2011 16:02, schrieb satish patel: Hey Guy, I have quick question. I am purchasing Sangoma A102D card but i am confused between PCI and PCI Express. Which card would be good for me. Definitely PCI Express is advance but i just want to know is there any major difference, like quality, performance etc.. As far as I know you should prefer PCI Express. There should be less problems with IRQ-Sharing and IRQ-Overruns. We use a A104D (PCIe) and have no problems with the current driver set. -- _ -- 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] Loudness of recorded wav-audio
Danny Nicholas wrote: In sip.conf, add rxgain=-4.0 to the peer. The last I knew, rx/tx gains are only for dahdi/zaptel devices. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Friday, March 04, 2011 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudness of recorded wav-audio Danny Nicholas wrote: In sip.conf, add rxgain=-4.0 to the peer. The last I knew, rx/tx gains are only for dahdi/zaptel devices. Doug #1 You are probably correct #2 As copper usage continues to drop and Asterisk progresses, I expect this capability to be incorporated into the sip channel (if it isn't already there in 1.8.X) -- _ -- 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] Testing from where number is...
Piotr Górski pi...@prnet.pl writes: So how to bill customers? Number portability makes it pretty impossible... In the US, you pay the same to call a cell phone as you pay to call any other phone. The callee pays for the airtime. This is a sensible arrangement, as it allows for number portability and price competition. Alas, Europe chose to pass the costs onto the caller, without even making it reasonably possible for the caller to know whether he is calling a cell phone or not! The Danish number plan in particular is completely insane. /Benny -- _ -- 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
I feel your pain On Fri, Mar 4, 2011 at 9:29 AM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro Sent: Friday, March 04, 2011 8:07 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Lync / Call Center Transfer / Refer Hey all, Alright. So we decided to not go with Avaya for our next PBX and we are now full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP gateway and call center and Lync is our internal UC and IP-PBX server. I've already got Asterisk tied with our Nortel/Merridian Option 11 with QSig and all is beautiful (except for the Opt11 not receiving names from * but that's another topic). So, my problem now is with the call center. This setup may be a bit convoluted at first but it'll make sense I hope. I've created the queues in Asterisk via FreePBX. I then created a ring group for each Lync extension so we get the Confirm Calls option and dodge the voice mail problem. The agents the login via their Lync phone with the Ring Group extension as their Agent ID. It kind of looks like this: Queue 2001 Agent 4001 Agent 4002 Agent 4003 Ring Group 4001 - Lync Extention 5001 Ring Group 4002 - Lync Extention 5002 Ring Group 4003 - Lync Extention 5003 This all works beautifuly! The problem I have is on transfers. If Lync extension 5001 trasnfers to Lync extension 5010, Asterisk is unaware of the transfer and shows that 5001 is still active with the call. We're using OrderlyStats to monitor the queue so I watch the Talking counter just keep counting instead of being aware the transfer took place. Now to me, that says to me that the transfer took place within Lync so Asterisk is unaware of the transfer. So my next step was to enable Refer support in Lync so Lync sends the refer message back to Asterisk to transfer the call so Asterisk is fully aware of what's going on. It seems like the refer message is trying to work and Lync is sending it and Asterisk is receiving it but the Refer-To is changing between the two so I'm at a loss. (Logs are below signature) Lync says it's sending the following message with a Refer-to: sip:us...@domainname.com Asterisk is seeing the following and the refer-to changed, it's now REFER-TO: sip:Lyncserver.internal.domain:5067;transport=Tls;ms-opaque=3bb3da5834ad278 7?REPLACES=aa6f8871-4151-4149-ad5a-29ab941bf4d0%3Bfrom-tag%3D9227b8a39d%3Bto -tag%3D8be38bb187. At first it seems like Lync is sending a true SIP URI so I need to get Asterisk to know how to handle that SIP URI and then secondly, it seems like Asterisk doesn't even receive the same REFER-TO message that Lync sent. Is this because Asterisk doesn't know how to handle the SIP URI? So I guess I'm left with wondering if fixing the REFER message stuff is going to fix my problem even? The end goal is for Asterisk to be aware that a call was transferred to another extension in Lync. Thanks in advance everyone! Louis snip First of all, I assume you are using 1.8.X. Regardless, Queueing and referring have some known issues. If you look at chan_sip.c, you'll see that REFER is considered broken at this time (I know this to be the case in 1.4.37 and at least 1 flavor of 1.8). So my suggestion is that you either devise some workaround for this or set up multiple queues so you can feed calls to these phantom-busy folks. My Expertise (such as it is) is at the AGI level; I only fool with the portions of the actual tree code that are patently obvious (usually tweaks to patches). -- _ -- 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
[asterisk-users] server performance....
Hi every one, I am doing some experiments on asterisk server performance.. How can we know server performance? can any one explain me plz I have 2 doubts regarding the asterisk server performance... 1. When can we know asterisk server performance? 1. when server is in idle state ? 2. when the server is in busy state? can any one please tell me when can the server performance is known i mean when server is busy or in idle state? Best Regards, viswavardhanreddy -- _ -- 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] server performance....
On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna viswavardhanre...@gmail.com wrote: Hi every one, I am doing some experiments on asterisk server performance.. How can we know server performance? can any one explain me plz I have 2 doubts regarding the asterisk server performance... 1. When can we know asterisk server performance? 1. when server is in idle state ? 2. when the server is in busy state? can any one please tell me when can the server performance is known i mean when server is busy or in idle state? Best Regards, viswavardhanreddy Many people test their servers with call-setups and call tear-downs. Using another tool like sipp you can send 100-1000s of call-setups and then do call tear-downs. You can also use transcoding loops to test the load. If you have 1 call that is sent to a context where it dials exten+1 and continues the loop until a target number, you can then set the codec for each dialed number. I know that there are many methods of testing and this is just a common one. ~~~ Andrew lathama Latham lath...@gmail.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] server performance....
HI, The way you said is correct, we are using SIPp to generate as many calls as it can send and and the server is able is to take simultaneously of 560 - 570 calls 1. when we kept server for some time as idle it took 575 calls 2. when we kept again server as busy by continous calls back to back it is taking 560-570 between i am not knowing which boundary should i take in this should i take the boundary of max successfull calls when server is in busy state or when server is in idle state? Best Regards, viswavardhan On Fri, Mar 4, 2011 at 6:20 PM, Andrew Latham lath...@gmail.com wrote: On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna viswavardhanre...@gmail.com wrote: Hi every one, I am doing some experiments on asterisk server performance.. How can we know server performance? can any one explain me plz I have 2 doubts regarding the asterisk server performance... 1. When can we know asterisk server performance? 1. when server is in idle state ? 2. when the server is in busy state? can any one please tell me when can the server performance is known i mean when server is busy or in idle state? Best Regards, viswavardhanreddy Many people test their servers with call-setups and call tear-downs. Using another tool like sipp you can send 100-1000s of call-setups and then do call tear-downs. You can also use transcoding loops to test the load. If you have 1 call that is sent to a context where it dials exten+1 and continues the loop until a target number, you can then set the codec for each dialed number. I know that there are many methods of testing and this is just a common one. ~~~ Andrew lathama Latham lath...@gmail.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 -- _ -- 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] server performance....
Hi, I mean when the cpu history is in idel and in busy state... i have one more doubt that we are doing experiments on server performance(only on software) it does not depends on hardware or even on systemm/... knowing the server performance only the software side includes any cpu history like when the server is busy or idle BR, viswavardhan On Fri, Mar 4, 2011 at 6:25 PM, viswavardhanreddy karna viswavardhanre...@gmail.com wrote: HI, The way you said is correct, we are using SIPp to generate as many calls as it can send and and the server is able is to take simultaneously of 560 - 570 calls 1. when we kept server for some time as idle it took 575 calls 2. when we kept again server as busy by continous calls back to back it is taking 560-570 between i am not knowing which boundary should i take in this should i take the boundary of max successfull calls when server is in busy state or when server is in idle state? Best Regards, viswavardhan On Fri, Mar 4, 2011 at 6:20 PM, Andrew Latham lath...@gmail.com wrote: On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna viswavardhanre...@gmail.com wrote: Hi every one, I am doing some experiments on asterisk server performance.. How can we know server performance? can any one explain me plz I have 2 doubts regarding the asterisk server performance... 1. When can we know asterisk server performance? 1. when server is in idle state ? 2. when the server is in busy state? can any one please tell me when can the server performance is known i mean when server is busy or in idle state? Best Regards, viswavardhanreddy Many people test their servers with call-setups and call tear-downs. Using another tool like sipp you can send 100-1000s of call-setups and then do call tear-downs. You can also use transcoding loops to test the load. If you have 1 call that is sent to a context where it dials exten+1 and continues the loop until a target number, you can then set the codec for each dialed number. I know that there are many methods of testing and this is just a common one. ~~~ Andrew lathama Latham lath...@gmail.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 -- _ -- 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
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=3bb3da5834ad2787?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. - [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 0 [ 53]: REFER sip:1820@10.10.10.10:5060;transport=TCP SIP/2.0 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 1 [ 78]: FROM: sip:1...@lyncserver.internal.name:5068;epid=E5790B0758;tag=15826bef52 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 2 [ 41]: TO: sip:1820@10.10.10.10;tag=as41bacc0b [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 3 [ 13]: CSEQ: 2 REFER [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 4 [ 58]: CALL-ID: 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 5 [ 16]: MAX-FORWARDS: 70 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 6 [ 59]: VIA: SIP/2.0/TCP 20.20.20.20:5068;branch=z9hG4bK70e8a145 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 7 [107]: CONTACT: sip:lyncserver.internal.name:5068;transport=Tcp;maddr=20.20.20.20;ms-opaque=09aa43d8a2a895b9 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 8 [ 17]: CONTENT-LENGTH: 0 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 9 [200]: REFER-TO: sip:lyncserver.internal.name:5068;transport=Tcp;maddr=20.20.20.20;ms-opaque=09aa43d8a2a895b9?REPLACES=a9b5f241-5e9d-4439-b347-2cac9384a627%3Bfrom-tag%3Daa19f11d4f%3Bto-tag%3D7a9abe27a5 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Header 10 [ 40]: USER-AGENT: RTCC/4.0.0.0 MediationServer [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: --- (11 headers 0 lines) --- [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: = Looking for Call ID: 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 (Checking From) --From tag 15826bef52 --To-tag as41bacc0b [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Received REFER (9) - Command in SIP REFER [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: Call 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 got a SIP call transfer from caller: (REFER)! [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Attended transfer: Will use Replace-Call-ID : a9b5f241-5e9d-4439-b347-2cac9384a627 F-tag: aa19f11d4f T-tag: 7a9abe27a5 [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: SIP transfer to extension lyncserver.internal.name:5068@from-internal-xfer by (null) [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: SIP attended transfer: Transferer channel SIP/Lync-0003, transferee channel SIP/1820-0002 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Got SIP transfer, applying to bridged peer 'SIP/1820-0002' [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: --- Transmitting (no NAT) to 20.20.20.20:5068 --- SIP/2.0 202 Accepted Via: SIP/2.0/TCP 20.20.20.20:5068;branch=z9hG4bK70e8a145;received=20.20.20.20 From: sip:1...@lyncserver.internal.name:5068;epid=E5790B0758;tag=15826bef52 To: sip:1820@10.10.10.10;tag=as41bacc0b Call-ID: 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 CSeq: 2 REFER Server: FPBX-2.8.1(1.8) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:1820@10.10.10.10:5060;transport=TCP Content-Length: 0 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Trying to put 'SIP/2.0 202' onto TCP socket destined for 20.20.20.20:5068 [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Looking for callid a9b5f241-5e9d-4439-b347-2cac9384a627 (fromtag aa19f11d4f totag 7a9abe27a5) [Mar 4 12:54:53] DEBUG[11296] chan_sip.c: Strict routing enforced for session 655e28eb45e0db7639856ec92ca88909@10.10.10.10:5060 [Mar 4 12:54:53] VERBOSE[11296] chan_sip.c: set_destination: Parsing sip:lyncserver.internal.name:5068;transport=Tcp;maddr=20.20.20.20 for address/port to send to [Mar 4 12:54:53] DEBUG[11296] netsock2.c: Splitting 'lyncserver.internal.name:5068' gives... [Mar 4 12:54:53] DEBUG[11296] netsock2.c: ...host 'lyncserver.internal.name' and port '5068'. [Mar 4 12:54:53] DEBUG[11293] manager.c: Examining event:
[asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why?
I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :) Which do you use and why? -- 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] OT: OpenSIPS vs Kamailio -- which do you use and why?
Hi, I have been working on a project with asterisk and kamailio. I would prefer using kamailio because i have personally met with the developers and it has more active users and rapid developments. The developers are also very friendly and helpful. And well open ser is not gone, the name is changed to kamailio I guess. It had a fork, but now they have merged together. Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net On 3/4/2011 11:49 AM, Steve Edwards wrote: I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :) Which do you use and why? -- _ -- 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] server performance....
On Fri, Mar 4, 2011 at 11:28 AM, viswavardhanreddy karna viswavardhanre...@gmail.com wrote: Hi, I mean when the cpu history is in idel and in busy state... i have one more doubt that we are doing experiments on server performance(only on software) it does not depends on hardware or even on systemm/... knowing the server performance only the software side includes any cpu history like when the server is busy or idle Have a look at munin, or maybe cacti or even mrtg. -- 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] chan_skinny and Cisco 793X (7936) support in 1.8
Does anybody have an answer to this? - Original Message From: Alfred Monticello ajmce...@yahoo.com To: asterisk-users@lists.digium.com Sent: Wed, March 2, 2011 9:59:20 PM Subject: chan_skinny and Cisco 793X (7936) support in 1.8 Is there any way to make a Cisco 7936 conference phone work in version 1.8? -- _ -- 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] chan_skinny and Cisco 793X (7936) support in1.8
Skinny? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alfred Monticello Sent: Friday, March 04, 2011 2:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] chan_skinny and Cisco 793X (7936) support in1.8 Does anybody have an answer to this? - Original Message From: Alfred Monticello ajmce...@yahoo.com To: asterisk-users@lists.digium.com Sent: Wed, March 2, 2011 9:59:20 PM Subject: chan_skinny and Cisco 793X (7936) support in 1.8 Is there any way to make a Cisco 7936 conference phone work in version 1.8? -- _ -- 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] OT: OpenSIPS vs Kamailio -- which do you use and why?
Hi, We use Opensips and like the results. The forks are similar, docs from one can help in the other. The opensips mailing list is monitored by one of the main developers. He is even in the IRC chat in the mornings. The docs are kept current on the opensips webpage. They like to change modules a bit, so really watch your versions. The commercial PDF Building Telephony Systems with OpenSIPS 1.6 is excellent.(duck) Yum is nice for the dependencies, but I would use a compile for Opensips. Most of the docs are Debian specific. I love Debian, but our clients love Centos. I have some Centos Opensips compile docs if needed. There are a few GUI's, but I prefer Opensips-cp. To put opensips-cp on a remote server, you need the xmlrpc module loaded on opensips. This works in Debian but fails on Centos (64 bit ONLY). Good luck, Adrian On 03/04/2011 01:49 PM, Steve Edwards wrote: I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :) Which do you use and why? -- _ -- 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] 2 ip phones and 1 normal, can't neither send nor receive calls at all...
I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco spa8800, all them are internal lines. 1.- spa921, 401 ext 2.- spa921, 402 ext 3.- normal phone connected to spa8800 404 ext. It had a very strange behavior when I was configuring call transfer and call pickup. These are steps to repeat it: 1.- from 401 call to 404 2.- from 404 don't answer it. 3.- from 402 press *8 and wait 10 seconds 4.- 402 says that it is connected. 5.- 404 stops to sound. 6.- 401 keeps ringing 7.- Hang up 402 8.- Hang up 401 After these steps I can not neither send nor receive calls from anyone of 401, 402 or 404 until I restart asterisk. /var/log/asterisk/messages, doesn´t show anything strange. ¿what's happening with my phones? Thank you for your kind help. -- _ -- 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] server performance....
Hi, We have worked out another approach for load testing: - generate using sipp certain number of test calls and that go to PBX echo server playing and receiving back pre-defined audio - generate +1 test call, which also plays and receives back an audio file Then we test the audio we received from the +1 test call using AQuA (Audio Quality Analyzer) and obtain a MOS score (AQuA is doing perceptual audio quality assessment, it's not calculating MOS as in G.107, but more likely in P.862, although the algorithms are absolutely different). In this way we can always know how many calls can the PBX under test handle before actual call quality goes down. The whole test suit is put together with other testing (loop back call testing, conference bridge testing) capabilities into what we call Asterisk VQM. If my previous message goes through moderation you will be able to see screenshots as well :) Best Regards, Sevana Oy http://www.sevana.fi - Original Message - From: Andrew Latham lath...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2011 8:20 PM Subject: Re: [asterisk-users] server performance On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna viswavardhanre...@gmail.com wrote: Hi every one, I am doing some experiments on asterisk server performance.. How can we know server performance? can any one explain me plz I have 2 doubts regarding the asterisk server performance... 1. When can we know asterisk server performance? 1. when server is in idle state ? 2. when the server is in busy state? can any one please tell me when can the server performance is known i mean when server is busy or in idle state? Best Regards, viswavardhanreddy Many people test their servers with call-setups and call tear-downs. Using another tool like sipp you can send 100-1000s of call-setups and then do call tear-downs. You can also use transcoding loops to test the load. If you have 1 call that is sent to a context where it dials exten+1 and continues the loop until a target number, you can then set the codec for each dialed number. I know that there are many methods of testing and this is just a common one. ~~~ Andrew lathama Latham lath...@gmail.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 -- _ -- 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] How is Libpri developped ?
On Fri, Mar 4, 2011 at 12:50 AM, Olivier oza_4...@yahoo.fr wrote: Hi, Can you explain the main differences between Libpri 1.4.11 and 1.4.12 as both seem to receive additions and patches ? Do they target different asterisk versions ? Can they both be considered as production-ready ? 1.4.12 is just a newer version than 1.4.11 and any released version is as production-ready as can be reasonably be expected AFAIK. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.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
[asterisk-users] can anyone tell me how to set asterisk to record all phonecall
Hi all, I need to use asterisk to record all phonecall I have test using mixmonitor to record a call. Now I need to set the configure file to let asterisk auto record all calls. I have searched many document but still can not succeed. My version is 1.8beta and I prefer using mixmonitor. Regards! -- _ -- 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] [announce] jkSMS
For those interested, I have released a first version of jkSMS, which is a simple package that lets cell phones text messages to asterisk. Note it's not real SMS, it makes heavy use of email-to-sms gateways, but it seems to work well. I have had the code running 12 hours, but haven't found any issues. it's not for the faint-of-heart and might require a bit of hacking (really minimal though) if you're not running the same tools that i'm running (like editing the code's DSN if you dont have sqlite installed) http://jeremy.kister.net/code/asterisk/jkSMS/ enjoy, -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] can anyone tell me how to set asterisk to record all phonecall
On 3/4/2011 9:49 PM, John Wu wrote:
I need to use asterisk to record all phonecall I have test using
mixmonitor to record a call.
this is one way it can be done
make sure you have 'lame' installed.
- in your extensions.conf:
[global]
VSA=/var/spool/asterisk
[outbound-or-wherever-you-dial]
exten = _XXX,1,Macro(Snoop,${EXTEN})
exten = _XXX,n,Dial(SIP/${EXTEN},${TIMEOUT})
exten = _XXX,n,StopMixMonitor
; above in case you're in some loop Dial fails,
; e.g., swift+monitor crash asterisk
[macro-Snoop]
; ${ARG1} channel
exten = s,1,GotoIf($[${SNOOPING} = 1]?snooping)
exten = s,n,Set(SNOOPING=1)
exten = s,n,Set(=${STRFTIME(${EPOCH},,%Y)})
exten = s,n,Set(MM=${STRFTIME(${EPOCH},,%m)})
exten = s,n,Set(DD=${STRFTIME(${EPOCH},,%d)})
exten = s,n,Set(HMS=${STRFTIME(${EPOCH},,%H%M%S)})
exten = s,n,Set(FILENAME=${HMS}-${CALLERID(num)}-${ARG1}-${UNIQUEID})
exten = s,n,Set(MIXMON_ARGS=mkdir -p ${VSA}/monitor/${}/${MM}/${DD}
nice -n 19 /usr/local/bin/lame --silent --resample 11.025 -b 16 -t -m
m ${VSA}/monitor/${FILENAME}.wav
${VSA}/monitor/${}/${MM}/${DD}/${FILENAME}.mp3 rm -f
${VSA}/monitor/${FILENAME}.wav)
exten = s,n,MixMonitor(${FILENAME}.wav,,${MIXMON_ARGS})
exten = s,n(snooping),NoOp(snooping on ${CHANNEL})
that'll end up putting a mp3 of the call in
/var/spool/asterisk/monitor//MM/DD/HHMMSS-CALLERID.mp3
don't forget any legal issues you might have to work around, recording
the fact that you declared the message is being recorded.
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] TLS/SRTP calls go to circuit busy.
Once again, thanks for your reply. I had done some research already but forget to include it in my previous email. I did find a bug that is remarkably similar to the issues that I'm having. The bug number is 18674. Thanks, Mitch Johnson Message: 8 Date: Fri, 04 Mar 2011 00:34:45 -0600 From: Terry Wilson twil...@digium.com Subject: Re: [asterisk-users] TLS/SRTP calls go to circuit busy. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4d708805.3060...@digium.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 03/03/2011 02:22 PM, Mitch Johnson wrote: Thanks so much for pointing this out. I was curious why the commands in the documentation differed to the commands I was using. That problem is fixed, but now I have a new issue. I can call with no issues, however, as soon as I answer one of the calls I see the error: ast_srtp_unprotect: SRTP unprotect: authentication failure. Below is a snippet of the debug as the call is answered. The best thing to do at this point would be to file a bug report with the info at which point it will eventually probably be assigned to me (unless some awesome person comes up with a fix first!) to look at. If I have a bit of free time, I'll try to take a peek at it. If you can post the sip debug output of the entire offer/answer exchange to the bug report, it will help greatly. Terry -- _ -- 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 Asterisk / API / Perl
Hi
i want use the API on my asterisk 1.6, but i have a small problems :
In extension, i start it :
exten = _X.,3,AGI(My-Script.agi)
The perl agi file are started without problems
but i want get into this script a lot of variable:
Type (SIP or IAX)
src (from cdr)
but that's don't work:
use Asterisk::AGI;
use lib /var/lib/asterisk/agi-bin;
$AGI = new Asterisk::AGI;
$typ = $AGI-get_variable('agi_type');
$typ don't have SIP or IAX, same test without succes:
$typ = $AGI-get_variable('type');
anyone know this problems ?
thanks
Olivier
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[asterisk-users] Asterisk, Sent accountcode between 2 asterisk
Hi
I have two Asterisk Server:
The first server A, all phone are connected
The Second server B only route call to a lot of SIP supplier
the server A sent:
; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR
exten = _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW)
exten = _X.,2,Dial(IAX2/SERVERB/${EXTEN},180,rt)
exten = _X.,3,Hangup
anyone know if it's possible to add the CDR Accountcode to this process
for get it on the second server B ?
i want the same accountcode on the 2 servers
thanks
Olivier
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