Re: [asterisk-users] Record() Cmd and My SQL
Why not write the file to /tmp using MixMonitor, then use the command option to trigger an AGI script that will move the data into your database then delete the original file? John On 24 September 2010 04:23, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: The reason is when doing a load balancing , We cannot confine the recording to a particular asterisk machine ( If we have more than one asterisk machine in the topology ). So a centralized mechanism might be better . So that any machine can access the recording . Regards Mahesh -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext David Backeberg Sent: Thursday, September 23, 2010 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Record() Cmd and My SQL On Thu, Sep 23, 2010 at 2:21 AM, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: HI , Is there Any way is there so that I can store my recordings directly to a database rather storing the same to a file . Please, please, please tell us why you would want to do that. -- _ -- 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 -- JA Taylor MA VetMB MRCVS Mansion Hill Veterinary Practice 133-137 Main Road Middleton Cheney OX17 2PP 01295 712110 -- _ -- 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] How to test BIG traffic through DAHDI/WANPIPE interfaces
Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- 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] tcpdump auto stats script
Before I reinvent the wheel, I'm looking for a script then when run will - launch tcpdump (or equivalent) on the server and capture all SIP and UDP traffic to an IP address - then, rather than me manually analysing with wireshark, will analyze the cap file and produce stats on jitter, lag, delta etc. Thanks for any help John -- _ -- 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] RDNIS not passed from one box to another with BRI access
Hi,
I've configured a new Asterisk 1.6.1 box to replace an old Bristuffed 1.2
Asterisk.
Since then, it happens that forwarded calls are not presented the way they
used to be.
It seems that now, some endpoints are displaying the original caller id
(that's what I'm trying to achive), while some are displaying the
redirecting number :
so if A calls B, B forwards to C
depending on where C is located, C will either display that the call is
coming from A or from B.
(in this case, A, B and C are phones, each connected to a different PBX)
So I dived into voip-info.org and reached the following conclusion :
If everything is correctly configured, the last PBX (the one to which phone
C is connected) should receive an incoming call with the following
properties set to
CALLERID(num) set to A's number
CALLERID(ANI) set to B's number
CALLERID(RDNIS) not set
An acceptable alternative would be:
CALLERID(num) set to B's number
CALLERID(ANI) set to B's number
CALLERID(RDNIS) set to A's number
I've got a lot of questions.
0. Is my understanding (see above) correct ?
1. In general, is the ability to send and receive RDNIS a feature telcos
bundle with BRI access or is this feature an option or dedicated to PRI ?
2. Is the setting of RDNIS a task for Dial application or this configurable
from within the dialplan ?
This comes from this extract from http://www.voip-info.org/wiki/view/RDNIS :
Asterisk only supports RGN, and at that, it's wrongly placed in the
channel's Caller*ID information. The RGN can be set or retrieved using the
CALLERID http://www.voip-info.org/wiki/view/Asterisk+func+callerid(rdnis)
function, such as Set(CALLERID(rdnis)=5551212). The Dial application also
sets the RGN to the current extension, unless called within a macro, in
which case the contents of ${MACRO_EXTEN} are used instead. This overwrites
anything else set within the dialplan, which may not be what a dialplan
author would expect. Although IAX2
http://www.voip-info.org/wiki/view/IAX2supports RGN in IE 27
(actually misnamed RDNIS), it does not appear to be
properly passed to a peer when placing outgoing calls.
3. Is there options in chan_dahdi.conf (or elsewhere) that trigger on or off
this RDNIS capability ?
Cheers
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Re: [asterisk-users] Record() Cmd and My SQL
Thanks , I was not knowing about Mix Monitor . Whether MixMonitor is faster than record ? Both uses same mechanism to write to the file . Regards Mahesh -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext John Taylor Sent: Friday, September 24, 2010 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Record() Cmd and My SQL Why not write the file to /tmp using MixMonitor, then use the command option to trigger an AGI script that will move the data into your database then delete the original file? John On 24 September 2010 04:23, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: The reason is when doing a load balancing , We cannot confine the recording to a particular asterisk machine ( If we have more than one asterisk machine in the topology ). So a centralized mechanism might be better . So that any machine can access the recording . Regards Mahesh -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ext David Backeberg Sent: Thursday, September 23, 2010 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Record() Cmd and My SQL On Thu, Sep 23, 2010 at 2:21 AM, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: HI , Is there Any way is there so that I can store my recordings directly to a database rather storing the same to a file . Please, please, please tell us why you would want to do that. -- _ -- 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 -- JA Taylor MA VetMB MRCVS Mansion Hill Veterinary Practice 133-137 Main Road Middleton Cheney OX17 2PP 01295 712110 -- _ -- 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] tcpdump auto stats script
Hi! traffic to an IP address - then, rather than me manually analysing with wireshark, will analyze the cap file and produce stats on jitter, lag, delta etc. This is what RTCP was made for. Philipp -- _ -- 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] Fax On Demand - Asterisk 1.4.29
Hi All, Is there anyone who ever implemented successfully Fax On Demand on Asterisk 1.4.29 ? I've tried to look from Google about this issue and could not find any satisfying about this. Thanks in advance for all of you who willing to help And Sorry if there's any mistake in my question, cause this is my first time asking question in this mailing list. Thanks Regards, Zoel Hairi -- _ -- 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 to test BIG traffic through DAHDI/WANPIPEinterfaces
Hi DD, We usually use loopback cables and use the open source SIP test tool SIPp to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Dias Verzonden: vrijdag 24 september 2010 11:05 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- 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 to test BIG traffic through DAHDI/WANPIPEinterfaces
ummm but how do you do that? SIPp is only for SIP calls...i need to check in some way the dahdi driver, i need in someway stress de card, is that possible? may be it has no sence at all :( Thanks! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- 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] Fax On Demand - Asterisk 1.4.29
i don't see any mistakes in your question.. but i still don't get it.
what do you need exactly from Fax on demand? sending faxes? receiving faxes?
From: zoelha...@yahoo.co.id
To: asterisk-users@lists.digium.com
Date: Fri, 24 Sep 2010 17:27:57 +0700
Subject: [asterisk-users] Fax On Demand - Asterisk 1.4.29
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Hi All,
Is there anyone who ever implemented successfully Fax On Demand
on Asterisk 1.4.29 ?
I’ve tried to look from Google about this issue and could not
find any satisfying about this.
Thanks in advance for all of you who willing to help
And Sorry if there’s any mistake in my question, cause this is
my first time asking question in this mailing list.
Thanks
Regards,
Zoel Hairi
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Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
As the previous poster said use the sip software to make test calls. Have the number it dials go out of the sangoma card and back into another port via a crossover cable to an extension which answers and plays back a file for a second or so before hanging up. You can then make lots of calls which constantly make outgoing calls on 4 ports and incoming calls on another 4 ports. By being able to change the diration of the call to can load the box very well. Danny Dias wrote: ummm but how do you do that? SIPp is only for SIP calls...i need to check in some way the dahdi driver, i need in someway stress de card, is that possible? may be it has no sence at all :( Thanks! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl mailto:i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- 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] rtp problem with 1.8.0-rdc1
On 10-09-23 05:40 PM, cov...@ccs.covici.com wrote: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Any assistance here would be appreciated. We're probably going to need some sort of debugging information such as a console trace and SIP (I assume chan_sip) debug. More information here: doc/HOWTO_collect_debug_information.txt Leif. -- _ -- 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 1.6.2.13 - have asterisk reply from same IP address
On 10-09-23 07:01 PM, Mike wrote: Hi, I have a server with multiple IP address, Asterisk binding with all of them. I'd like Asterisk to reply to a SIP peer from the same IP address as the peer used to register to Asterisk (as opposed to using the main IP address all the time regardless of how the peer communicated with Asterisk). Is this possible? I know it wasn't with 1.4, but I was told 1.6 had something like this (something to do with not breaking SIP over TCP) This has been a requested feature for quite a while now, but I don't think Asterisk does this yet. Some time in the past Jared Smith had a patch that would do this which was considered hacky but did seem to work in his particular situation. If this has been implemented, I am not aware of it. Leif. -- _ -- 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 to test BIG traffic throughDAHDI/WANPIPEinterfaces
Hi, We usually stress test with asterisk using dialplans like: [sipp] exten = service,1,1,Dial(DAHDI/r1/12345678) [incoming-1] exten = 12345678,1,Dial(DAHDI/r2/12345678) [incoming-2] exten = 12345678,1,Answer() exten = 12345678,n,WaitMusicOnHold(30) exten = 12345678,n,HangUp() The sipp peer is configured with context=sipp, DAHDI group 1 (spans 1-4) has context=incoming-1, DAHDI group 2 (spans 5-8) has context=incoming-2. By letting SIPp set up a call it will dial out on the first group which is looped back to the second group which then plays hold music for 30 seconds. Met vriendelijke groet, Ingmar Steen Teleknowledge Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Dias Verzonden: vrijdag 24 september 2010 12:38 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] How to test BIG traffic throughDAHDI/WANPIPEinterfaces ummm but how do you do that? SIPp is only for SIP calls...i need to check in some way the dahdi driver, i need in someway stress de card, is that possible? may be it has no sence at all :( Thanks! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool SIPp to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Dias Verzonden: vrijdag 24 september 2010 11:05 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- 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] rtp problem with 1.8.0-rdc1
cov...@ccs.covici.com writes: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Why is it a problem? It sounds like Asterisk does silence suppression. /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
[asterisk-users] Redirecting a Channel more than three times...
Hi folks, could someone please try to confirm the following (mis)behaviour of my asterisk? Imagine the following scenario: Caller A calls the central. Central picks up, talks to Caller A which wants to be connected to employee X. Central puts Caller A on hold by Redirecting the Channel to a Queue. Central calls emplyee X and bridges both channels... everybody is happy. But..: Caller A calls the central. Central picks up, talks to Caller A which wants to be connected to employee X. Central puts Caller A on hold by Redirecting the Channel to a Queue. Central calls emplyee X and X doesn´t want to talk with Caller A Central and employee hang up.. Central pulls Caller A back from Queue (again, with Redirecting the channel to its own extension) Caller A now want to talk with employee Y and so on This game works exactly three times... when the central wants to pull back the Caller from the Queue for the third time, the call is hungup. I searched and searched, but could not find anything about a redirect-limit or so... what, if there is no such limit, am I doing wrong? If there is such a limit.. where is it configured? thank you anyways, yves -- _ -- 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 to test BIG traffic through DAHDI/WANPIPEinterfaces
Garet, MANY thanks my friend...can you believe that my brain was stucked :( So simple ;) THANKS for your valuable help! DD 2010/9/24 Gareth Blades list-aster...@skycomuk.com As the previous poster said use the sip software to make test calls. Have the number it dials go out of the sangoma card and back into another port via a crossover cable to an extension which answers and plays back a file for a second or so before hanging up. You can then make lots of calls which constantly make outgoing calls on 4 ports and incoming calls on another 4 ports. By being able to change the diration of the call to can load the box very well. Danny Dias wrote: ummm but how do you do that? SIPp is only for SIP calls...i need to check in some way the dahdi driver, i need in someway stress de card, is that possible? may be it has no sence at all :( Thanks! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl mailto:i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- 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] rtp problem with 1.8.0-rdc1
Leif Madsen leif.mad...@asteriskdocs.org wrote: On 10-09-23 05:40 PM, cov...@ccs.covici.com wrote: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Any assistance here would be appreciated. We're probably going to need some sort of debugging information such as a console trace and SIP (I assume chan_sip) debug. More information here: doc/HOWTO_collect_debug_information.txt Leif. I certainly can do a sip set debug, is that what you need? I did do an rtp set debug and this is how I found out that when I hit the mute button on the soft phone all rtp traffic ceased between the phone and the asterisk box. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] rtp problem with 1.8.0-rdc1
Benny Amorsen benny+use...@amorsen.dk wrote: cov...@ccs.covici.com writes: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Why is it a problem? It sounds like Asterisk does silence suppression. But it surpresses in both directions! I still want to hear the other end. For a test is there a way to turn off that feature to see if that is the cause? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] Redirecting a Channel more than three times...
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Friday, September 24, 2010 6:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Redirecting a Channel more than three times... Hi folks, could someone please try to confirm the following (mis)behaviour of my asterisk? Imagine the following scenario: Caller A calls the central. Central picks up, talks to Caller A which wants to be connected to employee X. Central puts Caller A on hold by Redirecting the Channel to a Queue. Central calls emplyee X and bridges both channels... everybody is happy. But..: Caller A calls the central. Central picks up, talks to Caller A which wants to be connected to employee X. Central puts Caller A on hold by Redirecting the Channel to a Queue. Central calls emplyee X and X doesn´t want to talk with Caller A Central and employee hang up.. Central pulls Caller A back from Queue (again, with Redirecting the channel to its own extension) Caller A now want to talk with employee Y and so on This game works exactly three times... when the central wants to pull back the Caller from the Queue for the third time, the call is hungup. I searched and searched, but could not find anything about a redirect-limit or so... what, if there is no such limit, am I doing wrong? If there is such a limit.. where is it configured? thank you anyways, yves #1. Have you looked at the CLI output for this scenario #2. Why don't you use Parking instead of queue? -- _ -- 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] Fwd: Can't cross compile asterisk 1.6.2.13 on arm using ltib
No ideas ? Just give me the way if possible Sebastien Hi, Excuse me if I'm late to reply but my first response has been blocked by the moderator (message too big) So I've created an account on rapidshare to share my config.log and menuselect/config.log Hope it will help. The link : http://rapidshare.com/users/Z8SX25 Thanks for any help ! Sebastien On Wed, Sep 22, 2010 at 9:21 AM, IMS ims77@gmail.com wrote: Do you have any ideas of the problem ? config.log don't give me more explanations. Attach your config.log so we can see what is going on. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com Hi, I can cross compile asterisk 1.4.21 on arm (imx27) using ltib I want to cross compile the new version 1.6.2.13 but there is an error when I execute the commands : ./configure --build=i686-pc-linux-gnu --host=arm make menuselect The configure seems ok, I have the result info : *configure: Package configured for: configure: OS type : none configure: Host CPU : arm configure: build-cpu:vendor:os: i686 : pc : linux-gnu : configure: host-cpu:vendor:os: arm : unknown : none : configure: Cross Compilation = YES * But when I try to execute make menuselect I have the message : *CC=cc CXX= LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent makeopts make[1]: Entering directory `/home/m/ltib/rpm/BUILD/asterisk-1.6.2.13/menuselect' configure: error: in `/home/m/ltib/rpm/BUILD/asterisk-1.6.2.13/menuselect': configure: error: cannot run C compiled programs. If you meant to cross compile, use `--host'. See `config.log' for more details. make[1]: *** [makeopts] Error 1 make[1]: Leaving directory `/home/m/ltib/rpm/BUILD/asterisk-1.6.2.13/menuselect' make: *** [menuselect/makeopts] Error 2* Do you have any ideas of the problem ? config.log don't give me more explanations. With google i found the problem should be corrected from the revision 268052 (Build menuselect with the build environment's compiler, not the host (target)'s compiler) here : http://svnview.digium.com/svn/asterisk/branches/1.6.2?view=revisionrevision=268052 Thanks for your help. Sebastien -- _ -- 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] Fax On Demand - Asterisk 1.4.29
Hi Tarek,
what do you need exactly from Fax on demand? sending faxes? receiving
faxes?
In simple explanation is like this, Caller goes through IVR (After having
been validated), Then Caller Choose Fax On Demand option and hang up, and
then Asterisk Send the Caller a Fax that already been prepared.
That's the plan that I had in mind.
Thanks
Regards,
Zoel Hairi
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah
Sent: Friday, September 24, 2010 5:42 PM
To: Asterisk Users
Subject: Re: [asterisk-users] Fax On Demand - Asterisk 1.4.29
i don't see any mistakes in your question.. but i still don't get it.
what do you need exactly from Fax on demand? sending faxes? receiving faxes?
From: zoelha...@yahoo.co.id
To: asterisk-users@lists.digium.com
Date: Fri, 24 Sep 2010 17:27:57 +0700
Subject: [asterisk-users] Fax On Demand - Asterisk 1.4.29
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Hi All,
Is there anyone who ever implemented successfully Fax On Demand
on Asterisk 1.4.29 ?
Ive tried to look from Google about this issue and could not
find any satisfying about this.
Thanks in advance for all of you who willing to help
And Sorry if theres any mistake in my question, cause this is
my first time asking question in this mailing list.
Thanks
Regards,
Zoel Hairi
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[asterisk-users] differential billing
Hi All, How can we develop a differential charging setup using asterisk like for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec charge @15cent, etc? Any idea, suggestion. -- Regards, Abdul Basit | +92 32 1416 4196 -- _ -- 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] differential billing
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Abdul Basit Sent: Friday, September 24, 2010 8:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] differential billing Hi All, How can we develop a differential charging setup using asterisk like for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec charge @15cent, etc? Any idea, suggestion. -- Regards, Abdul Basit | +92 32 1416 4196 Since the CDR records the call duration in seconds, this should be a relative no-brainer, assuming you are billing post-call. If you are wanting to generate the charges during the live calls, AMI would be your best option for getting a running duration of the connection. -- _ -- 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-pakistan] differential billing
It has nothing to do with asterisk. A separate billing system has to be made, where the billing / rate policies are defined. I can help you out further, so feel free to contact me. Regards, Nasir. 0333-2302834 On 24-09-2010 18:13, Abdul Basit wrote: Hi All, How can we develop a differential charging setup using asterisk like for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec charge @15cent, etc? Any idea, suggestion. -- Regards, Abdul Basit | +92 32 1416 4196 __._,_.___ Reply to sender mailto:basit.e...@gmail.com?subject=differential%20billing | Reply to group mailto:asterisk-pakis...@yahoogroups.com?subject=differential%20billing | Reply via web post http://groups.yahoo.com/group/asterisk-pakistan/post;_ylc=X3oDMTJwY2xiMjRuBF9TAzk3MzU5NzE0BGdycElkAzIyMDY4OTkyBGdycHNwSWQDMTcwNTAwNDcyNgRtc2dJZAM2NjkEc2VjA2Z0cgRzbGsDcnBseQRzdGltZQMxMjg1MzM0MDIy?act=replymessageNum=669 | Start a New Topic http://groups.yahoo.com/group/asterisk-pakistan/post;_ylc=X3oDMTJmbnI2djd2BF9TAzk3MzU5NzE0BGdycElkAzIyMDY4OTkyBGdycHNwSWQDMTcwNTAwNDcyNgRzZWMDZnRyBHNsawNudHBjBHN0aW1lAzEyODUzMzQwMjI- Messages in this topic http://groups.yahoo.com/group/asterisk-pakistan/message/669;_ylc=X3oDMTMzc3NtdW0zBF9TAzk3MzU5NzE0BGdycElkAzIyMDY4OTkyBGdycHNwSWQDMTcwNTAwNDcyNgRtc2dJZAM2NjkEc2VjA2Z0cgRzbGsDdnRwYwRzdGltZQMxMjg1MzM0MDIyBHRwY0lkAzY2OQ-- (1) Recent Activity: Visit Your Group http://groups.yahoo.com/group/asterisk-pakistan;_ylc=X3oDMTJmZmR2c29yBF9TAzk3MzU5NzE0BGdycElkAzIyMDY4OTkyBGdycHNwSWQDMTcwNTAwNDcyNgRzZWMDdnRsBHNsawN2Z2hwBHN0aW1lAzEyODUzMzQwMjI- pCopyright © 2009 a href=http://www.asteriskpakistan.com; title=Asterisk PakistanAsterisk Pakistan/a/p Yahoo! Groups http://groups.yahoo.com/;_ylc=X3oDMTJlMTEzMDVoBF9TAzk3NDc2NTkwBGdycElkAzIyMDY4OTkyBGdycHNwSWQDMTcwNTAwNDcyNgRzZWMDZnRyBHNsawNnZnAEc3RpbWUDMTI4NTMzNDAyMg-- Switch to: Text-Only mailto:asterisk-pakistan-traditio...@yahoogroups.com?subject=change%20delivery%20format:%20Traditional, Daily Digest mailto:asterisk-pakistan-dig...@yahoogroups.com?subject=email%20delivery:%20Digest . Unsubscribe mailto:asterisk-pakistan-unsubscr...@yahoogroups.com?subject=unsubscribe . Terms of Use http://docs.yahoo.com/info/terms/ . __,_._,___ -- _ -- 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] Fwd: Can't cross compile asterisk 1.6.2.13 on arm using ltib
On Fri, Sep 24, 2010 at 9:11 AM, IMS ims77@gmail.com wrote: No ideas ? Just give me the way if possible Download the latest asterisk version (1.4.36) and retry, if it fails create a new issue on https://issues.asterisk.org -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] rtp problem with 1.8.0-rdc1
Benny Amorsen wrote: cov...@ccs.covici.com writes: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Why is it a problem? It sounds like Asterisk does silence suppression. /Benny 1) With no rtp traffic, the nat device will drop the connection in it's nat table and thus disconnecting the softphone from Asterisk. (after the router's timeout period of course) 2) The other issue is you are connected to a conference call and you want to mute your transmitter while listening to the conference. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] differential billing
Thank you Danny. I am thinking for AMI events. Do we need some code level change? As i want asterisk to push events to some listener rather than i ask via AMI. For hight call volume read from AMI may be an over head on asterisk, i think. On Fri, Sep 24, 2010 at 6:19 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Abdul Basit *Sent:* Friday, September 24, 2010 8:13 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] differential billing Hi All, How can we develop a differential charging setup using asterisk like for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec charge @15cent, etc? Any idea, suggestion. -- Regards, Abdul Basit | +92 32 1416 4196 Since the CDR records the call duration in seconds, this should be a relative “no-brainer”, assuming you are billing post-call. If you are wanting to generate the charges during the live calls, AMI would be your best option for getting a running duration of the connection. -- _ -- 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 -- Regards, Abdul Basit | +92 32 1416 4196 -- _ -- 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] Fwd: Can't cross compile asterisk 1.6.2.13 on arm using ltib
First I've tryed with the version 1.4.36 But it didn't worked so I supposed it should be ok with the last version 1.6.2... but not = I will create a new issue for this if you think it should be. Just hope it will not be too long to have a correction. Thanks a lot. Sebastien On Fri, Sep 24, 2010 at 9:11 AM, IMS ims77@gmail.com wrote: No ideas ? Just give me the way if possible Download the latest asterisk version (1.4.36) and retry, if it fails create a new issue on https://issues.asterisk.orghttp://192.168.49.5/exchweb/bin/redir.asp?URL=https://issues.asterisk.org -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://192.168.49.5/exchweb/bin/redir.asp?URL=http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hellohttp://192.168.49.5/exchweb/bin/redir.asp?URL=http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usershttp://192.168.49.5/exchweb/bin/redir.asp?URL=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] Redirecting a Channel more than three times...
Hi Danny, I decided against Parking Calls, because it seemed quite complicated and useless for me... as far as i remember, parkedcalls return automagically after a timeout which was not desirable. I would have to rewrite a lot of code, if i have to change... but there must be a reason for this misbehaviour, and i think its hardcoded in the asterisk-source. somewhere seems to be a counter that counts the redirects... it maybe useful in some case, maybe to avoid loops or something similar to bounces in emails, but in my case its undesired... because i am using trixbox / freepbx the dialplan is very complicated, but it showed me no hint of beeing responsible for this... the cli-output gives no hint. yves Am 24.09.2010 15:10, schrieb Danny Nicholas: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Friday, September 24, 2010 6:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Redirecting a Channel more than three times... Hi folks, could someone please try to confirm the following (mis)behaviour of my asterisk? Imagine the following scenario: Caller A calls the central. Central picks up, talks to Caller A which wants to be connected to employee X. Central puts Caller A on hold by Redirecting the Channel to a Queue. Central calls emplyee X and bridges both channels... everybody is happy. But..: Caller A calls the central. Central picks up, talks to Caller A which wants to be connected to employee X. Central puts Caller A on hold by Redirecting the Channel to a Queue. Central calls emplyee X and X doesn´t want to talk with Caller A Central and employee hang up.. Central pulls Caller A back from Queue (again, with Redirecting the channel to its own extension) Caller A now want to talk with employee Y and so on This game works exactly three times... when the central wants to pull back the Caller from the Queue for the third time, the call is hungup. I searched and searched, but could not find anything about a redirect-limit or so... what, if there is no such limit, am I doing wrong? If there is such a limit.. where is it configured? thank you anyways, yves #1. Have you looked at the CLI output for this scenario #2. Why don't you use Parking instead of queue? -- _ -- 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] rtp problem with 1.8.0-rdc1
Lyle Giese l...@lcrcomputer.net wrote: Benny Amorsen wrote: cov...@ccs.covici.com writes: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Why is it a problem? It sounds like Asterisk does silence suppression. /Benny 1) With no rtp traffic, the nat device will drop the connection in it's nat table and thus disconnecting the softphone from Asterisk. (after the router's timeout period of course) 2) The other issue is you are connected to a conference call and you want to mute your transmitter while listening to the conference. This is my issue, I am on a conference and mute myself, but I still want to hear the other end and asterisk is cutting off both ends audio. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] differential billing
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Abdul Basit Sent: Friday, September 24, 2010 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] differential billing Thank you Danny. I am thinking for AMI events. Do we need some code level change? As i want asterisk to push events to some listener rather than i ask via AMI. For hight call volume read from AMI may be an over head on asterisk, i think. snip You can actually use AMI as a listener to track the progress of all calls (use uniqueid to keep the calls separated). In PERL, I have a module that I copied down from voip-info.org that I can use to give me all or selected events for all calls in the listening period I choose. -- _ -- 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] rtp problem with 1.8.0-rdc1
Hi! Why is it a problem? It sounds like Asterisk does silence suppression. 1) With no rtp traffic, the nat device will drop the connection in it's nat table and thus disconnecting the softphone from Asterisk. (after the router's timeout period of course) 2) The other issue is you are connected to a conference call and you want to mute your transmitter while listening to the conference. Set internaltiming to yes in asterisk.conf and see if that helps. In addition you might also be able to change the mute behaviour of your SIP clients so that it keeps on sending silent RTP packets. Philipp P.S.: You could also mute the conference user, not the SIP UA. -- _ -- 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] rtp problem with 1.8.0-rdc1
Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! Why is it a problem? It sounds like Asterisk does silence suppression. 1) With no rtp traffic, the nat device will drop the connection in it's nat table and thus disconnecting the softphone from Asterisk. (after the router's timeout period of course) 2) The other issue is you are connected to a conference call and you want to mute your transmitter while listening to the conference. Set internaltiming to yes in asterisk.conf and see if that helps. In addition you might also be able to change the mute behaviour of your SIP clients so that it keeps on sending silent RTP packets. I cannot change the soft phone, so this is why I need asterisk to behave properly or at least have an option to behave differently -- and it did work up to a point and then they fixed something. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] Debug compile fails
Somehow I can't get 1.6.2.13 to compile with DEBUG_CHANNEL_LOCKS. Downloaded latest tgz and extracted $ ./configure $ make menuselect (select the needed options from compiler flags) $ grep DEBUG_CHANNEL_LOCKS menuselect.makeopts MENUSELECT_CFLAGS=DONT_OPTIMIZE LOADABLE_MODULES DEBUG_CHANNEL_LOCKS MALLOC_DEBUG $ make make install $ asterisk asterisk -rx core show locks No such command 'core show locks' (type 'core show help core' for other possible commands) Am I missing something? -- Daniel Tryba -- _ -- 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] should trixbox system hang when ISP drops connection?
NEWBIE alert: i'm a linux person, not an asterisk person so i'm certainly capable of handling any linux-flavoured solution you can suggest. here's a note i got from a local company i know (some proper names removed): = start = Now and again our ISP goes down and when it does give us a hicup, the Asterisk system shuts down (not very forgiving). When it shuts down our phone system as well goes down. This will need some visitation as to a resolution however would it be possible for now to have a heartbeat happening on the ISP so that should it go down, a email is generated to myself advising that this has occurred. This will then notify me to reboot the tribox. Let me know if this is doable. = end = so, is there an easy fix for this? if the ISP goes down, does that necessarily mean that trixbox has to go down as well? or should i be asking this question on a trixbox-specific list? thanks. rday -- Robert P. J. Day Waterloo, Ontario, CANADA Top-notch, inexpensive online Linux/OSS/kernel courses http://crashcourse.ca Twitter: http://twitter.com/rpjday LinkedIn: http://ca.linkedin.com/in/rpjday -- _ -- 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] should trixbox system hang when ISP dropsconnection?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert P. J. Day Sent: Friday, September 24, 2010 9:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] should trixbox system hang when ISP dropsconnection? NEWBIE alert: i'm a linux person, not an asterisk person so i'm certainly capable of handling any linux-flavoured solution you can suggest. here's a note i got from a local company i know (some proper names removed): = start = Now and again our ISP goes down and when it does give us a hicup, the Asterisk system shuts down (not very forgiving). When it shuts down our phone system as well goes down. This will need some visitation as to a resolution however would it be possible for now to have a heartbeat happening on the ISP so that should it go down, a email is generated to myself advising that this has occurred. This will then notify me to reboot the tribox. Let me know if this is doable. = end = so, is there an easy fix for this? if the ISP goes down, does that necessarily mean that trixbox has to go down as well? or should i be asking this question on a trixbox-specific list? thanks. rday -- Robert P. J. Day Waterloo, Ontario, CANADA Trixbox (Asterisk) should not go down when the ISP does. Of course the actual process wouldn't go down, just the connectivity (assuming SIP connections/trunks). The BOBW solution I would suggest is that you run your Trixbox/Asterisk using a local DCHP provider/server so you aren't as vulnerable to how efficient your ISP is at staying up. -- _ -- 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] should trixbox system hang when ISP drops connection?
On Fri, Sep 24, 2010 at 9:55 AM, Robert P. J. Day rpj...@crashcourse.cawrote: so, is there an easy fix for this? if the ISP goes down, does that necessarily mean that trixbox has to go down as well? or should i be asking this question on a trixbox-specific list? thanks. rday Try installing a local caching nameserver on the same box that runs asterisk, and have that handle DNS queries for you. I remember at one point that trixbox would hang if you had any SIP trunks configured and you lost internet connectivity, but a caching nameserver on the same box tended to help. -- Thanks, --Warren Selby 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] should trixbox system hang when ISP dropsconnection?
On 24 Sep 2010, at 16:09, Danny Nicholas wrote: The BOBW solution I would suggest is that you run your Trixbox/Asterisk using a local DCHP provider/server so you aren't as vulnerable to how efficient your ISP is at staying up. DNS. Not DHCP. S -- _ -- 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] should trixbox system hang when ISP drops connection?
Is your ISP doing DNS resolutions for you? If yes, then I also think it has something to do with the DNS queries which hangs asterisk. But it should not bring the server down. On CentOS, caching name server should be very easy to install by doing: yum install caching-nameserver I don't remember if it also sets up the required config files. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-24 11:15 AM, Warren Selby wcse...@selbytech.com wrote: On Fri, Sep 24, 2010 at 9:55 AM, Robert P. J. Day rpj...@crashcourse.ca wrote: so, is there... Try installing a local caching nameserver on the same box that runs asterisk, and have that handle DNS queries for you. I remember at one point that trixbox would hang if you had any SIP trunks configured and you lost internet connectivity, but a caching nameserver on the same box tended to help. -- Thanks, --Warren Selby 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 -- _ -- 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] should trixbox system hang when ISP drops connection?
On Fri, 24 Sep 2010, Warren Selby wrote: Try installing a local caching nameserver on the same box that runs asterisk, and have that handle DNS queries for you. I remember at one point that trixbox would hang if you had any SIP trunks configured and you lost internet connectivity, but a caching nameserver on the same box tended to help. so, just to be clear, if you were counting on your ISP for DNS, and the ISP went down, that would not just *pause* your trixbox, it might genuinely take it down? in any event, i suspect installing the caching nameserver certainly wouldn't hurt anything. thanks. rday -- Robert P. J. Day Waterloo, Ontario, CANADA Top-notch, inexpensive online Linux/OSS/kernel courses http://crashcourse.ca Twitter: http://twitter.com/rpjday LinkedIn: http://ca.linkedin.com/in/rpjday -- _ -- 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] Record() Cmd and My SQL
On Thu, Sep 23, 2010 at 11:23 PM, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: The reason is when doing a load balancing , We cannot confine the recording to a particular asterisk machine ( If we have more than one asterisk machine in the topology ). Yes you can. You can record the file wherever the call takes place. In fact, you can make the recording on any network segment the packet traverses as well. So a centralized mechanism might be better . So that any machine can access the recording . Regards Mahesh Recordings are formatted data, typically stored as files. You can put them into a database, but you haven't provided a reason why that would be a good idea. There are these things called shared filesystems. You should take a look at them. They work well. Options include NFS, iscsi, sans, etc. Or you can record the file in-place, and when the recording completes, copy it off to your shared filesystem. That's what I do. Or you can take a look at something like OrecX, which let's you do network spanning on your entire subnet, and it doesn't matter where your call takes place because all RTP streams get written to disk. None of what you've explained would be a good reason to put your recordings into a database. Don't do that. -- _ -- 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] differential billing
A quick answer? A2billing. It has what you call it differential billing.. but they call it progressive billing.. 3 steps .. for 3 different rates .. Go for it.. easy to setup and quick to learn and use. Regards From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, September 24, 2010 4:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] differential billing _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Abdul Basit Sent: Friday, September 24, 2010 8:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] differential billing Hi All, How can we develop a differential charging setup using asterisk like for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec charge @15cent, etc? Any idea, suggestion. -- Regards, Abdul Basit | +92 32 1416 4196 Since the CDR records the call duration in seconds, this should be a relative no-brainer, assuming you are billing post-call. If you are wanting to generate the charges during the live calls, AMI would be your best option for getting a running duration of the connection. -- _ -- 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] differential billing
A quick answer? A2billing. It has what you call it differential billing.. but they call it progressive billing.. 3 steps .. for 3 different rates .. Go for it.. easy to setup and quick to learn and use. Regards From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, September 24, 2010 4:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] differential billing _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Abdul Basit Sent: Friday, September 24, 2010 8:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] differential billing Hi All, How can we develop a differential charging setup using asterisk like for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec charge @15cent, etc? Any idea, suggestion. -- Regards, Abdul Basit | +92 32 1416 4196 Since the CDR records the call duration in seconds, this should be a relative no-brainer, assuming you are billing post-call. If you are wanting to generate the charges during the live calls, AMI would be your best option for getting a running duration of the connection. -- _ -- 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] best format for playback/generation
Greetings fellow listers, I have an application where I have approximately 300 files that I playback individually or in blocks to simulate text-to-speech in a less mechanical voice than normal Allison files provide. These files are presently in GSM format and sound pretty good when I play them on my computer speakers or on my in-house Polycom 501's over SIP connections. The problem I have is that the intended use of the application is going to be over SIP/DAHDI trunks that will connect to VM's over IAX trunks. What is your best suggestion for maintaining the quality of the audio as much as possible? Best Case presently - SIP phone in-house to IAX Worst Case presently - Cell phone calls Asterisk 1 on TDM400P which connects to VM Asterisk 2 via IAX. Asterisk version is 1.4.30 Thanks in Advance Danny Nicholas -- _ -- 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] best format for playback/generation
The best format would be in whatever format asterisk is sending the final audio out in. Even if you store it in the highest quality asterisk may have to transcode it on the fly so its best to store it in an already transcoded format to reduce the cpu load. For dahdi you would want to use the native .sln format. For sip use whatever coded you use over the sip connection. Danny Nicholas wrote: Greetings fellow listers, I have an application where I have approximately 300 files that I playback individually or in blocks to simulate “text-to-speech” in a “less mechanical” voice than normal Allison files provide. These files are presently in GSM format and sound pretty good when I play them on my computer speakers or on my in-house Polycom 501’s over SIP connections. The “problem” I have is that the intended use of the application is going to be over SIP/DAHDI trunks that will connect to VM’s over IAX trunks. What is your best suggestion for maintaining the quality of the audio as much as possible? Best Case presently – SIP phone in-house to IAX Worst Case presently – Cell phone calls Asterisk 1 on TDM400P which connects to VM Asterisk 2 via IAX. Asterisk version is 1.4.30 Thanks in Advance Danny Nicholas -- _ -- 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] Record() Cmd and My SQL
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Friday, September 24, 2010 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Record() Cmd and My SQL On Thu, Sep 23, 2010 at 11:23 PM, Govind, Mahesh (NSN - IN/Bangalore) mahesh.gov...@nsn.com wrote: The reason is when doing a load balancing , We cannot confine the recording to a particular asterisk machine ( If we have more than one asterisk machine in the topology ). Yes you can. You can record the file wherever the call takes place. In fact, you can make the recording on any network segment the packet traverses as well. So a centralized mechanism might be better . So that any machine can access the recording . Regards Mahesh Recordings are formatted data, typically stored as files. You can put them into a database, but you haven't provided a reason why that would be a good idea. There are these things called shared filesystems. You should take a look at them. They work well. Options include NFS, iscsi, sans, etc. Or you can record the file in-place, and when the recording completes, copy it off to your shared filesystem. That's what I do. Or you can take a look at something like OrecX, which let's you do network spanning on your entire subnet, and it doesn't matter where your call takes place because all RTP streams get written to disk. None of what you've explained would be a good reason to put your recordings into a database. Don't do that. Don sez: I don't know how to make Outlook indent. I usually top-post, but I don't like getting yelled at. Why do you say Don't do that? Is there a real reason that it would be bad? I'd like to put the recordings in a database so they are available to another application that has no other relationship to the Asterisk server. The application uses the database to determine if the recording has been listened to, by whom and if it needs additional attention. --Don -- _ -- 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] best format for playback/generation
If your sip provider supports gsm, then it is fine to send them your existing format, but I am sure by the time voice reaches an end user, it is transcoded at least once or twice again, so you can never guarantee what quality the end user is getting. I would stay with ulaw, as it has more chances to retain a better quailty even after a few transcodings, plus almost every sip provider will be able to receive it as it is and pass it on as received. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-24 1:02 PM, Gareth Blades list-aster...@skycomuk.com wrote: The best format would be in whatever format asterisk is sending the final audio out in. Even if you store it in the highest quality asterisk may have to transcode it on the fly so its best to store it in an already transcoded format to reduce the cpu load. For dahdi you would want to use the native .sln format. For sip use whatever coded you use over the sip connection. Danny Nicholas wrote: Greetings fellow listers, I have an ... -- _ -- 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] Losing local SIP phones when internet goes down?
Still I have the connection loss when internet goes down, I have to restart the Asterisk machine or need to remove the VoIP trunk accessing internet... DNSmasq is the only option by losing the connection when internet goes down...is there any other way... Thanks On Fri, Feb 12, 2010 at 4:20 AM, Matt Riddell li...@venturevoip.com wrote: On 9/02/10 12:59 PM, Tilghman Lesher wrote: add to the top of /etc/resolv.conf nameserver 127.0.0.1 If you're using DHCP on any of your interfaces, you'll need to configure dhclient (or whatever dhcp client you're using) to prepend in the configuration with (e.g. /etc/dhcp3/dhclient.conf): prepend domain-name-servers 127.0.0.1; Otherwise, your entry in resolv.conf will be overwritten on each DHCP lease renewal. Yeah, although if you're using DHCP, then dnsmasq is possibly a better option. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopalakrishnan A.N, -- _ -- 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] Record() Cmd and My SQL
On Fri, Sep 24, 2010 at 1:32 PM, Don Kelly d...@donkelly.biz wrote: Don sez: I don't know how to make Outlook indent. I usually top-post, but I don't like getting yelled at. Why do you say Don't do that? Is there a real reason that it would be bad? Performance is a real reason. Multiple simultaneous write streams into a database sounds like a disaster. While trying to read from the db and use it to listen to recordings sounds like a bigger disaster. /path/to/the/recording is a short varchar string the actual recording is a massive, usually multi-megabyte, potentially multi-gigabyte blob. http://en.wikipedia.org/wiki/Blob_(computing) If you're not actually taking advantage of the recording being in the database, doing computing that is easier because of the database, such as nearest neighbor searches, indexing, and the like, you're just slowing down your ability to store and retrieve recordings. I'd like to put the recordings in a database so they are available to another application that has no other relationship to the Asterisk server. Sounds like a filesystem. I can store my pdf file with my web browser, and read it on another computer after I store it to my shared filesystem. The application uses the database to determine if the recording has been listened to, by whom and if it needs additional attention. Database can maintain metadata (as can a filesystem, owner, creation date, access date), but you could still just store a pointer to the actual file in the db. If you were paranoid about the filesystem and db getting out of step you could do referential integrity checks in the application. If you want to do something wholesale to all the recordings, like carve off the first five seconds, it's quite straightforward with a batched sox call against the filesystem. If you want to do that in a db, it's a select, write output to a file, convert the file, and replace on the value to store it back into the db. -- _ -- 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] Losing local SIP phones when internet goes down?
Its a long and old thread, haven't read it all, but just to let you know this happens when there is no reply from the DNS. So change DNS or install it locally on your asterisk server. At least caching name server should be installed. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-24 1:51 PM, Gopalakrishnan A.N sai...@gmail.com wrote: Still I have the connection loss when internet goes down, I have to restart the Asterisk machine or need to remove the VoIP trunk accessing internet... DNSmasq is the only option by losing the connection when internet goes down...is there any other way... Thanks On Fri, Feb 12, 2010 at 4:20 AM, Matt Riddell li...@venturevoip.com wrote: On 9/02/10 12:59 ... -- Thank you with regards, Gopalakrishnan A.N, -- _ -- 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] Record() Cmd and My SQL
snip To add to this laundry list #1. It is much simpler to get a path from a database and load that file than to try and process a MYSQL BLOB of any size. #2. If you should eventually leave MYSQL, blobs don't always play nicely (no pun intended) with other DB's like PostgreSQL. #3. You can always use SOX to make a file into a smaller format - good luck doing that with a BLOB. -- _ -- 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] Record() Cmd and My SQL
I hadn't considered writing to the db real-time; was actually planning on recording locally and moving it to the db. Thanks for the suggestions. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Friday, September 24, 2010 12:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Record() Cmd and My SQL On Fri, Sep 24, 2010 at 1:32 PM, Don Kelly d...@donkelly.biz wrote: Don sez: I don't know how to make Outlook indent. I usually top-post, but I don't like getting yelled at. Why do you say Don't do that? Is there a real reason that it would be bad? Performance is a real reason. Multiple simultaneous write streams into a database sounds like a disaster. While trying to read from the db and use it to listen to recordings sounds like a bigger disaster. /path/to/the/recording is a short varchar string the actual recording is a massive, usually multi-megabyte, potentially multi-gigabyte blob. http://en.wikipedia.org/wiki/Blob_(computing) If you're not actually taking advantage of the recording being in the database, doing computing that is easier because of the database, such as nearest neighbor searches, indexing, and the like, you're just slowing down your ability to store and retrieve recordings. I'd like to put the recordings in a database so they are available to another application that has no other relationship to the Asterisk server. Sounds like a filesystem. I can store my pdf file with my web browser, and read it on another computer after I store it to my shared filesystem. The application uses the database to determine if the recording has been listened to, by whom and if it needs additional attention. Database can maintain metadata (as can a filesystem, owner, creation date, access date), but you could still just store a pointer to the actual file in the db. If you were paranoid about the filesystem and db getting out of step you could do referential integrity checks in the application. If you want to do something wholesale to all the recordings, like carve off the first five seconds, it's quite straightforward with a batched sox call against the filesystem. If you want to do that in a db, it's a select, write output to a file, convert the file, and replace on the value to store it back into the db. -- _ -- _ -- 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] Debug compile fails
On Fri, Sep 24, 2010 at 10:47 AM, Daniel Tryba dan...@tryba.nl wrote: Am I missing something? DEBUG_THREADS -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] Asterisk 1.62.13 - CPU spikes every 10 minutes
Hi, I've been getting regular CPU usage spikes(50%-80%), due to asterisk (according to top). I never noticed this on 1.4, and I have top running in the background pretty much all the time. In between those spikes Asterisk stays under 10% CPU usage (I have a transcoder card, which helps). It's very regular, never any missed spike, or any spike in between the regular spikes. I don`t have cron job running every 10 minutes (asterisk-related or not). Because it's so regular, I don`t think it's anything in my dialplan. If it was it would be more random. SIP peers are reregistering every 60 seconds, so the spikes would be every minute if that was the problem. There is no scheduled attack from outside that I know of, network traffic (calculated from the switch port and the server) does NOT spike. Memory is very stable at a low value. Does anyone have an idea of where to start looking? Can Asterisk report somehow what is causing this? 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] Asterisk 1.62.13 - CPU spikes every 10 minutes
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, September 24, 2010 2:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes Hi, I've been getting regular CPU usage spikes(50%-80%), due to asterisk (according to top). I never noticed this on 1.4, and I have top running in the background pretty much all the time. In between those spikes Asterisk stays under 10% CPU usage (I have a transcoder card, which helps). It's very regular, never any missed spike, or any spike in between the regular spikes. I don`t have cron job running every 10 minutes (asterisk-related or not). Because it's so regular, I don`t think it's anything in my dialplan. If it was it would be more random. SIP peers are reregistering every 60 seconds, so the spikes would be every minute if that was the problem. There is no scheduled attack from outside that I know of, network traffic (calculated from the switch port and the server) does NOT spike. Memory is very stable at a low value. Does anyone have an idea of where to start looking? Can Asterisk report somehow what is causing this? Mike I would start with /var/log/asterisk/messages and /var/log/asterisk/full to see what is (was) happening (if anything) on the console at these spike times. -- _ -- 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 1.62.13 - CPU spikes every 10 minutes
sip / other registrations... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Fri, Sep 24, 2010 at 3:40 PM, Mike l...@net-wall.com wrote: Hi, I've been getting regular CPU usage spikes(50%-80%), due to asterisk (according to top). I never noticed this on 1.4, and I have top running in the background pretty much all the time. In between those spikes Asterisk stays under 10% CPU usage (I have a transcoder card, which helps). It's very regular, never any missed spike, or any spike in between the regular spikes. I don`t have cron job running every 10 minutes (asterisk-related or not). Because it's so regular, I don`t think it's anything in my dialplan. If it was it would be more random. SIP peers are reregistering every 60 seconds, so the spikes would be every minute if that was the problem. There is no scheduled attack from outside that I know of, network traffic (calculated from the switch port and the server) does NOT spike. Memory is very stable at a low value. Does anyone have an idea of where to start looking? Can Asterisk report somehow what is causing this? 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 -- _ -- 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 1.62.13 - CPU spikes every 10 minutes
Thanks guys for caring enough to write. Danny: I did check /var/log/messages/full . Nothing out of the ordinary. Andrew: many hundreds of SIP peers are registering every 60 seconds (and have done so since 1.4). No problem there and it doesn't coincide with the 10 minute spikes anyways. Core show threads doesn't show how busy the threads are, unfortunately. But I can't see a difference between normal and spiked CPU looking at that output. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Friday, September 24, 2010 15:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes sip / other registrations... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Fri, Sep 24, 2010 at 3:40 PM, Mike l...@net-wall.com wrote: Hi, I've been getting regular CPU usage spikes(50%-80%), due to asterisk (according to top).� I never noticed this on 1.4, and I have top running in the background pretty much all the time. In between those spikes Asterisk stays under 10% CPU usage (I have a transcoder card, which helps). It's very regular, never any missed spike, or any spike in between the regular spikes.� I don`t have cron job running every 10 minutes (asterisk-related or not). Because it's so regular, I don`t think it's anything in my dialplan.� If it was it would be more random. SIP peers are reregistering every 60 seconds, so the spikes would be every minute if that was the problem.� There is no scheduled attack from outside that I know of, network traffic (calculated from the switch port and the server) does NOT spike. Memory is very stable at a low value. Does anyone have an idea of where to start looking? Can Asterisk report somehow what is causing this? 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 -- _ -- 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 1.62.13 - CPU spikes every 10 minutes
snip Check out this (old) link about 1.6.1 https://issues.asterisk.org/view.php?id=16158 you might want to recreate /dev/null and /dev/random and see if that helps. -- _ -- 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 1.62.13 - CPU spikes every 10 minutes
I found that bug before I wrote, and I was hoping you were right, but recreating those two missing files didn't help. I wasn't running 1.6.1 anyways, but I figured I'd try. There must be a way (Linux or Asterisk-centric) to see if a particular thread/module is doing this? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, September 24, 2010 16:24 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes snip Check out this (old) link about 1.6.1 https://issues.asterisk.org/view.php?id=16158 you might want to recreate /dev/null and /dev/random and see if that helps. -- _ -- 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] rtp problem with 1.8.0-rdc1
cov...@ccs.covici.com writes: But it surpresses in both directions! I still want to hear the other end. For a test is there a way to turn off that feature to see if that is the cause? Ah, so it isn't Asterisk doing silence suppression, it's Asterisk being unable to handle that other devices do. If you switch to 1.6.2.x and enable internal-timing, you should have a shot at getting it working. /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] Can't turn debug on in a 1.2 box
On Thu, Sep 23, 2010 at 10:06 AM, khalid touati khalidtou...@gmail.com wrote: do you guys know how i can turn debug on or just know why it's not getting enabled? Thanks a lot for your help! Abdullah *CLI set debug 15 *CLI reload -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] Asterisk 1.62.13 - CPU spikes every 10 minutes
On 09/24/2010 03:52 PM, Mike wrote: I found that bug before I wrote, and I was hoping you were right, but recreating those two missing files didn't help. I wasn't running 1.6.1 anyways, but I figured I'd try. There must be a way (Linux or Asterisk-centric) to see if a particular thread/module is doing this? Not always. It may be that you are seeing a side effect of how the kernel accounts for time in each timeslice. What versions (DAHDI / asterisk / kernel) are you using? The 10 minute interval could be because of something like the kernel charging neighbour table scanning to the asterisk process. These are just some thoughts though. Regarding the time accounting specifically, there's been some talk recently on LKML to fix this behaviour [PATCH 0/6] Proper kernel irq time accounting http://thread.gmane.org/gmane.linux.kernel/1037168 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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] Can't turn debug on in a 1.2 box
On Thu, Sep 23, 2010 at 10:06 AM, khalid touati khalidtou...@gmail.com wrote: do you guys know how i can turn debug on or just know why it's not getting enabled? On Fri, 24 Sep 2010, Paul Belanger wrote: *CLI set debug 15 *CLI reload If you change these lines in the '[logfiles]' section of logger.conf and enter 'logger reload' at the Asterisk CLI, you will get more than enough debugging info on the console and in your syslog file (probably /var/log/messages). console = debug,dtmf,error,event,notice,verbose,warning syslog.local0 = debug,dtmf,error,event,notice,verbose,warning Please remember to change them back and reload when you have identified your problem(s). -- 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] rtp problem with 1.8.0-rdc1
All I have to do to make it work is to use 1.8.0 revision 281875 -- after that something is broke. I was hoping someone could look and see what changed just after that rev and see if it makes sense. Benny Amorsen benny+use...@amorsen.dk wrote: cov...@ccs.covici.com writes: But it surpresses in both directions! I still want to hear the other end. For a test is there a way to turn off that feature to see if that is the cause? Ah, so it isn't Asterisk doing silence suppression, it's Asterisk being unable to handle that other devices do. If you switch to 1.6.2.x and enable internal-timing, you should have a shot at getting it working. /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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] Asterisk 1.62.13 - CPU spikes every 10 minutes
Thanks Shaun, but I'm not sure I understand everything you wrote...I can understand that blaming Asterisk might be a Linux error, but it still doesn't explain what does make the CPU usage shoot up like this. I am using 2.6.18-194.3.1.el5 (64 bits, CentOs), Asterisk 1.6.2.13 and DAHDI Version: 2.3.0.1 Echo Canceller: MG2. On that server I have a 4-port PCIE PRI card and a TCE400B card. A few hundred SIP peers, but nothing 1.4 couldn't handle easily (same server). Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Friday, September 24, 2010 18:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes On 09/24/2010 03:52 PM, Mike wrote: I found that bug before I wrote, and I was hoping you were right, but recreating those two missing files didn't help. I wasn't running 1.6.1 anyways, but I figured I'd try. There must be a way (Linux or Asterisk-centric) to see if a particular thread/module is doing this? Not always. It may be that you are seeing a side effect of how the kernel accounts for time in each timeslice. What versions (DAHDI / asterisk / kernel) are you using? The 10 minute interval could be because of something like the kernel charging neighbour table scanning to the asterisk process. These are just some thoughts though. Regarding the time accounting specifically, there's been some talk recently on LKML to fix this behaviour [PATCH 0/6] Proper kernel irq time accounting http://thread.gmane.org/gmane.linux.kernel/1037168 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
[asterisk-users] can call internal branch , but can not call external numbers with avaya , always get return message : Q931IncompatibleDestination
Hi Gurus,
We have configured asterisk to trunk with avaya with ooh323 channel driver. The
sip phone registered on asterisk
can dial the extensions registered on avaya via this trunk , and vice versa
works too. Even we can make the avaya branch to dial asterisk’s extension and
then this extension dial back to another avaya’s extension.
But if we dial the external DID number via this trunk from asterisk extension ,
it always fails.
Have anyone experienced such issue, and share me the experiences?
On avaya , it report the below message,
-
timedata
16:47:30 Calling party trunk-group 7 member 1 cid 0xef2
16:47:30 Calling Number Name 7002 Tony
16:47:30 active trunk-group 7 member 1 cid 0xef2
16:47:30 dial 03 route:ARS
16:47:30 denial event 1367: BCC incompatibility D1=0x830007 D2=0xef2
16:47:30 dial 03 route:ARS
16:47:30 term trunk-group 7cid 0xef2
16:47:30 idle trunk-group 7cid 0xef2
16:47:42 Calling party trunk-group 7 member 1 cid 0xef7
16:47:42 Calling Number Name 7002 Tony
16:47:42 active trunk-group 7 member 1 cid 0xef7
16:47:42 dial 03 route:ARS
16:47:42 denial event 1367: BCC incompatibility D1=0x830007 D2=0xef7
16:47:42 dial 03 route:ARS
16:47:42 term trunk-group 7cid 0xef7
list trace tac #07 Page 2
LIST TRACE
timedata
16:47:42 idle trunk-group 7cid 0xef7
16:47:47 Calling party trunk-group 7 member 1 cid 0xefa
16:47:47 Calling Number Name 7002 Tony
16:47:47 active trunk-group 7 member 1 cid 0xefa
16:47:47 dial 03 route:ARS
16:47:47 denial event 1367: BCC incompatibility D1=0x830007 D2=0xefa
16:47:47 dial 03 route:ARS
16:47:47 term trunk-group 7cid 0xefa
16:47:47 idle trunk-group 7cid 0xefa
Or
list trace previousPage 1
LIST TRACE
timedata
17:46:16 Calling party trunk-group 7 member 1 cid 0x1289
17:46:16 Calling Number Name 7002 Tony
17:46:16 active trunk-group 7 member 1 cid 0x1289
17:46:16 dial 03 route:ARS
17:46:16 term trunk-group 3cid 0x1289
17:46:16 G729 ss:off ps:20
rgn:1 [192.168.0.12]:7968
rgn:1 [192.168.3.100]:2232
17:46:16 xoip options: fax:Relay modem:off tty:US (igc)
xoip ip: [192.168.0.12]:7968
17:46:16 xoip options: fax:Relay modem:off tty:US (igc)
xoip ip: [192.168.3.100]:2232
17:46:16 G729 ss:off ps:20
rgn:1 [192.168.0.12]:7968
rgn:1 [192.168.3.100]:2232
17:46:16 dial 039129051 route:ARS
17:46:16 route-pattern 2 preference 1 cid 0x1289
17:46:16 seize trunk-group 3 member 17 cid 0x1289
17:46:16 Setup digits 39129051
17:46:16 Calling Number Name 7002 Tony
17:46:16 Proceed trunk-group 3 member 17 cid 0x1289
17:46:16 denial event 1204: Bearer cap not implem D1=0x830007 D2=0x241
17:46:16 idle trunk-group 3 member 17 cid 0x1289
The below is the configuration logs,
--
Avaya : AVAYA G650 S8800
Asterisk: 1.4.31
Ooh323 config:
Objective Open H.323 Channel Driver's Config:
IP:Port:0.0.0.0:1720
H.225 port range: 12030-12230
FastStart yes
Tunneling yes
CallerIdasterisk
MediaWaitForConnect no
Gatekeeper: No Gatekeeper
H.323 LogFile: /var/log/asterisk/h323_log
Context:default
Capability: 0xf (g723|gsm|ulaw|alaw)
DTMF Mode: rfc2833
AccountCode:ast_h323
AMA flags: Unknown
Aliases:
100 ObjSysAsterisk
Avaya codecs setting:
---
1 G729
2 g711a
3 g711mu
/etc/asterisk/ooh323.conf
---
[avaya]
type=peer
context=default
ip=192.168.0.14 ; UPDATE with appropriate ip address
port=1720; UPDATE with appropriate port
e164=101
disallow=all
allow=ulaw
allow=alaw
/var/log/asterisk/h323_log
-
18:44:37:802 Using configured media info (outgoing, ooh323c_o_10)
18:44:37:802 Created new logical channel entry (outgoing, ooh323c_o_10)
18:44:37:802 Building reverse olc. (outgoing, ooh323c_o_10)
18:44:37:802 Receive channel of type audio started (outgoing, ooh323c_o_10)
18:44:37:802 FastStart Element = {
18:44:37:802 forwardLogicalChannelNumber = {
18:44:37:8021003
18:44:37:802 }
18:44:37:802 forwardLogicalChannelParameters = {
18:44:37:803dataType = {
18:44:37:803 nullData = {
18:44:37:803 NULL
18:44:37:803 }
18:44:37:803}
18:44:37:803multiplexParameters = {
18:44:37:803 none = {
18:44:37:804 NULL
Re: [asterisk-users] Asterisk- speech to text(Voicemail totext message)
В Чтв, 23/09/2010 в 14:21 -0500, Danny Nicholas пишет: FWIW, the current state of Speech-to-text will let you do a 70-95% accurate translation of incoming voicemails depending on clarity/dialect/training. Also depends on language of native speakers. For 100% reliability, this still requires Human intervention. I'd like to do this too. Poking around, it looks like res_speech.so is the library to enable it, but an actual separate program to convert from voice to text is needed, like Sphinx or VXI? I haven't found anything yet that describes how to connect it to voicemail. Examples are welcome, if anyone has one to point at/paste. Looking at Sphinx and the available documentation, I think these things to be true. #1 - res_speech.so isn't necessary since Sphinx operates as a external module as opposed to the resident modules of Vestec and Lumenvox. #2 - Didn't really find a good on-the-fly example of processing the file as it came in. Hello guys I've created a little HOWTO about voicemail transcription with Asterisk and pocketsphinx here: http://nsh.nexiwave.com/2010/09/voicemail-transcription-with.html try it. If you have any other questions just ask --- Nexiwave - Speech Mining For Call Centers http://nexiwave.com signature.asc Description: This is a digitally signed message part -- _ -- 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] Losing local SIP phones when internet goes down?
Yes I read the one more thread http://lists.digium.com/pipermail/asterisk-users/2010-February/244256.html also.. Thanks for your comments...:) On Fri, Sep 24, 2010 at 11:27 PM, Zeeshan Zakaria zisha...@gmail.comwrote: Its a long and old thread, haven't read it all, but just to let you know this happens when there is no reply from the DNS. So change DNS or install it locally on your asterisk server. At least caching name server should be installed. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-24 1:51 PM, Gopalakrishnan A.N sai...@gmail.com wrote: Still I have the connection loss when internet goes down, I have to restart the Asterisk machine or need to remove the VoIP trunk accessing internet... DNSmasq is the only option by losing the connection when internet goes down...is there any other way... Thanks On Fri, Feb 12, 2010 at 4:20 AM, Matt Riddell li...@venturevoip.com wrote: On 9/02/10 12:59 ... -- Thank you with regards, Gopalakrishnan A.N, -- _ -- 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 -- Thank you with regards, Gopalakrishnan A.N, -- _ -- 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 1.8.0 Release Candidate 2 Now Available
At 01:14 PM 9/23/2010, you wrote: The Asterisk Development Team has announced the second release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ I downloaded this, ran ./configure followed by make menuselect and I don't seem to have SIP as an available protocol. Is there something I can do to make it available? It works fine on the most recent 1.6 version and it's worked on most of the prior 1.8 versions. Ira -- _ -- 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 1.8.0 Release Candidate 2 Now Available
On Fri, Sep 24, 2010 at 10:25:01PM -0700, Ira wrote: At 01:14 PM 9/23/2010, you wrote: The Asterisk Development Team has announced the second release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ I downloaded this, ran ./configure followed by make menuselect and I don't seem to have SIP as an available protocol. Is there something I can do to make it available? It works fine on the most recent 1.6 version and it's worked on most of the prior 1.8 versions. You probably need to install libssl-dev then rerun ./configure. At least I did (Debian Lenny). Seems chan_sip needs res_crypto which needs libssl. -- Barry -- _ -- 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
