[asterisk-users] Experience with LLDP
Hello, LLDP is more and more available on various network elements (endpoint, switches, ...). It seems to ease network configuration. Do you have any experience with it ? How would you rate LLDP ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] distribute free call minutes over different channels
Hi, I have 4 ISDN channels (2 lines) and each line may do calls of up to 360 minutes/month for free. As I understand asterisk will pick the first available line so the probability is big that the other lines will not use their free minutes and the firs line will exceed the free minutes. How can I configure asterisk in a way that it looks up in the CDR which ISDN line has lest calling time in the present month and chosse this? Kind regards Eckhard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR for asterisk
Anyone can recommend a commercial large scale IVR with easy + pro management for asterisk? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] distribute free call minutes over different channels
Couldnt you do this by calling MySql? Compare who has the least minutes used and then send it out the appropriate channel? --Matt On Tue, Nov 24, 2009 at 7:07 AM, Eckhard Jokisch e.joki...@orange-moon.dewrote: Hi, I have 4 ISDN channels (2 lines) and each line may do calls of up to 360 minutes/month for free. As I understand asterisk will pick the first available line so the probability is big that the other lines will not use their free minutes and the firs line will exceed the free minutes. How can I configure asterisk in a way that it looks up in the CDR which ISDN line has lest calling time in the present month and chosse this? Kind regards Eckhard ___ -- 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 -- Matthew Desbiens 603.581.3160 //* EOF *// ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1.10 Music On Hold
Hello again, I just tried version 1.6.1.9, and the MOH works well there. It seems to be a bug introduced in 1.6.1.10. Best regards, Örn 2009/11/23 Örn Arnarson o...@arnarson.net Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some configuration change I need to implement for this to work properly? Was there a conscious change in Asterisk's behavior? Best regards, Örn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cianet channel bank with noise and echo
Hello. I'm running some asterisks in a small voip provider in Brazil and we're having some problems with a analogic channel bank. When I make calls using analogic extensions, I have a crystal clear quality, but the receptor have a lot of noise and echo. We tested the situation using SIP, E1 links, GSM links, analogic trunks (FXO ports) and we always have the same symptons My configurations for /etc/asterisk/chan_dahdi.conf and /etc/dahdi/system.conf: http://www.pastebin.org/56678 Versions of software: CentOS 5.4 final Asterisk 1.6.2.0-rc2 Dahdi Linux 2.2.0.2 Dahdi Tool 2.2.0 Kernel 2.6.18-164.6.1.el5 Channel Bank Model: (Sorry, no english datasheet) http://www.cianet.ind.br/pt/channel_bank.php Any help is appreciated, Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] distribute free call minutes over different channels
On Tue, Nov 24, 2009 at 01:07:28PM +0100, Eckhard Jokisch wrote: Hi, I have 4 ISDN channels (2 lines) and each line may do calls of up to 360 minutes/month for free. As I understand asterisk will pick the first available line so the probability is big that the other lines will not use their free minutes and the firs line will exceed the free minutes. If you use chan_zap or chan_dahdi, you can use r/R instead of g/G to go round-robin through the channels rather than start from the first/last (respectively). This would give you a rather even spreading of the outgoing calls. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] keep asterisk in RAM
Is there a way to keep asterisk in RAM and tell linux not to swap it out (ever). There are times when delays are noticed and I presume its due to linux swapping out the program. As if I call right back in then everything responds right away. Wait awhile and the same thing might occur. How can I keep asterisk always in RAM? I use CentOS 5. Thanks, jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
Disable swap space. swapoff -a Jerry Geis wrote: Is there a way to keep asterisk in RAM and tell linux not to swap it out (ever). There are times when delays are noticed and I presume its due to linux swapping out the program. As if I call right back in then everything responds right away. Wait awhile and the same thing might occur. How can I keep asterisk always in RAM? I use CentOS 5. Thanks, jerry ___ -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cianet channel bank with noise and echo
On 24 Nov 2009, at 13:05, jefferson alexandre wrote: I'm running some asterisks in a small voip provider in Brazil and we're having some problems with a analogic channel bank. When I make calls using analogic extensions, I have a crystal clear quality, but the receptor have a lot of noise and echo. Does the channel bank have onboard echo cancellation? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
Next question will be How can I keep my server from crashing? :) (add more RAM... which may have been a good answer for question 1...) j On Tue, 24 Nov 2009, Alex Balashov wrote: Disable swap space. swapoff -a Jerry Geis wrote: Is there a way to keep asterisk in RAM and tell linux not to swap it out (ever). There are times when delays are noticed and I presume its due to linux swapping out the program. As if I call right back in then everything responds right away. Wait awhile and the same thing might occur. How can I keep asterisk always in RAM? I use CentOS 5. Thanks, jerry ___ -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
On Tue, Nov 24, 2009 at 08:21:43AM -0500, Jerry Geis wrote: Is there a way to keep asterisk in RAM and tell linux not to swap it out (ever). There are times when delays are noticed and I presume its due to linux swapping out the program. As if I call right back in then everything responds right away. Wait awhile and the same thing might occur. How can I keep asterisk always in RAM? Theoretically - yes. Practically - some other component of the system will swap out and cause basically the same performance issues. What else takes much memory on that system? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
I was proceeding from the give them enough rope to hang themselves theory of technical support, which calls for doing just that when users insist on framing their question in terms of a solution they have already made up their mind on without examining whether they are asking the right question to begin with, or considering their problem in a larger context. Jeff LaCoursiere wrote: Next question will be How can I keep my server from crashing? :) (add more RAM... which may have been a good answer for question 1...) j On Tue, 24 Nov 2009, Alex Balashov wrote: Disable swap space. swapoff -a Jerry Geis wrote: Is there a way to keep asterisk in RAM and tell linux not to swap it out (ever). There are times when delays are noticed and I presume its due to linux swapping out the program. As if I call right back in then everything responds right away. Wait awhile and the same thing might occur. How can I keep asterisk always in RAM? I use CentOS 5. Thanks, jerry ___ -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
On Tue, Nov 24, 2009 at 11:21 AM, Jerry Geis ge...@pagestation.com wrote: Is there a way to keep asterisk in RAM and tell linux not to swap it out (ever). There are times when delays are noticed and I presume its due to linux swapping out the program. As if I call right back in then everything responds right away. Wait awhile and the same thing might occur. How can I keep asterisk always in RAM? How much RAM your system have? Concurrent calls? Write or encode calls on disk? You have other softwares (like databases) running on same machine? Maybe, you shoul run the vmstat, ps, top and other tools to detect the bootleneck on your system, and read about kernel swappiness: http://kerneltrap.org/node/3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.1.10 Music On Hold
Hi, I think it can be related to https://issues.asterisk.org/view.php?id=16268 Best regards, Santi 2009/11/24 Örn Arnarson o...@arnarson.net Hello again, I just tried version 1.6.1.9, and the MOH works well there. It seems to be a bug introduced in 1.6.1.10. Best regards, Örn 2009/11/23 Örn Arnarson o...@arnarson.net Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some configuration change I need to implement for this to work properly? Was there a conscious change in Asterisk's behavior? Best regards, Örn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cianet channel bank with noise and echo
On Tue, Nov 24, 2009 at 11:29 AM, Steve Howes steve-li...@geekinter.netwrote: On 24 Nov 2009, at 13:05, jefferson alexandre wrote: I'm running some asterisks in a small voip provider in Brazil and we're having some problems with a analogic channel bank. When I make calls using analogic extensions, I have a crystal clear quality, but the receptor have a lot of noise and echo. Does the channel bank have onboard echo cancellation? Steve, the hardware don't have echo cancellation. Is a TDMoE hardware, plugged directly in asterisk using a straight cable. In asterisk side, the network card haven't auto negotiation, the speed is adjusted to 100mbps full duplex. All electrical sources and ground are ok. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
On Tue, Nov 24, 2009 at 2:36 PM, jefferson alexandre jefferson.alexan...@gmail.com wrote: On Tue, Nov 24, 2009 at 11:21 AM, Jerry Geis ge...@pagestation.com wrote: Is there a way to keep asterisk in RAM and tell linux not to swap it out (ever). On a closely related note, has anyone built a normal (not embedded) system on SSD? It might help if it works well with linux. It seems to make a huge difference with OS like OS X. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cianet channel bank with noise and echo
On 24 Nov 2009, at 13:48, jefferson alexandre wrote: Steve, the hardware don't have echo cancellation. Thats probably it You're relying on Asterisks software echo canceling I have seen mixed results. Have you tried adjusting gains? I'd do the following 1. Turn off echo canceler (makes it more obvious whilst you're trying to remove it) 2. Turn down both gains 3. listening' inside your network (i.e. listening to audio coming to your network from the PSTN), adjust the gain upwards until it sounds suitable 4. 'listening' outside of your network (i.e. listening to audio coming from your network to PSTN) do the same. 5. Test for echo. Adjust the gains down for the sound you hear back (i.e. if you hear person inside your network echoing, adjust the gain in '4'). 6. Try and get it as close to echo free as you can using this method 7. Enable any echo canceling you can find to 'tidy up' the leftover echo, try various ones if you need to. voip-info.org/wiki may help 8. Buy hardware echo cancelers next time ;) S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk trunk CURL hangs in the dialplan
On Monday 23 November 2009 21:30:10 Eric Chamberlain wrote:
We've encountered a strange issue with the trunk version of asterisk.
Our dialplan makes CURL calls and occasionally CURL stops working.
The dialplan looks something like this:
[macro-curl]
; ${ARG1} CURL URL
; ${ARG2} CURL POST
exten = s,1,NoOp(CURL)
...
exten =
s,n(post),Set(RF_CURL_POST=userID=${RF_DIALER_USERID}password=${RF_PASSWOR
D}${ARG2}) exten = s,n,Set(CURLOPT(httptimeout)=5)
exten = s,n,Set(CURLOPT(conntimeout)=5)
exten = s,n,NoOp(CURL(${RF_URL}/${ARG1}?${RF_CURL_POST}))
exten =
s,n,Set(RF_CURL_RESPONSE=${CURL(${RF_URL}/${ARG1},${RF_CURL_POST})})
At this point, CURL either works or it will occasionally hang for a few
minutes. tcpdump doesn't show any traffic from the asterisk box to the web
server.
Something seems to be causing CURL to hang, before it sends out the http
request and the CURLOPT timeouts have no effect on the behavior.
Once CURL hangs, any additional calls to CURL also hang.
After a few minutes, tcpdump will show the CURL traffic going to the web
server.
And CURL begins functioning normally for a while.
Has anyone else seen this? Or have any suggestions on how to debug this?
Sounds like your local DNS resolver isn't answering queries promptly.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] Cianet channel bank with noise and echo
Ok Steve. I will follow these instructions until i get some results. The next time, I will consider buy a hardware with echo canceller ;) On Tue, Nov 24, 2009 at 12:05 PM, Steve Howes steve-li...@geekinter.netwrote: On 24 Nov 2009, at 13:48, jefferson alexandre wrote: Steve, the hardware don't have echo cancellation. Thats probably it You're relying on Asterisks software echo canceling I have seen mixed results. Have you tried adjusting gains? I'd do the following 1. Turn off echo canceler (makes it more obvious whilst you're trying to remove it) 2. Turn down both gains 3. listening' inside your network (i.e. listening to audio coming to your network from the PSTN), adjust the gain upwards until it sounds suitable 4. 'listening' outside of your network (i.e. listening to audio coming from your network to PSTN) do the same. 5. Test for echo. Adjust the gains down for the sound you hear back (i.e. if you hear person inside your network echoing, adjust the gain in '4'). 6. Try and get it as close to echo free as you can using this method 7. Enable any echo canceling you can find to 'tidy up' the leftover echo, try various ones if you need to. voip-info.org/wiki may help 8. Buy hardware echo cancelers next time ;) S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
On a closely related note, has anyone built a normal (not embedded) system on SSD? I've been running Asterisk on a 20GB SSD drive for a while now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
I recently implemented a vmware host using SSDs for the VM storage. I wish you could see the grin on my face right now. It's so fast. Remember thought that all SSDs are NOT created equal... Be careful what you buy. snip On a closely related note, has anyone built a normal (not embedded) system on SSD? I've been running Asterisk on a 20GB SSD drive for a while now. /snip ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experience with LLDP
If you have a network that doesn't support CDP (such as an all Juniper network), LLDP will do the job for you, as long as your phone supports it. The latest Polycom sip firmware supports it (but none of their older phones can run the new firmware, just the newer ones), as well as the latest bootrom. The Cisco 79x1 series and newer support it, but not the older 79x0 series. I believe I spoke with Aastra and Snom at the Astricon tradeshow and they said they support it on their newer models as well. LLDP is fine if you can't run CDP for whatever reason, the main sticking point is the phone support. Thanks, --Warren Selby On Nov 24, 2009, at 2:49 AM, Olivier oza-4...@myamail.com wrote: Hello, LLDP is more and more available on various network elements (endpoint, switches, ...). It seems to ease network configuration. Do you have any experience with it ? How would you rate LLDP ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
On Tue, Nov 24, 2009 at 3:42 PM, Richard Kenner ken...@gnat.com wrote: On a closely related note, has anyone built a normal (not embedded) system on SSD? I've been running Asterisk on a 20GB SSD drive for a while now. And? Noticed any significant performance advantage? /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
On Tuesday 24 November 2009 07:21:43 Jerry Geis wrote: Is there a way to keep asterisk in RAM and tell linux not to swap it out (ever). There are times when delays are noticed and I presume its due to linux swapping out the program. As if I call right back in then everything responds right away. Wait awhile and the same thing might occur. How can I keep asterisk always in RAM? http://asterisk.drunkcoder.com/patches/20091124__dontswap.diff.txt And run with -S flag. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
On Tue, 24 Nov 2009, Richard Kenner wrote: On a closely related note, has anyone built a normal (not embedded) system on SSD? I've been running Asterisk on a 20GB SSD drive for a while now. What mft/model? I was recently quoted a 4GB Compact Flash drive as part of a small system we plan to run asterisk on. Loosely tieing this to the recent thread on swap configuration, assuming a small number of SIP phones and no PSTN hardware, we were planning on 1GB of RAM to avoid swapping to this CF device. I know that CF cards have a limited number of writes before frying. If we keep it from using swap am I really only concerned about voicemail and logs? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
What mft/model? Actually, it's 16GB, not 20GB. It's a Transcend TS16GSSD25S-S. I know that CF cards have a limited number of writes before frying. If we keep it from using swap am I really only concerned about voicemail and logs? That number is quite large, though. I'm taking backups and figure that it has a limited lifetime, but I have no idea if that's 2 years, 10, or 100. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
On Tue, 24 Nov 2009 14:56:32 + (UTC), Jeff LaCoursiere wrote: On Tue, 24 Nov 2009, Richard Kenner wrote: On a closely related note, has anyone built a normal (not embedded) system on SSD? I've been running Asterisk on a 20GB SSD drive for a while now. What mft/model? I was recently quoted a 4GB Compact Flash drive as part of a small system we plan to run asterisk on. Loosely tieing this to the recent thread on swap configuration, assuming a small number of SIP phones and no PSTN hardware, we were planning on 1GB of RAM to avoid swapping to this CF device. I know that CF cards have a limited number of writes before frying. If we keep it from using swap am I really only concerned about voicemail and logs? That will work fine if you select an Asterisk distro intended for such applications, like Astlinux or Askozia. Both of these systems manage disk acces is such a way that they maximize the lifespan of flash media. Both are in fact intended to boot run from much small flash devices that are commonplace these days. They harken back to a time when 1 GB was huge and expensive CF card. There are many Astlinux systems out there that have been running on 64 MB CF cards in IDE adapters, and storing their configs VM to either a separate partition in the CF or a USB stick. I wrote this up back in Jan 2006. http://www.mgraves.org/voip/?p=1092 Michael Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
And? Noticed any significant performance advantage? I never ran it any other way, so have no comparison point. I didn't do it for performance reasons, but reliability. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] keep asterisk in RAM
snip And? Noticed any significant performance advantage? /snip Massive increase in performance on mysql VMs with database sizes that exceed memory size (file caching). Boot times on VMs (windows and linux) under 10 seconds. There is no noticeable change in performance for normal operations on normal VMs because most of the files they're IO blocked by are already cached in memory. I actually went with consumer-grade SSDs (4x OCZ 120gb models) in a raid 10. I know most people say 'those aren't good enough for me'. They are! And as long as you plan for some of them to fail over time, you're still ahead on cost and performance vs enterprise-grade SSDs (read: intel). Synthetic testing with hdparm (sdb is the SSD array, sda is the spinning disk array) is below. This comparison is against 7200rpm disks; I don't have hdparm installed on a box running 15k rpm disks: hdparm -tT --direct /dev/sdb /dev/sdb: Timing O_DIRECT cached reads: 1128 MB in 2.00 seconds = 563.48 MB/sec Timing O_DIRECT disk reads: 1276 MB in 3.00 seconds = 425.01 MB/sec hdparm -tT --direct /dev/sda /dev/sda: Timing O_DIRECT cached reads: 138 MB in 2.03 seconds = 68.03 MB/sec Timing O_DIRECT disk reads: 364 MB in 3.00 seconds = 121.32 MB/sec ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please some enlightment on ENUM !!
Norbert Zawodsky wrote: Leif Neland schrieb: - Original Message - *From:* Norbert Zawodsky mailto:norb...@zawodsky.at *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Monday, November 23, 2009 3:15 PM *Subject:* [asterisk-users] Please some enlightment on ENUM !! Hello all you Gurus out there! Please could you explain something to me: Currently I try to get ENUMLOOKUP() working. Naturally I do all the testing with my own number. I registered my number at e164.org I paid for registration of my number at a registration agent for e164.arpa (I know, I don't need both. I just did the .arpa registration first and later discoverd the free .org service) Assume my number was +4311234567 dig 7.6.5.4.3.2.1.1.3.4.e164.org and dig 7.6.5.4.3.2.1.1.3.4.e164.arpa both return NAPTR records. Now for the less clearer points: Your'e supposed to register your number without any extension. If I have some extensions here, how can the calling party get the correct sip uri to the requested extension? Do I have to run my own DNS server in that case? If for example if someone wants to call extension 10, is the ENUMLOOKUP(431123456710) request forwarded to my local DNS server by the e164.arpa server? Or how does that work? If everybody supported enum, it might be usefull to register extension 10 in enum, otherwise: Your extension 10 must have its own phonenumber, to be able to dial it directly. Just as with ordinary pabx. Eg: 123 555 is the reception 123 555 0010 is extension 10 Just some ideas: Is there free (as in not connected to a voisp) numbers, which can be registered in enum? Then you could use those numbers for extensions. But they would only be callable by enum. If the calling of extensions is only to be used by knowledgeable friends you could have them add your own enum-domain to their setup. Leif Hi Leif! No, I cannot believe that this was the right way. It would mean that I would have to register ( pay !!) for every single extension. BTW the How-To, the registration agent I'm using provides on his website, states, that if you're operating a PBX, you should only register your main number (=without any extensions). I *assume* that if I do an ENUMLOOKUP() of a number which includes some extension at the end, the DNS request is somehow delegated to that sub-server which is authorative over this sub-domain. This leads me to the next *assumption* that the right way would be to run an own DNS server which returns the sip-uri's for my extensions. Can someone confirm this? Norbert Yes... you would have to register (and possibly pay for, dependent on the ENUM registrar) each individual number. The idea behind ENUM is that it's an E164 number that is already yours that maps to whatever you want it to map to (email, SIP, etc). The key point here is that you already own the E164 number. If you do, then you could register them all at e164.org for free. If you don't own the individual numbers, you shouldn't be allowed to register them as your own. That sort of breaks the ENUM concept of a number you take with you as a personal identifier. N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP traffic through Asterisk??
Can you move the transfer functionality to the end device rather than through Asterisk? That's what we do - John On Tue, 2009-11-17 at 14:07 +0100, Ignacio wrote: Thank you very much to both of you. My problem was that I used transfer in the dialplan. I have read that If I have Tt, wW, or hH, then asterisk will always stay in the path. So I have to redefine what I want to do know. Allowing transfers is an useful feature, but I wanted all rtp traffic went p2p. Is there any intermediate solution? Thanks. Regards Ignacio On Mon, Nov 16, 2009 at 7:52 AM, Leonja Cerebro lio...@gmail.com wrote: see the DTMF method on both phones. 2009/11/14 Ignacio sanfermi...@gmail.com Ok, thank you very much. I will try to find any information in asterisk documentation about RTP. On Fri, Nov 13, 2009 at 3:03 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote: I have just established a call between 2 sip phones and I have noticed that all RTP traffic goes through Asterisk Server. I was expecting RTP traffic went to one phone to another phone directly. I set canreinvite=yes in sip.conf in both sip peers. I also tested it with 2 mgcp phones and same result, all rtp traffic goes through Asterisk. Is there any way to force traffic to go from one phone to another? snip I don't recall where it is off-hand but, somewhere in the Asterisk documentation, there is an explanation of how Asterisk makes a decision about reinvites. You may want to look at that to see if your environment satisfies all the requirements and how it can be adapted if it does not - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- We never did too much talking anyway So don't think twice, it's all right -- There are more things in heaven and earth, Horatio, Than are dreamt of in your philosophy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ring group issue
Hi All, I am having a problem with the ring group where when an incoming call comes it rings all the 3 extensions associated to that, but intermittently it rings one extension only once, but the others will be continuously ringing and the goes to generalized voicemail. When I check the log using debug, I could see that the phone/extension that rang only once has sent a busy messageI was not able to figure out what the problem could be.and why it rang only once... Thanks for all your help. Regards Sandesh ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring group issue
What is the channel technology in use? das sandesh wrote: Hi All, I am having a problem with the ring group where when an incoming call comes it rings all the 3 extensions associated to that, but intermittently it rings one extension only once, but the others will be continuously ringing and the goes to generalized voicemail. When I check the log using debug, I could see that the phone/extension that rang only once has sent a busy messageI was not able to figure out what the problem could be.and why it rang only once... Thanks for all your help. Regards Sandesh ___ -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] audio cuts out during IVR
Hi all, I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the audio vanishes in the middle of listening to an IVR background prompt. This happens with both analog (Digium card) and IAX2 incoming calls. The prompts are stored in ulaw format (and the IAX2 calls use ulaw). The asterisk console claims that the IVR prompts are proceeding in the expected fashion, but I can't hear anything. The logs don't report anything interesting. Has anyone seen anything like this? Suggestions? - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring group issue
I am talking about the endpoints (extensions). SIP? DAHDI? IAX? H.323? das sandesh wrote: Hi Alex, I am using Ring All channel strategy... Thanks Sandesh On Tue, Nov 24, 2009 at 10:21 AM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: What is the channel technology in use? das sandesh wrote: Hi All, I am having a problem with the ring group where when an incoming call comes it rings all the 3 extensions associated to that, but intermittently it rings one extension only once, but the others will be continuously ringing and the goes to generalized voicemail. When I check the log using debug, I could see that the phone/extension that rang only once has sent a busy messageI was not able to figure out what the problem could be.and why it rang only once... Thanks for all your help. Regards Sandesh ___ -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring group issue
Hi Alex, I am using Ring All channel strategy... Thanks Sandesh On Tue, Nov 24, 2009 at 10:21 AM, Alex Balashov abalas...@evaristesys.comwrote: What is the channel technology in use? das sandesh wrote: Hi All, I am having a problem with the ring group where when an incoming call comes it rings all the 3 extensions associated to that, but intermittently it rings one extension only once, but the others will be continuously ringing and the goes to generalized voicemail. When I check the log using debug, I could see that the phone/extension that rang only once has sent a busy messageI was not able to figure out what the problem could be.and why it rang only once... Thanks for all your help. Regards Sandesh ___ -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Change the FROM filed username and From Calling id in asterisk
Hello Guys, Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the calling ID as asterisk and username same , What i want is that it should forward some CLI in From Field , I have done my best effort but still not resolved i am adding a callerid in script still same please help me if some one can IP1:5060 - IP2:5060 INVITE sip:0423347871...@ip2:5060 SIP/2.0..Via: SIP/2.0/UDP IP1:5060;branch=z9hG4bK-966123148--16781 75694--693700493-4-..Via: SIP/2.0/UDP IP1:5065;branch=z9hG4bK00749b6d;rport..Call-Id: 4f8a33b207cd65f060b083b57 804d...@ip1..to 804d...@117.20.20.234..to: sip:3347871...@ip1.. *From: asterisksip:aster...@ip1:5065;tag=as0cae0b* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring group issue
We are using SIP channel technology... On Tue, Nov 24, 2009 at 11:03 AM, Alex Balashov abalas...@evaristesys.comwrote: I am talking about the endpoints (extensions). SIP? DAHDI? IAX? H.323? das sandesh wrote: Hi Alex, I am using Ring All channel strategy... Thanks Sandesh On Tue, Nov 24, 2009 at 10:21 AM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: What is the channel technology in use? das sandesh wrote: Hi All, I am having a problem with the ring group where when an incoming call comes it rings all the 3 extensions associated to that, but intermittently it rings one extension only once, but the others will be continuously ringing and the goes to generalized voicemail. When I check the log using debug, I could see that the phone/extension that rang only once has sent a busy messageI was not able to figure out what the problem could be.and why it rang only once... Thanks for all your help. Regards Sandesh ___ -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect Two Asterisk's using isdn Cards
Which cards exactly? It's 2 T1/E1 cards! Specifically, on of it is a TE110P and the other is a TE122! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] snapgear/mcafee sg560 rebooting
Hi all, Does anyone else use the SG560 firewall with Asterisk? I do, and it normally works great, except when it randomly reboots. Has anyone else experienced this annoyance? Did you fix it? - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk trunk CURL hangs in the dialplan
On Nov 24, 2009, at 6:17 AM, Tilghman Lesher wrote: Sounds like your local DNS resolver isn't answering queries promptly. Thanks, I'll look into it. Our CURL function only calls one hostname over and over. Would setting CURLOPT dnstimeout be of use in this situation? -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk trunk CURL hangs in the dialplan
On Tue, 24 Nov 2009, Eric Chamberlain wrote: On Nov 24, 2009, at 6:17 AM, Tilghman Lesher wrote: Sounds like your local DNS resolver isn't answering queries promptly. Thanks, I'll look into it. Our CURL function only calls one hostname over and over. Would setting CURLOPT dnstimeout be of use in this situation? Put that host in /etc/hosts if it is static and under your control... j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Crosstalk - Is there a debug option for logging this?
Hi All, I'm struggling with an intermittent crosstalk issue resulting in a caller's audio being broadcasted to other calls (only one way as they are unable to hear the others listening in). Doing do diligence I've scoured the web in hopes of triaging the issue. So my thoughts are leaning towards this NOT being an Asterisk issue, but instead being related to the telco (PRI) config...however without proper logging this is a guess. Is there a debug option in which I can see how Asterisk is routing the audio for callers? This would at least allow me to capture logging of the call routing to determine if Asterisk is doing this or if it's occurring outside of my control. -JT ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR for asterisk
On Tue, Nov 24, 2009 at 7:12 AM, B.Masoud @ SH i...@saudihome.com wrote: Anyone can recommend a commercial large scale IVR with easy + pro management for asterisk? Those words don't mean anything to anybody except you. For instance, large scale is meaningless. You need to say out loud how large 'large' is. We define 'large' as number of simultaneous phone calls the service needs to support. Are you planning on being the phone voting platform for American Idol? Commercial just means you paid money for it. You can pay money for just about anything. 'Easy' and 'pro' don't usually go together. FreePBX is free, and asterisk is free. You can use them to make IVRs with browser-based management that is easy enough to turn over to Windows users who cannot operate a command-line interface. The pro users can go to a lower level and make customizations or debug with the CLI. I think FreePBX on top of asterisk is the closest I can think of to match 'easy' with 'pro' for asterisk. You can pay Digium for professional support. Or you can pay a third party for professional services to build something like this. Or you can buy an asterisk appliance that does most of the work for you, but you lose some of the control to do things the way you want. You can post to the asterisk-biz list if you are looking to hire professional services to build something like this for you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio cuts out during IVR
On Tue, Nov 24, 2009 at 11:47 AM, Dr. Michael J. Chudobiak m...@avtechpulse.com wrote: Hi all, I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the audio vanishes in the middle of listening to an IVR background prompt. This happens with both analog (Digium card) and IAX2 incoming calls. The prompts are stored in ulaw format (and the IAX2 calls use ulaw). The asterisk console claims that the IVR prompts are proceeding in the expected fashion, but I can't hear anything. The logs don't report anything interesting. Has anyone seen anything like this? Suggestions? Are you playing with the system clock? I've seen cases where putting a clock backwards during a call causes problems with playback. The system seems to think the playback won't have completed until x seconds from the time when the playback began. Or something like that. dramatic ntp changes? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crosstalk - Is there a debug option for logging this?
On Tue, 2009-11-24 at 14:05 -0500, JT wrote: I'm struggling with an intermittent crosstalk issue resulting in a caller's audio being broadcasted to other calls (only one way as they are unable to hear the others listening in). Crosstalk like this isn't a common occurrence, especially on digital audio paths. So my thoughts are leaning towards this NOT being an Asterisk issue, but instead being related to the telco (PRI) config...however without proper logging this is a guess. I would lean that same direction, but it's not a common problem to see on a PRI, so don't be surprised if your telco's first reaction is It can't be us... go talk to your PBX vendor. Is there a debug option in which I can see how Asterisk is routing the audio for callers? This would at least allow me to capture logging of the call routing to determine if Asterisk is doing this or if it's occurring outside of my control. Type core show channels at the Asterisk CLI to see each channel, and what it's being bridged to. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect Two Asterisk's using isdn Cards
mos...@infolog.mr escribió: Which cards exactly? It's 2 T1/E1 cards! Specifically, on of it is a TE110P and the other is a TE122! Hi, That would be a very special need, I'm wondering why connect two asterisk with expensive E1/T1 cards when you can connect them with simple network cards and use SIP or IAX2? Anyway, the way to do it is to define one asterisk as the master (network side) and the other one as the slave (CPE side). You can achieve that configuring one box with signalling = pri_net and the other one with signalling = pri_cpe in chan_dahdi.conf. You can see the rest of the configurations on this link: http://www.voip-info.org/wiki/view/chan_dahdi.conf Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio cuts out during IVR
On 11/24/2009 02:14 PM, David Backeberg wrote: The asterisk console claims that the IVR prompts are proceeding in the expected fashion, but I can't hear anything. Are you playing with the system clock? ... dramatic ntp changes? No, that shouldn't be happening. But I'll keep it in mind while debugging... - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experience with LLDP
2009/11/24 Warren Selby wcse...@selbytech.com If you have a network that doesn't support CDP (such as an all Juniper network), LLDP will do the job for you, as long as your phone supports it. The latest Polycom sip firmware supports it (but none of their older phones can run the new firmware, just the newer ones), as well as the latest bootrom. The Cisco 79x1 series and newer support it, but not the older 79x0 series. I believe I spoke with Aastra and Snom at the Astricon tradeshow and they said they support it on their newer models as well. LLDP is fine if you can't run CDP for whatever reason, the main sticking point is the phone support. Lately, I've seen more and more phones (Aastra, Thomson, ...) support LLDP (in fact LLDP-MED) but you're right to underline some older models can't support it. Thanks, --Warren Selby On Nov 24, 2009, at 2:49 AM, Olivier oza-4...@myamail.com wrote: Hello, LLDP is more and more available on various network elements (endpoint, switches, ...). It seems to ease network configuration. Do you have any experience with it ? How would you rate LLDP ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snapgear/mcafee sg560 rebooting
On 11/24/2009 01:19 PM, Dr. Michael J. Chudobiak wrote: Does anyone else use the SG560 firewall with Asterisk? I do, and it normally works great, except when it randomly reboots. Has anyone else experienced this annoyance? Did you fix it? Oops, never mind. The SG560 was fine. The AC power to it wasn't! - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Route Non-Call Data to Agent Through Queue
Hello, I was wondering if their is a way to use the Asterisk ACD to initiate a call that will route variables through the ACD, which can then be read at the other end by an application. The idea here is instead of terminating a call to an agent I would be terminating some variables/text data. Thanks! Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio cuts out during IVR
On 11/24/2009 02:14 PM, David Backeberg wrote: I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the audio vanishes in the middle of listening to an IVR background prompt. Are you playing with the system clock? Actually, setting the internal_timing option seems to have fixed the problem. https://issues.asterisk.org/view.php?id=15932 - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experience with LLDP
On Tue, Nov 24, 2009 at 12:49 AM, Olivier oza-4...@myamail.com wrote: Hello, LLDP is more and more available on various network elements (endpoint, switches, ...). It seems to ease network configuration. Makes Voice VLAN assignment much easier for sure. Do you have any experience with it ? I work with customers that have mixed environments for access level switches (Cisco, Linksys, Extreme, Juniper, etc) and prefer to use LLDP when the phones support it. It makes sense if you are in an all Cisco environment to use CDP. How would you rate LLDP ? I would rate LLDP as a very useful vendor-agnostic protocol. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')
Hi, I am using codec g729 on two asterisk machines, but when call is forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1 outputs following error and there is no audio. Also the IVRs being played have choppy voice. Insufficient information for SDP (m = 'audio RTP/AVP 18 127', c = '') It is running fine when codec gsm is in RTP traffic. Also I have another server 3 which is also running g729, call from server 3 to server 2 is established but still choppy voice. Earlier I integrated server 3 to server 1 and it was a smooth run. Any idea what could be the possible reasons! /ag ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')
ast guy escribió: Hi, I am using codec g729 on two asterisk machines, but when call is forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1 outputs following error and there is no audio. Also the IVRs being played have choppy voice. Insufficient information for SDP (m = 'audio RTP/AVP 18 127', c = '') It is running fine when codec gsm is in RTP traffic. Also I have another server 3 which is also running g729, call from server 3 to server 2 is established but still choppy voice. Earlier I integrated server 3 to server 1 and it was a smooth run. Any idea what could be the possible reasons! /ag Please provide the asterisk version and g729 codec that is installed on each server, so people can have a clue of what's happening. Maybe could be a known bug or something. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio cuts out during IVR
On 25/11/09 5:47 AM, Dr. Michael J. Chudobiak wrote: Hi all, I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the audio vanishes in the middle of listening to an IVR background prompt. This happens with both analog (Digium card) and IAX2 incoming calls. The prompts are stored in ulaw format (and the IAX2 calls use ulaw). The asterisk console claims that the IVR prompts are proceeding in the expected fashion, but I can't hear anything. The logs don't report anything interesting. Has anyone seen anything like this? Suggestions? Is the machine running a GUI? I.E. Gnome/KDE/XFCE etc -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')
On Tue, Nov 24, 2009 at 10:49 PM, Miguel Molina mmol...@millenium.com.cowrote: ast guy escribió: Hi, I am using codec g729 on two asterisk machines, but when call is forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1 outputs following error and there is no audio. Also the IVRs being played have choppy voice. Insufficient information for SDP (m = 'audio RTP/AVP 18 127', c = '') It is running fine when codec gsm is in RTP traffic. Also I have another server 3 which is also running g729, call from server 3 to server 2 is established but still choppy voice. Earlier I integrated server 3 to server 1 and it was a smooth run. Any idea what could be the possible reasons! /ag Please provide the asterisk version and g729 codec that is installed on each server, so people can have a clue of what's happening. Maybe could be a known bug or something. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center I am running Asterisk 1.2.13. I need to look for the actual source from where I got the codec. /ag ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1950's UK rotary dial phone
Folks, I've got one of those GPO 1950's rotary dial phones that I'm trying to get working in the UK. I've got pretty much everything working with my TDM400, the phone rings and I can receive calls but I cannot dial with the rotary dialer. I have set pulsedial=true or whatever the exact setting is and I can dial from the phone by lifting the receiver and tapping out the number on the hook. However, using the rotary dialer does not work (works fine plugged into my phone line). I have read about the possibilty that the pulse settings may need adjusting in kernel.h in the dahdi driver but I have no idea what to set them to. I have tried tweeking them to various extents but I've not been able to bring it to life yet. Does anyone have any experience getting this to work? Does anyone know the specs for UK pulse dial? How long should the pulses be and what is the gap between them? Thanks, Mike. signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Where are documented channel-dependant Dial options ?
Hi,
I've recently discovered Dial examples such as Dial(DAHDI/g4d/${EXTEN})
but I wonder where I can get an uptodate doc.
Is there any CLI option such as core show channel dialoption that would
explain what g4d exactly means ?
core show application dial doesn't explain much about those
channel-dependant Dial options.
Regards
___
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Re: [asterisk-users] Experience with LLDP
2009/11/24 Jonathan Thurman jthurma...@gmail.com I would rate LLDP as a very useful vendor-agnostic protocol. -Jonathan So I guess, the next item on my todo list is to test LLDP ! Thanks for the advice. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR for asterisk
On Wed, Nov 25, 2009 at 1:12 AM, B.Masoud @ SH i...@saudihome.com wrote: Anyone can recommend a commercial large scale IVR with easy + pro management for asterisk? I don't know what you mean by pro management but you can write IVR applications in any language you want. Personally I like ruby but you can do it java, pyton, php, perl, erlang etc. Drop me a private email if you want to know more. Writing IVRs is not difficult but there are issues you need to think about especially if you are concerned about uptime, load balancing etc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1950's UK rotary dial phone
Have you tested the dialer mechanism to confirm that it actually works? Sounds like your dialer mechanism MAY not be opening/closing the loop properly. Have you tried using the phone on a Telco-provisioned loop? -K - Original Message - From: Mike asterisk-us...@norgie.net To: asterisk-users@lists.digium.com Sent: Tuesday, November 24, 2009 5:03 PM Subject: [asterisk-users] 1950's UK rotary dial phone ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't get pap2 to register from outside the LAN.
On Tue, Nov 24, 2009 at 4:16 PM, Landy Landy landysacco...@yahoo.com wrote: How about adding: insecure=invite,port That didn't work. How weird. I have reset the device to factory settings too. Nothing seems to work. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't get pap2 to register from outside the LAN.
On Tue, Nov 24, 2009 at 3:38 PM, Michael Wyres mwy...@cdm.com.au wrote: I would without the deny and permit directives in the SIP, and rule out some sort of clash there that is rejecting the address the registration is coming from, and take it from there. It made no difference to remove those entries. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1950's UK rotary dial phone
On Tue, Nov 24, 2009 at 06:55:16PM -0600, Karl Fife wrote: Have you tested the dialer mechanism to confirm that it actually works? Sounds like your dialer mechanism MAY not be opening/closing the loop properly. Have you tried using the phone on a Telco-provisioned loop? -K Yeah, I've plugged the phone directly into the phone line and the dialer works just fine. Plug it into the TDM400 and it doesn't work, although I can tap the number usin the hook. Mike. signature.asc Description: Digital signature ___ -- 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
