[asterisk-users] DAHDI Dummy for Linux VServers

I'm running dahdi on the host system, and have added the /dev/dahdi/ devices to the guest vserver as recommended in Beave's "Virtual Private Asterisk" whitepaper (http://www.telephreak.org/papers/vpa/). I tried copying libtonezone.so and libtonezone.h to the guest, but I couldn't anything to rep

Re: [asterisk-users] Best Firewall Suggestions?

Hello, we are using vyatta, a linux based router. the software is more focused on routing capabilities, than on firewall rules, but it works fine an there is a very good support. for ha you can use it in a cluster. bye, patrick -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Tel. +49 2151

Re: [asterisk-users] Asterisk in the Cloud

This may help: http://www.amoocon.de/talks/27 - Original Message - From: Dan Journo To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, October 14, 2009 02:22 Subject: [asterisk-users] Asterisk in the Cloud Hi, I was wondering if anyone is s

Re: [asterisk-users] Asterisk in the Cloud

http://en.wikipedia.org/wiki/Cloud_computing An example: http://en.wikipedia.org/wiki/Amazon_Elastic_Compute_Cloud - Original Message - From: "C F" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, October 14, 2009 02:44 Subject: Re: [asterisk-users] Aste

Re: [asterisk-users] T38 negotiations in RTP

Ujjval Karihaloo wrote: > T38 rfc does not detail the sequence of t30 negotiations.. > Like v21 preamble, nsf, Dsi, what the correct sequence of t30 > negotiation. Is. T.38 is not an RFC, it's an ITU recommendation, as is T.30. The T.30 and T.38 specifications (along with T.4 and a few others) f

Re: [asterisk-users] T38 negotiations in RTP

T38 rfc does not detail the sequence of t30 negotiations.. Like v21 preamble, nsf, Dsi, what the correct sequence of t30 negotiation. Is. On Oct 13, 2009, at 8:57 PM, "Kevin P. Fleming" wrote: > Ujjval Karihaloo wrote: >> Is there any t38 spec which details the T30 negotiations that occur

[asterisk-users] ChanSpy on asterisk 1.6

I have read about that on asterisk 1.6, there will be a parameter "o" (Only listen to audio coming from this channel), I have tried, but I still get inbound and outbound audio from the spied channel. Has anyone used this feature? Is it working? Is there any work-around? I will like to only spy the

Re: [asterisk-users] AMI input streams limit?

I will try that tomorrow, as well; thank you. At first it was not a problem when there were just a few entries. Matt Riddell wrote: > On 14/10/09 12:56 PM, Jose P. Espinal wrote: >> Hello List, >> >> >> I was writing something in PHP that connects to AMI and sends a data >> stream ( example of i

Re: [asterisk-users] Best Firewall Suggestions?

>> My customer has a outdated firewall that is also presenting a NAT nightmare > for getting the Asterisk server reachable from the internet. > > What firewalls work good with VOIP? I really want to steer away from any ALG > supported firewall. I just want a good firewall that works well wit

Re: [asterisk-users] T38 negotiations in RTP

Ujjval Karihaloo wrote: > Is there any t38 spec which details the T30 negotiations that occur in > the Media/RTP PAth like the NSF, DSI etc frames that are exchanges > between T38 gateways/endpoints. The T.38 recommendation defines how T.30 negotiations are carried in IFP (Internet Fax Protocol) p

Re: [asterisk-users] AMI input streams limit?

Yes, as you can see in the example stream http://slackware-es.com/ami-input.txt (I can not use pastebin, sorry, emails trigger the spam filter there) I added a context at the end, and then appended some lines (mailboxes). It stops writing at the same point always, but without displaying error m

Re: [asterisk-users] AMI input streams limit?

On 14/10/09 12:56 PM, Jose P. Espinal wrote: > Hello List, > > > I was writing something in PHP that connects to AMI and sends a data > stream ( example of it: http://slackware-es.com/ami-input.txt ), but the > file (voicemail.conf , in this case) does not get fully written. > > I tried pasting the

Re: [asterisk-users] AMI input streams limit?

On Tue, Oct 13, 2009 at 7:56 PM, Jose P. Espinal wrote: > I was writing something in PHP that connects to AMI and sends a data > stream ( example of it: http://slackware-es.com/ami-input.txt ), but the > file (voicemail.conf , in this case) does not get fully written. > > I tried pasting the strea

Re: [asterisk-users] AMI input streams limit?

Sir. I am not trying to 'copy/transfer' a file, I'm making changes to voicemail.conf file via AMI. The error is, (As I wrote before) that voicemail.conf is not being written completely. AMI fails to write all of the entries that I am trying to 'append' to it. It is not reporting any error mes

Re: [asterisk-users] AMI input streams limit?

On Tue, Oct 13, 2009 at 7:56 PM, Jose P. Espinal wrote: > Hello List, > > > I was writing something in PHP that connects to AMI and sends a data > stream ( example of it: http://slackware-es.com/ami-input.txt ), but the > file (voicemail.conf , in this case) does not get fully written. > > I tried

Re: [asterisk-users] Asterisk in the Cloud

You mind elaborating what you mean by cloud? On Tue, Oct 13, 2009 at 8:22 PM, Dan Journo wrote: > Hi, > > > > I was wondering if anyone is successfully running Asterisk in a cloud > environment. > > If you could state which cloud you are using, I’d appreciate it. > > > > Many thanks > > Dan Journ

[asterisk-users] T38 negotiations in RTP

Is there any t38 spec which details the T30 negotiations that occur in the Media/RTP PAth like the NSF, DSI etc frames that are exchanges between T38 gateways/endpoints. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCo

Re: [asterisk-users] outgoing sip calls work; incoming calls fail

On Sat, 10 Oct 2009 18:02:04 -0700 listm...@websage.ca wrote: > On Sun, 11 Oct 2009 02:11:47 +0200 > Ivan Stepaniuk wrote: > > > listm...@websage.ca wrote: > > > > > > On the LAN side I can see the INVITE and OKAY messages which end > > > with a CANCEL, apparently initiated by the Asterisk gatew

[asterisk-users] Asterisk in the Cloud

Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, I'd appreciate it. Many thanks Dan Journo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.co

[asterisk-users] AMI input streams limit?

Hello List, I was writing something in PHP that connects to AMI and sends a data stream ( example of it: http://slackware-es.com/ami-input.txt ), but the file (voicemail.conf , in this case) does not get fully written. I tried pasting the stream directly through telnet to AMI, and everything

Re: [asterisk-users] Best Firewall Suggestions?

- Original Message - From: "Michiel van Baak" To: Sent: Tuesday, October 13, 2009 5:24 PM Subject: Re: [asterisk-users] Best Firewall Suggestions? > On 23:52, Tue 13 Oct 09, Hans Witvliet wrote: >> On Tue, 2009-10-13 at 14:42 -0500, Karl Fife wrote: >> > I think one of the very best o

Re: [asterisk-users] Best Firewall Suggestions?

Interesting on my asterisk box I have installed Virtualbox and I run my firewall/router in a vm, stripped down linux box with iptables, I have snapshoted the image to a working image. it only does ip forwading/vpn/iptable stuff ends up being a low foot print, 256M + 8G / Alex On Tue, Oct 13, 200

Re: [asterisk-users] Best Firewall Suggestions?

On 23:52, Tue 13 Oct 09, Hans Witvliet wrote: > On Tue, 2009-10-13 at 14:42 -0500, Karl Fife wrote: > > I think one of the very best options is pfSense. Free Open-source, > > but it's BSD based, rather than LINUX based. As such it has a lower > > risk of external exploits. The user-interface mak

Re: [asterisk-users] wrond DTMF detection on Zap channel

for disabling the hardware DTMF you intend to recompile zaptel with vpmdtmfsupport=0? Thanks On Fri, Oct 9, 2009 at 6:54 PM, C F wrote: > are you using chan_local? > try disabling the hardware DTMF. > > Sent using my wired Blueberry. > > On 10/9/09, nik600 wrote: >> Dear all >> >> i have a TE20

Re: [asterisk-users] Best Firewall Suggestions?

On Tue, 2009-10-13 at 14:42 -0500, Karl Fife wrote: > I think one of the very best options is pfSense. Free Open-source, > but it's BSD based, rather than LINUX based. As such it has a lower > risk of external exploits. The user-interface makes it incredibly > simple to set up and maintain. The

Re: [asterisk-users] Best Firewall Suggestions?

On 10:04, Tue 13 Oct 09, David Wathen wrote: > Hi, > > My customer has a outdated firewall that is also presenting a NAT nightmare > for getting the Asterisk server reachable from the internet. > > What firewalls work good with VOIP? I really want to steer away from any ALG > supported firewall.

Re: [asterisk-users] tealtime static

On Mon, 2009-10-12 at 05:24 -0700, Robor Oghene wrote: > Dear All, > > Can I mix realtime conf and static configuration files? > Thanks for responses. > The rule is that you can use Realtime and the text config file at the same time. If you use Realtime Static you cannot use the text fil

Re: [asterisk-users] Best Firewall Suggestions?

Steve Totaro wrote: > > > On Tue, Oct 13, 2009 at 2:41 PM, SIP > wrote: > > David Wathen wrote: > > > > Hi, > > > > My customer has a outdated firewall that is also presenting a NAT > > nightmare for getting the Asterisk server reachable from the >

Re: [asterisk-users] Dial Delay

On 14/10/09 4:07 AM, Ron wrote: > Hi All, > > Would just like to know why it takes so long before asterisk process the > call. > > From the time i press send on the phone, it takes about 7 seconds > before i see activity on the asterisk console that asterisk is already > processing the call. > >

Re: [asterisk-users] Best Firewall Suggestions?

On Tue, Oct 13, 2009 at 3:53 PM, Grygoriy Dobrovolskyy wrote: > Allmost your solutions require second server or some hardware, why do you > use shorewall ? Its a iptables rule generator with a friendly config files. > Mine was up and running in 30 min or reading some docs. And you can trace > all

Re: [asterisk-users] Best Firewall Suggestions?

Allmost your solutions require second server or some hardware, why do you use shorewall ? Its a iptables rule generator with a friendly config files. Mine was up and running in 30 min or reading some docs. And you can trace all traffic live. Good day. ___

Re: [asterisk-users] Best Firewall Suggestions?

Check out ClarkConnect and SmoothWall. Regards, Juan Steve Totaro wrote: On Tue, Oct 13, 2009 at 2:41 PM, SIP wrote: David Wathen wrote: > > Hi, > > My customer has a outdated firewall that is also presenting a NAT > nightmare for getting the Asterisk serve

Re: [asterisk-users] Best Firewall Suggestions?

BlankI think one of the very best options is pfSense. Free Open-source, but it's BSD based, rather than LINUX based. As such it has a lower risk of external exploits. The user-interface makes it incredibly simple to set up and maintain. There is an embedded versions of it available to run on

Re: [asterisk-users] Best Firewall Suggestions?

On Tue, Oct 13, 2009 at 2:41 PM, SIP wrote: > David Wathen wrote: > > > > Hi, > > > > My customer has a outdated firewall that is also presenting a NAT > > nightmare for getting the Asterisk server reachable from the internet. > > > > What firewalls work good with VOIP? I really want to steer awa

Re: [asterisk-users] Best Firewall Suggestions?

On Tue, Oct 13, 2009 at 11:22 AM, Gordon Henderson wrote: > On Tue, 13 Oct 2009, David Wathen wrote: > >> ... What firewalls work good with VOIP? > > ...I use Draytek Vigor 2820's these days. ... > > Gordon > I've had great luck with Untangle. Open Source... can shape traffic for SIP / IAX / etc.

Re: [asterisk-users] Best Firewall Suggestions?

David Wathen wrote: > > Hi, > > My customer has a outdated firewall that is also presenting a NAT > nightmare for getting the Asterisk server reachable from the internet. > > What firewalls work good with VOIP? I really want to steer away from > any ALG supported firewall. I just want a good firewa

Re: [asterisk-users] Best Firewall Suggestions?

On Tue, 13 Oct 2009, David Wathen wrote: > Hi, > > My customer has a outdated firewall that is also presenting a NAT nightmare > for getting the Asterisk server reachable from the internet. > > What firewalls work good with VOIP? I really want to steer away from any ALG > supported firewall. I jus

[asterisk-users] Bridge command in 1.6

Hi. I'm using Debian's 1.6.2.0 version of asterisk and trying to get an understanding of bridging two lines together. I've seen the documentian on the bridge command but a good example would probably help me more. At the moment we have a number of SIP lines, one DAHDI line, and an IAX2 line t

Re: [asterisk-users] Best Firewall Suggestions?

Cisco PIX and/or ASA work great. Buy them used on eBay. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Wathen Sent: Tuesday, October 13, 2009 11:04 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [a

Re: [asterisk-users] Call Recording and Posting

On Tue, 13 Oct 2009, Elliot Otchet wrote: > To Steve's other point, you could put all of this into an AGI > program/script, but you'll still also need a fallback mechanism to > actually copy the files to the remote server in the event that it is > unavailable/unreachable. To me, having two lin

Re: [asterisk-users] Dial Delay

That's what I thought... When I changed it in DAHDI, it also affected my in-house SIP calls. Ignore at your risk... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Sent: Tuesday, October 13, 2009 11:33 AM T

Re: [asterisk-users] Best Firewall Suggestions?

pfsense On Tue, Oct 13, 2009 at 12:12 PM, Doug Lytle wrote: > David Wathen wrote: > > > > Hi, > > > > My customer has a outdated firewall that is also presenting a NAT > > nightmare for getting the Asterisk server reachable from the internet. > > > > What firewalls work good with VOIP? I really

Re: [asterisk-users] Dial Delay

thank you sir. but i think tonedur is only set at zapata. but my calls are basically SIP and IAX. call from sip user to asterisk is SIP. then asterisk to another asterisk via IAX. then from this asterisk to a SIP trunk the call the number. first asterisk is simply used as a registrar using real

Re: [asterisk-users] Best Firewall Suggestions?

David Wathen wrote: > > Hi, > > My customer has a outdated firewall that is also presenting a NAT > nightmare for getting the Asterisk server reachable from the internet. > > What firewalls work good with VOIP? I really want to steer away from > any ALG supported firewall. I just want a good fire

[asterisk-users] Best Firewall Suggestions?

Hi, My customer has a outdated firewall that is also presenting a NAT nightmare for getting the Asterisk server reachable from the internet. What firewalls work good with VOIP? I really want to steer away from any ALG supported firewall. I just want a good firewall that works well with Asterisk.

Re: [asterisk-users] Call Recording and Posting

Steve's spot on about wanting to move the calls when they're available. We're the instant gratification society. Anytime I've linked multiple independent systems together I've always had to plan for one system being offline - either for maintenance, upgrades, etc. so please consider that in yo

Re: [asterisk-users] Dial Delay

This may or may not be related, but if your tonedur is set too high, this will cause this - a value of 300 caused this for me, changing to 80 resolved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Sent: T

[asterisk-users] Dial Delay

Hi All, Would just like to know why it takes so long before asterisk process the call. From the time i press send on the phone, it takes about 7 seconds before i see activity on the asterisk console that asterisk is already processing the call. But, if i do a sniff, i immediately see the INV

Re: [asterisk-users] G729 in asterisk upgrade issue

Kevin P. Fleming wrote: > Luis Silva wrote: >> I have an asterisk in 1.2 version with 30 g729 licenses. I what to >> upgrade it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3. >> >> For what I understand I can make the backup of the license files in /var/lib/asterisk/l

Re: [asterisk-users] strange transcoding values

On Tue, Oct 13, 2009 at 10:19:19AM +0200, Gianluca Baù wrote: > Hello guys, > i have a question about a voip gateway we use. > I saw those values typing in cli: > > core show translation > >g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex > ilbc g726 g722 slin16 >

Re: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4

We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on "attended transfers". What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed o

Re: [asterisk-users] strange transcoding values

>> On Tue, 13 Oct 2009, Gianluca Baù wrote: >> Hello guys, >> i have a question about a voip gateway we use. >> I saw those values typing in cli: >> >> core show translation >> >> g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex >> ilbc g726 g722 slin16 >> g723 -

Re: [asterisk-users] strange transcoding values

On Tue, 13 Oct 2009, Gianluca Baù wrote: Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - -

[asterisk-users] missing CDR records in cdr while kick from meetme

dear all, i facing one big issue in CDR , i have one user connected in meetme and then and after some time one user are joined meetme bu this user was originally dialed by , AMI via Originate and when remote user picked up call he joined meetme room everything is fine , but now 1 st user kicked 2

[asterisk-users] strange transcoding values

Hello guys, i have a question about a voip gateway we use. I saw those values typing in cli: core show translation g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - - - -- - - - - - -

Re: [asterisk-users] [SOLVED] Asterisk to Asterisk access voicemail - not working

Joseph wrote: > On 10/12/09 13:29, Ivan Stepaniuk wrote: >> Check the "relaxed" dtmf setting on yhe linksys, and also check the >> Impedance, Rx and Tx gain, As well as the DTMF duration, All this >> "knobs" can mess up your DTMF tones if there is something wrong. Just >> > > Solved! > The pro

Re: [asterisk-users] Call Recording and Posting

Steve Edwards wrote: > On Tue, 13 Oct 2009, Dan Journo wrote: > > >> To avoid the problem of deleting/copying calls that are still being >> recorded, I could record the call into a temp directory. Then using the >> dial plan, I could copy the temp recording into the ftp root directory >> once

Re: [asterisk-users] Call Recording and Posting

Dan Journo wrote: > Thanks for that. > > I really appreciate it! > > Dan > As pointed by the follow-ups, note that the recordings are not taken from the monitor but from an upload folder inside, the dialplan takes care to move there the files for the ended files only issuing a 'System' comman