[asterisk-users] How do I tell if T.38 was used?

I am running Trixbox 2.4 which has Asterisk 1.4.18-1 I have kind of followed: http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 I added to sip_general_custom.conf ;NEEDED!!! t38pt_udptl = yes I did not add this to the actual SIP extension, as I assumed this being general it

[asterisk-users] How is reinvite triggered

Particularly WRT T.38 fax. Supposedly, when fax tones are detected, Asterisk is to do a reinvite asking for T.38. Here is what I am using in my dialplan: [custom-fax1] exten = s,1,Answer exten = s,n,StopPlayTones exten = s,n,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif) exten =

[asterisk-users] Anything like SipT38SwitchOver in Asterisk?

Is there something equivalent to SipT38SwitchOver in Asterisk (in callweaver)... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Asterisk 1.4.17 and RXFAX via T38

I was pointed to the following: http://asteriskforum.ru/viewtopic.php?t=1761 It is in Russian, which I don't speak, but it references an Asterisk patch. Is this patch in 1.4.17? Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?) Anyone work with this?

[asterisk-users] chan_mobile type=

What are the values for type for chan_mobile? headset and phone ??? I get my Treo650 to pair. hcitool scan shows the device. hcitool con comes up empty. I go into Asterisk cli. mobile search shows the device (while I am waiting for a response, I see the phone showing a connection being set

[asterisk-users] More detalis: Re: SIP URI question and NATs

, but this did not make a difference. Johansson Olle E wrote: 10 jan 2008 kl. 15.24 skrev Robert Moskowitz: I am seeing slight differences in URIs. In the case where things work, the URI is [EMAIL PROTECTED] where it does not work is [EMAIL PROTECTED]:5060 In the first case I suspect

Re: [asterisk-users] IEEE 802.1x capable sip phones

Olivier wrote: Mitel and Avaya support 802.1X with proprietary protocols. For Siemens, I'm not so sure. Two facts: Proprietary EAP methods that can actually complete in a reasonable amount of time. Many of these have small security holes and thus are not acceptable as standards. (I know, I

Re: [asterisk-users] IEEE 802.1x capable sip phones

Olivier wrote: I thought that : 1. 802.1X was mainly when you plug your hardphone into your network, 802.1X-2001 was written to secure ports on a 802.3 switch. Originally for PCs works just fine for phones. Really does NOT play with VLANs, but HP cheated (I know their lead engineers).

Re: [asterisk-users] Using Asterisk as an Fax-Gateway for analog Fax devices

I have been trying to do this. The only thing, really holding me up is IAXmodem 1.0 rpm for Centos5. Anyone want to build it for me (I am terrible at compiling; do it maybe once a year). Armin Krämer wrote: Hi, this problem could be a bit tricky. We´ve got some good old Fax-Machines

[asterisk-users] SIP URI question and NATs

I am seeing slight differences in URIs. In the case where things work, the URI is [EMAIL PROTECTED] where it does not work is [EMAIL PROTECTED]:5060 In the first case I suspect that Asterisk did something, perhaps at startup, where it 'decided' it was behind a firewall, so let the firewall

Re: [asterisk-users] IEEE 802.1x capable sip phones

Olivier wrote: When we were starting on 802.1AE (LinkSec), Norm Finn (a CISCO Fellow and long time worker on 802.1 and other layer 2 standards) said it well: Layer 2 security protects and addresses the liablities of the network owner Layer 3 security protects and

Re: [asterisk-users] IEEE 802.1x capable sip phones

Olivier wrote: 2008/1/10, Robert Moskowitz [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Jeronimo Romero wrote: Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated

Re: [asterisk-users] IEEE 802.1x capable sip phones

Olivier wrote: 2008/1/10, Robert Moskowitz [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Olivier wrote: 2008/1/10, Robert Moskowitz [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Jeronimo Romero

Re: [asterisk-users] Using Asterisk as an Fax-Gateway for analog Fax devices

/showfiles.php?group_id=148814 Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Moskowitz Sent: Thursday, January 10, 2008 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk

Re: [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?

Olivier wrote: So, I think in this case (Ethernet link), standard spandsp doesn't help as it needs a TDM board. Nope not the case at all. I have been doing fax--ATA--lan--Asterisk-email for quite some time without ANY zaptel interfaces. Zaptel creats the pseudo interface and that does the

Re: [asterisk-users] Using Asterisk as an Fax-Gateway for analog Fax devices

Jonn R Taylor wrote: I just built one. Give it a try. I will grab it and put it into my local repo after I get back from an errand. thanks! Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Moskowitz Sent: Thursday, January 10, 2008

[asterisk-users] No NAT, but firewall mangles Register SDP

Nailed it! TCPdump on Trixbox 2.4 (Asterisk 1.4.17-1) going out and public side of firewall (Linksys WRT54G running Sveasoft) Firewall is configued NOT to NAT (public addressing on internal network. I stop asterisk (amportal stop). wait 30 min to insure timeout. Start both tcpdumps. Start

Re: [asterisk-users] IEEE 802.1x capable sip phones

Jeronimo Romero wrote: Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. This is so unlikely. I worked on 802.1X and 802.11i. There is just too much overhead there. No way to meet the ITU 50ms

[asterisk-users] tale of two firewalls

I have a server behind a firewall. It is publicly addressed. Should NOT be trying to NAT (how would I know). The connection is a SIP trunk to Broadvoice. I am calling the Broadvoice # from my cell and the call is being routed to my server. With one firewall the INVITE contains information

[asterisk-users] 2 firewalls, different INVITES

I have a SIP trunk to Broadvoice. My Asterisk box (1.4.13) is on public addresses behind a firewall. Originally it was behind a Linksys WRT54G running sveasoft. Sveasoft really can't NOT do NAT even when you turn it off. My Asterisk box is defined as the DMZ box to Sveasoft and it seemed

[asterisk-users] What is the status and future of chan_mobile

I have been looking forward for months to get chan_mobile working. I am limited to using prepackaged Asterisk code, mostly Trixbox. I have recently heard that chan_mobile is considered 'beta' and there is no effort to move it into the main code of Asterisk. Not even for Asterisk 1.6. So

Re: [asterisk-users] What is voice format 8

Moises Silva wrote: You should not care for debug messages unless you are debugging. I have been debugging. My IAXmodem connection. core show codecs Will show you format 8 is ALAW thanks. - Moy On Nov 28, 2007 2:41 PM, Robert Moskowitz [EMAIL PROTECTED] wrote: The IAX2 channel

[asterisk-users] What is voice format 8

The IAX2 channel is to IAXmodem. The SIP extension is an ATA with a fax attached. Nov 28 15:30:20 DEBUG[2997] chan_sip.c: build_route: Contact hop: Nov 28 15:30:20 VERBOSE[3276] logger.c: -- SIP/2201-090995f0 answered IAX2/24729-2 Nov 28 15:30:20 DEBUG[2995] chan_iax2.c: Ooh, voice format

[asterisk-users] Lost setting up IAXmodem after drive crash

A few weeks ago, I lost my Trixbox that was all set up with Hylafax and IAXmodem. I am trying to set it up for email procmail faxmail iaxmodem asterisk sipext ATA (with attached fax). I have followed all the instructions on creating the IAX extension and configuring the IAXmodem config

Re: [asterisk-users] Lost setting up IAXmodem after drive crash

oops. I meant to post this to the Hylafax list :-[ And with a reboot, at least it made the call. Now to get the fax to print! Robert Moskowitz wrote: A few weeks ago, I lost my Trixbox that was all set up with Hylafax and IAXmodem. I am trying to set it up for email procmail faxmail

Re: [asterisk-users] Recommendations for 100 Wifi SIP phone setup

I would like to share some facts about wifi and wifi security vis-a-vis wifi phones. First off, it takes REAL time to negotiate the 4-way-handshake. Not even thinking about the 802.1X authentication. Thus a person walking at a normal rate, going through a door will find themselves

Re: [asterisk-users] spandsp as T.38 termination?

Olivier wrote: Robert, Do you mean T.38 passthrough ou T.38 to T.30 gateway ? The former is said to work with Asterisk 1.4 but the latter is not ... I know about what Asterisk 1.4 can do. And Asterisk 1.2 only does T.30 passthrough :) You need 'stuff' to handle fax. Stuff like spandsp,

Re: [asterisk-users] spandsp as T.38 termination?

extension, and it kicks off rxfax (I can provide mine here) should work just fine with T.38 as they do with T.30 over RTP. However digium refuses to include such a program with Asterisk. On Nov 21, 2007 6:13 PM, Robert Moskowitz [EMAIL PROTECTED] wrote: It seems that Spandsp has everything

[asterisk-users] spandsp as T.38 termination?

It seems that Spandsp has everything in it (when you include rxfax and txfax) to be a T.38 termination when used with Asterisk 1.4? And if so, what version of Spandsp? What version of IAXModem (so I don't have to also deal with T38Modem)? ___

[asterisk-users] Help in getting a dialplan to produce the right CDR info

I have been shaking down a dialplan for SIP fax to efax. The basic senario is an ATA on the same subnet as the Asterisk 1.2 box (avoid RTP packet lose and thus fax crash), calling a 'fax extension' and envoking rxfax then email. I leverage off of context: from-internal-additional-custom, so as

[asterisk-users] Bluetooth questions

I see that the term now is chan_mobile to use a bluetooth to cellphone trunk. (what is in a name? :) ) What I want to know is: Is there any restriction on the bluetooth chipset for the server? Can I use the dongle for PAN and chan_mobile at the same time? Can I use the dongle for headset (a

Re: [asterisk-users] surge protector?

Joe acquisto wrote: APC makes a two line unit. PTEL2. But it's two lines in one jack. I have always been a fan of Triplite. They use old tech when appropriate. I am big on Line Conditioners and UPSs with line conditioning. Of course power in my house is really bad. Anything big kicks on

[asterisk-users] extensions reload -- what impact?

I am putting some scripts together to allow a local admin to add extensions, then to reload the extensions, something like: asterisk -r -x extensions reload Are registered extensions forced to reauth? Are active calls disrupted? ___ --Bandwidth

[asterisk-users] DTLS availablity?

Is DTLS available for Asterisk on any Linux distro? I am most interested in Centos ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] DTLS availablity?

Noah Miller wrote: Is DTLS available for Asterisk on any Linux distro? Nope. I've read that the reSIProcate SIP stack has DTLS support. I found out that DTLS is in openSSL 0.9.8. This is available with Redhat/Centos 5. So the code is there. Perhaps just configuring it to some ports

Re: [asterisk-users] zap and fax

Giedrius Augys wrote: Hi, My situation is : I need to send fax from sip device attached fax over zap channel. Using G711, fax send ok, but is it posible to use t.38 protocol. Maybe someone can suggest me what software to use? You want to go to the spandsp site and read all the challenges.

[asterisk-users] ztmonitor in numeric mode

I have been having no success with my zap interface, and am trying to use ztmonitor to fix it. I am running Trixbox 1.1 and so far have one x100p card (shows up as 'Wildcard). I tried many manual attempts at setting rx and tx gain with no results. Then I read:

Re: [asterisk-users] ztmonitor in numeric mode

Tzafrir Cohen wrote: On Fri, Jul 07, 2006 at 10:57:02AM -0400, Robert Moskowitz wrote: I have been having no success with my zap interface, and am trying to use ztmonitor to fix it. I am running Trixbox 1.1 and so far have one x100p card (shows up as 'Wildcard). I tried many manual