Re: [asterisk-users] small homebrew pbx

On 15/06/15 07:46, lu...@sulweb.org wrote: Hello all, Given the requirements above, what's a cheap but working PCIe card / USB adapter I could buy for this kind of PBX? Do I need things like echo cancellation? Do I need FXS ports? Thanks in advance, Lucio. I would get hold of some lower-pow

Re: [asterisk-users] Updating to 11.7.0

On 05/03/14 12:56, Paul Hayes wrote: I appreciate that and I do understand why but that setting doesn't work as described, it seems to do nothing. While we're at it, what's the recommended alternative method to replace using "asterisk -rx" in bash scripts now? cheer

Re: [asterisk-users] Updating to 11.7.0

On 19/12/13 17:15, David Lee (digium) wrote: On Dec 19, 2013, at 10:34 AM, Jerry Geis mailto:ge...@pagestation.com>> wrote: [snip] Looking that up, it says add to asterisk.conf [options] live_dangerously = yes After doing this, and stopping and starting I still get the message. I'm havin

Re: [asterisk-users] Sip Registration Hijacking

On 20/01/12 01:36, eherr wrote: It is also register on an AudioCodes MP-118. Thanks, -E Is the Audiocodes gateway accessible online? Have you set a strong password for it's web interface (and cli if it has one)? It is possible someone is breaking into that and getting the SIP password o

Re: [asterisk-users] local channels and g729a voice quality

On 16/01/12 07:59, Roi Stork wrote: I also asked my provider to test call me using their Cisco as5300 system and g729 codec and compared it with ulaw. The difference is unnoticable. ^^ this doesn't make any sense, the difference *should* be very much noticeable. g729 is a lower quality codec

Re: [asterisk-users] Change port from 5060 on Snom phone

On 06/01/12 16:17, Ishfaq Malik wrote: Hi Does anyone know how to change the target port on a Snom phone. I have tried adding : to the end of the registrar but this doesn't work. It should do. Try putting : into Outbound Proxy and leave the Registrar box just set to . Can you email me off

Re: [asterisk-users] Connecting to an Old Phone System

On 06/01/12 13:14, Dan Journo wrote: Is there such a thing as an ISDN30e PCI card which can be used with a copy of Asterisk, that can act like a voip gateway between the old phone system, and our asterisk box? Yes Digium sell 2 port PRI cards that support E1. TE200 series. I use them like th

Re: [asterisk-users] Wanted a modified SIP message body

On 31/08/11 08:46, Jaime Lozano wrote: Hello, I agree with you, I'm not explaining the problem in a proper manner, because of my lack of Asterisk knowings. I send the Wireshark captures. 3com telephones take the timezone TZ:7200 from the 3Com PBX to show the time right. But what if I want a 3Com

Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?

On 27/08/11 10:14, Gordon Henderson wrote: On Sat, 27 Aug 2011, Alan Lord (News) wrote: On 26/08/11 19:02, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. We've been using the Siemens Gigaset 685IP range for over three years and I'm (still

Re: [asterisk-users] Trouble with *8 Pickup

On 15/08/11 15:41, Ishfaq Malik wrote: On Mon, 2011-08-15 at 15:32 +0100, Paul Hayes wrote: is this bug already reported at the issue tracker/jira? Is someone working on it? Karsten https://issues.asterisk.org/jira/browse/ASTERISK-18225 That's a different issue to what we have

Re: [asterisk-users] Trouble with *8 Pickup

is this bug already reported at the issue tracker/jira? Is someone working on it? Karsten https://issues.asterisk.org/jira/browse/ASTERISK-18225 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Trouble with *8 Pickup

On 12/08/11 08:46, Ishfaq Malik wrote: Have you seen it in any other versions of 1.8 or is it something that has happened in the latest release? I've not specifically seen this issue with other versions of Asterisk but then I've never tried to replicate it. The only time I've seen this with

Re: [asterisk-users] Trouble with *8 Pickup

2011/8/11 Ishfaq Malik mailto:i...@pack-net.co.uk>> On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote: > Ah, now this is interesting as one of our clients had the same problem the other day; in our case when they performed the *8 they got an extension unavailable from a comp

Re: [asterisk-users] snom and srtp

On 03/08/11 03:15, James Perkins wrote: Hi, I am running asterisk 1.8.5.0 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when tryi

Re: [asterisk-users] Lightning and thunder (Claude Hayn

On 27/07/11 19:41, Claude Hayn wrote: The office manager freaks out each time and starts randomly rebooting devices in no particular order including the UPS, PBX, Asterisk Gateway, firewall and router. Ahh that old chestnut. That's never a good thing, try to tell them not to do this, althou

Re: [asterisk-users] Strange network issue

On 28/07/11 02:58, Mike Diehl wrote: Any ideas? Mike. I'd go on site if possible and see what actually happens at 19:00. Set up a wireshark trace capturing all traffic through their router. -- _ -- Bandwidth and Colocatio

[asterisk-users] Fwd: Re: Securing Asterisk

Original Message Subject: Re: [asterisk-users] Securing Asterisk Date: Wed, 27 Jul 2011 09:28:54 -0700 From: Myles Wakeham To: p...@provu.co.uk On 07/27/2011 09:23 AM, asterisk-users-requ...@lists.digium.com wrote: On 23/07/11 18:38, CDR wrote: > I beg to differ. Digium is h

Re: [asterisk-users] Securing Asterisk

On 23/07/11 18:38, CDR wrote: I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or

Re: [asterisk-users] Securing Asterisk - How to avoid sending, "SIP/2.0 603 Declined"

On 23/07/11 04:48, Bruce B wrote: Quote,/"How do the users register to begin with, if their REGISTER requests won't be processed unless their IP is already known to be a registrant? :-)"/ Well, unfortunately I don't have the luxury of knowing their IP and the closest I know is their IP range.

Re: [asterisk-users] a=sendonly Music On Hold ignored

On 19/07/11 08:20, Michael wrote: On the AsteriskNow system, it gives an OK, but nothing happens, there's no music and after some time, the call even drops for empty RTP. That's the log there: What does the Asterisk CLI show when this happens on your AsteriskNow system? --

Re: [asterisk-users] Re : Re : Direct RTP with Asterisk

On 20/06/11 13:18, Eric Wieling wrote: If you can't ping between the two end points, then you can't do direct RTP. precisely. If 10.10.9.1 isn't reachable from the network that 10.10.8.1 is on then 10.10.8.1 isn't going to be able to send RTP to 10.10.9.1. You need to add routes to the ro

Re: [asterisk-users] PAP2T provisioning via SRV record?

On 13/06/11 19:44, Mike Diehl wrote: Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: _sip._udp.example.com However, the PAP doesn't seem to be able to find my server with this hostname.

Re: [asterisk-users] Connect intercom to Asterisk?

On 07/06/11 09:47, Gilles wrote: Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the

Re: [asterisk-users] Playing with sipvicious ..

On 01/06/11 16:13, Allen David Niven wrote: what does ossec give u that fail2ban does not ? thx and cheers Replied to list so others can find this in the future if they want to. I haven't spent a lot of time investigating fail2ban as I was already using ossec before I saw much talk about fa

Re: [asterisk-users] standalone PRI-to-SIP converter

On 27/05/11 16:10, Michelle Dupuis wrote: I'm looking for recommendations for standalond PRI to SIP converters. (Needs to be outside the asterisk box - so a PCIe card won't do) I've used redfone but this project doesn't need the redundancy features... Thanks! A 2nd Asterisk box with a PCIe

Re: [asterisk-users] Asterisk 1..8 multiple queue

On 26/05/11 23:18, Satish Patel wrote: Thanks, I went through this example before. I was confuse and wondering how should I add third queue in this picture? From the example: *CLI> database put queue_agent 0001/available_queues support^sales "support^sales" is a list of queues. Put

Re: [asterisk-users] Reporting Tool: To show who is login, queue, ... etc

On 26/05/11 15:03, Justin Sherrill wrote: Queuemetrics is neat-looking. However, it requires MySQL, and I'm using Postgres. Does anyone have a recommendation for a different product for reporting usage that's not tied to MySQL? It uses JDBC so it should work with any storage engine you can

Re: [asterisk-users] SIP per-call heartbeat?

On 24/05/11 12:08, Tony Mountifield wrote: OK, thanks. Sounds like there was some kind of issue at the ITSP then. I have seen this happen with broken SIP-ALGs in routers too. The ITSP sends the BYE but for some reason a broken SIP ALG will not deliver the packet to the right place. The IT

Re: [asterisk-users] Sending call to specific IP address

On 23/05/11 22:30, Elliot Murdock wrote: Hello, I am wondering how to send a call to a specific IP address that is different than the host of the URI. For example, an invite to the URI is "j...@phone.com " needs to be sent to the IP address 123.456.789.255, not to the IP

Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

Hi, It looks to me that the 401 unauth packets aren't getting back to the phones. Which suggests a network/router/nat issue rather than anything wrong with the asterisk or phone configuration. Cheers, Paul. On 8 May 2011, at 01:59, GNUbie wrote: > Hello all, > > I have installed the .deb

Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory

On 05/05/11 14:16, Olle E. Johansson wrote: We've had that for quite some time. There's an option to Dial() and one for Queue() to enable it. Check the documentation. /O yes my only problem with the 'c' option for the Dial command is that it still seems to add the Reason header if the ca

Re: [asterisk-users] audiohook.c: Failed to get 160 samples from write factory

On 05/05/11 14:04, Jonas Kellens wrote: Hello list, what does this mean : [May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause code, buddy. The cause code!!! [May 5 14:58:12] DEBUG[8770] chan_sip.c: This call

Re: [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2

On 05/05/11 13:41, Richard Kenner wrote: Asterisk does indeed send an Options before the OK but my 57i doesn't seem to mind. That's odd. It does for me. Perhaps you need to upgrade firmware on the Aastra phone? The problem occured when I DID upgrade it! Precisely to the one you mentioned.

Re: [asterisk-users] Issue with Asterisk & Aastra 57i at v3.2

On 05/05/11 04:37, Richard Kenner wrote: I recently tried to update my Aastra 57i to version 3.2 and ran into a problem. It won't properly register and says "contact mismatch". I added "sip contact matching: 2" to aastra.cfg, but that didn't help. When I look at the SIP trace, but I see is the

Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

On 05/05/11 00:02, Ira wrote: Not that it applies but I recently installed a Snom M3 and it seems to behave like you want. When I walk out of range and then back in the call is usually still there. I've not tested past that so it might hang up after an unknown timeout. Ira The difference her

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On 05/05/11 05:41, Cary Fitch wrote: Flavio E. Goncalves www.asteriskguide.com Compare to which version of Windows… Patches are a never ending process Cary Fitch I think this attitude is half the problem. Asterisk is not a desktop computer

Re: [asterisk-users] best current version and motherboard/CPU compatibilities

On 04/05/11 18:17, || dave cantera Mobile wrote: paul, doug, I had several AMD athlons 64bit... no problems running centos, suse. they seem solid on 1.4.xx... had a few intel celerons and P4s. they were good as well. guess I was Lucky back then! thanks for supporting the list! daveC don't get

Re: [asterisk-users] best current version and motherboard/CPU compatibilities

On 04/05/11 17:10, || dave cantera Mobile wrote: doug, why are you shaking!?!?... do you have a better recommendation? daveC AMD K6 CPU brings back some pretty bad memories from me too. Doug Lytle wrote: C F wrote: model name : AMD-K6(tm) 3D processor *shudder* Doug --

Re: [asterisk-users] Password to be ecrypted?

On 03/05/11 09:09, Robles Román, José Miguel wrote: Perhaps using one-way hash functions (http://en.wikipedia.org/wiki/Cryptographic_hash_function) like MD5 or SHA-x, even if you get the file with passwords and the code that checks them, it would be difficult to find a collision (a password th

Re: [asterisk-users] SIP Invite and Asterisk API/Variable

On 24/03/11 05:49, Olivier CALVANO wrote: The To, "To:", can i get it into a variable for sent it at a API ? You want the sip_header function: http://www.voip-info.org/wiki/view/Asterisk+func+sip_header cheers, Paul. -- _ -

Re: [asterisk-users] Why shouldn't I use 1.8?

On 25/03/11 14:36, Douglas Mortensen wrote: Now that we've hashed out some thoughts on the most stable version of asterisk, I'd like to ask the question as to why I should NOT use 1.8? What are specific reasons? For instance a few days back I was speaking with James at Rhino Equipment. He said

Re: [asterisk-users] SIP registration DoS but no logs in messages

On 17/03/11 05:37, Patrick wrote: Dear mailing list, I've a Asterisk 1.4.21.2~dfsg-3+lenny1 package installed on my debian and I've a strange behavior. After some days running normally, my asterisk is under heavy attack, however, there is nothing logged in the console (logging from debug -> err

Re: [asterisk-users] ADA: DOA?

On 06/10/10 20:25, Ken D'Ambrosio wrote: > Hey, all. While ADA can still be downloaded, that's about all that I see. > No development, no recent mention, and -- perhaps worst of all -- it > appears not to work properly under 64-bit systems. So, assuming Digium's > abandoned it, are there any su

Re: [asterisk-users] minimum card for dahdi timing source ?

On 02/10/10 17:24, mancyb...@gmail.com wrote: > Hi All, > > for a vicidial server which uses only voip, > which is the minimum telephony card which would provide the required clock > timing source for conferences to work properly ? > > Maybe the Digium TDM410PLF card > without any daughter card >

Re: [asterisk-users] SIP flood attacK

On 03/10/10 21:19, Greg Saunders wrote: > Hello all. I was recently the victim of a SIP flood attack. I'm > wondering what is the best method to prevent such things in the future. > Many thanks > Greg > do one of the following: - use deny & permit lines in sip.conf &/or iax.conf to restrict any

Re: [asterisk-users] Playing with sipvicious ..

On 18/08/10 17:10, Gordon Henderson wrote: > > ... using it as a tool and understanding what it does... > > So one part of it's toolset identifys valid SIP accounts - and I was under > the impression that alwaysauthreject=yes was supposed to stop this... > > However, it sends a request for a highly

Re: [asterisk-users] Asterisk Hardwares

On 16/08/10 11:46, Tino wrote: > Hello, > > Can antbody recommend devices that can be used along with my Asterisk > server > > Paging Amplifier > SIP enabled Paging Gateway > VOIP SIP loudspeaker > > Also , please recommend video phone sets that suppot paging, intercom > (autoanswer) > > Thanks >

Re: [asterisk-users] Small PC to build and run Asterisk

On 14/06/10 18:11, Gordon Henderson wrote: > On Mon, 14 Jun 2010, Chris Bagnall wrote: > >> Actually, the Atom seems to be surprisingly powerful. We have a couple of >> Atom boxes with transcoding and conferences enabled without issue. I >> wouldn't pretend it'll cope with hundreds of conference pa

Re: [asterisk-users] Dual Atom mobo - call capacity

On 11/06/10 01:19, Michelle Dupuis wrote: > I'm looking for a small formfactor mobo for an install that needs to handle > 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - > anyone know what kinds of call volume that will handle? > > MD Any of the Atom CPU systems will /eas

Re: [asterisk-users] OpenVPN/SNOM 820: a review.

--[ UxBoD ]-- wrote: >> > Would be nice if the VPN support could be back ported to the 360s. Never going to happen, there isn't enough flash memory to store the code. The Snom370 has had OpenVPN support for quite a while though. cheers, Paul. -- ___

Re: [asterisk-users] Odd bug in Siemens C460IP ?

Robert Lister wrote: > Hello, > > I think I have encountered an odd bug in Siemens C460 IP/dect handsets, > which is a bit annoying, and I'm not (yet) sure how to get round it without > lots of hacks. > > Basically, on all external incoming calls, we set: > > exten => s,n,SIPAddHeader(Alert-In

Re: [asterisk-users] Siemans SIP/PSTN phone S450

Adrian Marsh wrote: > Hi All, > > Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server, > and I see "Got SIP response 405 "Method Not Allowed" back from > 192.168.3.64" but the phone seems to work ok. > > Any ideas where it falls over in the SIP protocol? I've included this >

Re: [asterisk-users] Siemens Gigaset DECT base provisioning

Olivier wrote: > Hello, > > My goal is to provision C450IP or S450IP models. > Has anyone a hint to provision them from configuration files ? > > Usually, we use dedicated menu embedded inside Gigaset handset. > An http server also exists but I couldn't find any dhcp-tftp combination > to config

Re: [asterisk-users] POE injector

Noah Miller wrote: >> I'm looking for 24 or 48 port IEEE802.3af POE injector. >> Any recommendation? > > Yes. For the price of one of those multi-port injectors, you can come > close to the price of a new Netgear or 3Com PoE switch. The injectors > typically add power to the unused pairs (mode B

[asterisk-users] blind transfer on hook-flash from SIP phone

Hi, I have a SIP phone which does not natively support SIP transfers (REFER etc...). So far all that is possible is to enable blind transfers using the t and T arguments in Dial from the # DTMF key. The phone has an R button on it and this can be setup to either send an RFC2833 hook flash me

Re: [asterisk-users] wifi sip phone real-world experiences?

Alex Crow wrote: Alban, Thanks! Where on earth did you source this? I can't seen to find hide nor hair of it here in the UK :( Alex On Mon, 2007-06-04 at 16:01 +0200, Alban wrote: Hi, I've tested several wifi phone (UtStarcom, Hitachi 3000 and 5000, and one Siemens). The Siemens is the best

Re: [asterisk-users] wifi sip phone real-world experiences?

Zoa wrote: Gordon Henderson wrote: On Sun, 3 Jun 2007, Andrew Kohlsmith wrote: On Sunday 03 June 2007 4:30 pm, Alex Crow wrote: No frills, specs look good, price seems excellent! http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519 That's a terrible phone. I've tried them. t

Re: [asterisk-users] OT:spa942 provisioning

Benko wrote: Hello! Sorry for the OT-thread, but i don't know where else too ask... Has anyone done http-provisioning of a Linksys SPA942 with client side ssl-authentication? Where do i get the CA from? I'm aware of the Sipura mass deployment howto on voip-info.org, but it doesn't cover the auth

Re: [Asterisk-Users] F3000 registering to asterisk

Neil Cherry wrote: [snip] How did you get access to the web config? What user and is it the default password/access code? type it's IP address into a web browser. Username: admin, password: psw is the default. cheers, Paul. ___ --Bandwidth and Co

Re: [Asterisk-Users] dipura 2002 auto dial or intercom

This called "hot line" or "batphone" (as it's like the phone the commissioner used to have in Batman that went straight through to Bruce Wayne). Set the dialplan to this: (S0<:#>) where is the number/SIP address you want to dial. Note, that's a zero after the S. Anton K

Re: [Asterisk-Users] Development news :: T38 passthrough support

The SPA-2100 is the only one to support T.38 at the moment though.  SPA-2002 has the ability to support t.38 (i.e. it has the processing power required) but the firmware support isn't there yet. C F wrote: On 3/11/06, Kristian Kielhofner <[EMAIL PROTECTED]> wrote: Olle E Johansson

Re: [Asterisk-Users] SPA-3000: Dual Registrations?

Are you trying to register both lines to the same user account in *?  That wont work, a user can only be registered once at any time. Kerry Garrison wrote: We just posted an updated guide to the SPA-3000 a few days ago. The example uses AMP but all the settings are there: http://voipspeak.n

Re: [Asterisk-Users] Linksys SPA-941 Admin Guide

we should be getting a limited number in a couple of weeks time.  Proper stocks will be arriving in January - www.provu.com Paul. Senad Jordanovic wrote: [EMAIL PROTECTED] wrote: There is a review on the homepage at http://voipspeak.net It has been available for a few weeks, it

Re: [Asterisk-Users] Anyone using Parlay VoXip SIP Gateway with Asterisk ?

I've used one with a Snom SIP server system & it worked quite well but not tried it with * unfortunately.  Voxtream support team are excellent though & I'm sure they'll help you get it working. Robert Rozman wrote: Hi, we're having quite some problems with new hardware we're testing - Pa

Re: [Asterisk-Users] Linksys PAP2: supported codecs

yes that's what i'm lead to believe as well.  Only the SPA-2100 & SPA-2002 support two simultaneous g.729 calls, the older/lesser models don't have the processing power required to encode two g.729 streams. Rich Adamson wrote: I don't think they want to solve it. It's the same with the

Re: [Asterisk-Users] Sipura 2000

You could get in touch with the company who is providing the settings for the Sipura adaptor (should be able to find out who it is from the Settings URL) & ask them to change the settings to be user-changable. The permissions for each setting is configured through the http configuration & it s