[asterisk-users] [Asterisk 0013405]: [patch] T38 gateway (fwd)

testers needed -- Forwarded message -- Date: Wed, 11 Nov 2009 17:48:04 -0600 Subject: [Asterisk 0013405]: [patch] T38 gateway A NOTE has been added to this issue. == https://issues.asterisk.org/view.php?id=13405

Re: [asterisk-users] Asterisk keeps sending invite to sip phone "No response to critical packet"

Thanks Alex I suspected that no ack was being sent/received too. The invites are getting sent to the phone but nothing is coming back from the phone to the firewall. Does anybody know how I can sniff packets being sent and received to/from the phone and/or modem router the phone is connected to? I

[asterisk-users] Need Adapter/Gateway with PSTN-interface

I am looking for a gateway/ATA that can take conversations on the analogue line (PSTN) and send them to the Asterisk server on the private network. I was experimenting with the Atcom AG-188N but the "FXO"-port only supports lifeline, so it's not a real FXO-port that can send incoming calls to my p

Re: [asterisk-users] SendText

i have my own SMS provider as we sell SMS .. so i have setup my call center with SMS sending for several services and alerts like a Missed Call when i'm not registered it will send me an sms to alert me. it's pretty the same as Matt discribed.. you call an AGI which may use cURL to hit the HTTP

[asterisk-users] Cisco 7970 SIP endless ringing...?

Anyone know what would cause an endless ringing situation? I have a snom360 and cisco 7970 (sip 8.5.3). I have an incoming trunk which dials both phones: [gp710] exten => _[*1-9].,1,Dial(SIP/li...@cisco7970&SIP/li...@snom360,60) exten => _[*1-9].,n,Hangup If a call comes in, I can answer the cal

Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface

On Thursday, November 12, 2009, jonas kellens wrote: > Could someone advice on a gateway that can take analogue calls and > transfer them on my local network ?! FWIW, I've had a few recommendations for the Linksys SPA3000. However, I haven't tried this for myself yet since I'm still in the planni

Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface

I've read (through google) that the Linksys SPA-products do not have good voice quality on the PSTN-line. Grandstream HT486 is also just lifeline and EOL. The only I come up with is Patton-gateways but these are not at all cheap ! Jonas. On Thu, 2009-11-12 at 10:13 +, Steve Howes wrote: >

Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface

On 12 Nov 2009, at 09:33, jonas kellens wrote: > I am looking for a gateway/ATA that can take conversations on the > analogue line (PSTN) and send them to the Asterisk server on the > private network. > > I was experimenting with the Atcom AG-188N but the "FXO"-port only > supports lifeline

Re: [asterisk-users] SIP source address error

Quoting Matt Riddell : >> [Nov 11 14:29:47] WARNING[6365]: chan_sip.c:1787 __sip_xmit: >> sip_xmit of 0xb63d5694 (len 444) to 192.168.8.30:5060 returned -1: >> Operation not permitted > > Are you binding to an address that the box doesn't own? > > Check the top of sip.conf. It's set to bind t

[asterisk-users] Scheduling destruction of SIP dialog

Hello, I got situation which is unclear for me, hope somebody could explain this. A calls to B INVITE sent from A to B B responds with 100 Trying B responds with 183 Progress After 10 seconds: Asterisk CLI: Scheduling destruction of SIP dialog '..' in 32000 ms (Method: INVITE) Asterisk sends CAN

[asterisk-users] Incoming Call Ring

Hello, I have Asterisk set up with 6 extensions. When a call comes in, I use a Dial command to call all the extensions together until someone picks up. The problem is, when there is an incoming call and an extension is in use, if the extension puts down the phone while the incoming call is

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

2009/11/11 Tzafrir Cohen > On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote: > > 2009/11/10 Tzafrir Cohen > > > > > On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote: > > > > Hello, > > > > > > > > 1. How can specify in /etc/dahdi/genconf_parameters file that a port > from > > > a >

[asterisk-users] Termination Question

Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B & C, each one is located in a different country. Asterisk A is the main one, and both B & C are connected to it. My question is, when a call is originated from B to C, it will have to go through

[asterisk-users] BLF with SPA941?

Appearently SPA941 is less than a SPA942 without two ports, poe and backlight. There is less features too, it doesn't support BLF. Is it possible to hack 942-software into 941, or is there another workaround? Leif ___ -- Bandwidth and Colocation Provid

[asterisk-users] AST_CONFIG, MEETME_INFO and meetme.conf

Hello, To make my dialplan more robust, I thought I wouldn't include any meetme-specific rules and I would exlusively rely on meetme.conf data. For each dialed number, I would check if this number is used as a conference room number in meetme.conf. When I'm trying to implement this, I can see th

Re: [asterisk-users] Incoming Call Ring

- Original Message - From: Dan Journo To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 12, 2009 1:24 PM Subject: [asterisk-users] Incoming Call Ring Hello, I have Asterisk set up with 6 extensions. When a call comes in, I use a D

[asterisk-users] allowguest defaults to yes for SIP

In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general

Re: [asterisk-users] Incoming Call Ring

Depending on your phone, you can use CallWaitingRing to ring the phone anyway. I do this with Polycom 501's. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, November 12, 2009 6:24 AM To: Asterisk U

Re: [asterisk-users] RTPAUDIOQOS

OK, how do you get such information -- at times it would be very useful to know. Darryl Dunkin wrote: > Sorry to reply so late, I am months behind and catching up. > > > > I have been inspecting this on my own systems, and the results are > inconsistent to say the least. I’ve been dumping t

[asterisk-users] Codec interface

Hi All, I need to interface a codec-type device to Asterisk. The device uses a TI TLV320AIC1110 codec in 15 bit linear data mode with a 2.048 MHz clock supplied by the device. I am about to start on a custom hardware design to interface this device to the computer, but thought I'd ask here

[asterisk-users] Dell Poweredge T105

Hello, I someone successfully using Asterisk and Debian on an Opteron-enabled Dell Poweredge T105 ? If positive, which architecture (i386, amd, ...) w ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] Dell Poweredge T105

2009/11/12 Olivier > Hello, > > I someone successfully using Asterisk and Debian on an Opteron-enabled Dell > Poweredge T105 ? > If positive, which architecture (i386, amd, ...) w > If positive, which architecture (i386, amd, ...) was chosen ? Regards

Re: [asterisk-users] Termination Question

for the sake of bandwidth you are supposed to connect each two servers together.. otherwise calls between B && C will have to go through A . -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 From: i...@saudihome.com To: ast

[asterisk-users] my kernel is dazed and confused

Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason a0 on CPU 0. Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely on the PCI bus. Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue Would my Digium TDM410P cause an NMI, or is

Re: [asterisk-users] state_interface backport issue

Any takers? Still trying to get this resolved... Thanks! Robert Broyles wrote: > It's my understanding that the backport is available now in 1.4. > However, seem to be having some issues with it. Just wondering if I > have everything setup right. > > I'm running 1.4.26.2 realtime. > queue_membe

Re: [asterisk-users] allowguest defaults to yes for SIP

Lee Howard wrote: > Does anyone else agree with me that this is a poor default? I'd like to > see the default setting changed. > > I've always considered it to be good practice that something that may leave your system vulnerable, should be disabled by default. So yes, I would agree. Doug

Re: [asterisk-users] my kernel is dazed and confused

Dr. Michael J. Chudobiak wrote: > Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason > a0 on CPU 0. > Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely > on the PCI bus. > Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue > >

Re: [asterisk-users] allowguest defaults to yes for SIP

Lee Howard a écrit : > In your sip.conf file allowguest defaults to yes. This means that > anyone that can reach the SIP ports on that system has access to make > unauthenticated calls, by default. The administrator actually has to go > in and turn it off to prevent unauthenticated SIP calls (

Re: [asterisk-users] allowguest defaults to yes for SIP

Just my .02 - the guest context should torture or hangup instead of being empty. That might encourage a masochistic hacker though... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Th

Re: [asterisk-users] my kernel is dazed and confused

On 11/12/2009 09:42 AM, Francesco Peeters wrote: > Dr. Michael J. Chudobiak wrote: >> Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason >> a0 on CPU 0. >> Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely >> on the PCI bus. >> Nov 12 08:54:27 steerpike

Re: [asterisk-users] allowguest defaults to yes for SIP

Am I correct in saying that the without allowguest=no anyone can connect and make calls through the default context? If allowguest is set to no, how can I ensure that incoming calls can still be received from our DDI supplier? Many Thanks Dan -Original Message- From: asterisk-users-bou

[asterisk-users] "POTS 4K linear codec"

I am not sure what the problems are and the reasons for the basic 64K modems used in VOIP are. I understand the compressed codecs that get the bandwidth down to 20-30 K. And perhaps the 64K units give much better potential audio than you would get on a normal POTS line. But, as phone circuits VO

Re: [asterisk-users] BLF with SPA941?

Although I've never tested such feature on those devices, I know that it was only enabled in a recent firmware (6.1.3a/6.1.5a ?). Are you running it ? -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-use

Re: [asterisk-users] allowguest defaults to yes for SIP

Without the allowguest=no, Asterisk doesn't put up any defense against an unauthorized guest. You still have NAT/Firewall/IPTABLE "defenses", for what they are worth. The trick is to get what you need without allowing what you don't want. -Original Message- From: asterisk-users-boun...@l

Re: [asterisk-users] Termination Question

...and with a packet switched transport layer, the 'hairpin' route through A may create problematic levels of latency--latency that would perhaps NOT have been problematic on a classic circuit switched route, so it's definitely advisable to nail up a connection between b and c. -K - Origi

Re: [asterisk-users] allowguest defaults to yes for SIP

On Thursday 12 November 2009 08:59:16 Danny Nicholas wrote: > Without the allowguest=no, Asterisk doesn't put up any defense against an > unauthorized guest. You still have NAT/Firewall/IPTABLE "defenses", for > what they are worth. The trick is to get what you need without allowing > what you do

Re: [asterisk-users] allowguest defaults to yes for SIP

On Thursday 12 November 2009 09:00:45 Dan Journo wrote: > Am I correct in saying that the without allowguest=no anyone can connect > and make calls through the default context? > > If allowguest is set to no, how can I ensure that incoming calls can still > be received from our DDI supplier? You'r

Re: [asterisk-users] allowguest defaults to yes for SIP

On Thursday 12 November 2009 07:47:34 Lee Howard wrote: > In your sip.conf file allowguest defaults to yes. This means that > anyone that can reach the SIP ports on that system has access to make > unauthenticated calls, by default. The administrator actually has to go > in and turn it off to pre

Re: [asterisk-users] "POTS 4K linear codec"

On Thu, 2009-11-12 at 08:53 -0600, Cary Fitch wrote: > Digital 64K telco sounds very good as a phone conversation. Digital 64k audio coming across a T1 is essentially identical to the ulaw codec in VoIP. Digital 64k audio coming across an E1 is essentially identical to the alaw codec. -- Jared

[asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

Hello, I tried to install Asterisk + Asterisk addons + FreePBX (latest versions of all), but in the FreePBX screen, I don't have the option to set ring groups and IVRs . Can anyone tell me what I'm doing wrong? Thanks, Andreas ___ -- Bandwidth and C

[asterisk-users] solution for NAT issues?

Hi All, I been having issues on my users behind NAT, even if i hard set a specific port on the phone, there are some network that NAT's it out to a different port, in turn, some time later the phone could not be reached by the server. i think because on the server, e.g. the user is still regi

Re: [asterisk-users] allowguest defaults to yes for SIP

Tilghman Lesher wrote: > On Thursday 12 November 2009 07:47:34 Lee Howard wrote: > >> In your sip.conf file allowguest defaults to yes. This means that >> anyone that can reach the SIP ports on that system has access to make >> unauthenticated calls, by default. The administrator actually has

Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

On 12 Nov 2009, at 15:38, Cyprus VoIP wrote: > I tried to install Asterisk + Asterisk addons + FreePBX (latest > versions > of all), but in the FreePBX screen, I don't have the option to set > ring > groups and IVRs > > Can anyone tell me what I'm doing wrong? You are not posting on the FreeP

Re: [asterisk-users] Libpri-1.4.10.2 Released

Karl Fife wrote: > > Perhaps there's an arcane way to query lipbri the older releases from the > CLI? Can anyone speak to that? > Quick and dirty: strings /usr/lib/libpri.so That's CLI, tho' not the one you're talking about. Barry ___ -- Bandwi

Re: [asterisk-users] SendText

OK. Thanks On Thu, Nov 12, 2009 at 4:33 AM, Tarek Sawah wrote: > i have my own SMS provider as we sell SMS .. so i have setup my call center > with SMS sending for several services and alerts like a Missed Call when i'm > not registered it will send me an sms to alert me. > it's pretty the sam

Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

>> I tried to install Asterisk + Asterisk addons + FreePBX (latest >> versions >> of all), but in the FreePBX screen, I don't have the option to set >> ring >> groups and IVRs >> >> Can anyone tell me what I'm doing wrong? > > You are not posting on the FreePBX forums? ;) > I figured "Asteris

Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

On 12 Nov 2009, at 16:29, Cyprus VoIP wrote: > The problem is that the online module update is not working for me > (Cannot connect to online repository (mirror.freepbx.org). Online > modules are not available.) and I couldn't find online a working > solution :-( DNS/Gateway ok on server? S ___

Re: [asterisk-users] Can't connect to voip provider over NAT

On Thu, 2009-11-12 at 14:50 +1100, Michael Wyres wrote: > Have you tried "nat=yes" in the definition in sip.conf? > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy > Sent: Thursday, 12 Nove

Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

Dear Steve, Do you have your DNS settings ok? Otherwise include these settings(DNS1 DNS2) in your network configuration. Regards, Nelson Granados -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP Se

Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

>> The problem is that the online module update is not working for me >> (Cannot connect to online repository (mirror.freepbx.org). Online >> modules are not available.) and I couldn't find online a working >> solution :-( > > DNS/Gateway ok on server? > Yes. The problem is with the FreePBX modul

Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

On 12 Nov 2009, at 16:54, Nelson Granados wrote: > Dear Steve, > > Do you have your DNS settings ok? Yes, but its not me with the problem. ;) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

[asterisk-users] How to send DTMF on Zaptel with 50ms tone duration and 50ms gap between the digits?

Hi, After some testing I've found out that my client's hardware recognizes DTMF only if digits are sent 50ms apart with 50ms of tone duration. This was tested using a test device which generates DTMF. Now asterisk doesn't do it by default because digits going out from Asterisk are not being recog

Re: [asterisk-users] allowguest defaults to yes for SIP

On Thursday 12 November 2009 09:53:17 Lee Howard wrote: > Tilghman Lesher wrote: > > On Thursday 12 November 2009 07:47:34 Lee Howard wrote: > >> In your sip.conf file allowguest defaults to yes. This means that > >> anyone that can reach the SIP ports on that system has access to make > >> unauth

Re: [asterisk-users] allowguest defaults to yes for SIP

Tilghman Lesher wrote: On Thursday 12 November 2009 09:53:17 Lee Howard wrote: These people should need to deliberately use allowguest=yes. I would venture to guess that these people already know who they are and deliberately have this set. I would venture to guess that there are far, far

Re: [asterisk-users] allowguest defaults to yes for SIP

On Thursday 12 November 2009 12:08:39 Lee Howard wrote: > Tilghman Lesher wrote: > > On Thursday 12 November 2009 09:53:17 Lee Howard wrote: > >> And yet this point is not even made clear in the doc/security.txt file. > >> It says to not use "default" for anything you don't want to get abused, > >>

Re: [asterisk-users] allowguest defaults to yes for SIP

Tilghman Lesher wrote: > The issue in question was suspended, while the reporter makes the case on the > Asterisk-dev mailing list, which is not this list. The opinions there > amongst > contributors (meritocracy, not democracy) are that keeping the sample > configuration as it is now is probabl

Re: [asterisk-users] Asterisk 1.6.1.9 with FreePBX 2.5.2.1

On 12 Nov 2009, at 17:09, Cyprus VoIP wrote >> DNS/Gateway ok on server? > Yes. The problem is with the FreePBX modules. I forced the mirror file > to include version 2.5, and I get a list, but when I try to install > the > modules, it says that the modules need FreePBX version 2.5.0alpha or >

Re: [asterisk-users] SIP source address error

> It's set to bind to 0.0.0.0, which IIRC is nothing strange. > > The question remains: how can a remote Asterisk server be receiving > SIP packets that still contain the private net IP address of a client? It sounds to me as if the client hasn't been told to use its gateway's public IP address

Re: [asterisk-users] allowguest defaults to yes for SIP

"Gentlemens clubs" usually don't have any. While LH probably has a valid point, jumping on Til isn't the way to bring it home. You can't protect the stupid or lazy from themselves. If you can't do this right, pay someone else to. -Original Message- From: asterisk-users-boun...@lists.dig

Re: [asterisk-users] "POTS 4K linear codec"

On Thu, 12 Nov 2009, Cary Fitch wrote: > I am not sure what the problems are and the reasons for the basic 64K modems > used in VOIP are. I understand the compressed codecs that get the bandwidth > down to 20-30 K. And perhaps the 64K units give much better potential audio > than you would get

Re: [asterisk-users] allowguest defaults to yes for SIP

Danny Nicholas wrote: > "Gentlemens clubs" usually don't have any. While LH probably has a valid > point, jumping on Til isn't the way to bring it home. You can't protect the > stupid or lazy from themselves. If you can't do this right, pay someone > else to. You're suggesting that if I pay som

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

On Thu, Nov 12, 2009 at 01:43:48PM +0100, Olivier wrote: > 2009/11/11 Tzafrir Cohen > > > On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote: > > > What about adding per-span section headers like Asterisk .conf files ? > > > [span1] > > > group_lines 1 > > > pri_termtype > > >

Re: [asterisk-users] allowguest defaults to yes for SIP

Eh... if VoIP fraud weren't so rampant, and I didn't constantly see mailings to the Asterisk list about "How do I secure my system from the people who've been costing me tons of money lately," I would say that having a lax stance on security in exchange for additional usability might be a good thin

Re: [asterisk-users] allowguest defaults to yes for SIP

I did not mean to state or imply that you are lazy or stupid; It's just that some folks expect to spend 10 minutes reading a PDF, set up Asterisk and all is well - That's not what Open Source is about. If you want limited or no risk, you have to pay the piper. I'll bet there are thousands of pie

Re: [asterisk-users] my kernel is dazed and confused

On Thu, Nov 12, 2009 at 09:31:11AM -0500, Dr. Michael J. Chudobiak wrote: > Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason > a0 on CPU 0. > Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely > on the PCI bus. > Nov 12 08:54:27 steerpike kernel: Daze

Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface

At 02:25 AM 11/12/2009, you wrote: >FWIW, I've had a few recommendations for the Linksys SPA3000. However, >I haven't tried this for myself yet since I'm still in the planning >stage of replacing my current Asterisk machine. In my case, I >currently have a full-size tower and I'm planning to move t

Re: [asterisk-users] Codec interface

On Thu, Nov 12, 2009 at 09:22:41AM -0500, Bill Shaw wrote: > Hi All, > > I need to interface a codec-type device to Asterisk. The device uses a > TI TLV320AIC1110 codec in 15 bit linear data mode with a 2.048 MHz clock > supplied by the device. I am about to start on a custom hardware design

Re: [asterisk-users] allowguest defaults to yes for SIP

On 11:16, Thu 12 Nov 09, Lee Howard wrote: > Danny Nicholas wrote: > > "Gentlemens clubs" usually don't have any. While LH probably has a valid > > point, jumping on Til isn't the way to bring it home. You can't protect the > > stupid or lazy from themselves. If you can't do this right, pay some

Re: [asterisk-users] allowguest defaults to yes for SIP

Michiel van Baak wrote: > When I started working with asterisk, and found my first issue, I > created a patch, put it on the tracker, followed up on the comments, and > stuff got in. I'm sincerely pleased to know that you've had a different experience than have I. > If you read the page about co

Re: [asterisk-users] Trouble registering Cisco 7942

On Wed, Nov 11, 2009 at 9:34 PM, Warren Selby wrote: > The 7960 and 79x2 use different sip firmwares and as far a I have seen > the 7960 does not have the same port issue the 7941/2 seems to have > (which technically is not a problem, just an implementation of the sip > protocol that you don't typ

[asterisk-users] Request for Review: Building Queues with Asterisk

I have been working on some documentation for how to build queues for Asterisk. This is an introduction for getting device state working for queues, and building queues. It contains the documentation file (text format) and also has the .tar.gz file of the /etc/asterisk/ directory I was using for

Re: [asterisk-users] Request for Review: Building Queues with Asterisk

Leif Madsen wrote: > Please review and let me know how it goes for you! Where is it? Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.

Re: [asterisk-users] Trouble registering Cisco 7942

Just checked with my actual config file, and it's not a sanitation mistake, that's how I've actually got mine setup. Like I said earlier, I've never even messed with that section of my config before...I set mine up based on a combination of configs I've found around the net (I think you've already

Re: [asterisk-users] allowguest defaults to yes for SIP

On 13/11/09 3:59 AM, Danny Nicholas wrote: > Without the allowguest=no, Asterisk doesn't put up any defense against an > unauthorized guest. You still have NAT/Firewall/IPTABLE "defenses", for > what they are worth. The trick is to get what you need without allowing > what you don't want. A slig

Re: [asterisk-users] Request for Review: Building Queues with Asterisk

On 17:19, Thu 12 Nov 09, Leif Madsen wrote: > I have been working on some documentation for how to build queues for > Asterisk. > This is an introduction for getting device state working for queues, and > building queues. It contains the documentation file (text format) and also > has > the .t

Re: [asterisk-users] allowguest defaults to yes for SIP

On 13/11/09 8:30 AM, SIP wrote: > Eh... if VoIP fraud weren't so rampant, and I didn't constantly see > mailings to the Asterisk list about "How do I secure my system from the > people who've been costing me tons of money lately," I would say that > having a lax stance on security in exchange for a

Re: [asterisk-users] allowguest defaults to yes for SIP

On 13/11/09 9:37 AM, Lee Howard wrote: > Michiel van Baak wrote: >> When I started working with asterisk, and found my first issue, I >> created a patch, put it on the tracker, followed up on the comments, and >> stuff got in. > > I'm sincerely pleased to know that you've had a different experience

Re: [asterisk-users] allowguest defaults to yes for SIP

On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote: > Maybe the best way would be to make it that the default context only > provides the info from the examples unless you provide an option: > > read_security_document=yes Asterisk used to require that you set have 'TELEPHONY=yes' in /

Re: [asterisk-users] Can't connect to voip provider over NAT

> Have you tried "nat=yes" in the > definition in sip.conf? Yes, I have that definition in sip.conf. Now, I'm getting the following error -- SIP/voipprovider-094132d8 is making progress passing it to SIP/102-09423d58 -- Got SIP response 603 "Declined" back from 208.xx.xx.xx -- SIP/voip

[asterisk-users] Home line noise problem

I Have a home line connected to a tdm400p with 3 extensions and a siemens sip-dect , it seems to work fine but during a call there is always a digital squeal every so often does anyone know what this could be? Robb ___ -- Bandwidth and Colocation Provide

Re: [asterisk-users] allowguest defaults to yes for SIP

On 13/11/09 12:33 PM, Tzafrir Cohen wrote: > On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote: > >> Maybe the best way would be to make it that the default context only >> provides the info from the examples unless you provide an option: >> >> read_security_document=yes > > Asterisk use

[asterisk-users] TDM400p , asteriskNow and may other woes.....

Hello all, I am new to asterisk and have spent a good 4 or 5 days trying to get things sorted out. I initially installed it in Fedora Core 11 and compiled mods + asterisk. After much problems, I went with asteriskNow. The biggest problem I am have is getting some kind of base configuration going.

Re: [asterisk-users] Termination Question

So how can I let A makes a PEER connection between B & C, and ONLY log the call information? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Thursday, November 12, 2009 6:10 PM To: Asterisk Users Mailing Lis

Re: [asterisk-users] Need Adapter/Gateway with PSTN-interface

Grandstream HT503. For me works just fine. 1xFXO 1xFXS port. Each port has its own sip account. Martin - Original Message - From: jonas kellens To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, November 12, 2009 5:38 AM Subject: Re: [asterisk-users] N

Re: [asterisk-users] allowguest defaults to yes for SIP

>-Original Message- >From: asterisk-users-boun...@lists.digium.com >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard >Sent: Friday, 13 November 2009 06:16 >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] allowguest defaul

Re: [asterisk-users] Termination Question

I have no first-hand experience with the fussy idiosyncrasies, but the BIG PICTURE is to have server A set up the call, and then "reinvite" the media directly from B to C. The call control messages flow to server A, the media goes directly. If you don't have "NAT traversal Kung-Fu", I suggest

Re: [asterisk-users] Termination Question

That could work, but I have no control over server B, not server C ! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Friday, November 13, 2009 3:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Re: [asterisk-users] RTPAUDIOQOS

I add this line in our in/out contexts: exten => h,1,Noop(QOS=${RTPAUDIOQOS}) Then grep for 'QOS' in asterisk-verbose (assuming you have verbose logging on). I'm sure you could output it anwhere else as well with a system call/echo. -Original Message- From: asterisk-users-boun...@lists.d

Re: [asterisk-users] Request for Review: Building Queues with Asterisk

Barry L. Kline wrote: > Leif Madsen wrote: > >> Please review and let me know how it goes for you! > > Where is it? Ah yes, in my eagerness to get ready for dinner with the g/fs parents, I have forgotten to post where this exists :) I posted it to the issue tracker here: https://issues.asteri

Re: [asterisk-users] Need opinion about GSM codec for Internet

If you doesn't need transcoding, you doesn't need any licenses... Martin - Original Message - From: "Vinícius Fontes" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, November 06, 2009 11:43 AM Subject: Re: [asterisk-users] Need opinion about GSM codec for In

Re: [asterisk-users] RTPAUDIOQOS

OK, thanks -- will have to try and see what I get. Darryl Dunkin wrote: > I add this line in our in/out contexts: > exten => h,1,Noop(QOS=${RTPAUDIOQOS}) > > Then grep for 'QOS' in asterisk-verbose (assuming you have verbose logging > on). I'm sure you could output it anwhere else as well with

[asterisk-users] Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?

Digium has discontinued their ATA iaxy adapter; don't blame them, too expensive so they can not compete. The adapter is upgraded automaticaly when it is connected to new asterisk version; since this adapter is discontinued will it still work with asterisk 1.6 and beyond or will it be\ just a "

Re: [asterisk-users] solution for NAT issues?

i have also tried setting qualify='yes' but cpu usage spiked to 100%. Ron wrote: > Hi All, > > > I been having issues on my users behind NAT, even if i hard set a > specific port on the phone, there are some network that NAT's it out to > a different port, in turn, some time later the phone co

Re: [asterisk-users] Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?

On Thursday 12 November 2009 21:18:18 Joseph wrote: > Digium has discontinued their ATA iaxy adapter; don't blame them, too > expensive so they can not compete. > > The adapter is upgraded automaticaly when it is connected to new asterisk > version; since this adapter is discontinued will it still

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

2009/11/12 Tzafrir Cohen > On Thu, Nov 12, 2009 at 01:43:48PM +0100, Olivier wrote: > > 2009/11/11 Tzafrir Cohen > > > > > On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote: > > > > > What about adding per-span section headers like Asterisk .conf files > ? > > > > [span1] > > > > group_lin

Re: [asterisk-users] allowguest defaults to yes for SIP

On 12:38, Fri 13 Nov 09, Matt Riddell wrote: > On 13/11/09 12:33 PM, Tzafrir Cohen wrote: > > On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote: > > > >> Maybe the best way would be to make it that the default context only > >> provides the info from the examples unless you provide an op

[asterisk-users] Multimedia PBX Solution

We are planning to develop a Multimedia PABX to connect about 500 or more personnel for Voice, Video and Text Communication. www.*gvsc*net.net is a similar solution but we wish to have our own independent solution. Please advise if anyone can offer a ready to go end to end Asterisk based solution.

Re: [asterisk-users] Multimedia PBX Solution

Nazir Ahmed Vaid wrote: > We are planning to develop a Multimedia PABX to connect about 500 or > more personnel for Voice, Video and Text > Communication. www.*gvsc*net.net is a similar solution > but we wish to have our own independent solution. Please advise if > anyone can

Re: [asterisk-users] Health IVR Recordings

Nazir Ahmed Vaid wrote: > We are looking for Pre-Recorded IVRs for Health Services in English and > other languages. If anyone is aware of a source kindly advise. We are > launching a TRIAGE SERVICE and we need these Recorded IVRs for this purpose. What makes you think that generic recordings o

[asterisk-users] Health IVR Recordings

We are looking for Pre-Recorded IVRs for Health Services in English and other languages. If anyone is aware of a source kindly advise. We are launching a TRIAGE SERVICE and we need these Recorded IVRs for this purpose. -- السلام عليكم ورحمة الله وبركاته Nazir Ahmed Vaid Cell:+92300-828 eH

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