Re: [asterisk-users] names of SIP aware firewalls

Sonicwall, but I have no idea if it really works. Jerry Jones wrote: Intertex Not cheap, licensed per number of users But seem to work great and have some nifty tools very confusing picking models though On Nov 5, 2006, at 3:54 PM, Erick Perez wrote: Besides ranch networks and borderware,

Re: [asterisk-users] FXO Cards vs. Channel bank with T1

You'll find the cost of a PRI varies dramatically from one telco to another. I've heard numbers in one case where three analog pstn lines cost the same as a PRI, another case where 16 analog pstn lines cost the same as a PRI. And, having worked in the telecomm industry for many years, there are

Re: [asterisk-users] Stopping putgoing calls after working hours

I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920

Re: [asterisk-users] SPA942 quality for a Bank

Tom wrote: At 02:30 PM 10/15/2006, you wrote: Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. We have been using Cisco

[asterisk-users] SPA942 quality for a Bank

Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. Expecting to deploy these using existing cat5 cabling and both rj45 jack

Re: [asterisk-users] DID failover

Todd- Asterisk wrote: I'm setting up an asterisk server where an administrator will not always be available in case of problems. While I expect problems to be rare, I need to be prepared. We're thinking of VoIP DID's and SIP phones so it's an all TCP/IP network. We could get a second server

Re: [asterisk-users] VOIP with PSTN backup

Brian Candler wrote: I'm looking for a way to set up a VOIP network in branch offices where one or more phones have "lifeline" capability, i.e. can place calls if the IP network or VOIP service dies, or even if power goes down. (I'm thinking of business continuity here, not just emergency service

[asterisk-users] FYI - Polycom SoundPoint IP 301 Denial of Service]

FYI. TITLE: Polycom SoundPoint IP 301 Denial of Service SECUNIA ADVISORY ID: SA22266 VERIFY ADVISORY: http://secunia.com/advisories/22266/ CRITICAL: Less critical IMPACT: DoS WHERE: From local network OPERATING SYSTEM: Polycom SoundPoint IP 301 http://secunia.com/product/12229/ DESCR

Re: [asterisk-users] No Dialtone

Jay R. Ashworth wrote: On Sat, Oct 07, 2006 at 10:31:16AM -0500, Rich Adamson wrote: If you've messed up in connecting telephone lines to the wrong module, the ringing voltage sent to a fxs module will destroy it. You would need to replace the module. I'm going to stick my neck out

Re: [asterisk-users] No Dialtone

If you've messed up in connecting telephone lines to the wrong module, the ringing voltage sent to a fxs module will destroy it. You would need to replace the module. Eddie Johnson Jr wrote: Yes, I have and I received the following: In zapata.conf your first two channels should be fxs_ks be

Re: [asterisk-users] Outbound FXO call, getting "You must first dial..."

Nick Ellson wrote: I am not sure what I might be set up wrong, but dialing out with my Zap/1 port seems to alwyas get the "You must first dial a 1 when calling this number" message from what sounds like the actual PSTN. My zapatel.conf and extensions.conf bits below. Any advice? (I do receive

Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute

Noah Miller wrote: You can also change some settings in the zapta and zaptel config.. to reduce echo and interference on the line.. This is the most important thing here - what does your zapata.conf look like? zapta.comf switchtype=national This is not necessary in your case. It pertains

Re: [asterisk-users] CALEA support within asterisk?

Matthew Thompson wrote: On 3 Oct 2006, at 19:53, Colin Anderson wrote: I, for one, welcome our new Republican overlords. lol you are just full of pop culture references, aren't you? Abortions for some, miniature American flags for others. Seriously though - is anyone aware of a precis of C

Re: [asterisk-users] CALEA support within asterisk?

Inline... On Tue, Oct 03, 2006 at 12:13:27PM -0500, Rich Adamson wrote: Does anyone know if asterisk currently supports the US government's Communications Assistance for Law Enforcement Act (CALEA) regulations? If not, does anyone have this item on their To-Do list? Why in

[asterisk-users] CALEA support within asterisk?

Does anyone know if asterisk currently supports the US government's Communications Assistance for Law Enforcement Act (CALEA) regulations? If not, does anyone have this item on their To-Do list? For those that are not familiar with CALEA, it's the governement's way of intercepting or "monitori

Re: [asterisk-users] Right way to prevent analog channel from answering the phone?

Nick Ellson wrote: I am in the process of learning my A1200P, and i would like an elegant way to prevent it from answering the phone, but still make outbound calls. I tried zap destroy channel 1 (which worked, but pissed off Asterisk ;) Is there a more elegant way to tell it to answer/not a

Re: [asterisk-users] T1 timing errors (Frame Slips) on Nortel 61C to TE110P

Ronnie Jones wrote: I am setting up an asterisk box , my first with PRI T1 interface to a Nortel 61C. We have quite a bit of experience with the 61C and do most of the programming including maintaining several other PRI interfaces in this switch. The problem we are having is as soon as we tur

Re: [asterisk-users] Asterisk behind Sonicwall firewall

Barry Fawthrop wrote: Hi all I didn't change anything that's my point It has be running and working just fine then at 4:32 pm yesterday I could not make or recieve VoIP calls via our VoIP Provider They say the Invite packet was being rejected and thus there was no "real" connection even thoug

Re: [asterisk-users] Asterisk behind Sonicwall firewall

Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Yes, have multiple clients with asterisk behind a sonicwall. I don't understand from your wording

Re: [asterisk-users] Line Pickup Problem

Pato Valarezo wrote: Lacy Moore - Aspendora wrote: Wherever you have your exten => s,1,Answer statement, replace with: exten => s,1,Wait(30) ; or however long you want to wait to give someone else the chance to answer exten => s,n,Answer then continue on. Asterisk will then wait 30 secon

Re: [asterisk-users] How can I stop lost DNS from killing Asterisk?

Eric Bishop wrote: Hi All, When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? We just went through the s

Re: [asterisk-users] Re: Setting QOS settings in asterisk and/or CentOS?

Steven wrote: I found this command if your Cisco switches support it: "auto qos voip trust" You set this on each interface. It automatically prioritizes all SIP and skinny traffic, but not iax. There is also "auto qos voip cisco-phone". This one can detect a Cisco phone and prioritize it. I ju

Re: [asterisk-users] How can I stop lost DNS from killing Asterisk?

Eric "ManxPower" Wieling wrote: Use IP addresses instead of hostnames in your Asterisk config. It sucks, but that is the only way I know of. Eric Bishop wrote: When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We a

Re: [asterisk-users] TDM2400P vs Sangoma A200

I'm putting together a plan for a new Asterisk system and I'm trying to decided on an interface card to use. I was originally planning on using a Sangoma A200 but now I'm considering a Digium TDM2400P. The server is large enough to accommodate the full sized TDM and I'll be using 8 FXO channe

Re: [asterisk-users] voicemail greeting

unplug wrote: Hi, When I use Voicemail function, there is a default system greeting before voicemail recording. Is it possible to change that greeting? How? Call into voicemail as though you were going to listen to your messages, and press "0" for Mailbox Options. Then press "3" to record you

Re: [asterisk-users] Cisco 7970 - DTMF

Tomislav Parčina wrote: In sip.conf for one friend (Cisco 7970 phone) I have define this dtmfmode=inband And in xml.conf of that phone I have none 101 3 none But DTMF doesn't work for that phone. Phone establishes call using g711 alaw codec. How should I configure phone and sip.conf to make

Re: [asterisk-users] Re: Very high ping times from 7960 phones

Tomislav Parčina wrote: In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times > 150ms. Every single on

Re: [asterisk-users] Setting QOS settings in asterisk and/or CentOS?

Nick Hoffman wrote: On Sat September 23 2006 06:14, Bob Amen <[EMAIL PROTECTED]> wrote: which sets the TOS bit on all IAX, SIP and RTP packets. Using iptables means that we can set up our rules on the router without using ACLs. Our Cisco Cookbook (http://www.oreilly.com/catalog/ciscockbk/) has

Re: [asterisk-users] Setting QOS settings in asterisk and/or CentOS?

BerkHolz, Steven wrote: How would I go about setting the TOS bit to "RTP IP TOS Byte: 18 (hex)" for SIP and IAX traffic at the asterisk server? Also, Do you have a quick reference on how to configure a Cisco switch to prioritize SIP traffic? I check in various Cisco docs, and there are so man

Re: [asterisk-users] SPA941 -> Asterisk -> Voip provider -> PSTN -> ShoreTel garble

Cliff Brake wrote: I am using the following setup: Linksys SPA941 -> Asterisk -> NuFone -> PSTN -> ShoreTel system The system works great for the most part. Most people I call say it sounds good. However, every time I call a certain company that uses a ShoreTel system, they claim the sound is

Re: [asterisk-users] Asterisk Design Question

Remi Quezada wrote: Hi, Right now I am in the process of setting up an asterisk box. I was thinking of having two asterisk box, one that is hooked up to the PSTN using a digium TE405P card and the other asterisk box will be used to store all the sip user features and routing information. Do yo

Re: [asterisk-users] Cisco Distinctive ring using alert-info

lt in ring tones are not very impressive, and as I recall, are basically limited to sounds such as one-long, one-long & one short, etc. Lacy Moore - Aspendora wrote: Do some 7960s perform differently? On 9/15/06, *Eric ManxPower Wieling* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTE

Re: [asterisk-users] Why not g726-32?

RR wrote: On 9/16/06, Rich Adamson <[EMAIL PROTECTED]> wrote: RR wrote: > All, > > is there anyone who uses g726-32 ? If not, then does anyone know why > don't people use it? I use g726 on iax links between systems and to teliax.com for LD calls. Have no idea if its -

Re: [asterisk-users] Reliability of the newer IAXy's

The sipura stuff (and lots of other ata's) work just fine behind most nat boxes "if" the asterisk box is on a registered IP. John Novack wrote: Regarding the IAXy, newer model- S101i I have an application for one. Both the IAXy and the Asterisk would be behind routers ( cheap Linksys ones )

Re: [asterisk-users] Why not g726-32?

RR wrote: All, is there anyone who uses g726-32 ? If not, then does anyone know why don't people use it? I use g726 on iax links between systems and to teliax.com for LD calls. Have no idea if its -32 or what though. What ships with asterisk (in terms of g726) has been working very well for

Re: [asterisk-users] Cisco Distinctive ring using alert-info

Eric "ManxPower" Wieling wrote: Rich Adamson wrote: Julian Lyndon-Smith wrote: I've got a cisco 7960, with (amongst many others) the following in the RINGLIST.DAT file Foghorn foghorn.raw I can manually select this for the ringtone. However, I was wanting to use a

Re: [asterisk-users] Cisco Distinctive ring using alert-info

Julian Lyndon-Smith wrote: I've got a cisco 7960, with (amongst many others) the following in the RINGLIST.DAT file Foghorn foghorn.raw I can manually select this for the ringtone. However, I was wanting to use a normal ringtone, with foghorn being used if the call was coming in from

Re: [asterisk-users] 9 becomes 99 ? And other strangeness

Brian Candler wrote: On Thu, Sep 14, 2006 at 09:00:57AM -0500, Rich Adamson wrote: [outbound] exten => _9.,1,Dial(Zap/4/${EXTEN:1}) <<<< NOTE HERE exten => _9.,2,Congestion() exten => _9.,102,Congestion() Try replacing the first step above with: exten => _

Re: [asterisk-users] 9 becomes 99 ? And other strangeness

Brian Candler wrote: I'm getting a strange situation with the first digit being doubled on outbound dialling, and other oddities. I think something strange is going on in my dialplan, rather than a DTMF decoding issue, but see what you think. The platform is CentOS 4.4 plus Asterisk SVN trunk as

Re: [asterisk-users] Switch Experiences

Ben Gore wrote: Hello: I'm would like to get feedback before finalizing design of a VOIP network, in particular about people's experience with network (primarily 10/100/1000 twisted pair) ethernet switches. I have a number of candidates in mind, but I would like any and all opinions and sug

Re: [asterisk-users] PRI: sometimes Asterisk drop calls

Steve Davies wrote: On 9/12/06, Rich Adamson <[EMAIL PROTECTED]> wrote: Steve Davies wrote: > For the curious, can anyone tell me how this flag fixes the issue? - I > have seen the error before, but always assumed it was related to hung > channels. > > Thanks, > Steve

Re: [asterisk-users] PRI: sometimes Asterisk drop calls

Steve Davies wrote: For the curious, can anyone tell me how this flag fixes the issue? - I have seen the error before, but always assumed it was related to hung channels. Thanks, Steve On 9/12/06, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote: Problema solved! Just put resetinterval=never ins

Re: [asterisk-users] question...

laymen terms. thanks. */Rich Adamson <[EMAIL PROTECTED]>/* wrote: Christopher Corn wrote: > i plan on buying 4 residential lines for our small office and i was > giving some thought. we'd like to have one main number that can transfer > calls to the other l

Re: [asterisk-users] question...

Christopher Corn wrote: i plan on buying 4 residential lines for our small office and i was giving some thought. we'd like to have one main number that can transfer calls to the other lines. but seeing that i have 4 different individual lines with different numbers, im not seeing hows thats pos

Re: [asterisk-users] Context

I have two contexts how could I isolate context A from context B ,in other words I want to ban context A from calling context B In sip.conf, define phones/extensions something like this: [1000] type=friend context=cust-a [1001] type=friend context=cust-a [2000] type=friend context=cust-b [

Re: [asterisk-users] using residential voip for business?

Michael Graves wrote: Not to answer a question with a questionbut why do so many businesses focus so intently on the cost of their voip service? If we presume that a business intends to stay in business, and that phone service is crucial to actually being in business, then I've never seen

Re: [asterisk-users] using residential voip for business?

Rushowr wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Christopher Corn wrote: thanks for the reply. why are residential lines cheaper than businesses? say for unlimited, it always costs more for residential. */Michael Graves <[EMAIL PROTECTED]>/* wrote: I'd just use a service that'

Re: [asterisk-users] using residential voip for business?

Christopher Corn wrote: I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use it for a business and it is over-utilized, it will be canceled. is this true for all residential voip plans? i have a small office of about 4 or 5 phones. i tend to chose

Re: [asterisk-users] Polycom related question

John Marvin wrote: Rich Adamson wrote: If you look at the sample configs, you'll find: [EMAIL PROTECTED],[EMAIL PROTECTED] ; Subscribe to status of multiple mailboxes in the sip.conf.samples for v1.2 stable. That is the only way that I know of to turn on the mwi for two diff

Re: [asterisk-users] Voip providers and sip origination and termination?

Yair Hakak wrote: actually Rich, not to be picky or anything, but your first paragraph is backwards. There are some providers that allow you to originate calls to the US/World pstn network via their facilities, but do not provide any way for the US/World to call you from the pstn network. (eg,

Re: [asterisk-users] Zaptel-1.2.9 compile error

Bill Maidment wrote: Rich Adamson wrote: Bill Maidment wrote: Rich Adamson wrote: Have you tried: cd /usr/src/zaptel make update make install I've never used the tarball, however if the tarball is installed and the resulting code can't be "compiled", obviously

Re: [asterisk-users] Voip providers and sip origination and termination?

Christopher Corn wrote: can someone please explain the differnces to me??? I have an asterisk system im setting up for a small office (4 or 5 phones) and as im looking for a voip provider, i find that voip providers generally have unlimited plans, and those that offer sip origination and ter

Re: [asterisk-users] Polycom related question

Kevin Smith wrote: Hi everyone, While this isn't a true "asterisk" question, I know a lot of people here use Polycom phones. Anyway, I have two Polycom 601 phones that share the same voicemail box. Now it is intermittent, but sometimes both phones will have a notification there is a voice mai

Re: [asterisk-users] Zaptel-1.2.9 compile error

Bill Maidment wrote: Rich Adamson wrote: Have you tried: cd /usr/src/zaptel make update make install I've never used the tarball, however if the tarball is installed and the resulting code can't be "compiled", obviously there is a Makefile present. Part of the Makef

Re: [asterisk-users] Zaptel-1.2.9 compile error

Samy Antoun wrote: --- Bill Maidment <[EMAIL PROTECTED]> wrote: Hi I've just tried to compile the zaptel-1.2.9 release and I get the following error: Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors when compiling zap: make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octas

Re: [asterisk-users] Re: Asterisk hangs up after 10-15 minutes whenSIPPhone is on mute

Several Linksys models have had a problem in the past allowing multiple devices on the inside lan to nat properly with something on the outside wan. Ordinarily a sip phone on the inside of the lan attempts to register with an external asterisk box, and the Linksys keeps track of source IP, sou

Re: [asterisk-users] using SIP to connect remote other VoIP server(Attn:Elpidio)

Crazy Boy wrote: Hi Elpidio, I am Chandra from India. I have a doubt. I am trying to solve my problem from many days. But, I couldn't able to solve this problem. I am using Asterisk with Redhat Enterprise Linux 4.2. The problem is 5060 port is blocked. After stop my firewall (service iptables

Re: [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

Dan Serban wrote: I have a system running Asterisk 1.2.10 (Debian packaging) and about 50 Linksys SPA-942 phones, after the initial config and mass deployment of the phones everything looks like it's configured well. When an incoming call is answered and then attempted to be "xfer'ed" via the so

Re: [asterisk-users] ATA being used as a SIP Trunk to connect LegacyPbx to Main Asterisk Server

Marco Mouta wrote: Hi all, Do you think it could be an affordable solution using a two fxs ATA device to connect an old legacy pbx (with few users) with a main asterisk server. phonesanalogueSmallOfficeLegacyPBxATA-2FXS-SIP--MainOffice AsteriskServer This way also I w

Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK

Elpidio Ramos wrote: This seems to be an easy-to-solve problem but it may be again my lask of knowledge in linux: My linux fedora core 3 asterisk box has a public IP and a private IP (two NIC) I got the ports open in fedora core 3 (5060 and 1 thru 3) for both interfaces. I was abl

Re: [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes

The svn branch-1.2 is very stable, probably more stable then the rpms and other distro's out there, as fixes are applied when problems are identified and corrected. Sometime later, the svn branch-1.2 is used to create packages. Kevin Smith wrote: Well personally I am just glad I wasn't the on

Re: [asterisk-users] UK (BT) Problem with TDM 400P

Mark Muffett wrote: I'm trying to get my TDM400P to work with a BT POT line. I've done everything I can think of to get the uk settings right (in zapata.conf, zaptel.conf and options for the wctdm driver) - and they all look right (ie uk like) and look like they are working when I try diagnostic

Re: [asterisk-users] Sipura SPA3000

like this is invoked when the Ethernet link is down or registration fails. I don't have a SPA3000 up at the moment to look at what's required. Bob... On Fri, 2006-09-01 at 11:45 -0500, Rich Adamson wrote: If "pstn call ring thru line 1" is enabled, all incoming pstn calls

Re: [asterisk-users] Sipura SPA3000

ED] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all

Re: [asterisk-users] Sipura SPA3000

Steve Kennedy wrote: On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones

Re: [asterisk-users] Sipura SPA3000

I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call b

Re: [asterisk-users] iax vs. sip?

We've been using iax with teliax.com for a couple of years, and it seems the quality of calls varies with time. Sometimes it is good and next time its not so good. There has been changes occurring to iax and the jitterbuffer stuff over the last two years, and I'm reasonably certain that some po

Re: [asterisk-users] HP ProLiant and Digium 24xxp

Kevin P. Fleming wrote: Robert Roach wrote: I have a customer request to deploy an HP rack server (ProLiant DL series) as the base system for an Asterisk install. They also want to use the Digium 24xxp card. I have heard that the Digium card is oversized and does not fit in a normal size chass

Re: [asterisk-users] oddity with TDM400P / Asterisk setup

Ted Wallingford wrote: Hi List, I am working with an Asterisk server running on Fedora Core 4. It has two TDM400P cards installed. There are 6 trunk ports and 2 (unused) analog line ports. There are 5 Polycom SoundPoint 501 SIP phones connected to the server, and a Linksys 24-port powered sw

Re: RE : [asterisk-users] Problem with a TDM400P

Seems to me that someone posted something about unusual analog connections in the UK that required a jumper wire (or something like that) on the pstn analog connection to the fxo port jack. (I'm in the US, so don't have a clue what I'm taking about.) Might be worth doing a little more google se

Re: [asterisk-users] Can anyone recommend a large button sip phone for the elderley.

Chuck Bunn wrote: Hi, Can anyone recommend a large button/type sip phone (VOIP) that an older person could use. I have a client that needs to have large button phones for elderly residents in her facility. How about the old Grandstream BT100? Large buttons, requires a firm press (no nervous

Re: [asterisk-users] Missing number 2 in "advanced options" of VM

Doug Lytle wrote: Stefan-Michael. Guenther (in-put GbR) wrote: Why does Asterisk strip all digits except 4498 and why doesn't _X. match That I can't answer, I've never used the option. My VM works just fine by sending the callback through the same context as what your sip phones use. __

Re: [asterisk-users] lost packets when bridging zap and iax

Simone Cittadini wrote: We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called client server often the called can't hear the caller (both machines on

Re: [asterisk-users] Tracing audio problems

Avi Miller wrote: Avi Miller wrote: Does anyone have any suggestions on where to look next? My users are getting increasingly annoyed and I'm quickly running out of ideas. Replying to myself to note that this is now happening on outbound calls via ISDN, i.e. calls that don't use IAX2 or the i

Re: [asterisk-users] Re: GSM gateway and FXO ATA

Martin Joseph wrote: On 2006-08-22 01:59:09 -0700, Tomislav Parčina <[EMAIL PROTECTED]> said: Hi list! I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over Grandstream HT488 ATA. Personally I found the FXO port on the HT-488 to unworkable except as a backup for pow

Re: [asterisk-users] zap channel media volume

in on the fly based on the volume of the two channels. Probably not realistic though. Is there other hardware other than digium's that better deals with this issue? Rich Adamson wrote: The root cause of the low volume problem is the result of software echo cancellation software, and its ne

Re: [asterisk-users] DNS

Ola Lidholm wrote: On 25 aug 2006, at 20.18, Bill Gibbs wrote: Asterisk server is setup in /etc/resolv.conf to query my primary and backup NS. Had an issue with my primary NS and asterisk refused to complete any calls or forward inbound calls to extensions. I had to manually switch it to l

Re: [asterisk-users] zap channel media volume

The root cause of the low volume problem is the result of software echo cancellation software, and its need to insert a noticeable loss. If I recall correctly, the wctdm.c driver has a statically defined loss value of something like -6 db that is loaded into the TDM400 chipset at driver load ti

Re: [asterisk-users] DNS

Bill Gibbs wrote: Asterisk server is setup in /etc/resolv.conf to query my primary and backup NS. Had an issue with my primary NS and asterisk refused to complete any calls or forward inbound calls to extensions. I had to manually switch it to look at the backup NS first then reboot for it to

Re: [asterisk-users] Trunk with multiple IPs?

I don't believe that addresses the OP's original post since he was talking about limiting "incoming" calls from specific IP addresses. You might want to validate how secure your definitions are considering the type=friend approach. Lists @ EMS wrote: Hi, I've only just now seen this post. Th

Re: [asterisk-users] Trunk with multiple IPs?

Benjamin Lawetz wrote: Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Origin

Re: [asterisk-users] Modems dialing over sangoma a104d

Sean Cook wrote: I have a sangoma 104d that is our main pbx now( legacy system died ). I have replaced every phone in the building and things are going very well. We have fax working well and calls are routing properly... All is well... Except for our support modems... we have support people t

Re: [asterisk-users] Cisco Router QOS and IAX2

, I used your sample and tweaked it for my needs and everything started working fine. Bruce On 8/23/06, *Rich Adamson* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Bruce Reeves wrote: > I'm needing some pointers from anyone who has been able to get C

Re: [asterisk-users] using asterisk + sangoma a102 to simulate telco PRI: is possible?

Giorgio Incantalupo wrote: Hi, I have an asterisk box with a sangoma a102 (two PRI ports). Is is possible to connect port A to port B in order to use port B as a simulation of a telco PRI line? If yes, is there a special cable needed? How can I configure the card and zaptel.conf? Yes. You'll

Re: [asterisk-users] Trunk with multiple IPs?

ldn't work with deny and permit) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: August 23, 2006 10:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk with multiple IPs? Benjam

Re: [asterisk-users] Cisco Router QOS and IAX2

Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. You can either match on udp/4569, or, match on TOS header bits. I

[asterisk-users] VM - advanced options?

running v1.2.10 svn checkout... When I listen to the VM options, it says 'press 3 for advanced options', but after pressing '3', there is nothing there with the exception of pressing '*' to return to the main menu. Have I missed a config option, sound file, or is the advanced option not tota

Re: [asterisk-users] Calls over VPN

Joseph wrote: Is anybody making calls over VPN? If so what is the "penalty" as encryption is involved. I was planning to use VPN to register Sipura units to my local asterisk this way I don't have to deal with NAT issues. vpn's work just fine as long as the vpn end-points have enough horse

Re: [asterisk-users] Strange SIP response

Diego Andres Asenjo G. wrote: Hi, I am getting the following message on the CLI: -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I real

Re: [asterisk-users] LOUD MP3 Hold Music

David Freeman wrote: I have the opposite problem. I can hardly hear the hold music at all. On 8/22/06, *Dennis P. Clark* <[EMAIL PROTECTED] > wrote: How do you lower the volume of MP3 hold music? I'm certainly not an expert on MOH, but I don't believe there ar

Re: [asterisk-users] Re: Zaptel install - Fedora Core 5

Tzafrir Cohen wrote: On Tue, Aug 22, 2006 at 08:42:36AM -0500, Rich Adamson wrote: Tomislav Parčina wrote: In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... I did "yum update" last week and here is my current kernel: I had no problem at all with zaptel. I am only usin

Re: [asterisk-users] Re: Zaptel install - Fedora Core 5

Tomislav Parčina wrote: In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... I did "yum update" last week and here is my current kernel: I had no problem at all with zaptel. I am only using TDM400P though, in case that matters. Hi Anto! The thing is that I can't rely on yum update for

Re: [asterisk-users] Sipura 3000 dialplan strings.

Ken D'Ambrosio wrote: I'm trying to set up a dialplan that dials via PSTN for: All eight-digit calls that start with 9 All 911 calls All calls that start with 424 (the local exchange) I haven't tested 911 -- for obvious reasons. I may do so after I feel more confident. I've got the starts-wit

Re: [asterisk-users] No retry after DNS failure

John Marvin wrote: Today I had a brief power outage which caused the Asterisk server and DSL modem to reboot. The Asterisk server came up before the internet connection was working, so it failed when try to look up some of the hosts for my outbound voip providers in sip.conf. Asterisk never r

Re: [asterisk-users] Linksys SPA-941 Message Waiting Indicator

voiplist wrote: Greetings.. I have a few Linksys SPA-941 IP phones running the latest firmware 4.1.12(a). I tried turning on the Message Waiting indicator but it doesn't seem to work correctly for me. This phone is connecting to Asterisk 1.24 running Realtime. Not sure if it matters but rtca

Re: [asterisk-users] Linksys SPA-3102

Inline... Barry D. Hassler wrote: On Sat, 2006-08-19 at 00:12 -0500, Rich Adamson wrote: Barry D. Hassler wrote: > Any further experience with the 3102? I'm looking for a solution to > connect 2 CO lines and a set of 2-line phones to my asterisk server > (along with a bunch

Re: [asterisk-users] Linksys SPA-3102

Barry D. Hassler wrote: Any further experience with the 3102? I'm looking for a solution to connect 2 CO lines and a set of 2-line phones to my asterisk server (along with a bunch of SIP phones). Would 2 of these work well for that? Hopefully no echo problems! That would kill this project? I'm

[asterisk-users] Dial statement problem

Need a little assist by someone else's eyes; mine have gone blurry. Running v1.2.10 checked out from svn as of today. Problem: When dial statement is executed with a timeout value and no one answers the call, the next priority (#4) is not being executed as expected. When an incoming pstn call

Re: [asterisk-users] Digium TDM400P Vs Sangoma A200

Correction... it only plugs into "one" pci slot, but anything beyond four ports covers up additional pci slots (even though it doesn't plug into the pci connector). So, a 24 port a200 card would essentially render "all" pci slots unusable due to size. The tdm2400 card consumes a single pci slo

Re: [asterisk-users] Digium TDM400P Vs Sangoma A200

Jonathan Borden wrote: I was wondering which of these cards would be better for a 1-2 line SOHO. I would like room to grow as well as I am concerned with voice quality and life expectancy of the product. Any input into which one and why would be greatly appreciated. The sangoma a200d does a

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