[Asterisk-Users] Echo Madness

2005-05-07 Thread Sophus
Hi there, I'm experiencing an echo problem and dammed If I can sort it out. We're running Asterisk on Fedora Core 3 64bit, installed as per http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3. These are the specs of the Machine – 1 x AMD A64/3500+ CPU: Desktop Athlon64™ Retail w/fan SKT 1 x

[Asterisk-Users] Re: chan_zap.so: load_module fails: Fedora Core 3: SMP

2005-05-07 Thread Tony Mountifield
In article <[EMAIL PROTECTED]>, Vamsi Pottangi <[EMAIL PROTECTED]> wrote: > > I'm trying to install asterisk on Dell power edge 2800 running Fedora core 3. > I don't have have any zaptel cards, so trying to use ztdummy. > /dev/zap is successfuly created... but I see some problems while > starting

RE: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Gregory Wiktor - ADCom Corp.
I am a user of Teliax and voipjet. I find voipjet to be very reliable and good for outgoing, very low lag, etc. Teliax is good too, but I am finding high lag rates, to the point where there is a half-second delay. I ended up just ordering a pots line for incoming (since I am going to be doing ad

[Asterisk-Users] Good NAT Pnp Hardphone

2005-05-07 Thread Gregory Wiktor - ADCom Corp.
Hello All, I am looking for a sip phone that is capable of automatic nat. The Cisco ata186 for example works fine for natting with iconnecthere, but as for asterisk, both my 7960 and polycom ip600 require you to set the nat ip on the tftp. Does anyone know a good phone (or ata) that can do this a

[Asterisk-Users] MWI Suggestion

2005-05-07 Thread Gregory Wiktor - ADCom Corp.
I had been trying to get mwi working on a pingtel phone for some time, with no success... The solution was simple. When I made my voicemail.conf, I added the boxes to the end of the file. The problem was, at the end was in a different context, so mwi would not light. The solution, all I had to

[Asterisk-Users] Re: HINT

2005-05-07 Thread Thorben Jensen
> Could you please give us some more detail as to what you did, in terms of > configuring the hint, and specifically what changes in the behavior of the > running server-phone interaction as a result? You need to set the hint for the phone when the phone is being dialed like this: exten => 20

[Asterisk-Users] ChanIsAvail for MGCP

2005-05-07 Thread Daniel Nyström
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 ChanIsAvail does not work with MGCP channels, as said in the wiki. But other applications works simular, like Queue and Dial. What's really the problem with ChanIsAvail? Is it possible to use Queue and Dial to make a working ChanIsAvail? I will take a b

Re: [Asterisk-Users] Re: HINT

2005-05-07 Thread Brian Capouch
Thorben Jensen wrote: Could you please give us some more detail as to what you did, in terms of configuring the hint, and specifically what changes in the behavior of the running server-phone interaction as a result? You need to set the hint for the phone when the phone is being dialed like thi

[Asterisk-Users] DTMF detection with Adit 600

2005-05-07 Thread Daniel Nyström
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 It seems like Asterisk are having problems detecting DTMF digits when using an Adit 600 channel bank via MGCP. I've tried to turn on RFC 2833 on both Adit and Asterisk, but no digits at all are working then. Anyone experienced simular with Adit or other

Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

2005-05-07 Thread Eric Wieling aka ManxPower
Jay Milk wrote: You got your groups mixed up. Should be: [default] exten => _.,1,Dial(ZAP/g2/${EXTEN}) [outgoing] exten => _.,1,Dial(ZAP/g1/${EXTEN}) Means that anything coming in to channel-group 1 (default context) will be sent out through group 2, and vice versa. Watch the console and be amazed

Re: [Asterisk-Users] Music on Hold

2005-05-07 Thread Sahil Gupta
[EMAIL PROTECTED]:~# mpg123 -v High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. Uses code from various people. See 'README' for more! THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK! Regar

[Asterisk-Users] DTMF generated from phone or from gateway?

2005-05-07 Thread Kido NOAGBODJI
Hello all, I was wondering if the DTMF were generated from the phone or from the ATA? I have a cisco ATA 186. Thanks K. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCR

Re: [Asterisk-Users] CVS question

2005-05-07 Thread administrator tootai
Mark Johnson a écrit : Is there a way to get a download of asterisk from cvs-head as of like 3 weeks ago? Having some weird problems and most people say that alot of these things have been introduced over the last few weeks. cvs co -D 2005-02-15 asterisk will give you the 15 february 2005 versio

Re: [Asterisk-Users] Asterisk on SMP machine with ztdummy working ?

2005-05-07 Thread Vamsi Pottangi
That is nice to hear. Congrats. Wondering who could help me out with this unique zap channel problem of mine. Thanks, ~Vamsi On 5/7/05, Tim Connolly <[EMAIL PROTECTED]> wrote: > I've got three dual Xeon's running Redhat Enterprise 4 with 2.6.9 > and CVS-HEAD from about a month ago. I didn

Re: [Asterisk-Users] Re: chan_zap.so: load_module fails: Fedora Core 3: SMP

2005-05-07 Thread Vamsi Pottangi
Bulls Eye !!! Thanks for that Tony ! It worked. Initially I thought that default conf file would work like my previous installations. Thanks, ~Vamsi On 5/7/05, Tony Mountifield <[EMAIL PROTECTED]> wrote: > In article <[EMAIL PROTECTED]>, > Vamsi Pottangi <[EMAIL PROTECTED]> wrote: > > > > I'm tr

[Asterisk-Users] IAX service provider with account balance announcement

2005-05-07 Thread C W Nel
Can anyone tell me if any IAX service provider supply audible "minutes left"/account balance announcement? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 06/05/2005

Re: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Rich Adamson
> I started out happy as a clam with my new Broadvoice account and > asterisk machine. About 10 days ago things began to change. Inbound > calling has been down for 2 days. Beyond the "We are currently > experiencing in-bound call issues with a carrier partner in some areas. > We are aware o

Re: [Asterisk-Users] Re: HINT

2005-05-07 Thread Julian J. M.
But that only works when SIP/201 receives a call, right? What if SIP/201 is making a dialout call, does it show as busy in the phone's keypad? Julian J. M. On 5/7/05, Thorben Jensen <[EMAIL PROTECTED]> wrote: > > Could you please give us some more detail as to what you did, in terms of > > confi

RE: [Asterisk-Users] Re: HINT

2005-05-07 Thread Armin Lediger (HotZone GmbH)
Yes it does. Armin > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Julian J. M. > Sent: Saturday, May 07, 2005 1:48 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Re: HINT > > > But that only w

Re: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Andrew Kohlsmith
On May 7, 2005 12:22 am, JD wrote: > Who's happy with their voip service using asterisk? I am. Nufone. For the past 18 months. Totally happy. > Where do you get reliable DIDs? I have a PRI I get my DIDs on. I have not yet found a VOIP provider with DIDs available in a WIDE area with reliabl

Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

2005-05-07 Thread Andrew Kohlsmith
On May 7, 2005 05:16 am, Eric Wieling aka ManxPower wrote: > Watch the console and be amazed when _. matches extension "h", which is > called when the far side of the call hangs up. You get two calls to the > same number by only dialing once! Stop being lazy and at least use _X. > as your pattern

Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

2005-05-07 Thread Eric Wieling aka ManxPower
Andrew Kohlsmith wrote: On May 7, 2005 05:16 am, Eric Wieling aka ManxPower wrote: Watch the console and be amazed when _. matches extension "h", which is called when the far side of the call hangs up. You get two calls to the same number by only dialing once! Stop being lazy and at least use _X.

Re: [Asterisk-Users] Problem with Manager Originate and SIP extension

2005-05-07 Thread Terje Elde
Terje Elde wrote: [m197] type=friend username=m197 secret= qualify=200 nat=yes host=dynamic canreinvite=no context=from-sip qualify=200, when the server is in the US, and my phone is in Norway, might not have been the best idea. Problem solved. Terje

Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

2005-05-07 Thread Andrew Kohlsmith
On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote: > In much of the world the PSTN dialplan is not that simple. Yes, a more > specific dialplan than my _X. exmaple is a good idea, but the USA has a > VERY simple PSTN dialplan and is NOT like most of the world. Perhaps I am naive but I don

Re: [Asterisk-Users] IAXy Firmware Upgrade

2005-05-07 Thread Time Bandit
> I'd like to known what I have to do to upgrade > the firmware into a IAXy device. It does it automagically when it connect to Asterisk if a newer version is available. Look in /var/lib/asterisk/firmware/iax and you will see iaxy.bin. hth ___ Asterisk-U

[Asterisk-Users] Re: Re: HINT

2005-05-07 Thread Thorben Jensen
You need to put that in whether SIP/201 is recieving or making a call. This only work for SIP/201 - you will need to do the same for every phone you have. thorben "Julian J. M." <[EMAIL PROTECTED]> skrev i en meddelelse news:[EMAIL PROTECTED] But that only works when SIP/201 receives a call,

RE: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Andre Normandin
I've had Broadvoice for over a year now, and although their outages are really annoying, the fact that their service costs $20/month unlimited is what keeps me with them.. I have 2 Inbound #'s through them (same account), one in GA (678-253) and one in CT (203-935), and overall their inbound has b

[Asterisk-Users] Termination South America

2005-05-07 Thread JOAO CARLOS MOURA
We offer termination in:   Miami,USA  u$s 0.019  Buenos Aires,ARGENTINA  u$s 0.019 Fortaleza,BRAZIL  u$s 0.029   Check our rates in CHILE Santiago, PARAGUAY Asuncion, URUGUAY Montevideo, Punta del Este, BRAZIL Rio de Janeiro, San Pablo, Goiania, Puerto Alegre.   DID's u$s 5.50 each in all ou

[Asterisk-Users] Inexpensive FAX and 800 Number retail service

2005-05-07 Thread lonnie
Greetings All, I have a number of projects in the works at the moment and for one of them, I need to locate an inexpensive and reliable service that can provide small-office virtual services: 1. FAX to Email 2. Toll Free number with voicemail boxes for Tech Support, Billing Inquiries, Customer Se

[Asterisk-Users] end user gui

2005-05-07 Thread Jim Sturtevant
I’ve reviewed the wiki and other resources and haven’t been able to locate a tool which would allow an end user to make changes to their service.  The features and end-user might want to change is fairly limited (call fwd, number of rings, etc).   This might require real-time. Thanks in adv

RE: [Asterisk-Users] Inexpensive FAX and 800 Number retail service

2005-05-07 Thread Jim Sturtevant
I've been using www.maxemail.com for quite awhile and they provide great service. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, May 07, 2005 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

RE: [Asterisk-Users] Inexpensive FAX and 800 Number retail service

2005-05-07 Thread lonnie
Thanks, I'll look into that one as well. > I've been using www.maxemail.com for quite awhile and they provide great > service. > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: Saturday, May 07, 2005 8:52 AM > To: Ast

RE: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Chris Coulthurst
I tend to agree about the in-house being the 'stable part'. Like anything else on the internet, if you don't have control of all parts (trunks and phones and dialplans), there are bound to be issues with uptime, and how your equipment responds to 'their' downtime. It reminds me of the headaches I

RE: [Asterisk-Users] Re: Re: HINT

2005-05-07 Thread Anton Krall
Can you post a full dialplan example... Also, will this only work for certain phones and atas also? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Thorben Jensen |Sent: Sábado, 07 de Mayo de 2005 09:45 a.m. |To: asterisk-users@lists.digium.com |S

Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

2005-05-07 Thread Eric Wieling aka ManxPower
Andrew Kohlsmith wrote: On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote: In much of the world the PSTN dialplan is not that simple. Yes, a more specific dialplan than my _X. exmaple is a good idea, but the USA has a VERY simple PSTN dialplan and is NOT like most of the world. Perhaps I

Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

2005-05-07 Thread Eric Wieling aka ManxPower
Andrew Kohlsmith wrote: On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote: In much of the world the PSTN dialplan is not that simple. Yes, a more specific dialplan than my _X. exmaple is a good idea, but the USA has a VERY simple PSTN dialplan and is NOT like most of the world. Perhaps I

Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

2005-05-07 Thread Eric Wieling aka ManxPower
Andrew Kohlsmith wrote: On May 7, 2005 08:29 am, Eric Wieling aka ManxPower wrote: In much of the world the PSTN dialplan is not that simple. Yes, a more specific dialplan than my _X. exmaple is a good idea, but the USA has a VERY simple PSTN dialplan and is NOT like most of the world. Perhaps I

RE: [Asterisk-Users] Good NAT Pnp Hardphone

2005-05-07 Thread Nabeel Jafferali
> For example, > I want to give a phone to my brother, who is going to europe. His ICH > softphone is fine there. Both the poly and cisco though require you to > setup for nat. He would not be able to set this up though, so I want to > just give him a preconfig'ed phone and plug and go... My ex

RE: [Asterisk-Users] Inexpensive FAX and 800 Number retail service

2005-05-07 Thread Nabeel Jafferali
> 1. FAX to Email Check out TrustFax (http://tinyurl.com/8png8). $10/year for a toll-free fax number and $0.10 per page in/out. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990 ___ Asterisk-Use

Re: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Michael D Schelin
Isn't amazing what has happened in the last five or six years with the Internet. There is no design flaw with IPv4. It was created back when you were in diapers and with todays pda's having more power than the systems back then. An industry protocol that is going strong 30 or more years is am

[Asterisk-Users] h323.conf - Asterisk not routing incoming calls based on IP - Ignoring type=user + host= + context=

2005-05-07 Thread Deon
Ok, at the bottom of my h323.conf file on my 1st server I have this: ; - [test] type=user host=209.237.227.185 context=termination-test incominglimit=10 accountcode=005 ; - Using an Asterisk at the other IP, I have this: exten => _1NXXNXX,1,Dial(H323/[E

[Asterisk-Users] cant connect

2005-05-07 Thread Bernie
I just installed Asterisk @home 1.0 and its up and running, i added an extension with the web interface, but now when I try to connect with a sip client (x-lite) it just times out. here is the log from x-lite below. Is there any way to view a log on the asterisk side to see whats going on? or

[Asterisk-Users] two questions about the Sipura 841?

2005-05-07 Thread Joel Duffield
Ok my first question is I have seen messages about a patch for asterisk so that I can do auto answer on these phones. I found the message in the archives but I do not have that message as an email still, so I do not have the attachment. Can anyone tell me where to get it? Also on this phone how can

RE: [Asterisk-Users] two questions about the Sipura 841?

2005-05-07 Thread Jim Sturtevant
What is the purpose of the beeping? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joel Duffield Sent: Saturday, May 07, 2005 12:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] two questions about the Sipura 841? Ok my first question

[Asterisk-Users] WIP-5000 and DTMF

2005-05-07 Thread Jim Meehan
My WIP-5000 phone is working well with my Asterisk box now, except for DTMF. All DTMF key presses come across as clipped or just clicks on the remote side. I had this problem with my Sipura ATA as well, but fixed that by playing with the settings on the Sipura device. I've tried dtmfmode=inband a

[Asterisk-Users] WIP-5000 and DTMF

2005-05-07 Thread Jerry Geis
Jim, My (3) WIP -5000 phones work just fine with DTMF. I setup the user.ini file to the following and of course the same in the sip.conf. Hope this helps. The OpenSip is also in the browser config for the phone... [OpenSip] *T1 = 500 *T2 = 4000 ; DTMFType - 0 RTP ; DTMFType - 1 INFO ; DTMFType - 2

[Asterisk-Users] Polyco ip600 incoming ring time

2005-05-07 Thread Gregory Wiktor - ADCom Corp.
Hello All, Does anyone know how to reduce the incoming ringtime on the polycoms? What I mean is, When I have an incoming call, my 7960 and pingtel ring immediately, but the polycom seems to delay 2 seconds before ringing... Any ideas? Greg ___ Asterisk

RE: [Asterisk-Users] two questions about the Sipura 841?

2005-05-07 Thread Joel Duffield
The beeping is to tell you that the remote end has hungup, im sorry I don't know the technical term for it but it happens on your regular home phone if the other end was to hang up and you did not hang up your receiver. the web interface calls it the "Reorder". Thanks Joel -Original Message-

[Asterisk-Users] Cisco ATA & Call Waiting

2005-05-07 Thread Christopher Kenna
I currently have 2 Cisco 7960's and 2 ATA 186's connected to asterisk. The 7960's work just fine for call waiting, but the ATA's dont. I cant seem to get the ATA's to use the call waiting feature, the calls just go straight to voicemail instead of prompting with the usual tone.   Please help Chri

RE: [Asterisk-Users] CallerID name lookup AGI script

2005-05-07 Thread Nabeel Jafferali
Jim: I modified your script to first look up Google and then look up 411.com. It's better for me, because 411.com has Canadian listings too. I still left Google in because it's much faster and if it has information, I'd rather use that. I removed the area code thing because it's no use to me. I al

RE: [Asterisk-Users] CallerID name lookup AGI script

2005-05-07 Thread Nabeel Jafferali
BTW This does not do most business name lookups from 411.com correctly. Maybe someone who actually knows Perl can do that :) -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990 > -Original Message- > From: [EMAIL PROTECTED] [mailto:

Re: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread John Novack
Michael D Schelin wrote: Again in the the last few years VoIP has come a long way as the PSTN has had over 100 years to perfect theirs. If we did not have to interface with the PSTN don't you think we would be better off? They didn't have to interface with anybody else. Well, if one studies th

[Asterisk-Users] Cisco ATA 186 and Asterisk

2005-05-07 Thread Christopher Iarocci
Anyone have call waiting working on the ATA-186 connected to Asterisk? Other VoIP phones seem to work, but I can not get the ATAs to allow call waiting. Christopher M Iarocci Network Admin JD Posillico 631-249-1872 X244 ___ Asterisk-Users mailing list

Re: [Asterisk-Users] Cisco ATA 186 and Asterisk

2005-05-07 Thread Deon
I think it has to do with your CallFeatures. Callfeatures: 0x I have a screen shot of my converters config if you want it, it supports call waiting. I had to turn it off on one of my customers converters once, I had to change the last 2 digits or something to turn off call waiting. But it

Re: [Asterisk-Users] Transparently Routing German pri through Asterisk

2005-05-07 Thread Peter Svensson
On Sat, 7 May 2005, Andrew Kohlsmith wrote: > Perhaps I am naive but I don't think that diaplans would be that much more > complex if people matched more accurately at all. Granted most of my calling > is north american, but there's some south america and germany in there as > well, along with

Re: [Asterisk-Users] Cisco ATA 186 and Asterisk

2005-05-07 Thread Christopher Kenna
if you could set me up with your config, that would be great.   thanx Chris >>> [EMAIL PROTECTED] 5/7/2005 6:52 PM >>> I think it has to do with your CallFeatures.Callfeatures: 0xI have a screen shot of my converters config if you want it, it supportscall waiting. I had to turn it off on

RE: [Asterisk-Users] Good NAT Pnp Hardphone

2005-05-07 Thread Anton Krall
Any special settings on * or your nat firewalls? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Nabeel Jafferali |Sent: Sábado, 07 de Mayo de 2005 01:07 p.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Use

Re: [Asterisk-Users] 3 x TDM400P in one PC ??

2005-05-07 Thread C F
On 5/6/05, Ariel Batista <[EMAIL PROTECTED]> wrote: > I just setup one Dell SC420 with just one SATA drive and 512mg Ram ($ > 404.00) with 2 TDM04B in it that is 8 FXO ports. And a second system for > another customer with 3 TDM 2 TDM40B 8 FXS ports and one TDM01B for 4 FXO's. > > Both systems are

Re: [Asterisk-Users] SEND TEXT to an extension?

2005-05-07 Thread C F
On 5/6/05, Anton Krall <[EMAIL PROTECTED]> wrote: > Will this only work on polycoms? Do you need to be on an active call to send > text? > As far as I know polycoms are the only phones that support it, but there might be others. No you don't really need to be on an active call to send text messag

RE: [Asterisk-Users] Who's happy with their voip service?

2005-05-07 Thread Chris Coulthurst
To be more specific to my point -- Using the internet today, with the demands of streaming real-time applications, which require a level of QoS wasn't originally designed in to IPv4. With a wide array of mods, patches and additions, there is 'some' support for prioritization. We would be b

[Asterisk-Users] At home Asterisk via Broadvoice?

2005-05-07 Thread John Stegenga
Hi all - sorry if what I'm asking is FAQ by now - I only have 2789 digest messages that I've not read yet...   The local phone company (Bell South) has gotten completely out of hand with their rates, and with them suing anyone who wants to compete against them...  So, I'm thinking very hard

RE: [Asterisk-Users] Good NAT Pnp Hardphone

2005-05-07 Thread Nabeel Jafferali
> Any special settings on * or your nat firewalls? Nope. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com

Re: [Asterisk-Users] ZapBarge a PRI DDI

2005-05-07 Thread izo
On 5/6/05, Steve Rawlings <[EMAIL PROTECTED]> wrote: > I'm using a TE405p with all four spans enabled, two configured as pri_cpe > and two as pri_net, the asterisk is sitting between our ISDN (UK BT > EuroISDN30) and our phone system. We have 200 DDI numbers on the ISDN's and > I need to give one

Re: [Asterisk-Users] At home Asterisk via Broadvoice?

2005-05-07 Thread Andrew Kohlsmith
On May 7, 2005 11:04 pm, John Stegenga wrote: > Broadvoice will give me 2 lines, with 2 phone numbers each - distinctive > ring - for a reasonable fee... Please do a google search for "broadvoice problems site:lists.digium.com" and reconsider your choice of VOIP provider. That reasonable fee doe

RE: [Asterisk-Users] Good NAT Pnp Hardphone

2005-05-07 Thread Anton Krall
Is your * open on the internet? No firewalls? And on the nat firealls no need to open any ports or do port forwarding to your natted phone? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Nabeel Jafferali |Sent: Sábado, 07 de Mayo de 2005 10:09 p.m.

Re: [Asterisk-Users] At home Asterisk via Broadvoice?

2005-05-07 Thread Luki
My thoughts: 1) I would not run asterisk on a laptop, or on Windows (if you can get it to turn properly via emulation like Vmware). 2) A 586 *might* be enough to handle this low call volume with no transcoding. 3) I know nothing about a Azatel 2 port adapter, but you could acquire a Sipura 2000

[Asterisk-Users] Getting DTMF to work with SIP?

2005-05-07 Thread beonice
Folks, from googling, I see that the dtmfmode parameter is not valid in the [general] context. My problem is that my overseas DID through Libretel seems to want to come into the [general] context! And, having done that, I get my welcome message, but then the DID does not accept the DTMF when I t

[Asterisk-Users] Setting the jitter buffer in AIX

2005-05-07 Thread Bryce W Nesbitt
Are these things possible? 1) Set the local Asterisk jitterbuffer size, but only for a particular connection. I'd like to force Asterisk to use a particularly large buffer in certain cases. Should I expect this to work? [general] jitterbuffer=no register => username:[EMAIL PROTECTED] ;pa

Re: [Asterisk-Users] Good NAT Pnp Hardphone

2005-05-07 Thread Eric Wieling aka ManxPower
If Asterisk is on a public IP, nat=yes in sip.conf takes care of all the required magic. No port forwarding needed anywhere, no special NAT settings needed on the phone. Anton Krall wrote: Is your * open on the internet? No firewalls? And on the nat firealls no need to open any ports or do port

Re: [Asterisk-Users] Am I on the right track, and consultants

2005-05-07 Thread Andrew Latham
http://www.voip-info.org/tiki-index.php?page=Asterisk+consultants+USA On 5/6/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Hi, > > I've been lurking here for about a month and I've been putting together > our companies planned migration to a new office and a new phone system. > Could anybo

RE: [Asterisk-Users] Good NAT Pnp Hardphone

2005-05-07 Thread Anton Krall
And if asterisk is behind nat doing prot forwarding? Say you just forwarded udp 4569 5060 5004 1-2000? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Eric Wieling aka ManxPower |Sent: Sábado, 07 de Mayo de 2005 11:02 p.m. |To: Asterisk Users Ma

RE: [Asterisk-Users] Good NAT Pnp Hardphone

2005-05-07 Thread Nabeel Jafferali
> And if asterisk is behind nat doing prot forwarding? Say you just > forwarded > udp 4569 5060 5004 1-2000? You'd just need to set externip correctly, assuming you have a static public IP. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 4

[Asterisk-Users] Distinctive Ring

2005-05-07 Thread Anton Krall
Guys. How do you configure asterisk to recognize distingtive ringing using x100p cards? Can this be done and how? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or u

[Asterisk-Users] Asterisk@Home on OnComputers Show Sunday morning

2005-05-07 Thread Kerry Garrison
From somewhere in Cyberspace to all points on the compass its the On Computers Radio Show.  Sunday's show will be all tech talk with the team. Peter Kastner will start off the show by talking about this week's hot news stories then the gang will continue on with tech talk. During the secon

Re: [Asterisk-Users] Good NAT Pnp Hardphone

2005-05-07 Thread Eric Wieling aka ManxPower
Then you need to use externip= localnet=, portforward 5060 and whatever ports you are using for RTP. Check rtp.conf. I don't recall if rtp.conf controls incoming or outgoing RTP packets. You have to portforward whatever ports the incoming RTP is. This has been discussed to death on the maili

[Asterisk-Users] What is the Polycom 301, 501 & 601?

2005-05-07 Thread Matt Darnell
These phones are mentioned in the Sip 1.5 manuals, anyone know what the differences are? Aloha, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

[Asterisk-Users] Setting variable for a context for all extensions?

2005-05-07 Thread Mark Wormgoor
Hi, Is it possible to set a variable for a context for all extensions? I haven't been able to find it. I want something like this in extensions.conf: [from-iaxfwd] exten => .,1,RING=r3 exten => 123456,1,Goto(from-pstn,s,1) [from-internal] exten => .,1,RING=r2 include => ext-local [ext-local] e

RE: [Asterisk-Users] Good NAT Pnp Hardphone

2005-05-07 Thread Anton Krall
If ip is dinamyc, can exterip be set instead as externhost? Where host is pointing always to the current ip? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Nabeel Jafferali |Sent: Domingo, 08 de Mayo de 2005 12:15 a.m. |To: 'Asterisk Users Mailing