Re: [asterisk-users] Problem in making SIP call after compiling Asterisk server

Prathap, That response is not sent by Asterisk. What you are most likely getting this from is a packet capture, and what you are referring to is an ICMP message sent as a backward notification by an intermediate router or host. Basically, it sounds like the SIP UDP port (5060) on the Asterisk

[asterisk-users] Problem in making SIP call after compiling Asterisk server

Hi There, I have installed an Asterisk server on Fedora Core, I can able to run the Asterisk Server successfully. But the problem is, my softphone(Xlite) is not getting registered with Asterisk server. From softphone Register request were sent and the Asterisk respond with "Destination Unreachabl

Re: [asterisk-users] Hardware that can ring my phone?

On 8/1/07, Linux Lover <[EMAIL PROTECTED]> wrote: > > > This SOHO PBX box won't interop with Asterisk > > because it doesn't speak any > > of the protocols that Asterisk does. This box > > I tend agree with your evaluation. Still, I was > thinking that since all these el-cheapo SOHO PBX boxes > su

Re: [asterisk-users] Dropouts and echo

Hi all, Can I ask that you please keep my personal address in the To: or CC: in this thread as for some reason I'm only getting half of the list emails coming through, and they're not showing up on the digium pipermail archive either. The list archive on http://marc.info seems to have the

Re: [asterisk-users] multiple pbxes, multiple domains, same user ids?

Ouch. And I thought I had an answer to my query. I totaly agree abt the long disclaimer nonsense Schmaltz, but I swear by the powers up there, it's the admins over here at my workplace doing all that nonsensical magic, as the mails go out. I wish i had the freedom to use gmail(just like you), t

Re: [asterisk-users] Hardware that can ring my phone?

> You can use a Linksys SPA-3102 for both FXO (POTS) and FXS (phone) > connection instead of a Digium card. The price is around $90-100. > > Almost any old PC will do if it can run Linux. There are also other > alternatives to a PC such as the Linksys WRT54GL. The OpenWRT (on whatever supported "r

Re: [asterisk-users] Hardware that can ring my phone?

you would still need an fxo port of some sort for asterisk for it to "pretend" to be a phone. Quoting Linux Lover <[EMAIL PROTECTED]>: > James, thank you for your educating answer. > > --- James FitzGibbon <[EMAIL PROTECTED]> > wrote: > >> >> This SOHO PBX box won't interop with Asterisk >> b

Re: [asterisk-users] Polycom 320 - Can it actually be configured?

At 21:02 8/1/2007, Doug, wrote: >At 16:49 8/1/2007, Douglas Garstang wrote: > >Don't know about the 320, but we provisioned the 301's. They're > >provisioning is basically the same as the 501's and 601's. What problems > >are you having? > >Have no problems with 501s or 601s or 430s. > >I p

[asterisk-users] chan_sip.c error

Hello all, I downloaded and built the Asterisk v1.4.9 from the Debian Unstable repository on my Debian Etch GNU/Linux but when I checked the logs, I got some error messages from the chan_sip.c. You can find the logs below. # pwd /usr/src/debian/ # apt-get build-dep asterisk # exit $ cd /usr/src/

Re: [asterisk-users] Hardware that can ring my phone?

James, thank you for your educating answer. --- James FitzGibbon <[EMAIL PROTECTED]> wrote: > > This SOHO PBX box won't interop with Asterisk > because it doesn't speak any > of the protocols that Asterisk does. This box > appears to be a solid-state > (and I'd assume very feature restricted) a

Re: [asterisk-users] Polycom 320 - Can it actually be configured?

At 16:49 8/1/2007, Douglas Garstang wrote: >Don't know about the 320, but we provisioned the 301's. They're >provisioning is basically the same as the 501's and 601's. What problems >are you having? Have no problems with 501s or 601s or 430s. I punch in the provisioning server IP, but the phon

Re: [asterisk-users] Asterisk DTMF Tones

John Meksavan wrote: > > The DTMF tones are being sent twice. On SIP Peer side, I set the > DTMFMODE=RFC2833 and the PAP2T you can choose from INBAND, AVT, INFO, and > AUTO, so I chose Auto. Should change on Peer Side and the PAP2T side > to use > INBAND? > > >> From: Alex Balashov <[EMAIL PROT

Re: [asterisk-users] Problem with the dial command

Mike wrote: > Thanks. Tell me, how intensive is it to use qualify? What type of > packet/check is done with this? Is it reasonnable to use qualify for > thousands of devices? > > Once the device is considered to be unreachable for any number of > reasons, will another poll of the device be d

Re: [asterisk-users] Couple installation questions

Once again guys thanks so much! Baji - Took a look at your instruction page. Thanks for putting that together. I've bookmarked it!! Howard On 8/1/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > On Wed, Aug 01, 2007 at 05:31:59PM -0400, hugolivude wrote: > > Hi, > > > > I'm installing * 1.4.9

Re: [asterisk-users] Problem with the dial command

Thanks. Tell me, how intensive is it to use qualify? What type of packet/check is done with this? Is it reasonnable to use qualify for thousands of devices? Once the device is considered to be unreachable for any number of reasons, will another poll of the device be done to check if it became a

Re: [asterisk-users] Asterisk DTMF Tones

The DTMF tones are being sent twice. On SIP Peer side, I set the DTMFMODE=RFC2833 and the PAP2T you can choose from INBAND, AVT, INFO, and AUTO, so I chose Auto. Should change on Peer Side and the PAP2T side to use INBAND? From: Alex Balashov <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mai

Re: [asterisk-users] Asterisk DTMF Tones

Alex, The DTMF tones are being sent twice. On SIP Peer side, I set the DTMFMODE=RFC2833 and the PAP2T you can choose from INBAND, AVT, INFO, and AUTO, so I chose Auto. Should change on Peer Side and the PAP2T side to use INBAND? >From: Alex Balashov <[EMAIL PROTECTED]> >Reply-To: Asterisk

Re: [asterisk-users] 2 Digit Issue

Thanks for the help it was a provider issue Jason Anthony Cennami wrote: > Looking at your dialplan I don't see extension 8 anything (8XXX) -- > Are you sure you didn't have those extensions in another context that > you forgot to include? > > According to the dialplan it is catching the inva

Re: [asterisk-users] Couple installation questions

On Wed, Aug 01, 2007 at 05:31:59PM -0400, hugolivude wrote: > Hi, > > I'm installing * 1.4.9 and a couple questions have come up: > > 1) I read > here( > http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4

Re: [asterisk-users] Polycom 320 - Can it actually be configured?

On Wednesday August 01 2007 5:49 pm, Douglas Garstang wrote: > Don't know about the 320, but we provisioned the 301's. They're > provisioning is basically the same as the 501's and 601's. What problems > are you having? > > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-u

Re: [asterisk-users] Retail DID provider ?

On Thu, 2007-08-02 at 01:12 +0530, Mail list wrote: > I am looking for a retail DID provider which should provide unlimited > incoming calls something around 4-5 bucks . Nufone seemed like a good > choice at $5 but they are down again :( . Any suggestions please . I'm using www.les.net . Regard

Re: [asterisk-users] Couple installation questions

On 8/1/07, hugolivude wrote: > Hi, > > I'm installing * 1.4.9 and a couple questions have come up: > > 1) I read here ( > http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x) > that if you are using E1 cards you need to install LIBPRI. I'm not using > any cards on this syst

Re: [asterisk-users] Problem with the dial command

qualify=yes in the sip.conf context for that device will change the device to unreachable and should send you directly to voicemail. There could still be a brief period where the device is timed out and the system hasn't qualified it yet, but outside of that, it will just continue trying to send t

Re: [asterisk-users] 2 Digit Issue

Well, I don't see it in the [inside] context that you include, and the Background application, by design, will jump to the invalid context as soon as it can no longer match a valid extension. I am assuming you're dialing from [impact,s,3]? Can you include the entire context(s), since the email th

Re: [asterisk-users] Hardware that can ring my phone?

Quoting Linux Lover <[EMAIL PROTECTED]>: > John, thank you very much. Indeed, this is the > direction I was thinking of taking. I just needed a > "quick & dirty" solution for the short term - I didn't > realize that Asterisk is so complex. > > In fact, I am not sure I completely understand it: > W

Re: [asterisk-users] Retail DID provider ?

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of SIP > Sent: Wednesday, August 01, 2007 1:05 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Retail DID provider ? > > IdeaSIP, Voxbone, G

Re: [asterisk-users] Polycom 320 - Can it actually be configured?

Don't know about the 320, but we provisioned the 301's. They're provisioning is basically the same as the 501's and 601's. What problems are you having? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Doug > Sent: Wednesday, August 01

[asterisk-users] dtmf issues over sip and pri

I have a pri connection to the phone company. Sending DTMF out over the pri I hear on my phone when I call it. However, a second box uses a SIP connection to talk to the first box. When the second box is trying to do the function sendDTMF(1) over the SIP connection and then out the PRI I do not he

[asterisk-users] Polycom 320 - Can it actually be configured?

Just got one of these. Horrible to program. Trying to key in the FTP server. Won't even remember the info after rebooting. Anybody know the proper way to beat on this stupid beast so it will work? ___ --Bandwidth and Colocation Provided by http://www

[asterisk-users] Announcing free (GPL) VXML for Asterisk - Voiceglue

The first release of Voiceglue is now available. Voiceglue provides a VXML interpreter using Asterisk telephony and the OpenVXI VXML parsing suite. It is released under the GPL, and thus compatible with Asterisk and OpenVXI licensing. The first release is available at the project website: htt

Re: [asterisk-users] Hardware that can ring my phone?

John, thank you very much. Indeed, this is the direction I was thinking of taking. I just needed a "quick & dirty" solution for the short term - I didn't realize that Asterisk is so complex. In fact, I am not sure I completely understand it: Will using Asterisk force me to use an external VoIP ser

[asterisk-users] Couple installation questions

Hi, I'm installing * 1.4.9 and a couple questions have come up: 1) I read here( http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x) that if you are using E1 cards you need to install LIBPRI. I'm not us

Re: [asterisk-users] 2 Digit Issue

I do have it in the inside context. It is also doing the circle dance. I just gave an example. It seems as if it is just forgetting any digits over 2. like that is in the dialplan but it is not. Jason Anthony Cennami wrote: > Looking at your dialplan I don't see extension 8 anything (8XXX)

Re: [asterisk-users] Problems building zaptel 1.4.4

Just had to install the linux kernel source... All better now! Thanks for responding everyone! I have a few installation questions, but I'll post them in a separate thread. Hugh On 8/1/07, hugolivude <[EMAIL PROTECTED]> wrote: > asterisk-dev:/ # rpm -qa | grep kernel > kernel-default-2.6.16.13

Re: [asterisk-users] 2 Digit Issue

Looking at your dialplan I don't see extension 8 anything (8XXX) -- Are you sure you didn't have those extensions in another context that you forgot to include? According to the dialplan it is catching the invalid extension and should be passing it to the i (invalid) handler to loop back into yo

Re: [asterisk-users] Problem with the dial command

Thanks Jared. It answers most of my question. Now, when the device doesn't register, the behavior is as expected. But eventually, any device that registers successfully might be unplugged, leaving Asterisk to wonder where the device has gone. So, what's the best approach to this? Should I put a

[asterisk-users] 2 Digit Issue

I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2 digits into the dialplan. error -- Invalid extension '81' in context 'impact' on SIP/207.174.111.34-b77167f8 I pressed 8107 and ideas my dial plan is (part of it) [impact] exten=>s,1,Answer() exten=>s,n,Set(CALLERID(name)=Im

Re: [asterisk-users] pri "call by call" trunking?

Call Sangoma On 8/1/07, Erik Anderson <[EMAIL PROTECTED]> wrote: > On 8/1/07, John covici <[EMAIL PROTECTED]> wrote: > > I had some troubles -- try setting the timing parameter to 0 (second > > one in your span) and see if that helps. > > If I'm reading the docs correctly, this param should only b

Re: [asterisk-users] Problems using TE412P and TDM400B in a IBM x3650

James FitzGibbon wrote: > Another day, another apparant unexplained hardware incompatibility. > > I have a TE412P and a TDM400B living quite happily in a whitebox using an > Intel motherboard: > > http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm > > I tried to move to an IBM x365

Re: [asterisk-users] Problems building zaptel 1.4.4

asterisk-dev:/ # rpm -qa | grep kernel kernel-default-2.6.16.13-4 Thanks, Howard On 8/1/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Wed, Aug 01, 2007 at 06:41:42AM -0400, hugolivude wrote: > > > Right. autoconf.h is not necessarily generated. Please ignore that > > > warning. To be fixed in

Re: [asterisk-users] pri "call by call" trunking?

On 8/1/07, John covici <[EMAIL PROTECTED]> wrote: > I had some troubles -- try setting the timing parameter to 0 (second > one in your span) and see if that helps. If I'm reading the docs correctly, this param should only be set to 0 if you *never* want to use the T1 connected to this port for tim

Re: [asterisk-users] Retail DID provider ?

IdeaSIP, Voxbone, Gizmo Project, etc... if we're talking retail. N. Mail list wrote: > I am looking for a retail DID provider which should provide unlimited > incoming calls something around 4-5 bucks . Nufone seemed like a good > choice at $5 but they are down again :( . Any suggestions please

Re: [asterisk-users] pri "call by call" trunking?

I had some troubles -- try setting the timing parameter to 0 (second one in your span) and see if that helps. on Wednesday 08/01/2007 Erik Anderson([EMAIL PROTECTED]) wrote > On 8/1/07, C F <[EMAIL PROTECTED]> wrote: > > what channel are they putting the Dchannel on? > > Post your zapata.conf a

[asterisk-users] Retail DID provider ?

I am looking for a retail DID provider which should provide unlimited incoming calls something around 4-5 bucks . Nufone seemed like a good choice at $5 but they are down again :( . Any suggestions please . ___ --Bandwidth and Colocation Provided by http:

Re: [asterisk-users] Asterisk DTMF Tones

John, On Wed, 1 Aug 2007, John Meksavan wrote: > I am running Asterisk 1.2.13 on Debian Linux 2.6.18-4-amd64 and having > problems with DTMF Tones. I have sip service from Teliax and configure to > use rfc2833 for dtmfmode. The problem occurs, when I am using Linksys PAP2T > phone adapter wit

Re: [asterisk-users] pri "call by call" trunking?

On 8/1/07, C F <[EMAIL PROTECTED]> wrote: > what channel are they putting the Dchannel on? > Post your zapata.conf and zaptel.conf The D channel is on 24. zaptel.conf: loadzone=us defaultzone=us span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf lpdlnx04 asterisk # cat zapata.conf ;autogenerat

Re: [asterisk-users] Slightly OT: SNOM & PoE

I use on a regular basis the D-Link line, they work. With the SNOM you will want to set the ignore Ethernet unplug in case the Ethernet switch restarts (like a Netgear 7248 attached to a cheap fiber trans). Keep in mind that holding a GSM phone real close to some of the SNOM phones will cause the

[asterisk-users] Asterisk DTMF Tones

Asterisk Users, I am running Asterisk 1.2.13 on Debian Linux 2.6.18-4-amd64 and having problems with DTMF Tones. I have sip service from Teliax and configure to use rfc2833 for dtmfmode. The problem occurs, when I am using Linksys PAP2T phone adapter with a regular analog phone. Is this an

Re: [asterisk-users] pri "call by call" trunking?

what channel are they putting the Dchannel on? Post your zapata.conf and zaptel.conf On 8/1/07, Erik Anderson <[EMAIL PROTECTED]> wrote: > I've been working with a telco for the past two days trying to get a > PRI span up and running. This is a small-ish telco and I get the > feeling they don't d

Re: [asterisk-users] Problem with the dial command

On Wed, 2007-08-01 at 11:43 -0400, Mike wrote: > Aug 1 11:47:57 NOTICE[26107]: app_dial.c:1069 dial_exec_full: Unable > to create channel of type 'SIP' (cause 3 - No route to destination) This happens when Asterisk don't know where to find the peer (which is often the case if the device has faile

Re: [asterisk-users] not hearing dtmf tones

On 7/31/07, Jerry Geis <[EMAIL PROTECTED]> wrote: > > I am trying to re-create calling sendDTMF in an agi and not hearing the > digit either. The above seems to re-create that without the AGI. > ...you will have to configure your polycom / sip peer for inband DTMF if you want to hear the tones

[asterisk-users] perl script to generate new sip.conf users

Hi All, I remember some folks had put together a web page or perl script to generate many sip.conf entries from a file defining the users, vmbox, secret, CID and other variables. Can someone please point me in the right direction. Thanks. JR -- JR Richardson Engineering for the Masses ___

[asterisk-users] Slightly OT: SNOM & PoE

Hello All, I apologize for the slightly off-topic question, but I'm sure that the people best acquainted with the issue would be hanging around here. We recently deployed several Linksys POE switches for some smaller customers (10-24 station) and appear to be suffering from intermittent lock-ups

Re: [asterisk-users] Hardware that can ring my phone?

Linux Lover wrote: > But one thing that I forgot to mention is that my > business is only in its beginning stage and I need to > be as thrifty as possible. While $216 is a reasonable > price, I was wondering whether my (currently very > modest) goal can be achieved by spending much less > (under

[asterisk-users] pri "call by call" trunking?

I've been working with a telco for the past two days trying to get a PRI span up and running. This is a small-ish telco and I get the feeling they don't do this very often. Anyway, they specified a pretty standard setup: ni2 switchtype, esf framing, b8zs coding, etc. All of my b-channels are up

Re: [asterisk-users] Hardware that can ring my phone?

Do you think you'll outgrow 1 phone line any time soon. If so You'll want something that you don't have to completely redo when you add the next line. That digium card you linked to has 2 more expansion slots open for more lines or phones. The soho pbx you linked to looks like you can have mo

Re: [asterisk-users] Hardware that can ring my phone?

A phone system for under $100 is asking a lot. It can be done, but what is your time worth. You might want to consider some other phone system if all you need is IVR and analog support or look at hosted solutions. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

Re: [asterisk-users] Hardware that can ring my phone?

On 8/1/07, Linux Lover <[EMAIL PROTECTED]> wrote: > But one thing that I forgot to mention is that my > business is only in its beginning stage and I need to > be as thrifty as possible. While $216 is a reasonable > price, I was wondering whether my (currently very > modest) goal can be achieved b

Re: [asterisk-users] Hardware that can ring my phone?

This is what I have at home and it works okay. I also added an SPA-2002 (~$70) that adds another two FXS (phone) ports for a total of three. Godspeed, Phil Drew Gibson wrote: > Hi Lynn, > > You can use a Linksys SPA-3102 for both FXO (POTS) and FXS (phone) > connection instead of a Digium c

Re: [asterisk-users] Hardware that can ring my phone?

Lynn, I am unfamiliar with soho-pbx, so I cannot comment on quality, service, configurability, etc. They are based out of Hong Kong, and their box is probably already running some flavor of Asterisk, so you would need nothing additional except for the phone line coming in and the telephone.

Re: [asterisk-users] Hardware that can ring my phone?

Hi Lynn, You can use a Linksys SPA-3102 for both FXO (POTS) and FXS (phone) connection instead of a Digium card. The price is around $90-100. Almost any old PC will do if it can run Linux. There are also other alternatives to a PC such as the Linksys WRT54GL. I use a Linksys NSLU2 (Slug) at ho

Re: [asterisk-users] How to use stun server?

which stun server do you use? On 8/1/07, SIP <[EMAIL PROTECTED]> wrote: > > No... there's no STUN server built into Asterisk. Asterisk handles NAT > in a different way... and is an endpoint rather than a proxy, so it > doesn't really NEED STUN built into it. > > However, we run a STUN server on th

[asterisk-users] Agent Question

Hi, All, I have a question about agents and queues. Right now we have about 4 queues in our system. Some agents are in multiple queues. Our main queue is for technical support and it's by far our busiest queue as well. We have the autologoff feature set to 14 sec right now in the agents.conf

[asterisk-users] Problem with the dial command

Hi, I have an Asterisk 1.2 (can`t upgrade to 1.4 because of some makefile error on my particular system, bug report opened). That being said, I doubt my particular issue is a bug, I think it's me not understanding something. Let`s take a simple dialplan command, i.e. make the phone ring for 15

Re: [asterisk-users] Hardware that can ring my phone?

Yes, you understood correctly. Thank you - and all others who replied so quickly - for your precise and guiding answers. The Digium TDM11B looks looks like the perfect match for me: http://www.telephonyware.com/telephonyware/tw00068.html But one thing that I forgot to mention is that my business

Re: [asterisk-users] How to use stun server?

No... there's no STUN server built into Asterisk. Asterisk handles NAT in a different way... and is an endpoint rather than a proxy, so it doesn't really NEED STUN built into it. However, we run a STUN server on the same machine as an Asterisk server and see nothing in terms of load increase. S

Re: [asterisk-users] How to use stun server?

Ok thanx. One more thing to ask is: does asterisk has a stun server implemented in it or not. i mean does asterisk contain a stun server and provides any application which can be used for enabling the stun server in asterisk? On 8/1/07, SIP <[EMAIL PROTECTED]> wrote: > > STUN is a pretty simplisti

Re: [asterisk-users] Hardware that can ring my phone?

Lynn, If I understand you question correctly, you would need: A computer (preferably a server) to run Asterisk An analog interface card such as the Digium TDM400P An analog phone line (POTS) An analog (real) phone Calls would come in on the POTS line, get answered by Asterisk. Callers would he

Re: [asterisk-users] Hardware that can ring my phone?

Lynn, What you need is an ATA (analog telephone adapter). The ATA is a SIP or IAX extension on your Asterisk box, and your standard phone plugs into it. Asterisk sends the call to the SIP extension (the ATA), and the ATA rings your phone. On the flip side, your phone dials normally and the ATA dig

Re: [asterisk-users] Hardware that can ring my phone?

Quoting Linux Lover <[EMAIL PROTECTED]>: any of the various module based cards with one fxo and one fxs port will do what you want. > Hello, > > I am a small business owner in need for a solution > that automatically answers an incoming call, prompts > the caller via touch-tone menu ("press 1

Re: [asterisk-users] Hardware that can ring my phone?

On Wed, 2007-08-01 at 06:48 -0700, Linux Lover wrote: > I am a small business owner in need for a solution > that automatically answers an incoming call, prompts > the caller via touch-tone menu ("press 1 to leave a > message, press 0 to speak to a representative") and > will ring my (real) phone O

[asterisk-users] Can you specify a sip UA's codec based on IP?

Does anyone have any tricks to use some logic with SIP UA's codec negotiation based on the UA's IP? What I would like to do is have Cisco 7960's use g711u when they register with a local IP, and g729 when they register with a non-local IP. I was thinking about sip.conf and making two entries for ea

[asterisk-users] Hardware that can ring my phone?

Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu ("press 1 to leave a message, press 0 to speak to a representative") and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is cap

Re: [asterisk-users] How to use stun server?

I cannot really help except to say you may want to ask this question on the stund list (if they have one) since it relates more to the STUN software than it does Asterisk. Thanks, Steve Rizwan Hisham wrote: > Hi all, > This is the first time i am using stun with asterisk for nat problems. > I

Re: [asterisk-users] multiple pbxes, multiple domains, same user ids?

Can you please get rid of your awfull long nonsense disclaimer? On 8/1/07, Benjamin Jacob <[EMAIL PROTECTED]> wrote: > Hello good ppl, > A couple of questions for multiple pbxes > 1. Is it possible to support multiple pbxes in one Asterisk box(using > contexts, etc.)? > 2. Can we use the "domain"

[asterisk-users] Help on AsteriskNOW

Guys, please help me in understanding what I'm mistaking... Description: I've configured my AsteriskNOW (beta 6) server, in service providers section, with the parameters provided by my ITSP. Until now I've used this configuration with SJphone and all worked perfectly. Now I've decided to

Re: [asterisk-users] How to use stun server?

STUN is a pretty simplistic server. There's nothing else that needs to be configured on the STUN server side. It's pretty much either running or it's not. Just start plugging in the server to your clients and give it a whirl. It should work. N. Rizwan Hisham wrote: > Hi all, > This is the fi

Re: [asterisk-users] TE120P in Canada

On Wed, 2007-08-01 at 11:44 +1000, Klaverstyn, David C wrote: > I’m having problems trying to get a TE120P operational in Canada. [snip] > I’m not sure if the span line is correct. [snip] > span=1,1,0,ccs,hdb3,crc4 This would be right if you're configuring an E1 line, but since you're in Canad

[asterisk-users] How to use stun server?

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most peo

Re: [asterisk-users] TE120P in Canada

On 7/31/07, Klaverstyn, David C <[EMAIL PROTECTED]> wrote: > > span=1,1,0,ccs,hdb3,crc4 > I was under the impression that ccs/hdb3 was more typical of E1 service than T1. I ran across this when looking up something on span syntax yesterday (from http://www.voip-info.org/wiki/index.php?page=Zaptel

Re: [asterisk-users] Problems building zaptel 1.4.4

On Wed, Aug 01, 2007 at 06:41:42AM -0400, hugolivude wrote: > > Right. autoconf.h is not necessarily generated. Please ignore that > > warning. To be fixed in upcoming versions of Zaptel. > > > > (This is not related in any way to GNU autoconf, that is used to > > generate the ./configure script. T

[asterisk-users] multiple pbxes, multiple domains, same user ids?

Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the "domain" field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering tw

Re: [asterisk-users] Problems building zaptel 1.4.4

> ./configure Works > make menuselect Works > make > make install Does not work. Errors already posted. > menuselect select the modules to compile > > do you have problems with this? > I can select modules but make fails. Thanks, Hugh ___ --Band

Re: [asterisk-users] Problems building zaptel 1.4.4

> Right. autoconf.h is not necessarily generated. Please ignore that > warning. To be fixed in upcoming versions of Zaptel. > > (This is not related in any way to GNU autoconf, that is used to > generate the ./configure script. The program autoconf itself is not > needed to build Zaptel, unless you

Re: [asterisk-users] Turn off SIP 183 Session Progress in Asterisk 1.4.8

Hi Andrew Thanks for your response. > Yes, the option is progressinband in sip.conf As far as I can tell, the progressinband setting does not prevent the 183 from being sent altogether. For example, with progressinband=never, early audio from UA1 to UA2 after a 180 ringing will still trigger a 1

Re: [asterisk-users] 1and1 dedicated servers have been down for a few hours .

Am Dienstag, den 31.07.2007, 07:39 -0500 schrieb Asterisk guy: > 1and1 dedicated server's service has been down for a few hours , > unable to reach them by phone or email. do anyone know what is going > on there ? There were rumours they had trouble with an outdated version of the web administr

Re: [asterisk-users] Dropouts and echo

What kind of switch are you connecting the phones to? I've seen that behaviour with cheap Repotec switches (24+2Gigabit). Just replacing it with a different one fixed the problem. Julian J. M. On 7/31/07, Tom Lanyon <[EMAIL PROTECTED]> wrote: > The issues: > Dropouts - by far the most ser