Re: [Asterisk-Users] Asterisk-1.0 RC1

2004-07-16 Thread Jean-Yves Avenard
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello On 17/07/2004, at 4:17 PM, Mark Spencer wrote: ftp://ftp.digium.com/pub/asterisk Can someone grab a copy and put it on a mirror server? Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phon

Re: [Asterisk-Users] Asterisk-1.0 RC1

2004-07-16 Thread Jean-Yves Avenard
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Mark. thank you heaps for this. I also just updated our CVS source copy and it doesn't compile anymore it stops in the pbx directory with hundreds of GTK errors. I'm running RedHat 9.0 Linux (with 2.4.26 kernel) Now, will download the files from

Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-16 Thread Olle E. Johansson
Nicolas Gudino wrote: On Fri, 2004-07-16 at 18:28, Matthias Endler wrote: is it possible to receive SIP/IAX register and unregister events via the manager API (like in CLI)? I do receive all kinds of call events (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink). chan_sip2 support

[Asterisk-Users] Asterisk-1.0 RC1

2004-07-16 Thread Mark Spencer
We have officially made the first release candidate of Zaptel, Libpri, Asterisk and Gastman available. While there are still open major bugs, they are relatively limited, and it was time to go ahead and get the 1.0 ball rolling in earnest. ftp://ftp.digium.com/pub/asterisk Enjoy the code. Spe

Re: [Asterisk-Users] Pressing digits on SNOM phone results in letters on display

2004-07-16 Thread Jean-Yves Avenard
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 You should read the manual of your snom then (available from the internal website on the snom phone) Basically you press the a->1 soft-menu button at the top. Jean-Yves On 17/07/2004, at 1:23 PM, Rana Dutt wrote: My SNOM 200 phone got into a funny mod

[Asterisk-Users] CIsco 7905 and VLAN

2004-07-16 Thread Kevin
Has anyone had any success in getting the Cisco 7905/7912 phones working with a managed switch and VLAN's? I programmed a voice VLAN and changed the parameter in the telephones for the new VLAN. The individual ports are set up for tagging. It seems to work fine with the Cisco 7960/7940 telephone

[Asterisk-Users] Pressing digits on SNOM phone results in letters on display

2004-07-16 Thread Rana Dutt
My SNOM 200 phone got into a funny mode where if I dial any digit, a letter gets displayed and sent, so dialing no longer works. For example, if I dial "9", the letter "w" gets displayed and sent when I press OK. How do I get it out of this mode? ___ Ast

[Asterisk-Users] I already have a VAD frame?

2004-07-16 Thread Andrew Yager
Just shoved my 7960 onto one of my clients networks. I'm talking from outside the office to them (they are using a GS BT101). I'm getting the following message repeatedly in the log, whenever someone talks. Jul 17 12:29:40 NOTICE[96402352]: frame.c:120 ast_smoother_feed: Dropping extra frame of

[Asterisk-Users] Transmitting a hook-flash down an E&M DS-0?

2004-07-16 Thread Kris Boutilier
I'm trying to access feature codes remotely over a channelised T1 between a Norstar MICS (rev 4) and Asterisk. The timeslots are configured E&M and have been working fine under most circumstances except this one. There is mention of accessing the facility by calling Flash() from within extensions.c

RE: [Asterisk-Users] Asterisk Gui client

2004-07-16 Thread mattf
Hello, What version of the astguiclient suite are you using? What version of PHP are you using? Do you have GLOBAL_VARS turned on or off? It's very strange that being a POST all of the variables seem to be showing up on the URL like a GET would. also it doesn't sem to be submitting to the admin

[Asterisk-Users] Need configuration sample for VoIP(SIP) -> PSTN Gateway

2004-07-16 Thread Alejandro Sosa
Hello,   I’m very new with * and I would really appreciate some help to implement a SIP to PSTN Gateway. My current scenario includes an * box with a TE405P board. I have a 1.5Mb connection to the outside world (using a router with firewall capabilities) and channel banks that allow me

RE: [Asterisk-Users] 7960 Dynamic DNS?

2004-07-16 Thread Marty Mastera
> It's a way to specify a DNS via config file which has > priority over whatever is handed out from DHCP. > > (Optional) IP address of a new dynamic DNS server. If a new > DNS server address is specified, it is used for any further > DNS requests after the phone uses the initial DNS address >

Re: [Asterisk-Users] 7960 Dynamic DNS?

2004-07-16 Thread Shane Young
Quoting Marty Mastera <[EMAIL PROTECTED]>: > Hello everyone > > Searching the archives and google always comes up with entries regarding > the "dyn" dns option in the 7960, but I can't find answers to my > specific question It's a way to specify a DNS via config file which has priority o

Re: [Asterisk-Users] PSTN/phone/FXO/FXS cabling issue

2004-07-16 Thread Lyle Giese
Line 1 is always the center two pins. Line2 (on a two line analog jack) are the pins just to the outside of the center two pins. Now you know why the center two pins are pair 1 and the pins to the outside of them is pair 2. The rj11/12 jack will plug into the wider rj45 just fine. You just have

Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration

2004-07-16 Thread Christopher L. Wade
Please disregard, I have 'solved' the issue. Thank you, Chris Christopher L. Wade wrote: Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the ap

[Asterisk-Users] 7960 Dynamic DNS?

2004-07-16 Thread Marty Mastera
Hello everyone Searching the archives and google always comes up with entries regarding the "dyn" dns option in the 7960, but I can't find answers to my specific question My 7960 is connected via cable modem and is NAT'ed (everything is working fine). On the 7960 under SIP configuration\

Re: [Asterisk-Users] Offhook tone in channel OSS/dsp

2004-07-16 Thread Chris Foster
On Fri, 16 Jul 2004 16:36:34 +0200, Roger Schreiter <[EMAIL PROTECTED]> wrote: > Hi, > > I have to develop a phone application using asterisk's > chan_oss. > > When the phone is idle, i.e. the last command was a hangup, > one hears a "toot, toot, toot, ..." > But unforuntaly its use is in Germany

Re: [Asterisk-Users] DTMF issue --help

2004-07-16 Thread Andrew Yager
On 17/07/2004, at 3:24 AM, Eric Wieling wrote: Tony Nichols wrote: > After calling a bank, or cc processing center; you have to enter your social security number, or the cc number - followed by the # key. The lovely * voice responds "transfering" I'm sorry that was an invalade selection. Sometim

Re: [Asterisk-Users] Looking for WiFi phone recommendations

2004-07-16 Thread Andrew Yager
On 17/07/2004, at 7:13 AM, James H. Thompson wrote: Looking for WLAN - WiSIP - WiFi phone recommendations and experiences. What works, what doesn't. I'm using a ZyXel Prestige 2000W. It works, and sounds OK. Call hold, transfer, forward (and anything else that makes the phone worth using as a SIP

[Asterisk-Users] Sipura 3000 user guide is now available

2004-07-16 Thread Bill Reid
The "SPA User Guide" now covers configuring the 3000. http://www.sipura.com/support/index.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.

RE: [Asterisk-Users] PSTN/phone/FXO/FXS cabling issue

2004-07-16 Thread Sean Cheesman
Just use a standard phone cable. It will seat properly. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florin Andrei Sent: Friday, July 16, 2004 5:12 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PSTN/phone/FXO/FXS cabling issue I just received a

[Asterisk-Users] PSTN/phone/FXO/FXS cabling issue

2004-07-16 Thread Florin Andrei
I just received a Wildcard TDM400P by FedEx yesterday. I noticed that the FXO/FXS modules use connectors similar to Ethernet. Now, i want to connect the TDM400P to the PSTN connector in the wall, and also to a regular analog phone. Both the PSTN conn and the phone use smaller connectors, typical f

Re: [Asterisk-Users] SIP register and unregister events via Manager API

2004-07-16 Thread Nicolas Gudino
Hi Matthias, On Fri, 2004-07-16 at 18:28, Matthias Endler wrote: > Hi all, > > is it possible to receive SIP/IAX register and unregister events via the > manager API (like in CLI)? I do receive all kinds of call events > (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink). chan_s

[Asterisk-Users] SIP register and unregister events via Manager API

2004-07-16 Thread Matthias Endler
Hi all, is it possible to receive SIP/IAX register and unregister events via the manager API (like in CLI)? I do receive all kinds of call events (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink). My manager.conf looks like this: [general] enabled = yes port = 5038 bindaddr = 0

[Asterisk-Users] Looking for WiFi phone recommendations

2004-07-16 Thread James H. Thompson
Looking for WLAN - WiSIP - WiFi phone recommendations and experiences. What works, what doesn't.   Thanks.   Jim   James H. Thompson[EMAIL PROTECTED]

[Asterisk-Users] Asterisk Gui client

2004-07-16 Thread James Freire
I have installed the Asterisk gui client that is available off of sourceforge.net. I was curious if anybody here has used it and what experiences they have had with it. I am having a problem with it, I am able to use the admin page except when I try to submit information to the server to add ph

Re: [Asterisk-Users] When does the PUC become an issue?

2004-07-16 Thread rich allen
you may want to contact your state's PUC. sounds like you want to be a long distance carrier and each state will have different regulations/requirements in addition to federal rules. from everything i have read this all is still a very "grey" area - hcir On Jul 16, 2004, at 3:09 AM, John G

[Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration

2004-07-16 Thread Christopher L. Wade
Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration

RE: [Asterisk-Users] SIP channels UNKWN

2004-07-16 Thread brian
They will disappear in 15 seconds... they are waiting around just in case we get a late response on register. bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of M3 Freak > Sent: Friday, July 16, 2004 12:58 PM > To: [EMAIL PROTECTED]

Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION

2004-07-16 Thread Steve Underwood
Rich Adamson wrote: On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote: No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but

[Asterisk-Users] outgoing calls over SIP

2004-07-16 Thread Johannes van Hulst
I have a hard time to forward all my outgoing calls to my SIP connection. For incoming calls I set up the registry and it is working perfectly   How can I tell asterisk to forward all calls beginning with 9 to Gate.Sipserver.com User : userxx Password : pwxx   Best regards,   Ha

Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION

2004-07-16 Thread Rich Adamson
> On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote: > > > No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b > > or x100p running any Head cvs after June 23rd (totally stock install). > > > > Wouldn't necessarily recommend this box for any commercial production > > use, bu

Re: [Asterisk-Users] Bounty! For help with echo cancellation code.

2004-07-16 Thread Steve Underwood
[EMAIL PROTECTED] wrote: From the CLI and during a call I want to be able to: *** Pulse the outgoing line and record at least 50 ms of the incoming line. The pulse waveform must be specifiable as a series of amplitudes for each 1/8000 sec time slot. It would be best of these values

[Asterisk-Users] Patch to test: Mailbox path changes

2004-07-16 Thread Olle E. Johansson
http://bugs.digium.com/bug_view_page.php?bug_id=0001188 This patch unifies the code that decides on the location of a mailbox and stores voicemail in a tree-like structure, to be prepared for very large volumes of voicemailboxes in one file system. It's disputed whether this affects performance on

[Asterisk-Users] Problems with festival

2004-07-16 Thread Dan Fernandez
I cannot get Festival to work with asterisk. I have the following:   exten => 555,1,Answerexten => 555,2,Festival(mary has a little lamb)exten => 555,3,Hangup   I get the following from asterisk: "Festival returned ER" and the festival logs shows the following:   client(1) Fri Jul 16 15:35:54

[Asterisk-Users] Cisco Call Manager and Asterisk (AVVID) - Comparison

2004-07-16 Thread San Singhania
Hello All,   I have a large customer with close to 30 offices worldwide who want a VOIP solution. We have already implemented it in Singapore and India, but as usual the Corporate IT department which is in London is pushing for the Cisco call manager for all their locations.   I dont know a

Re: [Asterisk-Users] Fedora Core 2 softphone

2004-07-16 Thread defiance
Thanks for all the suggestions, unfortunately none of them worked. I could not for the life of me get phonegaim to compile. Had to change the automake calls in the 20 makefiles and it was still hosed up. I ended up just running xlite under wine. Works perfectly. Kind of a shame though. chris On T

[Asterisk-Users] zaptel red alarms with e&m wink

2004-07-16 Thread Glen Hinkle
I'm getting red alarms with my T100P card that last from 4-15 seconds. They seem to happen randomly, every couple of days. Has anyone seen this behavior, or does anyone have any ideas regarding what would be causing it? Thanks, Glen ___ Asterisk-

Re: [Asterisk-Users] SIP channels UNKWN

2004-07-16 Thread M3 Freak
On Fri, 2004-07-16 at 13:28, Eric Wieling wrote: > I'm having an oddball issue with a Polycom SoundPoint IP 500. As you > can see below Asterisk thinks there are 2 SIP channels active, but show > channels tells me there are no calls active. Anyone have any idea why > this is happening? The Po

[Asterisk-Users] Patch to test: Never say goodbye to an agent :-)

2004-07-16 Thread Olle E. Johansson
http://bugs.digium.com/bug_view_page.php?bug_id=0001693 This patch adds a lot of options for AgentLogin/AgentCallbackLogin Please test and respond in the bug tracker! /O - "This patch adds quite a few new features i

Re: [Asterisk-Users] Problem when using asterisk + gnugk

2004-07-16 Thread Roger Schreiter
Hi, maybe my fault was, that I named the gatekeeper in h323.conf. When I comment those lines, just tell in gnugk about the gateway ip address (asterisk), it's fine: | alias=myname | ;gatekeeper = 123.456.789.12 | ;AllowGKRouted = yes | | [myname] | type=h323 | prefix=0,1,2,3,4,5,6,7,8,9 Roger. Rog

[Asterisk-Users] Re: sendmail.cf and relaymail to a smtp server

2004-07-16 Thread David Cook
I am presuming your Asterisk box is behind your ISP. You don't actually need user/pw to send somebody email in the outside world, but your ISP has prevented you from _directly_ sending email to anybody and make you go through their SMTP server which forces you to authenticate with it like a Mail Us

Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION

2004-07-16 Thread Andrew Kohlsmith
On Friday 16 July 2004 12:43, W. Kevin Hunt wrote: > First let me say that normal cheapy PC hardware couldn't be made to > function with out echo. We tried on both the single port Digium T1 card > and the 4 port Digium T1 card. Even on a SuperMicro Dual PIII-933 w/ > hardware scsi raid we had ech

Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION

2004-07-16 Thread Scott Laird
On Jul 16, 2004, at 11:07 AM, Rich Adamson wrote: No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so c

[Asterisk-Users] SIP module with radius authentication support

2004-07-16 Thread Gildo J. Cáceres M.
Hi there,   I have been looking for information about radius authentication for SIP module on registration process, any help will be appreciated.   Regards     Conexion GroupGildo J. Cáceres M.Developer[EMAIL PROTECTED]www.conexiongroup

Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION

2004-07-16 Thread Joshua M. Thompson
On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote: > No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b > or x100p running any Head cvs after June 23rd (totally stock install). > > Wouldn't necessarily recommend this box for any commercial production > use, but... > > What

Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION

2004-07-16 Thread Joshua McClintock
We're using the "Supermicro X5DPE-G2" in both of our asterisk boxes. They have 3 (I think) PCI-X (133mhz) 64 bit slots. We're only using 2 of them (slots) and their not sharing IRQs. We have one pstn gateway that talks to the pri and the channel bank. The other machine is the pbx (and vm) where

[Asterisk-Users] SIP channels UNKWN

2004-07-16 Thread Eric Wieling
I'm having an oddball issue with a Polycom SoundPoint IP 500. As you can see below Asterisk thinks there are 2 SIP channels active, but show channels tells me there are no calls active. Anyone have any idea why this is happening? The Polycom occasionally stops accepting calls and requires a

Re: [Asterisk-Users] DTMF issue --help

2004-07-16 Thread Eric Wieling
Tony Nichols wrote: > After calling a bank, or cc processing center; you have to enter your social security number, or the cc number - followed by the # key. The lovely * voice responds "transfering" I'm sorry that was an invalade selection. Sometimes the IVR on the other end still gets the digi

Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION

2004-07-16 Thread Rich Adamson
> After speaking with several people, and even participating in a forum of > several other people with echo issues, I thought I'd share what we've > done (well actually what our chief R&D engineer, Brett Bourn has > done...) > > First let me say that normal cheapy PC hardware couldn't be made to >

RE: [Asterisk-Users] Astersik with g729 and 120 active channels with digium card ISDN PRI

2004-07-16 Thread Senad Jordanovic
Title: Message http://www.voip-info.org/wiki-Asterisk+dimensioning -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian NocettiSent: 16 July 2004 14:43To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Astersik with g729 and 120 acti

[Asterisk-Users] Hardware platform / features

2004-07-16 Thread matiaspinedo
I'm new to asterisk, I have some questions regarding the hardware required. For a complete E1 and something like 30 phones, what hardware would be required ? What is the bottleneck ? Memory, processor speed ? Any experiences of how much scalability does the asterisk have ? Can Asterisk have "vi

[Asterisk-Users] Echo problem update - POSSIBLE SOLUTION

2004-07-16 Thread W. Kevin Hunt
After speaking with several people, and even participating in a forum of several other people with echo issues, I thought I'd share what we've done (well actually what our chief R&D engineer, Brett Bourn has done...) First let me say that normal cheapy PC hardware couldn't be made to function with

[Asterisk-Users] sendmail.cf and relaymail to a smtp server

2004-07-16 Thread Johannes van Hulst
Hello,   Who can help me I am trying to setup the sendmail so that I can mail the voicemail’s to an internet SMTP mail server. I know that I have to setup the sendmail.cf and configured a relay to my normal SMTP server.   I am running RedHat 9 and my internet provider has a SMTP mail s

Re: [Asterisk-Users] Feature Group D

2004-07-16 Thread Mike Machado
Yes, this is working for me. The remote side must send *** for this to work. If asterisk does not detect the first *, then it prints the message you are seeing. On Fri, 2004-07-16 at 07:18, Joseph wrote: > Is anyone using Feature Group D with a digium 405P card > and get the ANI spill to work (ca

[Asterisk-Users] How to configure Asterisk as a VoIP(SIP) to PSTN Gateway?

2004-07-16 Thread Alejandro Sosa
Hello,   I’m very new with * and I would really appreciate some help to implement a SIP to PSTN Gateway. My current scenario includes an * box with a TE405P board. I have a 1.5Mb connection to the outside world (using a router with firewall capabilities) and channel banks that allow me t

[Asterisk-Users] Path to test: Sending HTML virus, no, VOICEMAIL!

2004-07-16 Thread Olle E. Johansson
http://bugs.digium.com/bug_view_page.php?bug_id=0002055 This patch adds the ability to send text and HTML messages as voicemail notificiations. Please test and respond to the bug tracker! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.d

[Asterisk-Users] Re: spa-3000 review?

2004-07-16 Thread Tom Neville
On Jul 16, 2004, at 12:22 AM, Randy Bush wrote: please be specific about dialplan hack and how you pointed to it in the config. thanks! Set Admin-Advanced->Line1->DialPlan-> ([2-9]xx<:@gw0>|[3469]11|0|00|[2-9]xx|1xxx[2 -9]xxS0|.) "[2-9]xx<:@gw0>" will send 211,311,411...911

[Asterisk-Users] Path to test: Czech localization

2004-07-16 Thread Olle E. Johansson
http://bugs.digium.com/bug_view_page.php?bug_id=0002013 If you use the Czech language, please test this and add your opinion, good or bad, to the bug tracker. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/as

[Asterisk-Users] Patch to test: Dynamic queues

2004-07-16 Thread Olle E. Johansson
http://bugs.digium.com/bug_view_page.php?bug_id=0001858 Constfilin writes: "The attached patch allows dynamic configuration of asterisk queues. Queue information is re-read from the configurable database in real time. Additional Information Right now implemented only for postgres, no mysql" Please

Re: [Asterisk-Users] Where can i get an UK SIP account with UK number?

2004-07-16 Thread Steve Kennedy
On Fri, Jul 16, 2004 at 10:49:57AM -0500, Chris Luke wrote: [stuff snipped] > http://www.btwholesale.com/content/binaries/service_and_support/pricing_information/carrier_price_list_browsable/b1_08.rtf > Tells you how much geographic number portability costs with BT, though > you may also need to

RE: [Asterisk-Users] Where can i get an UK SIP account with UK number?

2004-07-16 Thread Kevin Walsh
Chris Bond [EMAIL PROTECTED] wrote: > What I want to know is why you can port mobile numbers from network to > network but say you have a local std code and you wanted to use that with > a VoIP provider in the UK - in most instances you cant. Some are > available to offer STD codes in certain town

RE: [Asterisk-Users] Where can i get an UK SIP account with UK number?

2004-07-16 Thread Kevin Walsh
Mark Turner [EMAIL PROTECTED] wrote: > Dameon D. Welch-Abernathy wrote: > > > > There are probably others. > > > Such as www.intervivo.net. > > p.s. I work there. > You might want to change the FAQ: Q. Are you planning on offering non 0845 numbers? A. The industry is looking into various altern

Re: [Asterisk-Users] Re: VoicePulse changes

2004-07-16 Thread Chris Sullivan
On Jul 16, 2004, at 7:03 AM, <[EMAIL PROTECTED]> wrote: Yeah...that's believable. Mike, wake upthey made changes, the broke things. While they obviously "tried" to give everyone a month's notice, it just didn't work out that way. While the old config still works (now), I find it difficult

Re: [Asterisk-Users] Where can i get an UK SIP account with UK number?

2004-07-16 Thread Chris Luke
[EMAIL PROTECTED] wrote (on Jul 16): > On Fri, 16 Jul 2004, Chris Bond wrote: > > What I want to know is why you can port mobile numbers from network to > > network but say you have a local std code and you wanted to use that with a > > VoIP provider in the UK - in most instances you cant. Some ar

Re: [Asterisk-Users] where to sign up for fwd

2004-07-16 Thread John Galt
Thanks for the quick reply. That should do it. -Galt On Fri, 16 Jul 2004 17:17:29 +0200, Thomas Niesel <[EMAIL PROTECTED]> wrote: > Hallo John Galt > On Fri, 16 Jul 2004 07:48:23 -0700 you wrote: > > > Could someone please point me to the proper url to register for a fwd > > acount and get a fw

Re: [Asterisk-Users] where to sign up for fwd

2004-07-16 Thread Bodo Hahnke
Go here: http://account.freeworlddialup.com/index_new.php?section_id=94 or just press the "GET FWD"-Button on the page... At 16:48 16.07.2004, you wrote: Could someone please point me to the proper url to register for a fwd acount and get a fwd number. I couldn't find it at www.freeworlddialup.com

[Asterisk-Users] 3COM 3102 SIP Phone

2004-07-16 Thread Ben Bush
Can anyone point me in the right direction to learn how to make the 3COM 3102 phone work with Asterisk. I have looked everywhere that I can think of and I am not able to turn anything up. Do I need to have a login to 3COM's site to obtain a SIP image? Any help would be GREATLY appreciated. Than

Re: [Asterisk-Users] Where can i get an UK SIP account with UK number?

2004-07-16 Thread Steve Kennedy
On Fri, Jul 16, 2004 at 03:50:40PM +0100, Chris Bond wrote: > What I want to know is why you can port mobile numbers from network to > network but say you have a local std code and you wanted to use that with a > VoIP provider in the UK - in most instances you cant. Some are available to > offer

[Asterisk-Users] How to handle a macro that dials both international (011) and national

2004-07-16 Thread James W. Brinkerhoff
I have a macro that handles my outbound dialing... First it does an enum lookup, and if that fails it hands the call off to voicepulse for connection. The problem is... EnumLookup() requires that international numbers *NOT* be prefixed with 011, however VoicePulse does... Can someone provid

RE: [Asterisk-Users] Where can i get an UK SIP account with UK number?

2004-07-16 Thread steve
On Fri, 16 Jul 2004, Chris Bond wrote: > What I want to know is why you can port mobile numbers from network to > network but say you have a local std code and you wanted to use that with a > VoIP provider in the UK - in most instances you cant. Some are available to > offer STD codes in certai

Re: [Asterisk-Users] Anyone experience with early dial?

2004-07-16 Thread Olle E. Johansson
Holger Schurig wrote: I keep replying to myself quite often. As it turned out, this is a problem with incrementing CSEQ on the Grandstream. I don't have the clue if the SIP specification says that you have to increment it, but the GS sometimes sends a different SIP message with the same CSEQ. He

Re: [Asterisk-Users] 'Reverse Hold' feature prototype...

2004-07-16 Thread Jason Garland
Maybe another nice feature might be for the other end of this problem... Menu options while you are on hold to change the crappy music or mute it. Also an option to punch in a callback number and have the company ring your phone when it is your turn to have the call answered. - Jason > I have no

[Asterisk-Users] DTMF issue --help

2004-07-16 Thread Tony Nichols
I'm getting down to the last of my * issues ... After calling a bank, or cc processing center; you have to enter your social security number, or the cc number - followed by the # key. The lovely * voice responds "transfering" I'm sorry that was an invalade selection. Sometimes the IVR on the oth

Re: [Asterisk-Users] Anyone experience with early dial?

2004-07-16 Thread Holger Schurig
I keep replying to myself quite often. As it turned out, this is a problem with incrementing CSEQ on the Grandstream. I don't have the clue if the SIP specification says that you have to increment it, but the GS sometimes sends a different SIP message with the same CSEQ. Here's an example, only

RE: [Asterisk-Users] Where can i get an UK SIP account with UK number?

2004-07-16 Thread Chris Bond
What I want to know is why you can port mobile numbers from network to network but say you have a local std code and you wanted to use that with a VoIP provider in the UK - in most instances you cant. Some are available to offer STD codes in certain towns but not all. Surely someone in the UK nee

[Asterisk-Users] where to sign up for fwd

2004-07-16 Thread John Galt
Could someone please point me to the proper url to register for a fwd acount and get a fwd number. I couldn't find it at www.freeworlddialup.com or fwd.pulver.com Thanks, -Galt -- "They that would give up essential liberty for temporary safety deserve neither liberty nor safety."

Re: [Asterisk-Users] Where can i get an UK SIP account with UK number?

2004-07-16 Thread Mark Turner
Dameon D. Welch-Abernathy wrote: There are probably others. Such as www.intervivo.net. Cheers, Mark. p.s. I work there. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update opti

Re: [Asterisk-Users] some questions on uniden uip200

2004-07-16 Thread Ryan Courtnage
On July 16, 2004 09:02 am, Jan Goericke wrote: > hello, > > yesterday the uniden uip200 phone was recommended to someone. i am looking > for an alternative to grandstream bt-100 because i can not do a supervised > tranfer with it. here my questions: > > 1) does the uip200 support supervised transfe

[Asterisk-Users] Offhook tone in channel OSS/dsp

2004-07-16 Thread Roger Schreiter
Hi, I have to develop a phone application using asterisk's chan_oss. When the phone is idle, i.e. the last command was a hangup, one hears a "toot, toot, toot, ..." But unforuntaly its use is in Germany, where one expects a continous "too ..." before dialing. Is ther

[Asterisk-Users] RESOLVED: 'Dropping voice to exceptionally long queue on IAX2'

2004-07-16 Thread Matt Davies | MattDavies.Net
As far as I can tell I have this resolved. For those of you that may encounter this error I send this out to help. I have determined that the problem lay with my mail setup. I was attaching voicemail messages to email alerts. The problem was my server decided to take the backup MX instead of the

Re: [Asterisk-Users] Flag Bad PRI Channel

2004-07-16 Thread Steven Critchfield
On Fri, 2004-07-16 at 02:03, Shawn Lawrence wrote: > Is there a way to flag a bad or noisy PRI channel in Asterisk so it will > be skipped over? How can a DIGITAL channel be noisy? your big problem will come from the fact that your provider will still attempt to ship calls down the channel you d

[Asterisk-Users] Feature Group D

2004-07-16 Thread Joseph
Is anyone using Feature Group D with a digium 405P card and get the ANI spill to work (callerid)? I can get the dnis to work fine. But not the caller id. Here is the error I get: chan_zap.c:4628 ss_thread: Got a non-Feature Group D input on channel 3. Assuming E&M Wink instead zapata.conf: ***

Re: [Asterisk-Users] MWI on Grand Stream ATA-286

2004-07-16 Thread Andrew Yager
On 16/07/2004, at 11:57 PM, Dennis Cartier wrote: Sorry if this has been asked before, but does anyone have any pointers on getting the Message Waiting Indicator working on a GS ATA-286? I have tried both settings on the 286's web page for sending a SUBSCRIBE to the SIP server. I was expecting the

RE: [Asterisk-Users] Re: Really long first ring, then normal

2004-07-16 Thread Paul Mahler
I have a recent version installed. I am having problems with hangup detection on my zap channels. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training > -Original Message- > From: [

RE: [Asterisk-Users] Re: VoicePulse changes

2004-07-16 Thread daryl
> > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > Randy Bush > > Sent: Thursday, July 15, 2004 3:35 PM > > To: [EMAIL PROTECTED] > > Cc: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] Re: VoicePulse changes > > > > the message arrived here so

[Asterisk-Users] Re: Using Asterisk with fiber optic

2004-07-16 Thread skruigners
>The fact how your Server/LAN is connected to the 'outside' world has >little to do with the way Asterisk connects to the PST, imho. > >What kind of voice 'streams' pass on this fiber? How is this fiber >connected? The fiber is connected through a hag, in which voice stream is digitized in IP tra

[Asterisk-Users] MWI on Grand Stream ATA-286

2004-07-16 Thread Dennis Cartier
Sorry if this has been asked before, but does anyone have any pointers on getting the Message Waiting Indicator working on a GS ATA-286? I have tried both settings on the 286's web page for sending a SUBSCRIBE to the SIP server. I was expecting the LED on the 286 to flash when there are messages w

[Asterisk-Users] Problem with asterisk and zaphfc

2004-07-16 Thread Massimo De Nadal
Hi everybody, I have a problem using zaphfc. When I start asterisk after 8-10 seconds I get the message "Primary D-Channel on span 1 down" and my isdn modem stops to work. If I place or receive a call before this message all works really fine (even if the call is very long), but when I hang up, aft

[Asterisk-Users] Re: Really long first ring, then normal

2004-07-16 Thread Andreas Anderson
Hiya, I've been seeing this lately with our Asterisk PBX as well. With Asterisk CVS HEAD 05-21, the ringing was normal. After our last upgrade, CVS HEAD 07-03, the first ring is almost always one continuous ring that lasts about 2 to 3 seconds. We're noticing other problems too, such as hangup d

[Asterisk-Users] Astersik with g729 and 120 active channels with digium card ISDN PRI

2004-07-16 Thread Sebastian Nocetti
Hello, I want to know what kind of equipment I need to handle 120 simultaneous calls with a Digium 4E1 card... and using 120 G.729 licences some help?   thanks   Sebastian.

Re: [Asterisk-Users] some questions on uniden uip200

2004-07-16 Thread Eric Wieling
Jan Goericke wrote: Think nobody would like to use call parking via #700 etc if he is "spoilt" by the features of an old Siemens Hicom and my fellows are spoilt. Or is there a more elegant way to make a BT100 doing this feature. I read nearly all threads in archive concerning transfer and sip. Se

Re: [Asterisk-Users] Anyone experience with early dial?

2004-07-16 Thread Holger Schurig
Ok, this seems to be a bug in channel_sip. When I do put all debug message AND I dial 0-0914115, then I only see: $ grep DTMF /var/log/asterisk/messages Jul 16 13:13:19 VERBOSE[6150]: Receiving DTMF! Jul 16 13:13:19 VERBOSE[6150]: * DTMF received: '9' Jul 16 13:13:20 VERBOSE[6150

Re: [Asterisk-Users] sip phone configuration problem

2004-07-16 Thread gomer
What phone do you have? On Fri, 16 Jul 2004 11:59:39 +0500, atif <[EMAIL PROTECTED]> wrote: > I am configuring a sip-phone, receing calls, excellent voice quality. but it does > not place calls, please, can some one sort out. > > here is my debug output, and below that is sip-debug, > > Jul 16

Re: [Asterisk-Users] some questions on uniden uip200

2004-07-16 Thread Jan Goericke
On Fri, 16 Jul 2004, Jean-Yves Avenard wrote: > With the BT100 you can always use call parking to achieve something > similar, sure it's not the most elegant way.. > Think nobody would like to use call parking via #700 etc if he is "spoilt" by the features of an old Siemens Hicom and my fel

Re: [Asterisk-Users] VoiceMail fails to delete messages after emailing them

2004-07-16 Thread Daniel Jimenez
Chris Smales - Magenta Solutions wrote: I've configured voicemail.conf to delete voice mails after they are emailed. The email work ok but the message don't delete. The config is as follows: [default] 3000 >= ,CS,[EMAIL PROTECTED],,delete=yes Thanks, Chris It should be 3000 => ,CS,[EMAIL P

RE: [Asterisk-Users] ACD Issues

2004-07-16 Thread Steve Hanselman
Sorry, my mistake, I thought you'd said you had assigned priorities, let me go back and search for your original posting Steve -Original Message- From: Robert Jackson [mailto:[EMAIL PROTECTED] Sent: 16 July 2004 14:10 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ACD Issues T

Re: [Asterisk-Users] Line Display

2004-07-16 Thread John Galt
On Fri, 16 Jul 2004 11:56:32 +0200, Holger Schurig <[EMAIL PROTECTED]> wrote: > In newer Grandstream Firmware you can have different dialtones for > different incoming caller id's. Asterisk can set the callerid to > anything, based on the incoming line, so you can hear by the ring from > which line

RE: [Asterisk-Users] ACD Issues

2004-07-16 Thread Robert Jackson
That would certainly make sense, but I am not sure how to set an Agent's priority. The only information that I have been able to find is setting a QUEUE_PRIO value when queuing the calls (New as of July 2004). Thanks, Robert Jackson > -Original Message- > From: Steve Hanselman [mailto:[

[Asterisk-Users] VoiceMail fails to delete messages after emailing them

2004-07-16 Thread Chris Smales - Magenta Solutions
I've configured voicemail.conf to delete voice mails after they are emailed. The email work ok but the message don't delete. The config is as follows: [default] 3000 >= ,CS,[EMAIL PROTECTED],,delete=yes Thanks, Chris ___ Asterisk-Users mailing list

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