RE : [asterisk-users] New Asterisk StumbleUpon Group
Hello Matt, I have not seen how to add a site. Could you help me (us) ? Tks Francois Bergeret, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Matt Riddell (IT) Envoyé : vendredi 6 octobre 2006 11:40 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] New Asterisk StumbleUpon Group -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Just thought I'd let people know that I've created a new StumbleUpon group for Asterisk sites. If you have a site that is related to Asterisk and is not listed, feel free to add it. Alternatively, if you're new to Asterisk and want to find out what sites are out there pop on over and have a look: http://asterisk.group.stumbleupon.com/ - -- Cheers, Matt Riddell ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astcc help-pleasssssseeee
So what should I do? On 10/7/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: On Fri, Oct 06, 2006 at 10:39:18PM -0700, Ali wrote:> Hi,>> I am wondering if astcc has ever worked for someone because it always return > 0 for answeredtime! I tracked every bit of informaion on google and wiki and> finally found out that its because of asterisk returning to dial plan after> executing Dial, so astcc.agi runs through the end without wating for call > completion.>> Am I missing something crazy? please someone give me a hint.>>astcc doesn't use strict. This is perl code I wouldn't like to touch.--Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755 iax:[EMAIL PROTECTED]+972-50-7952406 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] http://www.xorcom.com___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] polycom auto cfg file
http://www.freedomphones.net/polycom/files/polycom.phone1cfg.pl-script does anyone know how to use this file? Can you give me some tips on how to create it and use it with a Trixbox? Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ftp server
Whats the best ftp server to upload Polycom phone cfg’s from? I’m finding it a bit hit and miss using BTF server. Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Dialtone
Thé new phone no longer has a dial tone. I connected a different analog phone and it does not work but I have the phone lines plugged into the correct ports. Thé tech wants to remote in and have a look. Regards, Ed Sent from my BlackBerry® wireless handheld -Original Message- From: Rich Adamson <[EMAIL PROTECTED]> Date: Sat, 07 Oct 2006 22:10:14 To:Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Dialtone Jay R. Ashworth wrote: > On Sat, Oct 07, 2006 at 10:31:16AM -0500, Rich Adamson wrote: >> If you've messed up in connecting telephone lines to the wrong module, >> the ringing voltage sent to a fxs module will destroy it. You would need >> to replace the module. > > I'm going to stick my neck out here, and opine that any FXS module that > would be destroyed by receiving ringing voltage is *incredibly* poorly > designed, and very probably wouldn't pass Part 68. "Shouldn't", certainly. Try it and see what happens, and report back. ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Real-time and priority "n"
Is it exclusive? Either Realtime or priority "n" ??? If so, what is the better way? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Dialtone
Jay R. Ashworth wrote: On Sat, Oct 07, 2006 at 10:31:16AM -0500, Rich Adamson wrote: If you've messed up in connecting telephone lines to the wrong module, the ringing voltage sent to a fxs module will destroy it. You would need to replace the module. I'm going to stick my neck out here, and opine that any FXS module that would be destroyed by receiving ringing voltage is *incredibly* poorly designed, and very probably wouldn't pass Part 68. "Shouldn't", certainly. Try it and see what happens, and report back. ;) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pop a web page with DID in url
On 10/5/06, Michael Sampson <[EMAIL PROTECTED]> wrote: I'm looking to do this. When a call comes in to an agent in a queue, pop a web page like this http://www.mydomain.com/cgi-bin/script.cgi?did=952900 Where did is the number the caller dialed to reach the system in the first place. I know Hudlite can do this we caller ID, but the DID feature is not there yet. Does anyone have any other software they know of that can do this? FOP can do that. http://www.asternic.org Best regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk: Can anybody forward anybody's extension?
Hi, On Asterisk based system, you press *72 and it asks to enter the extension you want to forward. This way you can forward anybody's extension to any number. Is there any control on this feature. Isn't this a security issue? How do I know that nobody is forwarding my extension without my permission? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] disabling hardware echo can on tdm2400p
Hey list, Short version: I have a need to disable the hardware can on the tdm24xxp I have. I figure it's something in zconfig.h in the zaptel directory, but I'll be damned if I can figure it out. Long version: I have a tdm2403e card which is experiencing an odd problem; When several lines are in use, there is a "bleeding" of lines. My users call it the 'ghost'. Regardless, they can hear other people's conversation on different lines. I've been told this has to do with the hardware echo can I have on there, and that I should disable it if I continue having problems. So that's where I stand. Answers and opinions welcome. Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CDR
On Fri, Oct 06, 2006 at 11:46:22AM +0500, Rizwan Hisham wrote: > Hi guys, > i just want know how do i enable CDR in asterisk. Should be enabled automatically. There are several CDR modules. Some of them log to a simple CSV file: one line per call. Others log to all sorts of databases. Does the directory /var/log/asterisk/cdr-custom exist? Is it writable to the asterisk user? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Xorcom Astribank and 64 bit linux
DOes anyone know if you can use an Astribank usb channelbank with a 64 bit linux distribution like CentOs? I saw a note that the driver is only built when you have an i386 processor and a kernel>= 2.6.10 but it does not mention if you can build it manually for other architecture/kernel. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanIsAvail() in 1.2.12.1
Am Friday 06 October 2006 23:03 schrieb Douglas Garstang:
> *CLI> -- Executing NoOp("SIP/3254101-0817a220", "*** Originated call
> "Chocolate Chip" <3254101> -> 3254103") in new stack -- Executing
> NoOp("SIP/3254101-0817a220", "FOO1") in new stack -- Executing
> ChanIsAvail("SIP/3254101-0817a220", "SIP/3254103") in new stack
>
> It never makes it past the call to ChanIsAvail(). Dialplan processing just
> completely stops at this point. What's up with that???
Asterisk SVN-tag-1.2.12.1
Its working fine. (Iam using Realtime)
best regards
Thomas
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Re: [asterisk-users] Requirements for Asterisk & SER integratio
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER~Vamsi On 10/8/06, Crazy Boy <[EMAIL PROTECTED]> wrote: Hi Friends,I would like have Asterisk and SER implementation. I have lot of experience with Asterisk. Can anybody tell me what I have to install to integrate SER with Asterisk? Looking forwrad to your response. Thank you. Regards,Chandra.Get your email and more, right on the new Yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail maintenance
On Wed, Oct 04, 2006 at 03:17:14PM -0500, Jordan Novak wrote: > > Has anyone created a GUI for this. I would like to implement a server > specifically for Voicemail using out of band signalling tied to a PBX. I > fear the management will be exhaustive though. Can you please define the task, please? /me wonders if a little cron job won't be better. If at all required. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Requirements for Asterisk & SER integration
Hi Friends,I would like have Asterisk and SER implementation. I have lot of experience with Asterisk. Can anybody tell me what I have to install to integrate SER with Asterisk? Looking forwrad to your response. Thank you.Regards,Chandra. Get your email and more, right on the new Yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HTTP Connection Closed on 7960 SIP
Very interesting. The port was open but there was an HTTP proxy entry in the SIP config still. Thanks! On Oct 6, 2006, at 9:21 PM, Aaron Daniel wrote: This happens if you have a logo_url configured for your phone and the phone can't access it. I'm guessing you don't allow 80 through the firewall to the server that's serving the image. -- Aaron Daniel On Fri, October 6, 2006 20:13, Robert Goodyear wrote: Anyone know why I get "HTTP Connection Closed" on the display of a 7960 running a SIP image? Only seems to happen when registering against my Asterisk box from the WAN. I have 1:1 NAT happening on my firewall. Phones function perfectly otherwise. TFTP working fine across the firewall as well. Odd! Thanks in advance. -Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?
On Thu, Oct 05, 2006 at 03:05:40PM -0500, Joe wrote: > Actually the xclients could run the SIP clients but they have firewall > restrictions. > > I want to SSH to the machines which aren't behind the firewall and > pull the SIP client interfaces back via X Windows. ssh is a very poor choice for such a firewall. Read http://tldp.org/HOWTO/ppp-ssh/introduction.html#DRAWBACKS Now re-read the recent thread we had here about the extra overhead of RTP packets. With an ssh tunnel you'll typically get a data packet of 512 bytes for every RTP packet. Nice overhead. Consider using openvpn or something similar instead. Also: You did not mention where the the X clients reside. Maybe your task is actually to work with a decent networked sound server just as display is networked with X. Some interesting, but quite linux-specific resources regarding that may be found at http://wiki.ltsp.org/ . -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] No Dialtone
Hello Jay, I initially had it setup properly and after speaking with two different support techs at digium without having them remotely connect to the server I still did not have a dial tone. I experimented here in the office. I took a phone where I do I have a dial tone and plugged it in the card with the proper changes and the phone does not have a dial tone. I plug the newly purchased analog phone into the jack of the other phone and I receive a dial tone. Since installing this brand new card it has always worked this way. Digium support changed the text files by instructing me to do so. Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: Saturday, October 07, 2006 12:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No Dialtone On Sat, Oct 07, 2006 at 10:31:16AM -0500, Rich Adamson wrote: > If you've messed up in connecting telephone lines to the wrong module, > the ringing voltage sent to a fxs module will destroy it. You would need > to replace the module. I'm going to stick my neck out here, and opine that any FXS module that would be destroyed by receiving ringing voltage is *incredibly* poorly designed, and very probably wouldn't pass Part 68. "Shouldn't", certainly. Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] No Dialtone
Hello, If I am to understand what you wrote "The TDM400P (TDM22) will be damaged if the I plug the analog telephone line into the port 3 (which by the way is for the outside line connection via (PSTN) when it should be in port 1 or 2. I initially had it setup properly and after speaking with two different support techs at digium without having the remotely connect to the server I still did not have a dial tone. I experimented here in the office. I took a phone where I do I have a dial tone and plugged it in the card with the proper changes and the phone does not have a dial tone. I plug the newly purchased analog phone into the jack of the other phone and I receive a dial tone. Since installing this brand new card it has always worked this way. Digium support changed the text files by instructing me to do so. Reply? Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Saturday, October 07, 2006 11:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Dialtone If you've messed up in connecting telephone lines to the wrong module, the ringing voltage sent to a fxs module will destroy it. You would need to replace the module. Eddie Johnson Jr wrote: > Yes, I have and I received the following: > > In zapata.conf your first two channels should be fxs_ks because the first > two modules are FXO mdoules. Your last two channels should be fxo_ks because > the second two modules are FXS modules. > > For the TDM400P(TDM 22) the FXS modules work with the phone. The 3 port is > for the line. So I unplugged it from port 3, and plugged the analog phone > in port 1, made the changes to the channels and set immediate=no, restart > the server and activated asterisk. Nothing, my friend. > > Any more suggestions, > > Ed > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Francesco > Francesconi > Sent: Friday, October 06, 2006 10:23 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] No Dialtone > > Did you set immediate=no in zapata.conf? > > Francesco > > Eddie Johnson Jr wrote: >> >> >> Hello, >> >> >> >> I have the following setup: >> >> >> >> 1. Ubuntu Dapper Server 6.06 plus latest patches >> >> >> >> 2. Asterisk 1.2.11 >> >> >> >> 3. libpri 1.2.3 >> >> >> >> 4. Zaptel 1.2.8 >> >> >> >> 5. Digium TDM22 (TDM400P) >> >> >> >> 6. Analog phone plugged in port 3 >> >> >> >> 7. The wctdm, zaptel modules load at startup, I type asterisk as root and >> >> it is activated. >> >> >> >> 8. I check the Channel Map and I have the following: >> >> >> >> >> >> Channel map: >> >> >> >> Channel 01: FXO Kewlstart (Default) (Slaves: 01) >> >> Channel 02: FXO Kewlstart (Default) (Slaves: 02) >> >> Channel 03: FXS Kewlstart (Default) (Slaves: 03) >> >> Channel 04: FXS Kewlstart (Default) (Slaves: 04) >> >> >> >> 4 channels configured. >> >> >> >> I can ssh into the server and remotely connect to the server. Great! >> The card is not connected to an outside line as of yet but I have no >> dialtone on the phone. I spoke with a rep. at digium and was told a >> dialtone should be there. >> >> >> >> Zaptel.conf : >> >> >> >> >> >> loadzone=us >> >> defaultzone=us >> >> fxoks=1,2 >> >> fxsks=3,4 >> >> >> >> Zapata.conf: >> >> >> >> ;FXS Modules >> >> signalling=fxo_ks >> >> channel => 1,2 >> >> ; >> >> ;FXO Modules >> >> signalling=fxs_ks >> >> channel => 3,4 >> >> >> >> I made sure the card is not sharing an IRQ, I checked the hard drive >> and all is well. I load zttool and get the following: >> >> >> >> cat /proc/zaptel/* >> >> Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" >> >> >> >> 1 WCTDM/0/0 >> >> 2 WCTDM/0/1 >> >> 3 WCTDM/0/2 >> >> 4 WCTDM/0/3 >> >> >> >> Any suggestions? >> >> >> >> Ed >> >> >> >> >> >> ___ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
Re: [asterisk-users] No Dialtone
On Sat, Oct 07, 2006 at 10:31:16AM -0500, Rich Adamson wrote: > If you've messed up in connecting telephone lines to the wrong module, > the ringing voltage sent to a fxs module will destroy it. You would need > to replace the module. I'm going to stick my neck out here, and opine that any FXS module that would be destroyed by receiving ringing voltage is *incredibly* poorly designed, and very probably wouldn't pass Part 68. "Shouldn't", certainly. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 Licence Consumption Problem
Hi List: I have the next diagram: GSM G729 G729 IdeFisk -- Asterisk A - [INTERNET] Asterisk B - PSTN ( Via Unicall / Zap ) The user at IdeFisk Login as Agents on Asterisk B at this moment we have the next Licence Use: A) 1/1 B) 1/0 When a Call from the QUEUE on Asterisk B is Bridge to the Agent I have the next Use: A) 1/1 B) 1/3Any one can explain me this ?, why the incress of licence consumptions.Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Dialtone
If you've messed up in connecting telephone lines to the wrong module, the ringing voltage sent to a fxs module will destroy it. You would need to replace the module. Eddie Johnson Jr wrote: Yes, I have and I received the following: In zapata.conf your first two channels should be fxs_ks because the first two modules are FXO mdoules. Your last two channels should be fxo_ks because the second two modules are FXS modules. For the TDM400P(TDM 22) the FXS modules work with the phone. The 3 port is for the line. So I unplugged it from port 3, and plugged the analog phone in port 1, made the changes to the channels and set immediate=no, restart the server and activated asterisk. Nothing, my friend. Any more suggestions, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Francesconi Sent: Friday, October 06, 2006 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Dialtone Did you set immediate=no in zapata.conf? Francesco Eddie Johnson Jr wrote: Hello, I have the following setup: 1. Ubuntu Dapper Server 6.06 plus latest patches 2. Asterisk 1.2.11 3. libpri 1.2.3 4. Zaptel 1.2.8 5. Digium TDM22 (TDM400P) 6. Analog phone plugged in port 3 7. The wctdm, zaptel modules load at startup, I type asterisk as root and it is activated. 8. I check the Channel Map and I have the following: Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. I can ssh into the server and remotely connect to the server. Great! The card is not connected to an outside line as of yet but I have no dialtone on the phone. I spoke with a rep. at digium and was told a dialtone should be there. Zaptel.conf : loadzone=us defaultzone=us fxoks=1,2 fxsks=3,4 Zapata.conf: ;FXS Modules signalling=fxo_ks channel => 1,2 ; ;FXO Modules signalling=fxs_ks channel => 3,4 I made sure the card is not sharing an IRQ, I checked the hard drive and all is well. I load zttool and get the following: cat /proc/zaptel/* Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" 1 WCTDM/0/0 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 Any suggestions? Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound FXO call, getting "You must first dial..."
I did have a bit of trouble with searching (what to search on), though
looking for the w in the dial command did return quite a few hits as you
described. Thank you so much for taking the time to reanswer a covered
subject.
I played with the settings and 1 w and removing the :1 after EXTEN (not
stripping the leading digit?) makes it reliable. Not stripping the first
digit worked about 2 in 5 attempts. I stumbled onto that idea when I
missdialed a number "921503<7 digit number>" and it worked! The 2 was a
fat finger mistake. So I tried "90xx" and that worked.
As I have some success now, I can tune this so it works as the HowTo's
list. :)
Thank you again!
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Sat, 7 Oct 2006, Rich Adamson wrote:
Nick Ellson wrote:
I am not sure what I might be set up wrong, but dialing out with my Zap/1
port seems to alwyas get the "You must first dial a 1 when calling this
number" message from what sounds like the actual PSTN. My zapatel.conf and
extensions.conf bits below. Any advice? (I do receive inbound calls, and
it does sound like I am getting the PSTN error. I do notice that when I
get an inbound call, I have 5 secs of sevear static before it suddenly
becomes clear.. could that be happening on the outboud as well munging the
first few digits?)
signalling=fxs_ks
language=us
context=inbound_qwest
sendcalleridafter=2
callerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
channel=>1
exten => _9.,1,Dial(Zap/1/${EXTEN:1},60)
You should probably do a little research before posting questions like this
as its been answered many many time.
The problem is that "some" pstn central offices are not ready to receive dtmf
digits as quickly as what asterisk sends them. So, an option "w" has been
added to the Dial command to instruct asterisk to wait about 200 milliseconds
before sending dtmf. Try something like this:
exten => _9.,1,Dial(Zap/1/w${EXTEN:1},60)
and notice that lower-case "w" in the string. If that doesn't fix the
problem, try two "ww"'s in a row.
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Re: [asterisk-users] Outbound FXO call, getting "You must first dial..."
Nick Ellson wrote:
I am not sure what I might be set up wrong, but dialing out with my
Zap/1 port seems to alwyas get the "You must first dial a 1 when calling
this number" message from what sounds like the actual PSTN. My
zapatel.conf and extensions.conf bits below. Any advice? (I do receive
inbound calls, and it does sound like I am getting the PSTN error. I do
notice that when I get an inbound call, I have 5 secs of sevear static
before it suddenly becomes clear.. could that be happening on the
outboud as well munging the first few digits?)
signalling=fxs_ks
language=us
context=inbound_qwest
sendcalleridafter=2
callerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
channel=>1
exten => _9.,1,Dial(Zap/1/${EXTEN:1},60)
You should probably do a little research before posting questions like
this as its been answered many many time.
The problem is that "some" pstn central offices are not ready to receive
dtmf digits as quickly as what asterisk sends them. So, an option "w"
has been added to the Dial command to instruct asterisk to wait about
200 milliseconds before sending dtmf. Try something like this:
exten => _9.,1,Dial(Zap/1/w${EXTEN:1},60)
and notice that lower-case "w" in the string. If that doesn't fix the
problem, try two "ww"'s in a row.
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Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
Noah Miller wrote:
You can also change some settings in the zapta and zaptel
config.. to reduce
echo and interference on the line..
This is the most important thing here - what does your zapata.conf look
like?
zapta.comf
switchtype=national
This is not necessary in your case. It pertains to PRI lines, and not
the POTS lines you have.
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
You may want to turn each of these off, in turn, for testing,
especially the "echocancewhenbridged".
You can also tune the "echocancel" setting in terms of taps (a tap is
one sample from the data stream per second). You can use the values:
16, 32, 64, 128, or 256 ('yes' just means 128).
Might also try echotraining=800. That parameter causes the zaptel code
to wait 800 milliseconds before pulsing the pstn line, and that pulse
return is used to preload the software echo canceller to some reasonable
starting point. Not usre if this will have any impact on your problem,
but might be worth a try.
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Re: [asterisk-users] astcc help-pleasssssseeee
On Fri, Oct 06, 2006 at 10:39:18PM -0700, Ali wrote: > Hi, > > I am wondering if astcc has ever worked for someone because it always return > 0 for answeredtime! I tracked every bit of informaion on google and wiki and > finally found out that its because of asterisk returning to dial plan after > executing Dial, so astcc.agi runs through the end without wating for call > completion. > > Am I missing something crazy? please someone give me a hint. > > astcc doesn't use strict. This is perl code I wouldn't like to touch. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outbound FXO call, getting "You must first dial..."
I am not sure what I might be set up wrong, but dialing out with my Zap/1
port seems to alwyas get the "You must first dial a 1 when calling this
number" message from what sounds like the actual PSTN. My zapatel.conf and
extensions.conf bits below. Any advice? (I do receive inbound calls, and
it does sound like I am getting the PSTN error. I do notice that when I
get an inbound call, I have 5 secs of sevear static before it suddenly
becomes clear.. could that be happening on the outboud as well munging the
first few digits?)
signalling=fxs_ks
language=us
context=inbound_qwest
sendcalleridafter=2
callerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
channel=>1
exten => _9.,1,Dial(Zap/1/${EXTEN:1},60)
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
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Re: [asterisk-users] AEL2 Catching on?
Rushowr wrote: Is it just me or am I seeing more AEL2 code in people's examples? Could If you're a C programmer, then yes. But, if you're like me, with very little programming skills; no. AEL to me, is too much like looking at C code. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pop a web page with DID in url
If you are using queues, you may want to test the free version of QueueMetrics. You can pass anything to it through the dialplan, and you can use any phone you want as it will take care of the URL opening on the client side. Yours l. On Fri, 06 Oct 2006 19:10:27 +0200, Michael Sampson <[EMAIL PROTECTED]> wrote: Yes I would be interested in testing out your product. Does anyone have any other recommendations. A softphone would work for me. I would like something that had a chat feature like eyebeam does. I found another product called SNAP that will pop a web page, but it can only pass cid info not did. This is for an inbound call center project. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 Catching on?
Is it just me or am I seeing more AEL2 code in people's examples? Could it be that AEL2 is starting to finally catch on? SKM -AEL2 Fanatic, Potato Eater, and General Lurker ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling Functions from AEL2
Douglas Garstang wrote:
> I am trying to call the DUNDILOOKUP dialplan function from ael2, like this:
>
> context route {
> Set(PATH=${DUNDILOOKUP(${EXTEN},DUNDIRegistr)});
> }
>
> The DUNDILOOKUP function returns no data. However, when I call it exactly the
> same way in a regular context, it DOES return data.
>
> [route]
>
> exten => _X.,n,Set(PATH=${DUNDILOOKUP(${EXTEN},DUNDIRegistr)})
>
> That works. Could this possibly be an AEL2 bug? This is Asterisk 1.4 beta2.
>
> Doug.
>
>
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I'll look into what I can find as soon as time permits (my company is
entering a beta release today), but my first suggestion to debug what's
happening is to do 'show dialplan route' and see what the "compiled"
dialplan shows. That could help you figure out if it's an AEL2 bug,
because it would show what the AEL2 compiler did with that line.
Hope this helps,
SKM
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[asterisk-users] SIP stuck channel soft hangup?
I am seeing occasional stuck SIP channels that seem to occur when the fricking Nokia E60 drifts out of WIFI range in the midst of a call. This is particularly annoying when the stuck channels include my PSTN gateway (wellgate 3701a), which leaves incoming and outgoing calls a busy signal. I see by googling that soft hangup is a good way to kill these channels and that works fine for me. I wonder if there is some way to automatically soft hangup these channels when the qualify fails? Any ideas on how to automate and or fix this issue? Thanks, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: AGI() in 1.2 and 1.4
In article <[EMAIL PROTECTED]>, Douglas Garstang <[EMAIL PROTECTED]> wrote: > I was experimenting with FastAGI in Asterisk 1.4 and wrote some code around > it. I was using > the AGISTATUS variable to determine if I had been able to connect to the fast > agi server, > and act accordingly. > > 1.2 appears to be different. It has no such AGISTATUS variable, but more > importantly, it > appears that if you fail to connect to your FastAGI server, all dial plan > processing just > stops dead. Is there a way around this? Yes, I modified my 1.2 code to set AGISTATUS instead of just returning -1. If you mean "is there any way around it without touching C?", I think the answer is "No". Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pop a web page with DID in url
On 5 Oct 2006, at 20:43, Time Bandit wrote: I'm looking to do this. When a call comes in to an agent in a queue, pop a web page like this http://www.mydomain.com/cgi-bin/script.cgi?did=952900 Where did is the number the caller dialed to reach the system in the first place. I know Hudlite can do this we caller ID, but the DID feature is not there yet. Does anyone have any other software they know of that can do this? Some softphones support handling URL when you pickup the call. You can set that URL to anything you want from the dialplan. My MediaX softphone (current beta version) support it. Let me know if you want to try it Evn more shameless plug:-) Ours supports it on _ring_ which means you can see who it is calling _before_ you pick up! It also allows you to pick up on a hardphone but have the softphone drive your web experience. I think the asterisk-java guys have a server side solution where you embed a javascript/AJAX thing in your web page and it polls a servlet - which in turn listens for manager events. - Nothing to install on the client side. I'm sure you can do the same trick in other (web)serverside languages. Tim. hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP client that runs on Linux or Solaris through X Windows?
On 5 Oct 2006, at 21:05, Joe wrote: Actually the xclients could run the SIP clients but they have firewall restrictions. Use IAX on the xclients (I'm semi-serious) Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
