Re: [Asterisk-Users] chan-sccp-easter2005 make error with stable 1.0.6?

Just installed and so far it works fine for a 30VIP, only issue is the Speed dials they dial out OK, but the cisco phone doesn't break dial tone even when the other party has answered. Do you have any suggestions Regards Robb ___ Asterisk-Users

[Asterisk-Users] Excternip and FWD

Hi I have updated from CVS about a week ago and got the externip working with FWD for outbound calls., but I'm having problems with inbound calls, I don't think they are even reaching the Asterisk box even though I have forwaorded 5060 and the rtp range specified, another thing I have

[Asterisk-Users] IAX hard phone

Has any one seen or heard of the lastest developments fo the Farfon IAX phone? the web site Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Distintive Ring on x100p

even print them 6 times instead of 3.. Ie if it prints DEBUG: 327 DEBUG: 0 DEBUG: 0 then that ring is 327,0,0 ( but to be safe lets round it to 5's) and make it 325,0,0 only have to hit within 10 +/- for it to match. bkw On Thu, 13 Nov 2003, Robert Boardman wrote: Thanks again Brian, one

[Asterisk-Users] HELP!!!! Having problems Starting Asterisk

I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I

[Asterisk-Users] HELP!!!! Having problems Starting Asterisk

I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have

Re: [Asterisk-Users] HELP!!!! Having problems Starting Asterisk

Tim Sailer wrote: On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote: I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4

[Asterisk-Users] Cisco 30VIP Phones

Hi Has anyone go the 30VIP phone to work with asterisk? If so how good us the usability of the Cisco 30VIP phone with asterisk either using chan_sccp or Chan_skinny? Thanks for your Help Robb -- Robert Boardman Tel:01617737929 FWD:86263

Re: [Asterisk-Users] HELP!!!! Having problems Starting Asterisk

John Fraizer wrote: Robert Boardman wrote: Tim Sailer wrote: On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote: I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe

[Asterisk-Users] Incomming Distinctive ringing

Hi I have had distinctive ringing working before the patch was applied to the CVS tree, now it doesn't work, Could anyone point me in the right direction to debug distinctive ringing? Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Chan_skinny

Hi All I have been working in getting 2 Cisco 12 sp phones working, they work fine over the lan, but I cannot get it to work across the Internet I only have one way voice Do es anyone have any advice Thanks for your help Robb ___ Asterisk-Users

Re: [Asterisk-Users] 12SP

mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Boardman Tel:01617737929 FWD:82623 ___ Asterisk

[Asterisk-Users] DPNSS and Asterisk

Hi Just one question do any of the Digium T1/E1 cards do DPNSS signaling? Robb -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 -- Robert Boardman Tel:01617737929 IAXTel:17007737929 FWD:82623 ___ Asterisk-Users mailing list [EMAIL

Re: [Asterisk-Users] PRI and Voicemail Memory increasing

Steven Critchfield wrote: On Wed, 2004-03-03 at 06:33, Robert Boardman wrote: Hi, With the current CVS as of last night 20:00GMT I was testing a asterisk with the e100p card using a PRI analyzer to excerise the 30 channels over and over, just going directly to voice-mail. Basically, I don't

[Asterisk-Users] Cisco VIP30

Hi Just got a brand new Box Cisco VIP30 off ebay, the standard phone functions work fine, just a couple of questions, 1) how do I program the other buttons not on the standard keypad part.. 2) When I hang up the display doesn't clear and keeps the numbers just dialed on screen, can this be

[Asterisk-Users] voicemail

Hi I have an asterisk voicemail system connected directly to a pri, there are no extensions connected to the asterisk box, anyway my questions is, can I get asterisk to call an associated phone number when a voicemail box has a message? Thanks in advance for your help Robb

[Asterisk-Users] Chan_sccp How-to

I wonder if anyone could post a how-to for the chan_sccp, I've downloaded and compiled the code, but I don't know where to go from here, any help would be appreciated Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] UK - 1471

In the UK we have a service that if you dial 1471, the last 6 calls are read out to you and you can pick which one you want by pressing 3, this means that 1471 shows in the cdr, has anyone created a script or an application that will read out the last callers and then dial the number? ( that

[Asterisk-Users] Budgettone 100 phone Configuration

Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] All extensions busy

Hi Firstly could I thnk everyone who has helped me so far, I just have a couple of queries I have not had chance to debug this much yet but When using the tdm40p all extesions busy themselves out, and * cannot rint the extensions for incoming calls is this because I don't have a hangup

[Asterisk-Users] Installing the wcfxs driver

Hi All when I modprobe the wcfxs drive and do a cat /proc/pci, it is sharing irq with my AGP and USB, I think this is causing the card to stop working, it would work for a couple of days or a couple of hours but then stop, I'm a complete linux newbie, how can I force the wxfxs driver onto

Re: [Asterisk-Users] Installing the wcfxs driver

Hi My Motherboard cannot disable the IRQ sharing, can I specify on with modprobe, the IRQ to be used with a particular module? Robb Quoting Emanuele Pucciarelli [EMAIL PROTECTED]: On Mon, Jun 16, 2003 at 12:05:34PM +0100, Robert Boardman wrote: for a couple of days or a couple of hours

[Asterisk-Users] limiting out going calls to a maximum duration

I want to limit my sons phone useage, by setting a 30min limit on out going calls from his room is there a simple way of doing this with asterisk? Thanks for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] OT: Grandstream power supplies..

When I was looking for a psu the only site I found for 5V ( and a decent price) was CPC, but I cannot remember the www address robb Quoting WipeOut . [EMAIL PROTECTED]: Hi, Quick question to all the electronics gurus out there.. I unpacked my second GS phone yesterday (had it for about

[Asterisk-Users] call Intrude

Hi I have looked through the wiki and search the mailing list, but I cannot find a way to intrude on a call, can asterisk do this feature? if so how? Thanks for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] ISDN TA

I have an ISDN TA that has 2 POTS interfases (FXS), can these be used with asterisk? Thanks in advance Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] UK Caller ID and X100p

Hi I really need caller id to work in the UK, I understand that the X100p uses a US chipset,two questions 1) is that a product that converts UK to US caller id in line or 2) would it be possible to have modem that supports CID in parallel with the line and the x100p.The modem reads the line

[Asterisk-Users] Dect Phone

Hi I have a problem with a new DECT phone I have bought The key pad works like a Mobile phone where you dial first then pick up the line, but it seems to dail too fast or spuriously, ie 012826736464 show on thew Asterisk console as 0012282677, could any one offer advice how to fix? Also when

[Asterisk-Users] Distinctive ringing

Hi I've just signedup for Distinctive ringing on my PSTN line in the UK, could anyone explain what I need to add in the conf files to detect and route based on in comming Distinctive ringing Thanks in advance for your help Robb ___ Asterisk-Users

Re: [Asterisk-Users] Distinctive ringing

Does asterisk know when each ring comes in or just the first ring, ie so the cadence can be worked out? say over two rings? Robb Martin Pycko wrote: The X100P together with asterisk does not support the distinctive ringing detection on the line. Asterisk however can generate the distinctive

[Asterisk-Users] Recall doesn't seem to work

Hi I'm having a problem where the recall button doesn't work If i press recall before I dial numbers it disconnects me which is what I would expect, but during a conversation if I want to transfer the TDM 400 just ignores the recall Any advice would be gratefully received Thanks Robb

Re: [Asterisk-Users] Recall doesn't seem to work

with asterisk) line 792 #define ZT_MAXPULSETIME (150 * 8) I moved it to (20 * 8) be sure not to set it under ZT_MINPULSETIME, that's (15 * 8) Matteo Il ven, 2003-09-19 alle 22:04, Robert Boardman ha scritto: Hi I'm having a problem where the recall button doesn't work If i press recall before I dial

[Asterisk-Users] MY Sql CDR

Could someone point me in the right direction for setting up the mysql cdr function Thanks robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Recall doesn't seem to work

Thanks that worked Robb Brancaleoni Matteo wrote: mmh... have you enabled threewaycalling = yes transfer = yes in zapata.conf ? matteo. Il sab, 2003-09-20 alle 10:51, Robert Boardman ha scritto: Thanks for the advice Matteo but it didn't work, anthink else I may of missed? Robb

[Asterisk-Users] CPU Optimisations For asterisk

How would I compile asterisk for the Athlon XP arch, would there be any advantage doing this? Thanks for your Help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] CPU Optimisations For asterisk

Thanks for your reply The system is a small installation but I was thinking about optimizations and wondered if there would be any particular benifit anyway thanks for the reply, your comments are very useful robb Quoting Alastair Maw [EMAIL PROTECTED]: Robert Boardman wrote: How would I

[Asterisk-Users] Caller Id AGI Script

As you my be aware the X100p cannot collect uk caller id, now I have a modem and a perl script that creates a file /etc/asterisk/callerid.txt on each incoming call in the format number,date,time,name over writing each time a new call comes in I can asterisk read this file and populate the

RE: [Asterisk-Users] Caller Id AGI Script

Hi Dave I've not completed the script yet, But you may not like this but I've had to use a win98 box for the zoom 3025C (important its the C model), the zoom modem is the only internal one I've found that can do uk caller id (but its not supported on the linux driver), and is still

[Asterisk-Users] Odd ringing conditions

I have two questions about incomming ring and extension ringing 1) When an incoming call is detected by asterisk it takes 2 or three rings before the internal phone ring does anyone know how I can fix this? 2) All internal phone ring on an incoming pstn call but after the call is answer all the

Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php

Hi I'm trying to install but I think I have a problem!!! Would I be correct in saying if I don't have the jp graph libs, the links on the form would be followed but nothing would be displayed Areski wrote: I made an Update, now don't need register_globals on anymore... By the way, I fix some

Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php

Yes php sysinfo say gd is complied inb any other clues? Robb Areski [EMAIL PROTECTED] said: Do your php support GD ? You can simply check it with a phpinfo ! More info about gd (configuration, installation) : http://www.php.net/image On Wed, 2004-03-24 at 21:12, Robert Boardman wrote

Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php

, 2004-03-24 at 21:12, Robert Boardman wrote: Hi I'm trying to install but I think I have a problem!!! Would I be correct in saying if I don't have the jp graph libs, the links on the form would be followed but nothing would be displayed Areski wrote: I made an Update

[Asterisk-Users] agi and stream_file

Hi, I trying to get agi with perl to stream a gsm file , and wait for a digit , the agi gets to the stream but doesn't play back, could some one explain how this works here is a snip it of code open(DAT,/etc/asterisk/1571.log) || die(Cannot Open File); while( $sth-fetch() ) { print DAT in

[Asterisk-Users] Compiling Zaptel 0.9.0 drivers

Hi I'm trying to compile the Zaptel Drivers, but I seem to be getting an error zaptel.c:131: warning: data definition has no type or storage class zaptel.c:132: error: parse error before config_must_be_included_before_module zaptel.c:132: warning: type defaults to `int' in declaration of

Re: [Asterisk-Users] small linux distro to run * in old boxes

Hi Victor I'm currently working in a Linux Distro, it is being internal alpha testing by my self and a couple of me my colleagues, over the next couple of weeks I'm hoping to release a beta version to the asterisk community., I'll keep you posted via asterisk users, about its features as it

[Asterisk-Users] recommend a Linux based TFTP server

Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Caller ID with BT CD50

First of all thanks for the patch it works great, but i think it breaks the distinctive ringing, I have 2 incoming numbers in one x100p in contexts home1 and home2 but 'default' is always chosen has anyone else seen this? if you need any more info just ask Robb Tony Hoyle wrote: David J Carter

Re: [Asterisk-Users] RE: bt communicator`

change it. There are probably unneeded lines above. Regards Peter -Original Message- From: Robert Boardman [mailto:[EMAIL PROTECTED] Sent: 09 October 2004 21:40 To: Whisker, Peter Subject: bt communicator` Hi Peter I have been following your post but didn't see the other emails about

Re: [asterisk-users] off-topic: Avaya 46xx, release 032207 ... help

Cesc Santa wrote: Hi, I am trying to use an Avaya 4602 phone, which I just updated from a very old SIP software to the latest I could find on avaya's site (032207). The upgrade went fine and it gets registered on the Asterisk server. Now, a couple of glitches, though. - The phone's web

[asterisk-users] Having problems posting to the list

Hi All I'm having problems posting to this list, no bounces the mails just dont show any advice how to get the postings through is there filtering? robb ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing

Re: [Asterisk-Users] Distintive Ring on x100p

me too being a patch newbie how do you apply the patch and are the three comma seperated values equivalent to the dron and drof on the modems? I ask because the dron and droff, using my modem arent always say 5, sometimes there 4 Robb --- Original Message --- From: John Vozza [EMAIL

Re: [Asterisk-Users] Distintive Ring on x100p

Thanks for your help Brian how would you come the the values required for the distincive ring? Robb Brian West wrote: cd /usr/src patch -p0 file.diff bkw On Thu, 13 Nov 2003, Robert Boardman wrote: me too being a patch newbie how do you apply the patch and are the three comma

Re: [Asterisk-Users] Distintive Ring on x100p

it 325,0,0 only have to hit within 10 +/- for it to match. bkw On Thu, 13 Nov 2003, Robert Boardman wrote: Thanks again Brian, one more question if i may ( soory for the hand holding) I've added the line below should the information show up when I am in asterisk gc, what do I have to do

Re: [Asterisk-Users] 4 Port FXO cards

will this port sort out UK caller id? --- Original Message --- From: Mark Spencer [EMAIL PROTECTED] Sent: Wed, 19 Nov 2003 17:58:01 -0600 (CST) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 4 Port FXO cards We *are* making progress, and i have a running prototype, however the production

RE: [Asterisk-Users] Re: [Asterisk] GSM access

Hi All Maybe this would be a beter solution, but you may have to buy directly from them http://www.artech.com.tw/html/gx100e/gx100e.htm Robb --- Original Message --- From: David Luyens [EMAIL PROTECTED] Sent: Mon, 24 Nov 2003 14:14:10 +0100 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users]

[Asterisk-Users] grand stream phone and double nat

Hi I'm trying to configur a grandstream BT101 to connect to asterisk, both behind different NATs, I realise that a double Nat is a problem, I have tried using fwd forwarding to iaxtel as a solution but cannt seem to get them to connect as I think there is a codec problem as IAXTEL doesn't

Re: [Asterisk-Users] Client for P800/P900

Peer Oliver schmidt wrote: Nicolas Bougues wrote: On Wed, Jan 07, 2004 at 01:42:30PM +, Andreas Anderson wrote: Hi Guys, is there a client which can be used on the SonyEricsson P800/P900...? IAX would be cool, but i take anything that can connect (via bluetooth) to an asterisk-server ;-).

[asterisk-users] MixMonitor fdiles

Hi, I have a load of files recorded with MixMonitor that are out of sync ie one leg of the call is 2-3 seconds behind the other, is this a bug in Asterisk 1.4.18, or am I possibly doing something wrong Is it possible to edit the file and re sync the a/b leg? Thanks for your help Robb

[asterisk-users] Outbound PRI ISDN 30 problems

Hi All I'm having problems with outboud ISDN calls, They setup OK , and ring the other end OK, but when the call is answered I get a disconnect cuase 17 with an error message in the console of [Apr 15 08:06:13] DEBUG[4361] chan_zap.c: Found empty available channel 0/31 [Apr 15 08:06:13]

[asterisk-users] ISDN Call Droping only for outgoing

I have been trying to sort this out for a while now but with no luck I have isdn - asterisk- pabx on a te205 incoming calls work fine outgoing calls seem to work fine but the call is dropped when answered I think it is to do with the line [May 8 17:51:55] WARNING[4762] channel.c: Unexpected

[asterisk-users] SERVICE CODES

Hi I'm trying to get the status of an extension that has DND set using the service code, or trying to disable the service codes altogether so that I can do them in the dialplan if needed any advice wout be appriciated Thanks Robb ___ -- Bandwidth and

Re: [asterisk-users] MWI with Siemens Gigaset S450IP

Olivier wrote: 2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi, 1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP it is mentioned MWI is now working. In my testings with lastest 02123 firmware, MWI is blinking when missed calls but

Re: [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)

Olivier wrote: 2008/10/5 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Kevin P. Fleming wrote: Olivier wrote: 2. R Hook-flash key is now available to transfer calls. In s450IP web management server, its defaults

[asterisk-users] Transfercapability DIGITAL

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for

Re: [asterisk-users] Transfercapability DIGITAL

yes and it is still set to speech I've even tried to port the old patch here http://bugs.digium.com/view.php?id=6251 to the system with no luck robb Melcon Moraes wrote: Have you tried: exten = s,n,SetTransferCapability(DIGITAL) ? []'s MM -Original Message- From: robert

Re: [asterisk-users] Transfercapability DIGITAL

seen the bearer capability in asterisk or is the call nat working? I've seen that a digital call shows up as speech. You are using Zap? Or are you using mISDN? Cause there you have to set an extra parameter in the dial statement. chris... robert boardman schrieb: yes and it is still set

Re: [asterisk-users] Help with asterisk and avaya SIP trunking

Krishna Sumanth Chava wrote: Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or

Re: [asterisk-users] Help with asterisk and avaya SIP trunking

the Asterisk and avaya talking to each other. Thanks Krishna On Fri, Nov 7, 2008 at 2:59 PM, Robert Boardman [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Krishna Sumanth Chava wrote: Hi * Users, I ran into a problem when I was trying to communicate an avaya

Re: [asterisk-users] * + Legacy PBX works but strange problem

Sriram wrote: Hi below are my configs: pstn(e1)---asterisk (span1)-legacy pbx(connected via span2)- legacy pbx analog extensions. my dial plan is like callers dial into asterisk(span1) , hear an IVR option and they are connected to the agents via the legacy pbx (which is in

[asterisk-users] ISDN Cause codes

Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).*

Re: [asterisk-users] ISDN Cause codes

-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Thursday, November 20, 2008 4:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ISDN Cause codes

Re: [asterisk-users] ISDN Cause codes

Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Thursday, November 20, 2008 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ISDN Cause codes Hi All Just been

Re: [asterisk-users] ISDN Cause codes

of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Friday, November 21, 2008 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

[asterisk-users] DTMF Tones

Hi All I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even on a zap to zap bridged call is this possible? Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] DTMF Tones

zap channel on one card to zap channel on another Robb Alex Balashov wrote: You mean a zap-to-zap call hairpinned into the same adaptor, or another one? Robert Boardman wrote: Hi All I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even

Re: [asterisk-users] DTMF Tones

thanks Found that but sometimes I need to detect dtmf ie when playing back a recording Robb Philipp Kempgen wrote: Robert Boardman schrieb: I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even on a zap to zap bridged call is this possible

[asterisk-users] Echo Cancelation

Hi All I Have an ISDN 30 circuit passing through an asterisk box to a legacy pbx, all is working well but I have had a problem that modems do not work, I thought of turning off echo cancelation but I cann t seem to find the ial switch do do it, could someone point me in the right direction to

[asterisk-users] HFC Single port Cards

Hi all Been messing about with the single port cards for a number of years, but never got good results, I was thinking of giving them another go over Christmas and was wondering if anyone would share there recent experience, as to which driver works best MISDN BRISTUFF etc with the latest

[asterisk-users] Warnings during a compile

Here is just one example of a warning when compiling asterisk on Ubuntu 8.10 manager.c:1760: warning: ignoring return value of âreadâ, declared with attribute warn_unused_result is this anything to worry about? can i safely ignore it? Thanks Robb

Re: [asterisk-users] Warnings during a compile

On 04/02/2009 00:24, Mark Michelson wrote: Robert Boardman wrote: Here is just one example of a warning when compiling asterisk on Ubuntu 8.10 manager.c:1760: warning: ignoring return value of âreadâ, declared with attribute warn_unused_result is this anything to worry about? can i

[asterisk-users] ISDN30 Channels Locking

Hi Had an issue today where all channels connected to the telco when dialed returned WARNING[15366] chan_zap.c: Call specified, but not found? in the logs, when I removed the isdn cable and reinserted everything was fine any ideas? software Versions asterisk-1.4.21.2 zaptel-1.4.12.1

Re: [asterisk-users] Asterisk and Data Modem

Jon Morgan wrote: Hi All, We have an Asterisk 1.4.21.2 box which uses a 2 port Digium card to bridge calls, as follows: ISDN Provider --- Span 1(pri_cpe) --- Span 2(pri_net) Phone System The company that looks after our internal phone system can no longer dial in using their data

[asterisk-users] Mix Monitor call quality

Hi trying to record calls using mixmonitor, but I'm having problems with call quality the call seems OK but then it drops frames with silence ( for less than 0.5 seconds) then call continues All I'm doing is bridging two zap channels and recording no transcoding or changes to the channels

[Asterisk-Users] BT Communicator (SIP???) and Asterisk

Hi All BT are providing a SIP gateway for PSTN through the BT communicator with Yahoo Messenger, I have done an ethereal trace and found that the BT Communicator side of the software is using SIP, so in theory I could add more PSTN lines to Asterisk for BT using SIP, but I am having problems

Re: [Asterisk-Users] BT Communicator (SIP???) and Asterisk

gARetH baBB wrote: On Fri, 20 Aug 2004, Robert Boardman wrote: BT are providing a SIP gateway for PSTN through the BT communicator with Yahoo Messenger, I have done an ethereal trace and found that the BT Communicator side of the software is using SIP, so in theory I could add more PSTN

Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p

should this work with the x101p? or just the tdm400? Thanks for your help Robb Edward Eastman wrote: Brilliant - thanks, took me half an hour but it's working now. Just for the record, settings as follows: The patch on http://bugs.digium.com/bug_view_page.php?bug_id=009

Re: [Asterisk-Users] UK Callerid bug #1719 TDM400p

. Dan On Fri, 2004-09-10 at 17:38, Robert Boardman wrote: should this work with the x101p? or just the tdm400? Thanks for your help Robb Edward Eastman wrote: Brilliant - thanks, took me half an hour but it's working now. Just for the record, settings as follows: The patch on http

Re: [asterisk-users] Asterisk and Avaya IP Office

Just done this for a client using an E1 Pri card in the avaya box and a sangoma a102, using qsig , works fine, I wouls recommend this to any oneits been up and stable for two months now Regards Robb housi mueller wrote: The main goal is that any extension from the Avaya PBX can make long

Re: [asterisk-users] Asterisk to make multiple extensions simultaneous calls on a single telephone line

hi vincent, In the UK you can have multiple pots lines with the same telephone number. but you would need more fxo lines for this. Regards Robb Vincent Li wrote: Hi Lists, I have one box with two FXO and two FXS ports, it is running asterisk inside. I have one sinle POTS line connected

[asterisk-users] Control playback

Hi All I have been asked if it is possible for an external application to be aware of the position of the playbcak of a file with control playback ie a file is playing and the user hits the fast forward button , is there a manager event that show how far into the file it has been played?

Re: [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken

Tzafrir Cohen wrote: On Wed, Jan 02, 2008 at 07:41:40PM +, Russell Brown wrote: Don't you just hate it when something was working and when you come to use it in anger it's broken :-( Something in the, fairly, recent series of Asterisk updates has broken DIGITAL call passthrough.

Re: [asterisk-users] 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken

Tzafrir Cohen wrote: On Thu, Jan 03, 2008 at 12:24:38AM +, robert boardman wrote: I have an outstanding problem with this,I have found that if you set overlapdial to no on the internal leg ie connected to the pabx it works, but you will have to set the pabx to dial en-block ie send

[asterisk-users] problems with zaptel and Udev

Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at Starting udev if I remove thew tdm400 it boots OK, but no zaptel has anyone seen this , and can offer any advice? Thanks Robb

Re: [asterisk-users] problems with zaptel and Udev

Tzafrir Cohen wrote: On Sun, Jan 13, 2008 at 05:33:58PM +, robert boardman wrote: Hi I have had a Centos 5 installed with asterisk and zaptel for a couple of weeks, I had to reboot eh machine today, and when it rebooted it got stuck at Starting udev if I remove thew tdm400 it boots

Re: [asterisk-users] problems with zaptel and Udev

thanks for the reply I'm already on 1.4.7.1 regards Robb Ed Nunez wrote: I had the same issue and updated my Zaptel drivers to version 1.4.17 and it's rebooting fine now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of robert boardman Sent

[asterisk-users] Linksys SPA922

Nearly got an SPA922 phone working behind a NAT, the phone registers, and I can dial out and have two way speech, on an incoming call the SPA922 rings I answer and the SPA922 shows Anwsering but never does and the far end continues ringing until the voicemail answers, this then show as a

Re: [asterisk-users] Linksys SPA922

and/or you’re timing out. Also sip.conf and user.conf would be helpful as well as Asterisk release. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *robert boardman *Sent:* Friday, August 07, 2009 9:01 AM

Re: [asterisk-users] Digium PRI cards for data usage?

Do you have to set aside kines for the data channels or can you have dynamic data channels, for example ISDN dialup internet backup? Robb 2009/9/1 Tim Nelson tnel...@rockbochs.com - Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Monday 31 August 2009 21:59:28 Tim Nelson

[asterisk-users] portech MV-378 SIP GSM Gateway

Hi All I having an intermittent problem with the above mobile gateway and would appriciate some advice basically 1 in 10 calls fail at some point during the call, the duration of the calls ate completely different call progression Call comes in from Zap channel and dials a mobile number on the

[asterisk-users] Home line noise problem

I Have a home line connected to a tdm400p with 3 extensions and a siemens sip-dect , it seems to work fine but during a call there is always a digital squeal every so often does anyone know what this could be? Robb ___ -- Bandwidth and Colocation

  1   2   >