RE: [Asterisk-Users] Connecting to Commander NT132
Hi Nick, The Commander NT132 is essentially a rebadged Nortel Norstar MICS - I believe with a slightly different firmware for Australian conditions. I have the smaller version - Commander NT40 (which is a rebadged Norstar CICS) connected up to Asterisk via the Norstar's built in Analog port. You can get analog ports for the Commander NT132 if you don't have any... I don't know how much they cost, but you can often find second hand parts for these systems relatively easily. Actually I wish I had a Commander NT132 for my setup as I ended up tagging 3 analog adapters off the NT40 system (which take up digital extension ports) and still need more analog ports. The external ATA's are so bulky and awkard, at least with the NT132 you can buy a proper analog card - I belive they have about 8 ports or so to plug directly into the system, great if you have a need for quite a few analog extensions. There was some good doco written up by David Gomillion on intergrating the Norstar MICS with Asterisk, but it's based on using PRI (Primary Rate Interface) cards. http://www.voip-info.org/wiki-Asterisk+Interop+Nortel+Norstar+MICS Integration with analog FXS port (on the NT40 anyway) is relatively straightforward - but a big downside of these smaller Nortel systems is they don't offer up any disconnect supervision on the FXS ports (and in my case I also have a 4port FXO trunk card in the system which also lacks disconnect supervision). This means when someone calls from the NT40 into my asterisk system (through a X101P card), if they hangup first the X101P doesn't get any signal that the line has hungup. It was pretty annoying to start with, but through ensuring my dialplans all correctly hangup the line from my end when they're supposed to, and watching for silence in voicemails, then much of the trouble is solved and I find it works quite well for what I need. Another problem with integrating via analog is that you don't receive any CallerID info from the NT40... So if someone calls from the NT40 and I miss the call, I have to do a bit of guesswork to figure out which extension to try calling back to find the person who was looking for me. As for integrating via ISDN with a Fritz card, I'm not sure, someone else might be able to answer that better... I asked about this quite some time ago about interfacing the NT40 via BRI - I believe you need a card with a HFC chipset like this one - http://www.junghanns.net/asterisk/page17.html My previous question - Norstar Integration with Asterisk via FXO or BRI ISDN http://lists.digium.com/pipermail/asterisk-users/2004-February/035729.ht ml It's not cheap, and hence why I haven't really pursued integration of my system via ISDN... from what I've read it seems PRI really is the best way to integrate, which is a downside if you only have BRI interfaces lines. Hope this helps some... The most important thing to know is that what you have is a Nortel Norstar MICS system, which makes searching for information much easier as they are very common systems in the states. Regards, Chris Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Cobley Sent: Friday, 22 October 2004 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Connecting to Commander NT132 Hi, I am looking at connecting Asterisk upto a Commander NT132. I need 2 lines, and initially was going to connect it up to some analog ports, which I have since discovered they don't have. So I am taking another look at other options rather than getting a couple single line cards. Now firstly, would I be able to install something like a Fritz ISDN card and hook that upto the PABX? I know this can be done with say an E1 card but was not sure if the same applies with an ISDN card connecting to the PABX rather than the carrier. If someone could perhaps summarise my options I would appreciate it. We will be installing a number of systems in exactly the same configuration, so I need to try and understand these areas a little more. BTW I am in Australia if that makes a difference. Kind regards Nick ___ Asterisk-Users 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 ___ Asterisk-Users 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
RE: [Asterisk-Users] Problems with voice menu
I having a lot of troubles to configure a simple voice menu. In extensions.conf I have the following. [incoming] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,DigitTimeout,10 exten = s,4,ResponseTimeout,20 exten = s,5,Background(itranser/msg_bienvenida) exten = 1,1,Goto(contexto_extensiones,s,1) exten = 2,1,Goto(contexto_operadora,s,1) The context refered by the menu. (each context play me a diferent message only ) [contexto_operadora] exten = s,1,Background(itranser/trans_operadora) exten = s,2,Dial(SIP/ismael,s,1) [contexto_extensiones] exten = s,1,Background(itranser/msg_pasar_ext) I've made the corrections to your context's above... Note in particular the Goto command and then using the 's' (start) extension in each extension line, also adjusted the priority numbers. For more info on Goto http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Goto Give that a try and see how you go. Regards, Chris Lee ___ Asterisk-Users 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
RE: [Asterisk-Users] Help with dialing out with TDM400P
Incoming calls work and the phone rings and can be answered
no problems, (although I wouldn't mind being able to adjust
the ring but that's not important), I can't ring out, I just
get a busy signal and nothing comes up on the console. I am
pretty sure its just a simple line missing from extensions.conf.
In your [internal] context try something like..
exten = _0.,1,Answer
exten = _0.,2,Dial(Zap/g1/${EXTEN:1})
exten = _0.,3,Hangup
This way Asterisk will send all the digits dialled after the 0 to the
zaptel card and you should be dialing out. You may not need the
answer/hangup lines for your setup.
2.
I am based in australia and when I have an incoming call with
callerid turned on then I get the following error on console.
-- Zap/1-1 is ringing
Sep 25 22:49:14 WARNING[-203428944]: chan_zap.c:3413 zt_handle_event:
Didn't finish Caller-ID spill. Cancelling.
I'm not sure if this is related with inbound CallerID on an FXO, but to
get Caller ID working on an FXS port I had to make this change to the
chan_zap.c file and recompile:-
http://lists.digium.com/pipermail/asterisk-users/2004-August/057349.html
In /usr/src/asterisk/channels/chan_zap.c
#define DEFAULT_CIDRINGS 2
The default is 1.. Seems we need this set to 2 in Australia, I dare say
making this change might get the inbound caller ID working for you also.
Hope this helps,
Chris Lee
___
Asterisk-Users 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
[Asterisk-Users] Atick Certification on FXO Modules (Australia)
Out of interest is there any estimated date for the TDM400 FXO modules receiving A-tick certification? And has anyone compared the FXO modules with the X100P on Australian exchanges/equipment? Do they perform any better than the X100? Cheers, Chris Lee ___ Asterisk-Users 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
RE: [Asterisk-Users] Called ID in Australia
The only change I believe I had to make was under /usr/src/asterisk/channels/chan_zap.c #define DEFAULT_CIDRINGS 2 The default is 1 Google search if you want some of the previous threads... http://www.google.com.au/search?q=asterisk+callerid+patch+australiaie=UTF-8 hl=enmeta= -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Robert Barnes Sent: Tuesday, 3 August 2004 8:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Called ID in Australia Hello All, Can any Australians who have any info or current patches relating to Caller ID in Australia please drop me a line? There is little or no info on the Wiki regarding this topic, although I am aware of a related patch mentioned in the bug tracker. Regards, Rob Barnes ___ Asterisk-Users 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 ___ Asterisk-Users 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
RE: [Asterisk-Users] VoIP hackers gut Caller ID
hi... here in Italy is almost impossible to set an invalid cid, if is out of your allowed space. ie. if you have X numbers on your PRI, you can only set one of these. nothing more. on bri you simply cannot do nothing. just my 2 cents. Indeed I've noticed here in Australia on BRI-ISDN (2x B channels) with DID I can't spoof numbers to the exchange... it's been a while since I toyed with the system, but from memory I could attempt to set any 9 digit number I wanted for the CallerID string, however the exchange would not allow that to go through and instead passed through the correct group directory number (primary number) for the service. However if I set the CallerID digits to anywhere within our 100-number block DID range, the exchange will happily pass on the specific number... guess it might be a combination of Euro ISDN standards and how the local telco's configure the exchanges. ___ Asterisk-Users 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
RE: [Asterisk-Users] Asterisk with Ericsson MD110 PBX
I don't have any direct experience with the MD110's and Asterisk, but I would envisage the MD110 digital phones are very much a proprietary protocol, as with Nortel digital phones, you can't mix and match between different vendors. It may be possible to get Ericsson (as well as Nortel and others) digital phones working with Asterisk, I doubt via ADSI, more likely via Dialogic voice boards from Intel. I know after some digging through intel.com for info on their Dialogic voice boards I found some technical info on the signaling used for Nortel digital phones. To successfully get a Ericsson/Nortel/etc digital phone working with Asterisk you'd need to firstly purchase the Dialogic voice board(s), then write the drivers for the Dialogic board for Linux (or maybe they already exist, I haven't checked), then some more drivers/plug-ins to get the Dialogic and the vendor-specific digital phones working with Asterisk. I imagine for the most part, depending on how many phones you have and budget, it really wouldn't be economically feasible - in the long run I think you'd find replacing the phones with SIP handsets and trying to sell off the old digital handsets to recoup some of the upgrade cost would be the way to go. Actually if memory serves, the main purpose of the Intel Dialogic boards is actually interfacing the PC (ie Asterisk or other software) to the digital ports of the proprietary PBX, rather than directly interfacing the PC to the proprietary digital phones. So for instance if you wanted to smoothly transfer calls between the Asterisk SIP extensions and the Ericsson MD110 handsets with all the caller ID details, or perhaps run a fancy IVR or auto-attendant system accessible to the MD110 handsets via Asterisk then they'd be ideal. Otherwise you have to interface via other digital trunk methods or Analog extensions and may not get access to as many features as you can through the digital extension ports. Even if you can use the dialogic boards directly with the proprietary handsets, I can't see the solution really scaling anywhere near as well as the proprietary digital cards that plug into the MD110 PBX itself. Of course it would be nice to see the Ericsson/Nortel phones recycled for use with Asterisk systems, but at this point in time I'm not sure how feasible this is. I do believe Nortel were working on (or perhaps have now released) a small black-box solution that plugs into the existing proprietary Meridian handset and then plugs into Ethernet to essentially turn the phone into a VoIP handset - not sure if it uses SIP protocol. If Ericsson have a black-box solution like this available, then it might be a feasible approach, depending on the cost per box and the existing network infrastructure, as ideally you'd have the black boxes powered over Ethernet so you can install UPS' in the communications cabinets to ensure the phones and network are available during power outages. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julian Pawlowski Sent: Wednesday, 2 June 2004 7:28 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk with Ericsson MD110 PBX I was just wondering if someone has experiences to use Asterisk in an existing Ericsson MD110 environment. Particulary I'd like to know if it is possible to use the MD110's system phones directly connected to Asterisk. I'm not very familiar with it but would it be possible to use ADSI with these phones? Are they more like analog or more like digital phones or is the protocol even more proprietary? Regards Julian Pawlowski ___ Asterisk-Users 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 ___ Asterisk-Users 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
RE: [Asterisk-Users] G.729a beta codec on old Pentiums - FIXED
Hi All, Just thought Id provide a quick summary on this thread as the problem is now resolved. I didnt actually hear what the cause of the problem was as Digium shelled into my machine last night (Australian time GMT+10 here :-) and installed the codec and checked the license registration for me. Today I decided to replace the old g729 codec with the G.729 beta codec on my local Asterisk box, and noticed that the Digium FTP server now has different codecs for each different platform (athlon, i586, i686 etc). I grabbed the codec in the athlon directory since the local machine is an Athlon XP 1800+ and went ahead and registered the codec without any problems, and restarted Asterisk and all is working fine. Im happy to report the beta g729 codec appears to be working nicely here with the Cisco 7940 (SIP 7.1 firmware) and over IAX2 to my remote Asterisk box, both boxes are running the latest CVS-HEAD (as at 28/05/04). Cheers, Chris Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Saturday, 22 May 2004 12:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G.729a beta codec on old Pentiums Hi, Ive been trying to get the G.729a beta codec running with my remote Asterisk box that talks IAX2 to my local Asterisk box. Digium fixed the problem I was having in registering the beta codec, so that now works fine. Ive removed the old codec_g729b.so from /usr/lib/asterisk/modules and put in place the codec_g729a.so beta from digium FTP. My CVS build of Asterisk is about a day old now. Everytime I try to execute /usr/sbin/safe_astersik with codec_g729a.so in place, it crashes and core dumps, not giving much indication of whats happening. I tried executing Asterisk directly with /usr/sbin/asterisk cvvg to get as much verboseness as possible, and have cut the last few lines (host ID and license intentionally blanked out):- [format_g729.so] = (Raw G729 data) == Registered file format g729, extension(s) g729 [codec_g729a.so] = (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: **masked** == Found license '**masked**' providing 2 channels == Found total of 2 G.729 licenses Illegal instruction (core dumped) The machine this is running on is rather old its a Pentium MMX (166Mhz according to Linux, I thought it was a 200Mhz but Im remote to the machine at the moment so I cant watch the BIOS boot to see). This is probably just a shot in the dark, but could this be related to the F00F bug in the older Pentiums? Has anyone else got the beta g729a codec running successfully on an older Pentium machine? Linux /proc/cpuinfo reports the following:- :~$ cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 5 model : 4 model name : Pentium MMX stepping : 3 cpu MHz : 167.049 fdiv_bug : no hlt_bug : no f00f_bug : yes coma_bug : no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr mce cx8 mmx bogomips : 333.41 Thanks, Chris Lee
[Asterisk-Users] G.729a beta codec on old Pentiums
Hi, Ive been trying to get the G.729a beta codec running with my remote Asterisk box that talks IAX2 to my local Asterisk box. Digium fixed the problem I was having in registering the beta codec, so that now works fine. Ive removed the old codec_g729b.so from /usr/lib/asterisk/modules and put in place the codec_g729a.so beta from digium FTP. My CVS build of Asterisk is about a day old now. Everytime I try to execute /usr/sbin/safe_astersik with codec_g729a.so in place, it crashes and core dumps, not giving much indication of whats happening. I tried executing Asterisk directly with /usr/sbin/asterisk cvvg to get as much verboseness as possible, and have cut the last few lines (host ID and license intentionally blanked out):- [format_g729.so] = (Raw G729 data) == Registered file format g729, extension(s) g729 [codec_g729a.so] = (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: **masked** == Found license '**masked**' providing 2 channels == Found total of 2 G.729 licenses Illegal instruction (core dumped) The machine this is running on is rather old its a Pentium MMX (166Mhz according to Linux, I thought it was a 200Mhz but Im remote to the machine at the moment so I cant watch the BIOS boot to see). This is probably just a shot in the dark, but could this be related to the F00F bug in the older Pentiums? Has anyone else got the beta g729a codec running successfully on an older Pentium machine? Linux /proc/cpuinfo reports the following:- :~$ cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 5 model : 4 model name : Pentium MMX stepping : 3 cpu MHz : 167.049 fdiv_bug : no hlt_bug : no f00f_bug : yes coma_bug : no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr mce cx8 mmx bogomips : 333.41 Thanks, Chris Lee
RE: [Asterisk-Users] G.729a beta codec on old Pentiums
Hi Andrew, Here's the results:- [snipped out most of the above symbols messages] Reading symbols from /usr/lib/asterisk/modules/format_g726.so...done. Loaded symbols for /usr/lib/asterisk/modules/format_g726.so Reading symbols from /usr/lib/asterisk/modules/format_g729.so...done. Loaded symbols for /usr/lib/asterisk/modules/format_g729.so Reading symbols from /usr/lib/asterisk/modules/codec_g729a.so...done. Loaded symbols for /usr/lib/asterisk/modules/codec_g729a.so #0 0x4044e862 in lsp_get_quant () from /usr/lib/asterisk/modules/codec_g729a.so (gdb) bt #0 0x4044e862 in lsp_get_quant () from /usr/lib/asterisk/modules/codec_g729a.so #1 0x in ?? () (gdb) x/5i $eip 0x4044e862 lsp_get_quant+210: fcomi %st(5),%st 0x4044e864 lsp_get_quant+212: jbe0x4044eb40 lsp_get_quant+944 0x4044e86a lsp_get_quant+218: fsubr %st,%st(2) 0x4044e86c lsp_get_quant+220: faddp %st,%st(1) 0x4044e86e lsp_get_quant+222: fxch %st(1) (gdb) info registers eax0x3 3 ecx0x3c 60 edx0x1 1 ebx0x15 21 esp0xbfffeef0 0xbfffeef0 ebp0xbfffef10 0xbfffef10 esi0x40459d40 1078304064 edi0x40458940 1078298944 eip0x4044e862 0x4044e862 eflags 0x10202 66050 cs 0x23 35 ss 0x2b 43 ds 0x2b 43 es 0x2b 43 fs 0x2b 43 gs 0x2b 43 fctrl 0x37f895 fstat 0x1132 4402 ftag 0x400f 16399 fiseg 0x23 35 fioff 0x4044e860 1078257760 foseg 0x2b 43 fooff 0x0 0 ---Type return to continue, or q return to quit--- fop0xcc 204 mxcsr 0x1f80 8064 orig_eax 0x -1 (gdb) And re the F00F bug, yep I figured it was probably very unlikely to be the root cause of the problem, as I have noticed messages when linux boots that it implements a workaround for the bug, so if that's the case then I guess all should be fine. Thanks, Chris Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, 22 May 2004 12:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729a beta codec on old Pentiums If you have gdb installed, try gdb /usr/sbin/asterisk (or whichever path is appropiate) name_of_core_file, and send back the results of bt x/5i $eip info registers and that should allow the developers or other people to work out what the problem is. As for the f00f bug, its unlikely that would be causing problems, the mention of f00f is there so that you can check to see if the machine is vulnerable. From memory, most os's now include work arounds for that bug. ___ Asterisk-Users 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
RE: [Asterisk-Users] G.729a beta codec on old Pentiums
Okay, doesn't mean so much to me, but it might help someone. Nor I. I've posted the results to the RT ticket I've already got open with Digium support in case it helps them as well. (gdb) x/5i $eip 0x4044e862 lsp_get_quant+210: fcomi %st(5),%st 0x4044e864 lsp_get_quant+212: jbe0x4044eb40 lsp_get_quant+944 0x4044e86a lsp_get_quant+218: fsubr %st,%st(2) 0x4044e86c lsp_get_quant+220: faddp %st,%st(1) 0x4044e86e lsp_get_quant+222: fxch %st(1) I'm wondering did this gdb command return the last 5 assembler instructions to execute before the program crashed? I did some searching for Pentium x86 instructions, and fcomi, fsubr, faddp, and fxch are all FPU instructions, so perhaps the module is indeed bombing out in the old Pentium FPU. But as above, I really don't know enough about the nitty gritty of GNU debugger and x86 assembler to fully appreciate what I'm looking at so perhaps I should leave it to the experts from here :-) If anyone has the beta G.729a codec successfully running on a Pentium (with or without MMX) CPU of similar vintage please drop me a line and let me know, as then I can rule out the machine itself and look harder at figuring out why my particular Asterisk installation isn't working. Cheers, Chris Lee ___ Asterisk-Users 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
RE: [Asterisk-Users] AArgh, * and the 7960
I have a 7940 running 6.3 SIP firmware and make the following type of calls:- 7940 = * = IAX2 = * = Digium X100P = Nortel CICS Analog FXS Both local and remote asterisk's run CVS-02/24/04 (built about 30mins apart). The IAX2 connection is over a VPN, and both sites are running 1500k/256k ADSL connections, about 75ms ping time between the sites. The only time I notice any problems is if one site has an application flooding its upstream, otherwise audio quality is very good. The odd packet might drop here and there, scrambling a word or two, which I usually attribute to upstream choking. 7940 is running G.729 over 100Mbps LAN to Asterisk, and IAX2 connection is presently running GSM (I've bought a couple of G729 licences for the remote asterisk but am waiting on the keys to install the beta codec). Unfortunately I don't have any spare 7940/60's at present to try out on the remote * box to see how a SIP-IAX2-SIP call would perform. ___ Asterisk-Users 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
RE: [Asterisk-Users] Re: G729 Segmentation fault
I'm having the exact same problems here - won't start with safe_asterisk.
I'm running a slightly dated CVS head (CVS-02/24/04-15:39:13) however I have
two machines running this date CVS, the other already has G.729 installed
and works fine - however it registered automatically with the voiceage
registration script, I have a suspicion that the manual voiceage
registration method does not work, I've received the va-certificate,
installed it and checked permissions on it as well as codec_g729b.so and
still no go.
I then noticed the g729/beta directory on ftp.digium.com and decided to give
that a go, but it tells me I've got an invalid registration key (which I cut
pasted directly from the email so there was no chance of making a typo).
I've written a detailed email to digium support about this and will await to
see what the solution is... I'm tempted to move up to the latest CVS, but
it's just that the current one has been quite stable and I was going to wait
for the 1.0 release of asterisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Bogan
Sent: Wednesday, 19 May 2004 3:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: G729 Segmentation fault
Are you using the safe_asterisk script to start up? G729 requires a
tty, which the script provides - at least so I've read... I can get
mine to segfault every time if I start up using just the asterisk
command, safe_asterisk works every time...
Nothing seems to want to work:
/usr/sbin/safe_asterisk: line 77: 21837 Segmentation fault (core
dumped) asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
Asterisk ended with exit status 1
Asterisk died with code 1. Aborting.
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
___
Asterisk-Users 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
___
Asterisk-Users 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
[Asterisk-Users] Caller ID from Call Pickup
Hi, Im wondering if its possible to get Caller ID information from a Call Pickup specifically on a Cisco 79xx SIP handset. Ive setup a speed dial line on my 7940 to dial *8 so I can quickly pickup a call, but because the 7940 initiates the call the information on the screen is To: *8 I gather because of the call setup its probably not possible to extract the caller ID details of the call that was picked up (ie change the display to From: 12345678 Fred), but just curious none the less if theres an option somewhere Ive missed that would allow you to do this? Cheers, Chris Lee
RE: [Asterisk-Users] no sound via playback
I'm running slackware 9.1 as well, and found that mpg123 doesn't come with slackware, so you may need to fetch and install it to get playback working... http://www.voip-info.org/tiki-index.php?page=mpg123 http://www.mpg123.de/ I know I couldn't get MOH working until I installed mpg123, can't recall though if playback/voicemail was working without it, but it wouldn't hurt to start looking in that direction anyway. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of John Brown (CV) Sent: Sunday, 28 March 2004 2:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] no sound via playback Hi List, I've just built a new * box (slackware 9.1) and I get no sound from a Playback(tt-weasels) command. I've got other slack9.1 boxes running. * Version is v1.0 stable exten = 213,1,Answer exten = 213,2,Playback(tt-weasels) exten = 213,3,Playback(tt-weasels) exten = 213,4,Hangup when I dial 213, it says its playing, but no audio. calls between SIP users on this box work fine. calls between SIP and IAX devices work fine. calls between SIP and PSTN work fine. any ideas ?? ___ Asterisk-Users 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 ___ Asterisk-Users 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
RE: [Asterisk-Users] Use of Alert_Info with C7960?
Unfortunately even though it would seem the phone should support the ability to play custom ring tones, at present it only supports the internal tones which are:- Bellcore-BusyVerify Bellcore-Stutter Bellcore-MsgWaiting Bellcore-dr1 Bellcore-dr2 Bellcore-dr3 Bellcore-dr4 Bellcore-dr5 --- Extract from the SIP 6.0 Firmware Release Notes --- http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp79 60/addprot/sip/relnote/phnrn60s.htm In RFC-3261, the Alert-Info header is specified as a URL. When the Alert-Info header is received, the phone downloads the file from the URL and plays it as the alternate ring tone. This release does not support any external ringers. Only the tones and ring patterns that are already internal to the phone can be selected and played as an alternate ring tone. In this release, the Alert-Info header consists of a name of an internal tone or ringing pattern that can be played, as shown in the following example: Alert-Info: Bellcore-Busy There is no need to add a file extension (.au, .wav) to these names because the names are internal to the phone. When an Alert-Info header is received, the software scans the list of known tones and ringing patterns to find a match. If the software finds a match, the phone plays that tone or ringing pattern. If the software does not find a match, the phone plays the alert ringing pattern as it does today. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sunday, 21 March 2004 9:28 AM To: Asterisk-a-users-list Subject: [Asterisk-Users] Use of Alert_Info with C7960? Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around with distinctive ringing, trying to make it work. Extensions.conf looks like: exten = 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer exten = 3010,2,Dial(SIP/3010,15) exten = 3010,3,Voicemail2(u3010) exten = 3010,102,Voicemail2(b3010) exten = 3010,103,Hangup Calling that extension, the CLI indicates: -- Executing SetVar(SIP/3002-39d1, ALERT_INFO=3) in new stack -- Executing Dial(SIP/3002-39d1, SIP/3010|15) in new stack -- Called 3010 -- SIP/3010-f848 is ringing On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old Style and Synth Low. The first three choices produce different ringing sounds when selected from the display. I expected Alert_Info=3 to cause the C7960 to ring with the Old Style ringer, but it doesn't and setting it to 2 or 3 doesn't make any difference. Am I doing something wrong? Rich ___ Asterisk-Users 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 ___ Asterisk-Users 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
RE: [Asterisk-Users] Use of Alert_Info with C7960?
The custom ring tones are selectable through the Ring Type option in the Settings menu. When the phone rings, it will play that custom ring tone. Perhaps it's a memory limitation or an issue with the way Cisco are implementing the SIP firmware as to why you can't select a custom ring tone, you'd be better asking a Cisco engineer (open a TAC case) The phone will only download the currently selected custom ring tone (defined in RINGLIST.DAT) from the TFTP server each time it boots, or when you select another custom ring tone from the Ring Type option. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Sunday, 21 March 2004 11:53 AM To: Asterisk Users Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960? Then what's the point of being able to upload custom ring tones? (as shown in http://www.loligo.com/asterisk/Cisco/79xx/current/RINGLIST.DAT ) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Saturday, March 20, 2004 8:50 PM To: Asterisk Users Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960? Unfortunately even though it would seem the phone should support the ability to play custom ring tones, at present it only supports the internal tones which are:- Bellcore-BusyVerify Bellcore-Stutter Bellcore-MsgWaiting Bellcore-dr1 Bellcore-dr2 Bellcore-dr3 Bellcore-dr4 Bellcore-dr5 --- Extract from the SIP 6.0 Firmware Release Notes --- http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon /english/ipp79 60/addprot/sip/relnote/phnrn60s.htm In RFC-3261, the Alert-Info header is specified as a URL. When the Alert-Info header is received, the phone downloads the file from the URL and plays it as the alternate ring tone. This release does not support any external ringers. Only the tones and ring patterns that are already internal to the phone can be selected and played as an alternate ring tone. In this release, the Alert-Info header consists of a name of an internal tone or ringing pattern that can be played, as shown in the following example: Alert-Info: Bellcore-Busy There is no need to add a file extension (.au, .wav) to these names because the names are internal to the phone. When an Alert-Info header is received, the software scans the list of known tones and ringing patterns to find a match. If the software finds a match, the phone plays that tone or ringing pattern. If the software does not find a match, the phone plays the alert ringing pattern as it does today. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sunday, 21 March 2004 9:28 AM To: Asterisk-a-users-list Subject: [Asterisk-Users] Use of Alert_Info with C7960? Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around with distinctive ringing, trying to make it work. Extensions.conf looks like: exten = 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer exten = 3010,2,Dial(SIP/3010,15) exten = 3010,3,Voicemail2(u3010) exten = 3010,102,Voicemail2(b3010) exten = 3010,103,Hangup Calling that extension, the CLI indicates: -- Executing SetVar(SIP/3002-39d1, ALERT_INFO=3) in new stack -- Executing Dial(SIP/3002-39d1, SIP/3010|15) in new stack -- Called 3010 -- SIP/3010-f848 is ringing On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old Style and Synth Low. The first three choices produce different ringing sounds when selected from the display. I expected Alert_Info=3 to cause the C7960 to ring with the Old Style ringer, but it doesn't and setting it to 2 or 3 doesn't make any difference. Am I doing something wrong? Rich ___ Asterisk-Users 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 ___ Asterisk-Users 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 ___ Asterisk-Users 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 ___ Asterisk-Users 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
RE: [Asterisk-Users] Use of Alert_Info with C7960?
Yes, I know it will definitely play custom ringtones, I was even using the 24ctu.raw ring tone for a while (I've gone back to Chirp 1 for now). But all incoming calls get the currently selected ring tone. I should have clarified on an earlier statement I made:- Perhaps it's a memory limitation or an issue with the way Cisco are implementing the SIP firmware as to why you can't select a custom ring tone, you'd be better asking a Cisco engineer (open a TAC case) This was in reference to playing a custom ring tone with Alert_Info (ie: Distinctive Ring). If you know how to make the 7960 play a different ring tone by setting the Alert_Info variable, please tell! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Sunday, 21 March 2004 1:17 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960? The 7960 will absolutely play custom ringtones. Paul Mahler [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Saturday, March 20, 2004 5:50 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Use of Alert_Info with C7960? Unfortunately even though it would seem the phone should support the ability to play custom ring tones, at present it only supports the internal tones which are:- Bellcore-BusyVerify Bellcore-Stutter Bellcore-MsgWaiting Bellcore-dr1 Bellcore-dr2 Bellcore-dr3 Bellcore-dr4 Bellcore-dr5 --- Extract from the SIP 6.0 Firmware Release Notes --- http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp 79 60/addprot/sip/relnote/phnrn60s.htm In RFC-3261, the Alert-Info header is specified as a URL. When the Alert-Info header is received, the phone downloads the file from the URL and plays it as the alternate ring tone. This release does not support any external ringers. Only the tones and ring patterns that are already internal to the phone can be selected and played as an alternate ring tone. In this release, the Alert-Info header consists of a name of an internal tone or ringing pattern that can be played, as shown in the following example: Alert-Info: Bellcore-Busy There is no need to add a file extension (.au, .wav) to these names because the names are internal to the phone. When an Alert-Info header is received, the software scans the list of known tones and ringing patterns to find a match. If the software finds a match, the phone plays that tone or ringing pattern. If the software does not find a match, the phone plays the alert ringing pattern as it does today. ___ Asterisk-Users 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
RE: [Asterisk-Users] Simulating the lighted line in use type of phone
Just a thought on this topic, if you're using Cisco 7940/7960 IP handsets it should be possible to write a Perl script on the Asterisk box that updates the XML directory to show the current status of extensions. Then when the receptionist is on the phone, they could hit the services button, scroll through the list of extensions; see what the persons status is, and even transfer the call right through by pressing the Dial button underneath the extension if they hit Transfer before the Services button. I don't have enough experience with coding the XML directory extension on the Cisco beyond a simple phone-directory, but Cisco do have a programming guide available and there are some advanced features you could use. In fact even with a simple XML phone-directory, all you need do is have the Perl script interrogate Asterisk for the extension status, and if the extension is busy, simply append BUSY in capitals onto the end of the persons name. Then when the extension frees, remove the BUSY message... this should work, and since the phone does a HTTP request every time the user hits the Services button, if it's a script that's executed on the server then they should have an up to date view of who's busy/free. Granted if someone who was busy hangs up before the receptionist can scroll to their name, it will still show as busy, but apart from that it should do the trick and could be the next best thing to a live LED busy lamp field. This would remove the need for a PC running status software; although at the end of the day a software solution could probably be a lot more user friendly and flexible than what you can do through XML. The Cisco IP phones may be a little pricier than some others on the market, but as far as I'm concerned they currently are the bee's knees as far as IP phones go. Awesome call quality and best speakerphone I've ever had, and coupled with the XML/HTTP capabilities of the phone you can't go too wrong. Before you jump in the deep end with deploying IP telephony in an office environment, get yourself set up with two (or just 1 if money is tight) Cisco 7940/7960's w/SIP firmware talking with Asterisk and see what can be done. Besides, the Cisco 7940/7960's look far sexier than 20 year old dinosaur key stations and you'll be the envy of everyone in the office :-) Cheers, Chris Lee -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Clifton Sent: Wednesday, 25 February 2004 6:50 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Simulating the lighted line in use type of phone That's fine for outbound lines, but what if I want to call the guy in the next office ? I have to call him and get redirected to his busy vm just to know that he's on the phone. This is a huge issue with the recepetionist with the 'master console'. How does he/she know whether a user is busy or not ? 20 year old phone systems offer this capability. I see this as a serious shortfall of asterisk / currently available ip phones right now. - Chris Clifton ___ Asterisk-Users 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
RE: [Asterisk-Users] Simulating the lighted line in use type of phone
Then when the receptionist is on the phone, they could hit the services button, scroll through the list of extensions; see what the persons status is, and even transfer the call right through by pressing the Dial button underneath the extension if they hit Transfer before the Services button. Um, no. Reception needs an 'at a glance' view of all extensions -- I don't think you've ever worked a switchboard or seen some of these people handle the call volumes they are used to. Anything involving button presses or scrolling would be useless. True, probably too much assumption there on my part. I figured the old switchboards of yesteryear were pretty much redundant, or had loads reduced these days with direct in dial through BRI/PRI interfaces. Anyway, in a high call volume environment you might as well go the software route and get setup with a computer and a nice colourful busy lamp display on screen. Cheers, Chris Lee ___ Asterisk-Users 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
RE: [Asterisk-Users] IAX2 Call menu handling problem with Norstar - Partial Solution
Well after a bit more googling, I've found the quick nasty fix to this problem. Users on the Norstar extensions need to dial Feature 808 to enable Long Tones so that when they press a key on their keypad, it's passed correctly to the Analog Terminal Adapter. I call this a partial solution, since this feature only works on a per-call basis. However it would seem to me that this was happening already, just that for some reason the Norstar extension then stops sending/receiving on the voice channel... maybe it's a bug or just a Norstar Feature. I read somewhere that they are one of the worst PBX's to try and integrate with, and my experiences so far definitely concur with that, particularly with the ATA's... no disconnect supervision, can't pass a DTMF properly from digital to analog, argh! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Monday, 23 February 2004 2:44 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] IAX2 Call menu handling problem with Norstar Ok, after much stuffing around with the configs to sort it, I've narrowed the problem down to DTMF passing from the Norstar extension as being what breaks my setup. If I'm on a call with someone on a Norstar extension from my system, and they press a key, I hear a split second of the DTMF signal and the line goes silent. Now I've just got to figure a way to get the Norstar Asterisk to work together in DTMF harmony :-) ___ Asterisk-Users 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
[Asterisk-Users] IAX2 Call menu handling problem with Norstar
A quick rundown of my setup...
Norstar FXS - X100P- Asterisk (Starlight) - IAX2 - Asterisk (Voipsrv)
- Cisco 7940 SIP
Now what I've done is to setup a simple menu system on my Asterisk (Voipsrv)
to allow the caller to select which extension to ring off Voipsrv. The main
benefit being that I can make the menu system pretty tight with timeouts so
that the Zaptel interface (X100P) on Starlight is hungup properly, since
there's no supervision on the Norstar.
What's _really_ got me stumped now is this. When I make a call from my
mobile phone, direct indial to the * Starlight box, the call is passed
neatly through to Voipsrv and I get the menus, can dial my extension, and
the call functions just great.
However, when someone on a internal Norstar extension dials Starlight, the
call is passed to Voipsrv, and then they get the menu, dial 1 for my
extension (Cisco 7940) and it rings. However when I answer, I cannot hear a
thing, and they cannot hear a thing!
I don't understand how a call from the internal Norstar extension can be
different from a direct indial call from the ISDN, as they both go through
the same analog port on the Norstar and the same X100P interface.
At first I figured the problem was not having an Answer line at the start of
my menu, which I've now added but still hasn't solved the problem.
I've noticed when a call comes in that was dialled from a Norstar extension,
the messages on the Voipsrv console are:
-- Accepting AUTHENTICATED call from 192.168.0.252, requested format = 4,
actual format = 4
-- Executing Goto([EMAIL PROTECTED]/1, from-aushot|s|1) in new
stack
-- Goto (from-aushot,s,1)
-- Executing Answer([EMAIL PROTECTED]/1, ) in new stack
And when I place a call from my mobile through the Norstar's ISDN DID trunk,
and onto starlight I see the following on Voipsrv:
-- Accepting AUTHENTICATED call from 192.168.0.252, requested format = 4,
actual format = 4
-- Executing Goto([EMAIL PROTECTED]/2, from-aushot|s|1) in new
stack
-- Goto (from-aushot,s,1)
-- Executing Answer([EMAIL PROTECTED]/2, ) in new stack
I'm wondering why the internal call seems to end with /1 and the outside
call /2?
Starlight handles the call from the X100P in the following context:
[inbound-analog]
exten = s,1,SetCallerID(917)
exten = s,2,SetCIDName(Aushot PBX)
exten = s,3,Dial(IAX2/user:[EMAIL PROTECTED]/591)
exten = s,4,Hangup
And Voipsrv then has a direct goto on extension 591 to the following
context:
[from-aushot]
exten = s,1,Answer
exten = s,2,Background(welcome-dchaos)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
exten = i,1,Playback(vm-goodbye)
exten = i,2,Hangup
exten = 1,1,Background(one-moment-please)
exten = 1,2,Dial(${PHONE1},20)
exten = 1,3,Playtones(busy)
exten = 1,4,Wait(5)
exten = 1,5,Hangup
exten = 2,1,Background(one-moment-please)
exten = 2,2,Dial(${PHONE2},20)
exten = 2,3,Playtones(busy)
exten = 2,4,Wait(5)
exten = 2,5,Hangup
exten = 3,1,Background(one-moment-please)
exten = 3,2,Dial(Zap/2,20)
exten = 3,3,Playtones(busy)
exten = 3,4,Wait(5)
exten = 3,5,Hangup
Any suggestions greatly appreciated. Before I implemented this menu system,
I had a simple direct dial to extension 501 from the starlight machine with
a timeout to stop the call going to voicemail, and this worked fine for both
internal and external calls.
Thanks,
Chris Lee
___
Asterisk-Users 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
RE: [Asterisk-Users] IAX2 Call menu handling problem with Norstar
Ok, after much stuffing around with the configs to sort it, I've narrowed the problem down to DTMF passing from the Norstar extension as being what breaks my setup. If I'm on a call with someone on a Norstar extension from my system, and they press a key, I hear a split second of the DTMF signal and the line goes silent. Now I've just got to figure a way to get the Norstar Asterisk to work together in DTMF harmony :-) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Monday, 23 February 2004 1:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAX2 Call menu handling problem with Norstar A quick rundown of my setup... Norstar FXS - X100P- Asterisk (Starlight) - IAX2 - Asterisk (Voipsrv) - Cisco 7940 SIP ___ Asterisk-Users 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
RE: [Asterisk-Users] Analog Cordless Phone Recommendations
I've gone out and purchased a Uniden XS916 cordless handset, which has a
CallerID display, although it wasn't really one of my requirements, I
couldn't find one with MWI so I figured CallerID could be useful.
I'm not sure if I've missed something in my * config's, or if I've run into
the CallerID compatibility issues.
Is there something else that needs to be added onto the dial command in
extensions.conf to get the CallerID passed through, or is it all in
Zapata.conf?
My Zapata.conf for channel 2 (FXS) is:
language=en
context=local
signalling=fxo_ks
usercallerid=yes
usecallingpres=yes
callwaitingcallerid=yes
transfer=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
mailbox=503
callerid= Home 503
channel = 2
And the dial plan in extensions.conf for 503 is:
;
; Extension 503 - FXS1 - Cordless Phone
; ---
exten = 503,1,Dial(${FXS1},30)
exten = 503,2,Hangup
Phone specs
http://www.uniden.com.au/v3/product.asp?ID=295Group=1
Thanks,
Chris Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Low, Adam
Sent: Tuesday, 17 February 2004 8:34 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Analog Cordless Phone Recommendations
I don't think there is really an issue with 'which' analog phone, the only
issue (I am aware of) with interoperability is in relation to CallerID. In
the US it seems FSCK (I understand from my Aussie colleague that FSCK is
also used in Australia) is always used and across Europe it seems to be
DTMF but the actual format of the DTMF varies from country to country.
Features such as stuttered dial tone are generated by the FXS interface.
Like Dan I use ATA186's which generates the stuttered dialtone when
messages are waiting and its completely separate from the handset. I have
no experience with the TDM10B but I am confident it can do all that the
ATA can do ...
FYI: I am lucky enough to be using the BO BeoCom 2 which works great with
the ATA.
Rgds, Adam
-Original Message-
From: Christopher Lee [mailto:[EMAIL PROTECTED]
Sent: 17 February 2004 09:00
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Analog Cordless Phone Recommendations
Hi all,
I've just added a TDM10B (1port FXS) to my Asterisk box and want to use
this
extension with a cordless phone.
In particular I'm just wondering if anyone has any suggestions for a phone
that will perhaps be able to detect voicemail waiting on the Asterisk
server?
I'm guessing I should be able to get asterisk to generate a stuttered dial
tone when a message is waiting, so it's just a matter of finding such a
cordless phone that can detect this.
Cheers,
Chris Lee
___
Asterisk-Users 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
* DISCLAIMER *
This message and any attachment are confidential and may be privileged or
otherwise protected from disclosure and may include proprietary
information. If you are not the intended recipient, please telephone or
email the sender and delete this message and any attachment from your
system. If you are not the intended recipient you must not copy this
message or attachment or disclose the contents to any other person
___
Asterisk-Users 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
___
Asterisk-Users 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
RE: [Asterisk-Users] Analog Cordless Phone Recommendations
usercallerid=yes = TYPO try with : usecallerid=yes Excellent, thanks for pointing that out... unfortunately it's still not working. I've loaded the Asterisk console with heaps of verboseness, and found the following message appears when calling the analog extension: Connected to Asterisk CVS-02/19/04-14:49:36 currently running on voipsrv (pid = 142) -- Executing Dial(SIP/501-1a5a, Zap/2|30) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing Feb 19 15:11:26 WARNING[409618]: chan_zap.c:3065 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Hungup 'Zap/2-1' == Spawn extension (local, 503, 1) exited non-zero on 'SIP/501-1a5a' What's Didn't finisher Caller-ID spill about? Compatibility issue between the Digium FXS card and the cordless phone? Thanks, Chris Lee ___ Asterisk-Users 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
RE: [Asterisk-Users] Analog Cordless Phone Recommendations
Problem solved, I found a previous posting about this... = http://lists.digium.com/pipermail/asterisk-users/2003-April/010591.html In asterisk/channels/chan_zap.c Change this from 1 to 2: #define DEFAULT_CIDRINGS 2 = Recompiled and the error message is gone, and CallerID is now displaying on the phone, although it doesn't appear until about the 2nd or 3rd ring, it's still better than nothing :-) Cheers, Chris Lee -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Thursday, 19 February 2004 3:14 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Analog Cordless Phone Recommendations usercallerid=yes = TYPO try with : usecallerid=yes Excellent, thanks for pointing that out... unfortunately it's still not working. I've loaded the Asterisk console with heaps of verboseness, and found the following message appears when calling the analog extension: Connected to Asterisk CVS-02/19/04-14:49:36 currently running on voipsrv (pid = 142) -- Executing Dial(SIP/501-1a5a, Zap/2|30) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing Feb 19 15:11:26 WARNING[409618]: chan_zap.c:3065 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Hungup 'Zap/2-1' == Spawn extension (local, 503, 1) exited non-zero on 'SIP/501-1a5a' What's Didn't finisher Caller-ID spill about? Compatibility issue between the Digium FXS card and the cordless phone? Thanks, Chris Lee ___ Asterisk-Users 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 ___ Asterisk-Users 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
[Asterisk-Users] Trouble emailing Digium
Is it just me or is everyone having problems with emailing digium? I've tried sending two emails, but they keep getting returned with the following errors:- - The following addresses had permanent fatal errors - [EMAIL PROTECTED] (reason: 554 [EMAIL PROTECTED]: Recipient address rejected: Relay access denied) - Transcript of session follows - ... while talking to digium.com.mail1.psmtp.com.: RCPT To:[EMAIL PROTECTED] 554 [EMAIL PROTECTED]: Recipient address rejected: Relay access denied 554 5.0.0 Service unavailable Thanks, Chris Lee ___ Asterisk-Users 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
RE: [Asterisk-Users] Dialling Hook Flash on Zaptel
I had some similar problems with the X100P and our ATA-2. I also couldn't ever get the Nortel to recognize the DTMF, or get Asterisk to recognize DTMF coming through the Nortel. I wish I could say that I figured out a really cool way to make it work, but instead I moved on and interconnected via PRIs. I did a little more testing here, I've found that from my Cisco 7940 dialing out to my mobile, I can dial DTMF tones and hear them on the mobile. I'm not sure if the Norstar is doing this, as no matter how long I press the button down for I only get a short beep of the DTMF tone on the mobile. Perhaps this means the Norstar can only pass along the tones but not actually interpret them, or maybe the DTMF tone length is too short for the Norstar. Either way, I've changed the station filter for this particular extension to allow a greater range of numbers to be dialled and will control it with the dial plans in Asterisk. I've also considered changing the interconnection method, unfortunately (although this may be a good thing) my system is only a baby CICS with a 4-port analog trunk module and a 4-port BRI module. To connect via the analog trunk would be really neat with a 4-port FXS digium card, but unfortunately this particular Nortel card is not a supervised card, so can't be setup in the Norstar for auto-answer (which was my main reason for installing the BRI card). Then in terms of connecting via BRI, I think it would probably be more effective in the long run to just replace the whole system with SIP handsets since there's only 7 extensions in use (although the cost of Cisco 7940's would quickly add up, but I wouldn't want to use anything less even though they may be cheaper, as these are fantastic phones and really worth it IMHO). Also apart from the handset replacement cost, I think it'll be somewhat hard to beat the near bullet-proof performance the current CICS system has given us. The only outages its ever had was to install the BRI card and the odd power outage that was long enough to fully drain the UPS batteries. Cheers, Chris Lee ___ Asterisk-Users 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
RE: [Asterisk-Users] Cisco 7960 quick dial
Out of interest, does anyone know if it's possible to get the 7960 to start accepting a number while on-hook, without having to press NewCall, the line button, or speaker button? This is just something I was used to with the Norstar extensions, I could immediately start dialing the numbers for an internal extension and it'd work. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jose Inzunza/YM/RWDOE Sent: Wednesday, 4 February 2004 2:21 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 quick dial Is there a way to make the Cisco 7960 SIP phone dial out automatically without having to press the dial button, once the numbers that you have entered match a specific pattern? This feature is present when the phone is working with a Cisco CallManager. For example, if all of my internal extensions begin with a '5' and are four digits long, if I dialed '5123' on the phone, the call would initiate once I pressed the '3'. Any help would be appreciated. Jose ___ Asterisk-Users 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 ___ Asterisk-Users 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
RE: [Asterisk-Users] Cisco 7960 quick dial
Then you've got to hand it to Nortel, they do know how to make a damn good phone extensions for lazy people like me :-) I actually believe this isn't the case with the Nortel Meridian systems, as I noticed when using one it wouldn't accept the numbers without first pressing that extensions DN key... perhaps it's just a Norstar CICS/MICS features. Anyway I've opened a TAC case with Cisco and will await their response, which I'm guessing already will be no, can't hurt to ask tho :-) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, 4 February 2004 11:58 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 quick dial I have one word for you... LAZY! bkw On Wed, 4 Feb 2004, Christopher Lee wrote: Out of interest, does anyone know if it's possible to get the 7960 to start accepting a number while on-hook, without having to press NewCall, the line button, or speaker button? This is just something I was used to with the Norstar extensions, I could immediately start dialing the numbers for an internal extension and it'd work. ___ Asterisk-Users 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
RE: [Asterisk-Users] Cisco 7960 quick dial
Absolutely no argument from me on that front, hands down the Cisco 7940/7960 are a damn good IP phone, and compared to the existing Norstar handsets we have, a far better phone overall. The handsfree functionality on the Cisco's is truly awesome, the mic pickup and clarity is far better than the Norstar and people can barely tell the difference between talking to them on handsfree or picking up the handpiece. Definitely worth every dollar, although I wouldn't say no to them lowering the price, which they appear to have done with the introduction of the 7970. The next phone I want to get is a Cisco 7920 WiFi... although once again, they're on the exy side, I'm sure they'll also be well worth it. Unfortunately they aren't available in Australia yet, hopefully not too far off. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Thursday, 5 February 2004 12:15 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 quick dial hehe ya I have to admit they are very featureful. :P Asterisk is still a baby i'm sure sip phones will get better with time. But you do have to admit that the cisco 7960's are damn good phones. ___ Asterisk-Users 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
RE: [Asterisk-Users] Cisco 7960 quick dial
Had a quick response from Cisco on this. The short and simple answer is NO, the phones cannot be made to dial while on-hook. It was mentioned to me that with SIP 6.0 firmware and onwards you can now manage your own personal directory on the phone (whereas before the phone did it for you), and by pressing the down arrow on the phone while idle gives you a fast was to access the personal directory. This still isn't what I was after, and I've setup an XML directory on the services button anyway :-) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Thursday, 5 February 2004 12:12 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 quick dial Then you've got to hand it to Nortel, they do know how to make a damn good phone extensions for lazy people like me :-) I actually believe this isn't the case with the Nortel Meridian systems, as I noticed when using one it wouldn't accept the numbers without first pressing that extensions DN key... perhaps it's just a Norstar CICS/MICS features. Anyway I've opened a TAC case with Cisco and will await their response, which I'm guessing already will be no, can't hurt to ask tho :-) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, 4 February 2004 11:58 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 quick dial I have one word for you... LAZY! bkw On Wed, 4 Feb 2004, Christopher Lee wrote: Out of interest, does anyone know if it's possible to get the 7960 to start accepting a number while on-hook, without having to press NewCall, the line button, or speaker button? This is just something I was used to with the Norstar extensions, I could immediately start dialing the numbers for an internal extension and it'd work. ___ Asterisk-Users 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 ___ Asterisk-Users 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
[Asterisk-Users] Dialling Hook Flash on Zaptel
Hi,
I'm trying to get my X100P to Dial the following sequence to gain access to
speed dial numbers on my Norstar PBX that the X100 is connected to...
[FLASH] [*] [0] [22] (where 22 is the speed dial number)
But so far I've had no luck, with the following extension:-
exten = 922,1,Flash(${DIALOUTANALOG})
exten = 922,2,Dial(${DIALOUTANALOG}/*022)
exten = 922,3,Congestion
exten = 922,4,Hangup
Looking at the console, Asterisk doesn't get past the Flash command, telling
me that it's not a valid Zap channel. The call is being made from my Cisco
SIP phone through my local Asterisk Box, then via an IAX2 channel to the
site with the Asterisk box+X100P connected to the Norstar.
CONSOLE LOG
-- Accepting AUTHENTICATED call from 192.168.1.1, requested format = 2,
actual format = 2
-- Executing Flash([EMAIL PROTECTED]/2, Zap/1) in new stack Feb 3
22:37:19 WARNING[1146896]: app_flash.c:85 flash_exec:
[EMAIL PROTECTED]/2 is not a Zap channel
== Spawn extension (local, 922, 1) exited non-zero on
'[EMAIL PROTECTED]/2'
-- Hungup '[EMAIL PROTECTED]/2'
CONSOLE LOG
Is there some other way to dial a flash with the dial command? I notice
there's a W to insert a wait sequence.
Thanks,
Chris Lee
___
Asterisk-Users 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
RE: [Asterisk-Users] Dialling Hook Flash on Zaptel
Just for fun, try this: exten = 922,1,Flash(Zap/1) exten = 922,2,Dial(Zap/1/*022) exten = 922,3,Congestion exten = 922,4,Hangup and see if it gives the same error. I'd be interested to see if there's perhaps some strange variable swapping going on. I gave that a try, but the same problem on the console that this is not a Zap channel. Also tried David's Gomillion's approach with a dial then flash, but still the same problem about not being a zap channel. And indeed, flash doesn't accept an argument so there's not much point in me placing it there, I was just trying out a non-working example/food for thought that was posted previously... http://www.mail-archive.com/[EMAIL PROTECTED]/msg23426.html I've been toying with the extension a little more in the hope of perhaps doing a pseudo flash as per David's dial then flash suggestion but tweaked as follows:- exten = 922,1,Dial(Zap/1/*022,1) exten = 922,2,Dial(Zap/1/*022) exten = 922,3,Congestion exten = 922,4,Hangup (I've tried a few variations on the above, including a 1 in place of the *, and first line dialing 0 instead of the full command) Indeed, Asterisk picks up the line, dials (hopefully correctly) *022 then hangs up after 1000ms. Then hopefully hammers the line open again so the Norstar sees it as a flash and continues to dial *022 again. Unfortunately it doesn't work, Asterisk seems to be doing it correctly, but the Nortel can be a cantankerous beast with it's analog ports. I just get the rapid congestion tone from it that somethings not right with the way I dialled. As per the further suggestions, my speed dials rarely ever change, and I think I will relent and take this approach... basically I was wanting to not have to change the restriction filters on the Nortel for that analog port (since the speed dials override restrictions), but I think I'll fine grain the dialing restrictions through my Asterisk dial plans, should be the most pain free approach. The other thing was I just wanted to learn a little more about what can be done on the X100P, as there's many other commands that can be sent to the Norstar that are prefixed with a flash, although I doubt I'll really ever need to use any of them. Thanks very much for all the suggestions though, much appreciated. And David I hope you continue on the list, if only as a reader, as your input and contributions are definitely appreciated. Cheers, Chris Lee ___ Asterisk-Users 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
[Asterisk-Users] Norstar Integration with Asterisk via FXO or BRI ISDN
Hi, I have a legacy Norstar system that I'm looking into integrating with my Asterisk setup. My first attempts have worked, which involves a Wildcard X100P FXO card in the * box connected to the Internal ATA (FXS port) on the Norstar system. Calling from SIP - Norstar works fine, since the SIP caller initiated the call and generally will be sane enough to hangup the phone when complete, making asterisk hangup the X100P. However in the opposite direction, Norstar - SIP, the problem begins, thanks to no disconnect supervision/provision on the FXS port from the Norstar, so the X100P doesn't know the Norstar caller has hangup. A thought that occurred was perhaps Asterisk could do some sort of soft disconnect supervision? So when the FXO card has seized the line, but there is no transmit/receive audio (or no major variation from the standard background radiation) then after a safe timeout of say a minute or two, it could disconnect the line? I'd be interested to hear others views on this, would it work (or does it already exist?). Another thought is that the Norstar has both an 4-port FXO trunk card and a 4-port BRI-ISDN trunk card. The 4-port FXO card is completely free for use, however it doesn't support disconnect supervision, so same problem there. However, the BRI-ISDN card obviously does support disconnect supervision _and_ Direct InDial, even better! Only two ports are in use, so I could easily borrow one. From what I understand of BRI ISDN, it's not possible to just make a crossover cable and connect this trunk card directly to an ISDN card in the Asterisk box. Doing some searching, it appears there are quite a number of ISDN simulators on the market people are using for Cisco exam preparations. I'm wondering if anyone out there would have done, or know if what I'm thinking is possible? - [Asterisk Box ISDN Card] --- [ISDN SIM] --- [Norstar BRI Trunk] Granted, there might be a need for NT-1's at each link for the ISDN Simulator, but apart from that, would this work? The way I see it, so long as the ISDN sim gives asterisk a dialtone, accepts a phone number, then passes that number (preferable a 3 digit number for DID) to the Norstar, then one can dial direct from Asterisk to a Norstar extension. Likewise, one can dial from the Norstar direct to the Asterisk box, perhaps without the need for a auto-attendent menu to direct users if Direct InDial numbering can be provided. Anyway, just some food for thought at this stage on how to get around it, the pricings for some of the simulators I've seen are prohibitively expensive. I think in my case it may be cheaper to replace the Norstar extensions with brand new Cisco SIP phones and plug the ISDN trunks direct into Asterisk, making the Norstar defunct and the whole system much more flexible... but that's just a pipe dream at this stage :-) Cheers, Chris Lee ___ Asterisk-Users 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
RE: [Asterisk-Users] Does anyone manage the wiki?
Hi Chris, I think it's a community resource, to be freely updated by anyone... I've added some info to a couple of pages. I do agree though, it's a strange feeling that all you need to do is register an account and you can freely edit pages, makes me wonder if these Wiki's are susceptible to bored people randomly changing pages/information to introduce errors or delete pages altogether? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Higgins Sent: Thursday, 29 January 2004 1:18 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Does anyone manage the wiki? I would like to correct some of the text on the GotoIf application page on the wiki. Does somebody actively manage changes like this, or should I fire away and make it myself? I'm actually surprised I have permission to edit a page without prior authorization, but it DOES state at the bottom of the generated pages to 'please update the page with new information...'. If I don't hear otherwise, I'll go ahead. -- Chris ___ Asterisk-Users 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 ___ Asterisk-Users 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
RE: [Asterisk-Users] Asterisk Indications
I am really having trouble with this. I have been making changes to indications.conf, but my changes are not taking effect. I have shut down Asterisk, re-run ztcfg, reloaded the zaptel modules, all to no avail -- I get the same tones consistently... What should I be doing to bring in a new indications.conf? No problems here with making changes to indications.conf and then doing a reload on the console, the changes then take effect. My testing involves calling from a SIP handset to a dummy extension setup to answer and playback the tones I want to check. ; Test Australian ringing tones - indications exten = 906,1,Answer exten = 906,2,Wait(1) exten = 906,3,Playtones(ring) exten = 906,4,Wait(12) exten = 906,5,Playtones(busy) exten = 906,6,Wait(5) exten = 906,7,Hangup It sounds like you must have FXS extensions your trying to test the indications on? I don't have an FXS card in my machine to test with, so I'm not sure how it works, but it should still be the same, as a reload definitely re-reads the indications.conf configuration. Regards, Chris Lee ___ Asterisk-Users 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
RE: [Asterisk-Users] Asterisk Indications
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Vic Cross Sent: Monday, 26 January 2004 9:31 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Indications On Sun, 25 Jan 2004, Steve Underwood wrote: Actually, nothing would use a 17Hz tone - it doesn't pass through a 300-3400Hz channel very well :-) It's not a 17Hz tone. Australian (and others) tones are single-frequency tones that are amplitude-modulated at a second, much lower, frequency. The x*y notation in indications.conf is supposed to reflect this: for example, 400*17 would be a 400Hz tone amplitude-modulated at 17Hz. So does that mean the second frequency is 400-17 = 383Hz ? I've tried 400+383, but it didn't sound right. I posted a few days ago that nothing seems to happen when I specify the modulation frequency in my indications.conf -- all I get is the constant 400Hz tone. I will try some of the other combinations mentioned and see if they produce something more suitable. I've now settled on my ring tone as being 400+425. Regards, Chris Lee ___ Asterisk-Users 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
RE: [Asterisk-Users] Asterisk Indications
For the benefit of anyone with the same questions or searching the archives, Ive solved my problem to the below. The Cisco 7940 (and other SIP devices) generate their own indication tones of ring etc., I found by placing an Answer before a dial, then Asterisk will answer the call and be able to provide indications.conf tones down the line. Eg: exten = 931,1,Answer exten = 931,2,Dial(SIP/931,20) exten = 931,3,Voicemail(u931) exten = 931,102,Voicemail(b931) exten = 931,103,Hangup Also, I found the ringing tone for Australia included in indications.conf doesnt sound quite like I expected Ive done a little toying with it, and found the following sounds a little bit closer to what youd expect, but its still not quite right: for [au] context in indications.conf == ring = 400+420/400,0/200,400+420/400,0/2000 The original indications has 400+17/400, but I find that sounds more like two beeps (which could possibly be confused with the Australian congestion/busy tones). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Sunday, 18 January 2004 2:04 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Indications Hi, Just wondering if someone could better explain how the indications.conf file actually affects Asterisk? I am using a Cisco 7940 from my Asterisk system, and have set in indications.conf country=au thinking that this would make the dialtones/call progress sound like the familiar Australian tones? However when I call another extension on my system, it still sounds like the American ring tone. Does the indications perhaps only effect Analog FXS cards and not SIP phones? Also, when loading the Asterisk configs as shown below, it displays a message about Removed default indication country au and at the end proceeds to set default indication country to au the Removed part has me thinking its forgotten all about the particular indications for au? ==cut from Asterisk console=== -- Unregistered indication country 'us' Jan 18 14:02:36 NOTICE[262161]: indications.c:390 ast_unregister_indication_coun try: Removed default indication country 'au' -- Unregistered indication country 'au' -- Unregistered indication country 'fr' -- Unregistered indication country 'de' -- Unregistered indication country 'nl' -- Unregistered indication country 'uk' -- Unregistered indication country 'fi' -- Unregistered indication country 'no' == Parsing '/etc/asterisk/indications.conf': == Parsing '/etc/asterisk/indic ations.conf': Found -- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Registered indication country 'fi' -- Registered indication country 'no' -- Setting default indication country to 'au' == Thanks, Chris Lee
RE: [Asterisk-Users] Asterisk Indications
I've had a closer listen to 400*17 through the handpiece rather than just on speaker phone, and I get the feeling that the Australian ringing tone must have been tweaked slightly, perhaps with the introduction of the newer Ericsson AXE exchanges? 400*17 sounds familiar, perhaps the older exchanges (cross-bar?) used that format? That said, the 400+420 isn't exactly how my current exchange sounds, but sounds good to me anyway :-) I'm looking at tweaking the sounds somewhat more and moving away from the exchange sounds... I'd actually like to get it sounding more like a Nortel Meridian system, but I don't yet have any example rings to work off to try and get it similar sounding. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Sunday, 25 January 2004 4:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Indications The correct tone is 400*17 (383 + 417) according to the ITU specs. Actually, nothing would use a 17Hz tone - it doesn't pass through a 300-3400Hz channel very well :-) Regards, Steve ___ Asterisk-Users 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
RE: [Asterisk-Users] mp3player not working
Problem solved found that Asterisk is calling mpg123 to playback mp3s which isnt installed on Slackware 9.1 by default. Downloaded mpg123 source from http://www.mpg123.de/ and compiled with make linux; make install and now working. Also discovered that mpg123 doesnt seem to playback mp3s with ID3 tags in them, so strip them out before copying them to your Asterisk box. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee Sent: Thursday, 22 January 2004 11:18 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] mp3player not working Hi, Im running the latest Asterisk (built last Saturday) and cant get mp3s to playback on my handsets (this includes music on hold). I setup a couple of extensions, 901 and 902 to playback an mp3 I loaded on, and the sample moh that is included with Asterisk. When I attempt to call either extension I dont hear any sound, and the following displays on the console: -- Executing Answer(SIP/931-0efa, ) in new stack -- Executing Wait(SIP/931-0efa, 1) in new stack -- Executing MP3Player(SIP/931-0efa, /var/lib/asterisk/mohmp3/sample-hold .mp3) in new stack Jan 22 11:12:32 WARNING[442386]: rtp.c:375 ast_rtp_read: RTP Read error: Resourc e temporarily unavailable Jan 22 11:12:35 NOTICE[442386]: app_mp3.c:93 timed_read: Selected timed out/erro red out with 0 -- Executing Wait(SIP/931-0efa, 20) in new stack == Spawn extension (local, 902, 4) exited non-zero on 'SIP/931-0efa' The IP phones Im using are Cisco 7940 running G.729a. I have successfully licenced and registered 2x channels of g729 codec (running the new_codec_binary from ftp.digium.com) today, and have no problems checking my voicemail on Asterisk or dialing out through IAXtel or receiving calls. Even when I was running g711ulaw codec on the phones I had the same problem. Is there another dependency that is required for mp3playback in Linux? Is a soundcard required? My Linux box is running Slackware Linux 9.1. Any help to point me in the right direction to getting mp3playback and my music on hold working would be greatly appreciated. Thanks in advance, Chris Lee
[Asterisk-Users] mp3player not working
Hi, Im running the latest Asterisk (built last Saturday) and cant get mp3s to playback on my handsets (this includes music on hold). I setup a couple of extensions, 901 and 902 to playback an mp3 I loaded on, and the sample moh that is included with Asterisk. When I attempt to call either extension I dont hear any sound, and the following displays on the console: -- Executing Answer(SIP/931-0efa, ) in new stack -- Executing Wait(SIP/931-0efa, 1) in new stack -- Executing MP3Player(SIP/931-0efa, /var/lib/asterisk/mohmp3/sample-hold .mp3) in new stack Jan 22 11:12:32 WARNING[442386]: rtp.c:375 ast_rtp_read: RTP Read error: Resourc e temporarily unavailable Jan 22 11:12:35 NOTICE[442386]: app_mp3.c:93 timed_read: Selected timed out/erro red out with 0 -- Executing Wait(SIP/931-0efa, 20) in new stack == Spawn extension (local, 902, 4) exited non-zero on 'SIP/931-0efa' The IP phones Im using are Cisco 7940 running G.729a. I have successfully licenced and registered 2x channels of g729 codec (running the new_codec_binary from ftp.digium.com) today, and have no problems checking my voicemail on Asterisk or dialing out through IAXtel or receiving calls. Even when I was running g711ulaw codec on the phones I had the same problem. Is there another dependency that is required for mp3playback in Linux? Is a soundcard required? My Linux box is running Slackware Linux 9.1. Any help to point me in the right direction to getting mp3playback and my music on hold working would be greatly appreciated. Thanks in advance, Chris Lee
[Asterisk-Users] X100P Configs for Australia
Hi, Just wondering if anyone else in Australia is using the X100P to connect to the PSTN, and what configs they have for it? Im finding at present when I make a call I get a fair bit of echo of myself speaking, and also the person on the other end cant hear me very well (perhaps need to up the digial Tx Gain? I dont have it configured at present) Asterisk is running on Slackware Linux 9.1 and I built from the latest CVS just last night (Saturday 17th Jan 04). The phone Im using to call from is a Cisco 7940 running the SIP 6.0 firmware. If I make calls between the two Cisco 7940s on my Asterisk system the voice quality is fine. The settings I have for now are:- Zapata.conf = [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=no immediate=no busydetect=no callprogress=no relaxdtmf=yes echocancel=yes echocancelwhenbridged=yes callerid=asreceived channel = 1 = Zaptel.conf = fxsks=1 loadzone=au defaultzone=au = Thanks, Chris Lee
[Asterisk-Users] Slackware 9.1 Install Help
Hi, I'm trying to install Asterisk onto a fresh install of Slackware 9.1... I've installed all packages in A, AP, D, E, F, K, L, N... So basically what's needed for a text based system with development, networking, docs, libraries.. No X-Windows, no games, no TCL/TEX etc. Following the commands on the Asterisk website I checked out the CVS source and started compiling... Zaptel and libpri combiled no problems, but when I got to Asterisk it ended up with the error cut out below. I'm guessing I've missed a dependency somewhere? I checked that I have the readline, openssl and openssl-solibs, along with the full kernel source installed. Can someone please point me in the right direction on how to get around this and get Asterisk to compile? Thanks in advance, Chris Lee =CUT= In file included from /usr/include/gtk-1.2/gtk/gtkarg.h:31, from /usr/include/gtk-1.2/gtk/gtkobject.h:31, from /usr/include/gtk-1.2/gtk/gtkaccelgroup.h:35, from /usr/include/gtk-1.2/gtk/gtk.h:32, from pbx_gtkconsole.c:38: /usr/include/gtk-1.2/gtk/gtktypeutils.h:163: warning: function declaration isn't a prototype In file included from /usr/include/gtk-1.2/gtk/gtk.h:80, from pbx_gtkconsole.c:38: /usr/include/gtk-1.2/gtk/gtkitemfactory.h:48: warning: function declaration isn' t a prototype pbx_gtkconsole.c: In function `__verboser': pbx_gtkconsole.c:101: warning: assignment discards qualifiers from pointer targe t type gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o `gtk-config --libs gthread` /usr/lib/gcc-lib/i486-slackware-linux/3.2.3/../../../../i486-slackware-l inux/bin /ld: cannot find -lXext collect2: ld returned 1 exit status make[1]: *** [pbx_gtkconsole.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/pbx' make: *** [subdirs] Error 1 [EMAIL PROTECTED]:/usr/src/asterisk# =CUT= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Slackware 9.1 Install Help
Thanks very for the rapid responses Andrew and Panny! I went and installed the base packages from the X set for x-windows, and recompiled asterisk successfully, so that's all good, but you did get me thinking of removing gtkconsole from the Makefile. However since it's compiled I won't try and break it :-) And yeah I guess installing everything from all of those packages really was overkill, but I was taking the lazy approach rather than trying to step through it manually and knowing my luck miss something important. It's only a test machine at present, celeron 500, 192mb ram, 13gb hdd and Wildcard X100P.. Once I get to know it I might get some decent hardware and rebuild it. Thanks once again for the rapid help! Cheers, Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, 22 October 2003 21:45 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Slackware 9.1 Install Help =CUT= In file included from /usr/include/gtk-1.2/gtk/gtkarg.h:31, from /usr/include/gtk-1.2/gtk/gtkobject.h:31, from /usr/include/gtk-1.2/gtk/gtkaccelgroup.h:35, from /usr/include/gtk-1.2/gtk/gtk.h:32, from pbx_gtkconsole.c:38: /usr/include/gtk-1.2/gtk/gtktypeutils.h:163: warning: function declaration isn't a prototype In file included from /usr/include/gtk-1.2/gtk/gtk.h:80, from pbx_gtkconsole.c:38: /usr/include/gtk-1.2/gtk/gtkitemfactory.h:48: warning: function declaration isn' t a prototype pbx_gtkconsole.c: In function `__verboser': pbx_gtkconsole.c:101: warning: assignment discards qualifiers from pointer targe t type gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o `gtk-config --libs gthread` /usr/lib/gcc-lib/i486-slackware-linux/3.2.3/../../../../i486-slackware-l inux/bin /ld: cannot find -lXext collect2: ld returned 1 exit status make[1]: *** [pbx_gtkconsole.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/pbx' make: *** [subdirs] Error 1 [EMAIL PROTECTED]:/usr/src/asterisk# =CUT= It wants GTK (and by extension, X), but I have * installed on a number of Slack91 boxes without X... This must be a recent thing... Try hacking the pbx/Makefile to eliminate pbx_gtkconsole or removing the GTK+ libraries from your system, as * must be autodetecting what it can build, and kdeconsole's not building. You've got a shitload of extra crap on your * system if you installed everything in those disksets... Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
