Re: [asterisk-users] zaptel/ssh interaction
On Sat, 14 Apr 2007, Greg Woods wrote: On Sat, 2007-04-14 at 14:17 -0500, Lacy Moore - Aspendora wrote: This was mentioned earlier: I suspect IRQ Sharing. I know. And I posted my /proc/interrupts showing that there were no shared IRQ's. I can't see either how an IRQ conflict would affect SSH. To me that's just bogus, and it seems that every time theres an issue with a Zaptel card/driver the easiest solution is to shout: Check your IRQs. If there was an IRQ conflict it would affect all ethernet traffic, wouldn't it? Why would it single out ssh... ? # cat /proc/interrupts CPU0 0: 670560442 IO-APIC-edge timer 1: 20079 IO-APIC-edge i8042 6: 2 IO-APIC-edge floppy 7: 0 IO-APIC-edge parport0 8: 1 IO-APIC-edge rtc 9: 0 IO-APIC-fasteoi acpi 12: 23344 IO-APIC-edge i8042 14:1073586 IO-APIC-edge ide0 15:5943428 IO-APIC-edge ide1 16: 5554 IO-APIC-fasteoi libata 17:1441409 IO-APIC-fasteoi uhci_hcd:usb1, uhci_hcd:usb2, uhci_hcd:usb3, uhci_hcd:usb4, ehci_hcd:usb5 18: 23387293 IO-APIC-fasteoi eth0 19: 613419658 IO-APIC-fasteoi wctdm 20:2592107 IO-APIC-fasteoi eth1 21: 2597 IO-APIC-fasteoi Ensoniq AudioPCI NMI: 0 LOC: 670560322 ERR: 0 MIS: 0 That looks fine to me. (if a little busy, if this were a production server, I'd go into the BIOS and disable as much as I could and compile up a custom kernel) Are you using a CentOS package? Have you tried compiling asterisk/zaptel from scratch? I'd think it's highly unlikely that Zaptel would be listeining on port 22 (ssh) though, but you might want to check with netstat, and I'd not have thought there would be any firewall issues (does CentOC come with a built-in firewall? Can you turn it off? (iptables -n -L to list) More questions that answers I'm afraid... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DISABLE 9?
Is there a way to make it so you do not have to dial 9 by default to dial a outside number? I would like it if we could just dial the number any pointers? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [asterisk-users] DISABLE 9?
Hi everybody ! I never use any prefix number to dial out. I prefer to do like any standard residential subscriber, not to force somebody to think : Oh no ! I have forgottent to input the 9 - or 0 - before to dial out !. Directly inputing the real number is more natural. Adding a prefix is an old way to go inherited from analog PABX integrators ;-) If a customer want that, ok, do it, to avoid to have to change his habits. If you have no obligation to do that, forget it ! Think a good dialplan instead of that... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de JNA Envoyé : dimanche 15 avril 2007 11:49 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [asterisk-users] DISABLE 9? Is there a way to make it so you do not have to dial 9 by default to dial a outside number? I would like it if we could just dial the number any pointers? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] agents and music on hold with autoanswer..
My colleague left our company, then I have to manage all our phones lines and asterisk: please, apologize me because I'm 'absolute beginner' about voip/asterisk!! Well... all seems work fine; we have some queues and some agents; the music on hold works fine when the agent press the hold button on the phone (thomson); the agents have the 'autoanwser' flag on. BUT if the agent have to go elsewhere for some minutes (coffe break, go to piss, and so on..), usually he press the 'hold' button on the phone; if a new call arrive, asterisk send the call to the agent's phone, since it seems 'free'; but the phone is 'on hold'; and the caller don't hear 'anything' (no music, nothing at all) until the agent press (of course) again the button (but usually the caller hang up since he don't hear anything) there is a way to send the 'music on hold' to the caller even with the asterisk send the call to the phone (autoanswer on) but the 'hold' button is already pressed? I have to search/manage the asterisk config or the phone one? We are using asterisk 1.2.1 with Thomson ST2030. this is the asterisk log: (...) -- Executing Queue(CAPI/ISDN4/-ce, coda_azienda|t| 3600) in new stack -- Started music on hold, class 'music', on channel 'CAPI/ISDN4/ **-ce' -- agent_call, call to agent '1005' call on 'SIP/barbaran-621c' -- Playing 'beep' (language 'it') -- Called Agent/1005 -- Agent/1005 answered CAPI/ISDN4/**-ce [WARNING: in the truth the Agent is in hold mode now; there is the autoanswer on] -- Playing 'wsa_from_coda_w' (language 'it') Apr 11 12:23:17 NOTICE[13534]: res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! -- Stopped music on hold on CAPI/ISDN4/*-ce [AND NOW THE CALLER DON'T HEAR ANYTHING until the agent will press the hold button again] I'd like to send some music to the caller now... :( thank you in advance for all! bye bye MAS! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DISABLE 9?
JNA wrote: Is there a way to make it so you do not have to dial 9 by default to dial a outside number? I would like it if we could just dial the number any pointers? the asterisk dialplan matches most specific entries first. So you could have one set for one or two ditgit internal numbers, one set for 7 digit local numbers, one set for 10 digit national numbers and one set for n digit international numbers all starting with an international prefix. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel/ssh interaction
On 4/15/07, Gordon Henderson [EMAIL PROTECTED] wrote: That looks fine to me. (if a little busy, if this were a production server, I'd go into the BIOS and disable as much as I could and compile up a custom kernel) Also maybe remove the sound card for a test just to see if that makes any difference. Although IRQ wouldn't seem to be the problem, I see no other reason Zaptel and ssh would conflict so the logic remains that since IRQ and other hardware hassles can wreak random havoc... I'd strip the hardware down to the minimum and turn off unused services as stated above. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] DISABLE 9?
Have you never run into a situation where you dial +15705551212 for a number, but also have an extention of 157 or something? The 9 is legacy, yes, but still important, in my opinion, to segregate the networks. You know that anything starting with a 9 is going to go outbound, and all of your extentions are then 1xx-8xx. 9anything is reserved for going to the PSTN. Otherwise, you are either going to have to have your callers dial 1areacode for everything (and then have your extentions 2xx-9xx), that is they can't just dial 5551212, which is a pain, or you are going to have overlap. The 9 may be legacy, but it is somewhat important! On 4/15/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi everybody ! I never use any prefix number to dial out. I prefer to do like any standard residential subscriber, not to force somebody to think : Oh no ! I have forgottent to input the 9 - or 0 - before to dial out !. Directly inputing the real number is more natural. Adding a prefix is an old way to go inherited from analog PABX integrators ;-) If a customer want that, ok, do it, to avoid to have to change his habits. If you have no obligation to do that, forget it ! Think a good dialplan instead of that... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de JNA Envoyé : dimanche 15 avril 2007 11:49 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : [asterisk-users] DISABLE 9? Is there a way to make it so you do not have to dial 9 by default to dial a outside number? I would like it if we could just dial the number any pointers? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] DISABLE 9?
The 9 may be legacy, but it is somewhat important! It's also geo or culture specific. I've been in may offices where you dial 0 for outside. I have a system in our office where there are provider codes to force use of one specific one, access codes that will give a ZAP dialtone of one of two lines and detection of anything else that is dialed as local extensions, long distance ones, cellphones (rates vary so LCR is important), etc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel/ssh interaction
Two longshots: On Thu, Apr 12, 2007 at 05:02:23PM -0600, Greg Woods wrote: I hope I don't get flamed the first time I post to a new list. I have spent a couple of hours poking around without seeing anything like this. The problem is, as soon as I load the Zaptel drivers (with a TDM-31B card), ssh into or out of the server is broken. Trying to ssh in, I get: RSA_public_decrypt failed: error:0407006A:rsa routines:RSA_padding_check_PKCS1_type_1:block type is not 01 key_verify failed for server_host_key If I try to ssh out, I get: hash mismatch key_verify failed for server_host_key This makes administering the server remotely impossible, so it's a fairly large problem for me right now. Anybody ever seen anything like this? It is easy to reproduce: modprobe zaptel and it's broken. Just zaptel? On its own? Do you see any special messages on dmesg? And if you modprobe zaptel with debug=1 ? Zaptel on its own should not cause problems if there are no spans. Are you sure no module is loaded? lsmod | grep zaptel If you run sshd on a different port in debug mode, what do you see? sshd -d -p And possibly -dd or -ddd . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fast busy on TDM400P
DR, Is the failed call scenario **always** terminating on the same X100M? IME, it's rarely the Telco line card at fault. :). John Treble From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David W. Rice Sent: April 14, 2007 7:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Fast busy on TDM400P Dont have an extra machine at the site. Im starting to suspect the FXO module or the telco. One FXO module hooked to another analog trunk works flawlessly. When the client is in the office next, Im going to have them switch the cables. Any other ideas in the meantime? Thanks, DR From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Segalowitz Sent: Saturday, April 14, 2007 3:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Fast busy on TDM400P for kicks lets try setting it up on another machine, if you have an extra one and see if it still does it. Astawerks VoIP Hardware sales and consulting http://www.astawerks.com 614-495-1400 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David W. Rice Sent: Saturday, April 14, 2007 1:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Fast busy on TDM400P I did take the NoOp out and there was no difference. *1.2.17 and zaptel 1.2.16. Thanks! DR From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Segalowitz Sent: Saturday, April 14, 2007 1:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Fast busy on TDM400P Did you try taking NoOP out? does it work fine then? What version of Linux are you using and what version of asterisk? Astawerks VoIP Hardware sales and Consulting www.astawerks.com 614-495-1400 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David W. Rice Sent: Saturday, April 14, 2007 12:38 PM To: [EMAIL PROTECTED] Subject: [asterisk-users] Fast busy on TDM400P Im getting a random fast busy signal after two rings on an incoming ZAP channel through an FXO interface on a TDM400P. I have a NoOp in the dial plan showing the callerID. When the random fastbusy happens, no CallerID is sent. When it works, the CallerID information is sent. Is this a card issue or a telco issue? I have a very upset client and need to figure out how to resolve this. Thanks, DR -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/760 - Release Date: 4/13/2007 8:04 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/760 - Release Date: 4/13/2007 8:04 PM -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/760 - Release Date: 4/13/2007 8:04 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DISABLE 9?
On Sunday April 15 2007 5:48 am, JNA wrote: Is there a way to make it so you do not have to dial 9 by default to dial a outside number? I would like it if we could just dial the number any pointers? In a number of my ATA's and IP Phones I have a delay in the pattern match so that if the user dials 4 digits the phone waits for 1 second to see if there will be a 5th or more digits. This eliminates the need to dial a 9 or a 0 to get outside dial tone. Yes 1 second can be a long time but if you don't make a big deal over it when talking to the users they usually do not notice and are more focused on the fact that they do not have to do anything to distinguish between extension dial and outside dial. John M ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing Applications
Ok Eric, Thank you again. Ronaldo On 4/14/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Ronaldo Zacarias Afonso wrote: Hi all, I was trying to set up a conference room using the MeetMe application and my asterisk is telling me that there is no MeetMe application available for the extension I've dialed. [Apr 14 20:57:11] WARNING[958] pbx.c: No application 'MeetMe' for extension (internal, 600, 1) So, I issued the command core show applications and it didn't show me a MeetMe application. I'd like to know how I can install that application or what I have to do to make it available for use. Thanks. MeetMe requires Zaptel to be installed. Install Zaptel, then rebuild and install Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] DISABLE 9?
Matt wrote: Have you never run into a situation where you dial +15705551212 for a number, but also have an extention of 157 or something? Of course, bu then again, a properly designed dialplan will have more specific entries for internal numbers _XXX which will match 157 but not 15705551212, so if you'd dial 15705551212, asterisk will have to find a less specific entry in your dialplan to match that to. The 9 is legacy, yes, but still important, in my opinion, to segregate the networks. You know that anything starting with a 9 is going to go outbound, and all of your extentions are then 1xx-8xx. 9anything is reserved for going to the PSTN. Otherwise, you are either going to have to have your callers dial 1areacode for everything (and then have your extentions 2xx-9xx), that is they can't just dial 5551212, which is a pain, or you are going to have overlap. The 9 may be legacy, but it is somewhat important! The 9 is legacy, american and IMNSHO completely obsolete. You can leave it in your dialplan as not to upset those users used to dialing a 9 for an outside line, but apart from that, it has no use anymore. -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] saydigits in another language
I want to rerecord the 1 2 3 ... 0 sounds, but not overwrite the defaults. So, I've recorded them into a custom directory /var/lib/asterisk/sounds/custom I was hoping to be able to do the following: exten = foo,1,Set(CHANNEL(language)=custom) exten = foo,2,SayDigits(1234567890) however, I get no errors, but still get the default Allison sounds for the digits. Anyone got any clues on what I'm doing wrong ? TIA Julian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail - digits/1F does not exist in any format
Carlos Chavez wrote: I am assuming you are using a language other that English? If so, do you have the language files installed in the correct place? For asterisk 1.2 you need a structure like this: No, I'm using English. The default setup that came with 1.4.1. The other sound files are in /var/lib/asterisk/sounds/digits: -rw-rw-r-- 1 per 1000 1353 Feb 20 23:05 0.gsm -rw-rw-r-- 1 per 1000 1089 Feb 20 23:05 1.gsm -rw-rw-r-- 1 per 1000 1023 Feb 20 23:05 10.gsm -rw-rw-r-- 1 per 1000 1353 Feb 20 23:05 11.gsm -rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 12.gsm -rw-rw-r-- 1 per 1000 1485 Feb 20 23:05 13.gsm -rw-rw-r-- 1 per 1000 1485 Feb 20 23:05 14.gsm -rw-rw-r-- 1 per 1000 1518 Feb 20 23:05 15.gsm -rw-rw-r-- 1 per 1000 1617 Feb 20 23:05 16.gsm -rw-rw-r-- 1 per 1000 1782 Feb 20 23:05 17.gsm -rw-rw-r-- 1 per 1000 1551 Feb 20 23:05 18.gsm -rw-rw-r-- 1 per 1000 1650 Feb 20 23:05 19.gsm -rw-rw-r-- 1 per 1000 990 Feb 20 23:05 2.gsm -rw-rw-r-- 1 per 1000 1254 Feb 20 23:05 20.gsm -rw-rw-r-- 1 per 1000 990 Feb 20 23:05 3.gsm -rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 30.gsm -rw-rw-r-- 1 per 1000 1089 Feb 20 23:05 4.gsm -rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 40.gsm -rw-rw-r-- 1 per 1000 1122 Feb 20 23:05 5.gsm -rw-rw-r-- 1 per 1000 1419 Feb 20 23:05 50.gsm -rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 6.gsm -rw-rw-r-- 1 per 10000 Feb 20 23:05 60.gsm -rw-rw-r-- 1 per 1000 1320 Feb 20 23:05 7.gsm -rw-rw-r-- 1 per 1000 1485 Feb 20 23:05 70.gsm -rw-rw-r-- 1 per 1000 891 Feb 20 23:05 8.gsm -rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 80.gsm -rw-rw-r-- 1 per 1000 1254 Feb 20 23:05 9.gsm -rw-rw-r-- 1 per 1000 1419 Feb 20 23:05 90.gsm /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] DISABLE 9?
On Sun, 15 Apr 2007, Matt wrote: Have you never run into a situation where you dial +15705551212 for a number, but also have an extention of 157 or something? The 9 is legacy, yes, but still important, in my opinion, to segregate the networks. You know that anything starting with a 9 is going to go outbound, and all of your extentions are then 1xx-8xx. 9anything is reserved for going to the PSTN. Otherwise, you are either going to have to have your callers dial 1areacode for everything (and then have your extentions 2xx-9xx), that is they can't just dial 5551212, which is a pain, or you are going to have overlap. The 9 may be legacy, but it is somewhat important! I've written about my situations before, but made the decision that I'd not force the dial-9 thing on my clients (unless they specifically asked for it!) I'm in the UK though, so things might not work this way elsewhere! Essentially, I treat numbers starting with 0 as outside numbers, and I include that leading zero in the outgoing number. Punters can still dial 0 to get the local operator though - this is handled correctly in the dialplan; exten = 0,1,Noop(Calling the Operator) has a higher priority than: exten = _0.,1,Noop(Outside line request: Dialled 0...) In the UK, we (still) have the concept of a local number and a national number - for a local number we don't need to dial the STD code (area code), however for some years now, it's not mattered if we do dial the STD code, so the issue with the above is that you need to dial the full number (including your local STD code, if it's a local call). I personally see this as no big deal - we've had to do this on mobiles for many years now, and incoming caller ID has always had the full number presented for some time, so if you have existing phones with displays and phone books then they just work when migrated over to PBXs. Additionally, a lot of my customers are migrating from a phone and a fax type scenarios to having a PBX and they're simply not used to dialling 9... There are cases where you need to dial 9 though. To call some local services - eg. 1570, (BT VoiceMail thing), 150 (faults) 118xxx (directory enquiries), so for these I have provided the 9 prefix, although I could simply hard-wire them into the dialplan (which I do for 999) Going back to the OPs request - I think you need to learn to write the dialplan rather than (I suspect) rely on some GUI doing it for you - or atleast work out how to compliment the GUI (or whatever is generating your dial-plan) with your own additions. Get the Asterisk book - either buy it off Amazon, or get the PDF - links to it have been posted here recently, so search the archives! Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : RE : [asterisk-users] DISABLE 9?
Hello again, They are many Asterisk servers outside of the US that use a different national plan... Here, in France, we are using _0Z for fixed national telephones lines, including _06 for national mobiles, _08 special (often higher) price calls, _00Z. For international calls, and few _XX and _ as specific services as Police, Firemen, our historical TELCO, some data only destinations, etc... Near all modern ATA, gateways, IP-Phones, Softphones are using delay before to send the complete numbered destination. The only problem could be a not so well built dialplan if you are using also legacy analog telephones behind Asterisk's FXS interfaces. What I was tempting to explain before (sorry if I was not so clear) is that it's possible to do as one wants, depending the target or volontee. This is why Asterisk is so powerfull and what I love it (realy) ;-) This was not to start a flamme war, sorry ! PS : hello to my friend Wilson ! Best Regards, Francois BERGERET, France. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf and blind xfer
I was wanting to automate entirely a blind transfer. We are not yet using a powerdialler, so when we hit an answermachine we have to manually leave a message. In order to make this a little quicker, I want to leave a standard message on the answermachine. attempt #1. Use the blind transfer feature. set blind transfer to be **. extension 22 is exten 22 = Goto (leavemessage,answermachine,1) during the call, agent presses **. Asterisk says transfer. Agent enters 22#. Call is transferred to extension 22. A Standard Message is left. cool. Agents say We always have to xfer to extension 22. Can't you automate that? attempt #2. Use the features application map doxfer = **,caller,goto,leavemessage|answermachine|1 during the call, agent presses **. CLI says Feature Found: doxfer exten: answermachine -- Goto (leavemessage,answermachine,1) but nothing happens. I've tried it with **,callee as well. Anyone been able to automate this type of thing ? Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] saydigits in another language
Julian Lyndon-Smith wrote: however, I get no errors, but still get the default Allison sounds for the digits. Anyone got any clues on what I'm doing wrong ? 1) Create a directory named your_country_iso_code (AR|MX|ES|ETC) [1] under the main sounds directory (/var/lib/asterisk/sounds/ ???); 2) Also remember to create the same subdirectory under every other main directory (letters, digits, phonetic etc); 3) Copy/move the newly recorded messages into these new directories - numbers into digits. exten = foo,1,Set(CHANNEL(language)=custom) exten = foo,2,SayDigits(1234567890) Instead of custom use the ISO code. [1] [1] http://preview.tinyurl.com/btkp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel/ssh interaction
Thanks to everyone so far for the suggestions; I have a few things to try. Some things I want to note: 1) Yes, I am certain that when I load and unload the modules, they are really loaded or unloaded. I did check with lsmod. I am also certain that the problems I am having are 100% correlated with whether the zaptel drivers are loaded. No driver, no problem, load driver, problems return. 2) I do not have an IRQ conflict (at least not directly; no shared interrupt numbers in /proc/interrupts) 3) The zaptel drivers do function properly when loaded. Asterisk works with my Digium card. 4) SSH itself works fine. It is only the initial key negotiation that fails when zaptel is loaded. If I stop asterisk and unload zaptel, then I can ssh in, and if I then reload zaptel and restart asterisk, my already-established ssh session works fine. The same thing happens with the ipsec-tools/racoon tunnel. If I start racoon on both ends and let it set up the SA, I can then reload zaptel and asterisk and the tunnel works fine. When the SA times out and racoon has to renegotiate the SA, it fails with a similar type of message, as though the packets it is seeing are malformed. 5) All other network-related traffic works fine, whether or not zaptel is loaded. It is only things related to RSA key negotiation that fail. 6) The zaptel driver also affects the sound. I get extraneous beeps and pops in the playback stream. Some things I will try: 1) Zaptel 1.4.1 2) See if Zaptel 1.4.1 will build with a newer kernel, and if so, whether the newer kernel makes a difference. 3) Disabling non-essential stuff in the BIOS (I'm pretty sure I already did that, but I do see a floppy in my interrupts list even though the system doesn't have a floppy drive installed, so I will check) 4) Unloading the sound driver and/or removing the sound card (#6 above suggests this might help) I will report back on the results. Thank you all very much for the suggestions. This is a problem that I must find a solution to. --Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel/ssh interaction
On Sun, Apr 15, 2007 at 09:40:35AM -0600, Greg Woods wrote: Thanks to everyone so far for the suggestions; I have a few things to try. Some things I want to note: 1) Yes, I am certain that when I load and unload the modules, they are really loaded or unloaded. I did check with lsmod. I am also certain that the problems I am having are 100% correlated with whether the zaptel drivers are loaded. No driver, no problem, load driver, problems return. The thing I specifically wondered about is that zaptel on its own causes problems (without modules that actually generate spans: ztdummy, wctdm, wcfxo, etc.) 2) I do not have an IRQ conflict (at least not directly; no shared interrupt numbers in /proc/interrupts) 3) The zaptel drivers do function properly when loaded. Asterisk works with my Digium card. 4) SSH itself works fine. It is only the initial key negotiation that fails when zaptel is loaded. If I stop asterisk and unload zaptel, then I can ssh in, and if I then reload zaptel and restart asterisk, my already-established ssh session works fine. The same thing happens with the ipsec-tools/racoon tunnel. If I start racoon on both ends and let it set up the SA, I can then reload zaptel and asterisk and the tunnel works fine. When the SA times out and racoon has to renegotiate the SA, it fails with a similar type of message, as though the packets it is seeing are malformed. So the problem is with /dev/random (the entropy pool)? What happens if you replace /dev/random with a link to /dev/urandom ? What do you get from an strace of the sshd around thetime it hangs? 5) All other network-related traffic works fine, whether or not zaptel is loaded. It is only things related to RSA key negotiation that fail. 6) The zaptel driver also affects the sound. I get extraneous beeps and pops in the playback stream. Not sure how this is related. Maybe this is aa separate issue. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Job listing on cisco.com for Asterisk...?
Replace your first word with Adtran and I think you will be predicting the future. Thanks, Steve Dean Collins wrote: NewsFlash Cisco Acquires Digium for $1.4 Gazillion dollars Mark Spencer seen flying off in a lear jet en-route for Barbados. * Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Friday, 13 April 2007 7:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Job listing on cisco.com for Asterisk...? I thought this was interesting, if you are in China and need a job, you might also... http://www.cisco.apply2jobs.com/index.cfm?fuseaction=mExternal.showJobR ID=7 71671CurrentPage=1 * Working knowledge : Asterisk PBX; SIP Proxy Servers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue report problem
Where can I get the meaning of each field in queue_log? On 4/15/07, Darryl Dunkin [EMAIL PROTECTED] wrote: You will probably find what you are looking for here: /var/log/asterisk/queue_log -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango Sent: Saturday, April 14, 2007 21:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] queue report problem HI all, I have a queue say 5000 and there are 10 member in the queue. When there is a call to the queue, the members will ring according to the defined strategy. In day end, I have to create a report about the queue and its member. But I found that it is very difficult to find the relation for the call to queue and the member who pick the call in CDR. Say, caller A calls the queue, queue member 9 pick the call. I want to know the caller A waiting time, conversion time for Caller A and member 9. Such relationship is very difficult to find in CDR. Anyone have such experience and how can I get such information? ango ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 301 questions
Lee Jenkins wrote: Hi all, I just purchased a Polycom 301 for my home office and I believe I have it setup correctly as I can dial out, receive calls in, etc. However, I'm having the following issue: When calling a local number over a Zap line, I hear a lot of feed back on the line. I had a Grandstream configured with the same information before I got the 301 and never had that kind of feedback noise. I've tried playing with rxgain/txgain and it doesn't seem to make a difference. If any one could offer a pointer on where to look to resolve this issue, I'd very much appreciate it. Thanks, Solved. It turns out that it was the lines coming into my home from Verizon. I plugged a analog phone directly into the terminating jack and can hear the humming noise quite clearly. I never heard it until I setup the Polycom. My guess is that the GrandStream is either too great of a phone or too poor quality of a phone (i'll let you determine that one) to have picked up the noise like the Polycom did. This is why I never checked the line directly, I assumed it was the new phone since nothing else had changed. On the plus side, playing with the rx/txgain settings, Asterisk did minimize the noise pretty well. ;) -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call tranfer drops 1st. digit
Hi list, I experiencing a strange behaviour when transferring a call. The use case is like this: - Incoming call from Zap/1-1 - Routed to SIP phone SIP/1001 - The called user (SIP/1001) wants to redirect the call and presses # - IVR (default setup) says Transfer and user gets dial tone - User dials 1002 - IVR says No such extension - please try again ??? It seems that the 1st digit gets canceled out? Debugging the server output I get (tried twice): snip -- Goto (incoming,s,70) -- Executing Goto(Zap/1-1, sip_incoming|s|1) in new stack -- Goto (sip_incoming,s,1) -- Executing Dial(Zap/1-1, SIP/1001||rtT) in new stack -- Called 1001 ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available -- SIP/1001-08d8c668 is ringing -- SIP/1001-08d8c668 answered Zap/1-1 -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on Zap/1-1 -- Unable to find extension '' in context 'local_extensions' -- Playing 'pbx-invalid' (language 'en') -- parse_srv: SRV mapped to host alpha2.callcentric.com, port 5060 -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on Zap/1-1 -- Unable to find extension '001' in context 'local_extensions' -- snip The extension.conf: snip -- [local_extensions] include = outgoing ; Local extensions exten = 1001,1,Dial(SIP/1001,20,rtT) exten = 1002,1,Dial(SIP/1002,20,rtT) exten = 1003,1,Dial(SIP/1003,20,rtT) -- snip ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue report problem
Here: http://www.voip-info.org/wiki/view/Asterisk+log+queue_log On 4/15/07, Rilawich Ango [EMAIL PROTECTED] wrote: Where can I get the meaning of each field in queue_log? On 4/15/07, Darryl Dunkin [EMAIL PROTECTED] wrote: You will probably find what you are looking for here: /var/log/asterisk/queue_log -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango Sent: Saturday, April 14, 2007 21:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] queue report problem HI all, I have a queue say 5000 and there are 10 member in the queue. When there is a call to the queue, the members will ring according to the defined strategy. In day end, I have to create a report about the queue and its member. But I found that it is very difficult to find the relation for the call to queue and the member who pick the call in CDR. Say, caller A calls the queue, queue member 9 pick the call. I want to know the caller A waiting time, conversion time for Caller A and member 9. Such relationship is very difficult to find in CDR. Anyone have such experience and how can I get such information? ango ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] transfering calls
Hello folks! I've added a tT statement in my extensions (softphones and hardphones) so both the caller and calling parts can transfer calls (the strange thing is that some extensions end up only transfering the calls they originated, even if the config is the same of all other working extensions, which are able to transfer both placed and received calls). However, my trouble comes with the PSTN. I've added a t on the POTS- incoming s extension, so when it dial an extension, that extension can transfer the PSTN to someone else. Similarly, I've add a T to the outgoing calls Dial, so I can transfer a call to another extension after I place it. However, what happens is this: when I transfer a PSTN call to an extension, the PSTN gains the ability to transfer it! How I can prevent that to happen? Cheers, Francis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue report problem
Queuemetrics is a nice commercial product if you want a very nice UI with all sorts of reporting and functionality including live monitoring and VNC. They also have a free script that will load your queue_log file into a database. Then you can run queries quite easily. If you do any of that, I suggest you put the DB on a separate box than your production Asterisk server. Thanks, Steve Yossi Ben Hagai wrote: Here: http://www.voip-info.org/wiki/view/Asterisk+log+queue_log On 4/15/07, *Rilawich Ango* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Where can I get the meaning of each field in queue_log? On 4/15/07, Darryl Dunkin [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You will probably find what you are looking for here: /var/log/asterisk/queue_log -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Rilawich Ango Sent: Saturday, April 14, 2007 21:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] queue report problem HI all, I have a queue say 5000 and there are 10 member in the queue. When there is a call to the queue, the members will ring according to the defined strategy. In day end, I have to create a report about the queue and its member. But I found that it is very difficult to find the relation for the call to queue and the member who pick the call in CDR. Say, caller A calls the queue, queue member 9 pick the call. I want to know the caller A waiting time, conversion time for Caller A and member 9. Such relationship is very difficult to find in CDR. Anyone have such experience and how can I get such information? ango ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax with Asterisk + Hylafax
Hi, Anybody lucky with this config inside an Asterisk server for dealing with FAX ? FXO_LINE ASTERISK 1.4.2 --- IAXMODEM -- HYLAFAX TDM400PZAPTEL 4.3.1 1 FXO port 1.41 I know Fax is not officially supported on TDM400P cards but I did not expect not being able of sending one single Fax. Actually when I try to send a Fax, the call is established between my * server and the remote Fax but after 30 secs Asterisk disconnects the call and Hylafax reports NO CARRIER DETECTED. Tried playing around with a few parameters such as no echocancellation, alaw (also slinear) codec, faxdetection =incoming in zaptel but with no luck. Regards, Jose Limeres ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax with Asterisk + Hylafax
From: Jose Limeres [EMAIL PROTECTED] Date: Sun, 15 Apr 2007 19:10:58 +0100 Hi, Anybody lucky with this config inside an Asterisk server for dealing with FAX ? FXO_LINE ASTERISK 1.4.2 --- IAXMODEM -- HYLAFAX TDM400PZAPTEL 4.3.1 1 FXO port 1.41 Search Asterisk forum. Yes, somebody posted positive results. Yuan Liu I know Fax is not officially supported on TDM400P cards but I did not expect not being able of sending one single Fax. Actually when I try to send a Fax, the call is established between my * server and the remote Fax but after 30 secs Asterisk disconnects the call and Hylafax reports NO CARRIER DETECTED. Tried playing around with a few parameters such as no echocancellation, alaw (also slinear) codec, faxdetection =incoming in zaptel but with no luck. Regards, Jose Limeres ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] LEDs on Polycom Expansion Modules misbehave when paging
I have a 601 with 3 expansion modules watching 39 buddies. When we page as per extensions.conf: exten = 560,1, SIPAddHeader(Alert-Info: Ring Answer) exten = 560,n,Pager(SIP/phone1SIP/phone2 ...removed... SIP/phone40) exten = 560,n,Hangup() When the receptioist hangs up, some of the LEDs have failed to turn on and some randomly fail to turn off. We are running sip version 1.6.6.0039 Has anyone else experienced this or have advice on how to make it more reliable? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel/ssh interaction
On Sun, 2007-04-15 at 19:07 +0300, Tzafrir Cohen wrote:
The thing I specifically wondered about is that zaptel on its own causes
problems (without modules that actually generate spans: ztdummy, wctdm,
wcfxo, etc.)
You are correct: I can load zaptel by itself and I do not see the
problem. I determined by trial and error that the minimal set of modules
that I need in order for my TDM-31B card to work are wctdm24xxp and
wctdm. Unfortunately, with only these two loaded, I still have the
problem. Is there a way to use this knowledge to my advantage?
So the problem is with /dev/random (the entropy pool)?
I had wondered whether or not this had something to do with it, if only
because I couldn't think of anything else that would cause only RSA
negotiation to fail while everything else still worked.
What happens if you replace /dev/random with a link to /dev/urandom ?
I tried that, but it had no effect: same problem.
What do you get from an strace of the sshd around thetime it hangs?
The problem is not specific to sshd, since outbound ssh exhibits the
same problem. So I tried strace there, and it doesn't help a whole lot.
Here's what the tail of the strace output looks like when it exhibits
the problem:
open(/root/.ssh/known_hosts, O_RDONLY|O_LARGEFILE) = 4
fstat64(4, {st_mode=S_IFREG|0644, st_size=1484, ...}) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1,
0) = 0xb7
f3e000
read(4, portal,192.168.1.251 ssh-rsa AAA..., 4096) = 1484
close(4)= 0
munmap(0xb7f3e000, 4096)= 0
write(2, hash mismatch\r\n, 15) = 15
write(2, key_verify failed for server_hos..., 39) = 39
exit_group(255) = ?
It reads the known_hosts file, then the next thing it does is crap out.
This means the computational error is occurring somewhere in user space,
due to god knows what that happens earlier. It does open and apparently
successfully read 32 bytes from /dev/urandom previous to this.
Without zaptel, this output looks like:
open(/root/.ssh/known_hosts, O_RDONLY|O_LARGEFILE) = 4
fstat64(4, {st_mode=S_IFREG|0644, st_size=1484, ...}) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1,
0) = 0xb7
fde000
read(4, portal,192.168.1.251 ssh-rsa AAA..., 4096) = 1484
close(4)= 0
munmap(0xb7fde000, 4096)= 0
write(3, \0\0\0\f\n\25\0\0\0\0\0\0\0\0\0\0, 16) = 16
write(3, \6\213\320M\236\315\\321|\212\327%\252\235\3\251A\261...,
48) = 48
File descriptor 3 is the previously-opened socket to port 22, and things
proceed normally from there. I'm not saying the evidence of what is
going on isn't in these strace outputs somewhere, I just don't know what
I should look for.
[sound issue]
Not sure how this is related. Maybe this is aa separate issue.
Could be, but maybe not. I do intend to try pulling the sound card out
of the machine in case what I'm looking at is some sort of driver
conflict. I thought those were a thing of the past, but I remember old
Mac systems having this sort of problem frequently. First I'm going to
try the newer version of the zaptel driver, then (if it will compile) a
newer kernel, because that's easier to try (doesn't require pulling all
the cables and opening the box).
--Greg
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[asterisk-users] 601 Rebooting/Crashing seems to be due to full directory
We were adding contacts to the directory successfully and when we got top approx. contact #45, the phone now reboots when we change a speed dial index or delete a contact from the directory. To gain more memory in sip.cfg, I set: directory dir.local.volatile.2meg=0 dir.local.nonVolatile.maxSize.2meg=20 dir.local.volatile.4meg=1 dir.local.nonVolatile.maxSize.4meg=42 dir.local.volatile.maxSize=100/ The phone's rebooting behavior persisits. I know that changes will be wiped out on reboot, right now I'm just trying to stabilize the phone. Do I need to reformat the file system to clear the nvram or something like that? Running sip 1.6.6.0039 Any help is really appreciated! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
Yuan LIU wrote: From: Jose Limeres [EMAIL PROTECTED] Date: Sun, 15 Apr 2007 19:10:58 +0100 Hi, Anybody lucky with this config inside an Asterisk server for dealing with FAX ? FXO_LINE ASTERISK 1.4.2 --- IAXMODEM -- HYLAFAX TDM400PZAPTEL 4.3.1 1 FXO port 1.41 Search Asterisk forum. Yes, somebody posted positive results. Yuan Liu I have had great results via PRI - Asterisk - IAXModem - Hylafax Post your log results and configs. Thanks, Steve I know Fax is not officially supported on TDM400P cards but I did not expect not being able of sending one single Fax. Actually when I try to send a Fax, the call is established between my * server and the remote Fax but after 30 secs Asterisk disconnects the call and Hylafax reports NO CARRIER DETECTED. Tried playing around with a few parameters such as no echocancellation, alaw (also slinear) codec, faxdetection =incoming in zaptel but with no luck. Regards, Jose Limeres ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel/ssh interaction
Greg Woods wrote: It reads the known_hosts file, then the next thing it does is crap out. This means the computational error is occurring somewhere in user space, due to god knows what that happens earlier. It does open and apparently successfully read 32 bytes from /dev/urandom previous to this. Did you make _any_ changes to zconfig.h when you built Zaptel, or did you make any changes to the Makefile or specify any special compilation arguments? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LEDs on Polycom Expansion Modules misbehave when paging
J French wrote: We are running sip version 1.6.6.0039 http://1.6.6.0039 Has anyone else experienced this or have advice on how to make it more reliable? That firmware release is quite old; I suspect most people would at least upgrade to the latest 1.x series before spending time on a problem like this, if not all the way to 2.x. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HTTP Connection Timeout Trouble with Cisco 7960 Phone
You should run a packet capture. Verify that a request is sent to the server and then verify if you see a response come back. I assume you have a cgi that generates the xml needed to display on the phone. So either, your cgi is not responding or the response is not formatted properly for the phone. Also check to make sure that your http proxy settings are correct in the phone. It's possible that you have a proxy set that is incorrect or that you need to set one in order to get to your webserver. These types of scenarios can cause the problem you described. Good luck. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hennessy Sent: Saturday, April 14, 2007 12:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] HTTP Connection Timeout Trouble with Cisco 7960 Phone Hello, I'm using two Cisco 7960 phones currently loaded and showing Firmware POS3-07-4-0 (Version 7.4?) and I'm having a strange problem. Whenever the phone is supposed to try to load anything over HTTP from my Apache 2.2.x web server, the connection just sits and times out. Nothing shows up in the Apache logs unless I hit cancel. What could the trouble be? -- Mark P. Hennessy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Job listing on cisco.com for Asterisk...?
Roflol. The chance of that happening are slim to none. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Friday, April 13, 2007 7:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Job listing on cisco.com for Asterisk...? NewsFlash Cisco Acquires Digium for $1.4 Gazillion dollars Mark Spencer seen flying off in a lear jet en-route for Barbados. * Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Friday, 13 April 2007 7:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Job listing on cisco.com for Asterisk...? I thought this was interesting, if you are in China and need a job, you might also... http://www.cisco.apply2jobs.com/index.cfm?fuseaction=mExternal.showJobR ID=7 71671CurrentPage=1 * Working knowledge : Asterisk PBX; SIP Proxy Servers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
I searched extensively on this and did not find anything conclusive except the general understanding that on TDM cards every instalation is different whereas for BRI /PRI it should work. As for the config, Asterisk just works perfectly well with voice conversations over the FXO line. I add one IAX extension in * for the IAXModem and then in the IAXmodem configure the default modem as the one that is registered in *. If I make a Fax call to my mobile, it rings and I can hear the Fax on the other side. So, the problem I believe is in the negotiation/sinchronization where something is broken at some point. I would like to know wether it is possible to solve this adjusting some parameter in some config file or I should just forget it and use a digital line instead. Thanks, Jose L. CLI -- Accepting AUTHENTICATED call from 127.0.0.1: requested format = slin, requested prefs = (), actual format = alaw, host prefs = (alaw), priority = mine -- Executing [EMAIL PROTECTED]:1] Dial(IAX2/200-1, Zap/g1/956855858) in new stack -- Called g1/956855858 -- Zap/4-1 answered IAX2/200-1 -- Hungup 'Zap/4-1' == Spawn extension (default, 956855858, 1) exited non-zero on 'IAX2/200-1' -- Hungup 'IAX2/200-1' CLI faxstat -s JIDPriSOwnerNumber PagesDials TTS Status 35 126 Sroot 9568558580:1 1:12 11.19 No carrier detected On 15/04/07, Steve Totaro [EMAIL PROTECTED] wrote: Yuan LIU wrote: From: Jose Limeres [EMAIL PROTECTED] Date: Sun, 15 Apr 2007 19:10:58 +0100 Hi, Anybody lucky with this config inside an Asterisk server for dealing with FAX ? FXO_LINE ASTERISK 1.4.2 --- IAXMODEM -- HYLAFAX TDM400PZAPTEL 4.3.1 1 FXO port 1.41 Search Asterisk forum. Yes, somebody posted positive results. Yuan Liu I have had great results via PRI - Asterisk - IAXModem - Hylafax Post your log results and configs. Thanks, Steve I know Fax is not officially supported on TDM400P cards but I did not expect not being able of sending one single Fax. Actually when I try to send a Fax, the call is established between my * server and the remote Fax but after 30 secs Asterisk disconnects the call and Hylafax reports NO CARRIER DETECTED. Tried playing around with a few parameters such as no echocancellation, alaw (also slinear) codec, faxdetection =incoming in zaptel but with no luck. Regards, Jose Limeres ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel/ssh interaction (SOLVED!)
On Sun, 2007-04-15 at 14:26 -0500, Kevin P. Fleming wrote: Did you make _any_ changes to zconfig.h when you built Zaptel, or did you make any changes to the Makefile or specify any special compilation arguments? Another good thought. To be honest, I do remember poking around in zconfig.h but I don't know if I actually ended up changing anything or not. However, what I do know is that when I recompiled zaptel with 1.4.1 and installed that, the problem is gone. I don't know if this was due to changes I made in the 1.4.0 zconfig.h file, or that there were fixes in 1.4.1. --Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is STP wire decent for analog phones?
I've got a run of Shielded Twisted Pair (4 conductors) which used to be a Token Ring Network drop and I'm not using it anymore. Would it be decent to replace the ends with normal analog phone connectors and use it for a phone extension, or is STP unsuitable for that? Thanks Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware
Hi, I'm looking for IBM hardware to support: 100 SIP hard phone users 10 fax machines on SIP ata's maybe later an additional 100 sip soft phones. Initially, all calls will be through PRI. Some conferencing. Don't know yet if this will even get used. Using 1.4 + ( probably business edition ) I'm looking for anyone who some experience / gotchas. I've google'd and voip-info but there's so much contradicting info. I know it isn't easy to say this one or that one but based on your experience what you think will work with the above configuration? Maybe just a XEON 3.0 processor with PCI or PCI-X would point me in the right direction. Would I need a riser card for PCI-X for a digium card ? I'll keep looking but any help would be very much appreciated! TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Optipoint 420std SIP Firmware
Hello, Im looking for Optipoint 420 Standard SIP Firmware to make my first tests with Asterisk and SIP, but Im unable to find it. Could someone help me? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
You could try to get it working but it may never be 100%. If your needs are 100% then I suggest using a standard fax and get an analog line and do it the old fashioned way. If you need Hylafax type features then buy a modem that is compatible with Hylafax and run it on a different box. Just suggestions. Thanks, Steve Jose Limeres wrote: I searched extensively on this and did not find anything conclusive except the general understanding that on TDM cards every instalation is different whereas for BRI /PRI it should work. As for the config, Asterisk just works perfectly well with voice conversations over the FXO line. I add one IAX extension in * for the IAXModem and then in the IAXmodem configure the default modem as the one that is registered in *. If I make a Fax call to my mobile, it rings and I can hear the Fax on the other side. So, the problem I believe is in the negotiation/sinchronization where something is broken at some point. I would like to know wether it is possible to solve this adjusting some parameter in some config file or I should just forget it and use a digital line instead. Thanks, Jose L. CLI -- Accepting AUTHENTICATED call from 127.0.0.1 http://127.0.0.1: requested format = slin, requested prefs = (), actual format = alaw, host prefs = (alaw), priority = mine -- Executing [EMAIL PROTECTED]:1] Dial(IAX2/200-1, Zap/g1/956855858) in new stack -- Called g1/956855858 -- Zap/4-1 answered IAX2/200-1 -- Hungup 'Zap/4-1' == Spawn extension (default, 956855858, 1) exited non-zero on 'IAX2/200-1' -- Hungup 'IAX2/200-1' CLI faxstat -s JID Pri S Owner Number Pages Dials TTS Status 35 126 S root 956855858 0:1 1:12 11.19 No carrier detected On 15/04/07, *Steve Totaro* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Yuan LIU wrote: From: Jose Limeres [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Date: Sun, 15 Apr 2007 19:10:58 +0100 Hi, Anybody lucky with this config inside an Asterisk server for dealing with FAX ? FXO_LINE ASTERISK 1.4.2 --- IAXMODEM -- HYLAFAX TDM400P ZAPTEL 4.3.1 1 FXO port 1.41 Search Asterisk forum. Yes, somebody posted positive results. Yuan Liu I have had great results via PRI - Asterisk - IAXModem - Hylafax Post your log results and configs. Thanks, Steve I know Fax is not officially supported on TDM400P cards but I did not expect not being able of sending one single Fax. Actually when I try to send a Fax, the call is established between my * server and the remote Fax but after 30 secs Asterisk disconnects the call and Hylafax reports NO CARRIER DETECTED. Tried playing around with a few parameters such as no echocancellation, alaw (also slinear) codec, faxdetection =incoming in zaptel but with no luck. Regards, Jose Limeres ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware
I found that the IBM X306 works well with Digium hardware. Sangoma works well too. I run RAID 1 on the hotswap SATA drives. Good luck with the fax machines though. You may want to look at a channel bank (I like Adtran 600s). You have to use one of your T1 ports to connect to it but I think your results for faxing will be much better than trying to use ATAs. Hylafax is another option. Thanks, Steve [EMAIL PROTECTED] wrote: Hi, I'm looking for IBM hardware to support: 100 SIP hard phone users 10 fax machines on SIP ata's maybe later an additional 100 sip soft phones. Initially, all calls will be through PRI. Some conferencing. Don't know yet if this will even get used. Using 1.4 + ( probably business edition ) I'm looking for anyone who some experience / gotchas. I've google'd and voip-info but there's so much contradicting info. I know it isn't easy to say this one or that one but based on your experience what you think will work with the above configuration? Maybe just a XEON 3.0 processor with PCI or PCI-X would point me in the right direction. Would I need a riser card for PCI-X for a digium card ? I'll keep looking but any help would be very much appreciated! TIA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is STP wire decent for analog phones?
STP should work just fine, even quad wire should work in most cases. On 4/15/07, Steve Prior [EMAIL PROTECTED] wrote: I've got a run of Shielded Twisted Pair (4 conductors) which used to be a Token Ring Network drop and I'm not using it anymore. Would it be decent to replace the ends with normal analog phone connectors and use it for a phone extension, or is STP unsuitable for that? Thanks Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] saydigits in another language
Not that custom shouldn't work, but you just need to place them in sounds/digits/custom not sounds/custom On 4/15/07, Hermann Wecke [EMAIL PROTECTED] wrote: Julian Lyndon-Smith wrote: however, I get no errors, but still get the default Allison sounds for the digits. Anyone got any clues on what I'm doing wrong ? 1) Create a directory named your_country_iso_code (AR|MX|ES|ETC) [1] under the main sounds directory (/var/lib/asterisk/sounds/ ???); 2) Also remember to create the same subdirectory under every other main directory (letters, digits, phonetic etc); 3) Copy/move the newly recorded messages into these new directories - numbers into digits. exten = foo,1,Set(CHANNEL(language)=custom) exten = foo,2,SayDigits(1234567890) Instead of custom use the ISO code. [1] [1] http://preview.tinyurl.com/btkp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Optipoint 420std SIP Firmware
On 4/15/07, Germán Rodríguez Vergara [EMAIL PROTECTED] wrote: I'm looking for Optipoint 420 Standard SIP Firmware to make my first tests with Asterisk and SIP, but I'm unable to find it. Could someone help me? My understanding is Siemens sells an optiPoint 420 and an optiPoint 420S. The 420S is the SIP version and if that's what you have you should be able to obtain the firmware The non-SIP phones use Siemens proprietary protocol which is based on an old version of H.323. Perhaps its possbile to cross-flash the phones but I haven't found too much clandestine documentation for the Siemens gear. Recados, Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G723 problems with TC400B
Hello asteriskers, I hope someone could help me ... !! I bought a TC400B, and I am testing doing calls with G729 and G723. When I used G729 it works fine, but when I try to use G723 the RTP has very low quality and is not possible to hear to the other person in the phone. I try to find out the problem but the only weird that I saw in the debug info is this line (using asterisk 1.2.17): asterisk*CLI Apr 15 19:42:44 WARNING[17321] chan_zap.c: Frame too large Apr 15 19:42:48 WARNING[17321]: chan_zap.c:4931 zt_write: Frame too large This is my scenario: - 1 Intel(R) Core(TM)2 CPU (Core 2 Duo with 2.13GHz) - 1 Gigabit Memory - kernel 2.6.18-1.2798.fc6 x86_64 - OS: Fedora Core 6 I tried first with asterisk 1.4.1 and its dependencies but the problem is there, too much noise when someone speeaks (poor voice quality). In asterisk 1.2.17 and its dependencies (libpri, zaptel and addons) is a little better but the voice quality is not good anyway. In this scenario appears the Warning : Frame too large. The show transcoders and show translations looks fine in asterisk CLI: asterisk*CLI show transcoder 0/0 encoders/decoders of 92 channels (G.729a / G.723.1 5.3 kbps) are in use. asterisk*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - 4 1 1 3 3 2 5 2 -14 gsm 3 - 2 2 2 2 1 4 3 -13 ulaw 1 3 - 1 2 2 1 4 1 -13 alaw 1 3 1 - 2 2 1 4 1 -13 g726 3 3 2 2 - 2 1 4 3 -13 adpcm 3 3 2 2 2 - 1 4 3 -13 slin 2 2 1 1 1 1 - 3 2 -12 lpc10 4 4 3 3 3 3 2 - 4 -14 g729 2 4 1 1 3 3 2 5 - -14 speex - - - - - - - - - - - ilbc 4 4 3 3 3 3 2 5 4 - - asterisk*CLI After of this I upgrade the kernel to 2.6.20-1.2944.fc6 x86_64, and the problem remains. Any help will be so much appreciated. -- Jovanny Saravia Solutions Manager e-solutions Ltda [EMAIL PROTECTED] +57-310-7676163 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 sluggish keys: found the problem!
Mike wrote: Here is what I had to change on the phone1.cfg file: Which means we caught you red-handed! Remember when I asked you about whether you were using the default configs from the new firmware package? I had this value in my 1.6.7 file, put in there following suggestions from the Wiki (http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501) : *reg.1.server.1.expires=30* That tells me right there that you didn't use the default sip.cfg and phone1.cfg files. Now, this worked flawlessly with 1.6.7. But with 2.x, this seizes up the phone with a huge CPU load (approaching 100% at times) and makes it unresponsive. I had to remove it, fixing the problem. I have learned the hard way that using old configs with new firmware is asking for trouble. It is much better to keep your custom configurations in a MAC specific overrides file and replace the sip.cfg and phone1.cfg files completely. This doesn't guarantee that you won't have problems, but it's a lot easier to troubleshoot an overrides file with a dozen items in it than to sift through big, customized sip.cfg files. Now, I have two follow-up questions: 1) Is that normal? (my guess is no) Well, you shouldn't need to re-register every 30 seconds, either. What was the reason for configuring your phones like that? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SIP phones to buy?
Hi: Salvatore Giudice wrote: Product selection is not cut and dry. What are your business requirements? So you need encryption? If so, what kind? No. Do they need support for outbound proxies? No. Are you going to use the same model for remote deployments? Yes. Do you need WAP capabilities? No. Do you need programmable speed dials? Yes. Do you need modular admin sidecars? Maybe. Do you need IPSEC capabilities built into the handset? No. Do you need advanced/specific codec support? Wideband (I think that's G.729) is a nice-to-have. Do you need guaranteed interoperability with specific vendor supplied components? Not at the moment. (No) Are you looking for a phone for 10 people, 100 people, or 1 people? If you are scaling, what does your provisioning system look like? 10 - 250; TFTP or FTP-based provisioning. Do you need phone features like video or quality speaker phone? Quality speaker phone. No demand for video. What is your budget for phones? up to 300 CAD per unit, preferably around 200 CAD Do you need an RTCP capable handset? If I knew what that was... :) Do you need a handset that support 802.11p for QoS? No. Will that help narrow things down? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium TE205P and channelbank
Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the PRI any hints / tips greatfully received thanks to Steve Totaro for the nudge in the right direction! Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Job listing on cisco.com for Asterisk...?
Salvatore Giudice wrote: Roflol. The chance of that happening are slim to none. And Slim just left town. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Loudspeaker
Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agents and music on hold with autoanswer..
MAS! wrote: My colleague left our company, then I have to manage all our phones lines and asterisk: please, apologize me because I'm 'absolute beginner' about voip/asterisk!! Well... all seems work fine; we have some queues and some agents; the music on hold works fine when the agent press the hold button on the phone (thomson); the agents have the 'autoanwser' flag on. BUT if the agent have to go elsewhere for some minutes (coffe break, go to piss, and so on..), usually he press the 'hold' button on the phone; Oh my word. Does the phone have a DND (Do not disturb) button? Are all the agents trained to press hold when they need to go the bathroom? If the answer to the last question is yes, you have more than a technology problem on your hands. Perhaps this is why your colleague left in the first place :) if a new call arrive, asterisk send the call to the agent's phone, since it seems 'free'; but the phone is 'on hold'; and the caller don't hear 'anything' (no music, nothing at all) until the agent press (of course) again the button (but usually the caller hang up since he don't hear anything) there is a way to send the 'music on hold' to the caller even with the asterisk send the call to the phone (autoanswer on) but the 'hold' button is already pressed? I have to search/manage the asterisk config or the phone one? We are using asterisk 1.2.1 with Thomson ST2030. The Thomson is the telephone set? this is the asterisk log: (...) -- Executing Queue(CAPI/ISDN4/-ce, coda_azienda|t|3600) in new stack -- Started music on hold, class 'music', on channel 'CAPI/ISDN4/**-ce' -- agent_call, call to agent '1005' call on 'SIP/barbaran-621c' -- Playing 'beep' (language 'it') -- Called Agent/1005 -- Agent/1005 answered CAPI/ISDN4/**-ce [WARNING: in the truth the Agent is in hold mode now; there is the autoanswer on] (!) Why? (FYI: Auto answer is normally enabled in the telephone configuration and not in Asterisk.) -- Playing 'wsa_from_coda_w' (language 'it') Apr 11 12:23:17 NOTICE[13534]: res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! -- Stopped music on hold on CAPI/ISDN4/*-ce [AND NOW THE CALLER DON'T HEAR ANYTHINGuntil the agent will press the hold button again] Well, that's to be expected. The phone has answered the call! A few bits of advice to start: 1. Agents shouldn't be using hold for bathroom breaks. Most phones have a button specifically for this purpose called Do not disturb. Asterisk then treats the station as busy. 2. Queue phones shouldn't answer automatically. That's just inviting disaster. What if somebody forgets to log out when they leave? Somebody is going to get silence if they're unlucky enough to be connected to that agent. Fix both those things and you won't have to worry about Music On Hold not playing for the caller :) -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
Steve Totaro wrote: You could try to get it working but it may never be 100%. If your needs are 100% then I suggest using a standard fax and get an analog line and do it the old fashioned way. If you need Hylafax type features then buy a modem that is compatible with Hylafax and run it on a different box. It's not entirely clear to me why people continue to cling to the idea that Asterisk should handle faxing also. What's the benefit? Hylafax is great, and you can even use it on the same machine. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loudspeaker
Once, I nailed a SIP phone on a wall, up high. Not something I was proud of but it did work. PaulH On Mon, 2007-04-16 at 11:53 +1000, Klaverstyn, David C wrote: Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loudspeaker
Use an ATA to a paging system On 4/15/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loudspeaker
On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? Do you already have the loud speaker? If not, I know there are various vendors of extension phone bells that do nothing more than plug into an analog line and ring the nice loud bell when a ring signal is received. You could easily combine one of those with a cheap ATA with FXS port. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Loudspeaker
Have you taken a look at the cyberdata VoIP loudspeaker? http://www.cyberdata.net/ Astawerks VoIP Hardware sales and consulting http://www.astawerks.com 614-495-1400 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, April 15, 2007 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudspeaker Use an ATA to a paging system On 4/15/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/761 - Release Date: 4/14/2007 9:36 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/761 - Release Date: 4/14/2007 9:36 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Loudspeaker
I'm not sure how that could help. At the moment when a call comes in, every phone in the office rings. I would prefer a loudspeaker to ring so it is not in everyone's face so to speak but they are able to hear it in the background. I want to do this for after hour calls and then if no one answers go to a recorded message. The previous system before Asterisk did have a loud speaker that rang so I would prefer to keep it the same if possible. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, 16 April 2007 12:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudspeaker Use an ATA to a paging system On 4/15/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Loudspeaker
This is what I want. Do you have any URLs to such a device as I cannot find any. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cb Sent: Monday, 16 April 2007 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudspeaker On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? Do you already have the loud speaker? If not, I know there are various vendors of extension phone bells that do nothing more than plug into an analog line and ring the nice loud bell when a ring signal is received. You could easily combine one of those with a cheap ATA with FXS port. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
Stephen Bosch wrote: Steve Totaro wrote: You could try to get it working but it may never be 100%. If your needs are 100% then I suggest using a standard fax and get an analog line and do it the old fashioned way. If you need Hylafax type features then buy a modem that is compatible with Hylafax and run it on a different box. It's not entirely clear to me why people continue to cling to the idea that Asterisk should handle faxing also. What's the benefit? Hylafax is great, and you can even use it on the same machine. -Stephen- I could have sworn that is what I just said. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SIP phones to buy?
Stephen Bosch wrote: Hi: Salvatore Giudice wrote: Product selection is not cut and dry. What are your business requirements? So you need encryption? If so, what kind? No. Do they need support for outbound proxies? No. Are you going to use the same model for remote deployments? Yes. Do you need WAP capabilities? No. Do you need programmable speed dials? Yes. Do you need modular admin sidecars? Maybe. Do you need IPSEC capabilities built into the handset? No. Do you need advanced/specific codec support? Wideband (I think that's G.729) is a nice-to-have. Do you need guaranteed interoperability with specific vendor supplied components? Not at the moment. (No) Are you looking for a phone for 10 people, 100 people, or 1 people? If you are scaling, what does your provisioning system look like? 10 - 250; TFTP or FTP-based provisioning. Do you need phone features like video or quality speaker phone? Quality speaker phone. No demand for video. What is your budget for phones? up to 300 CAD per unit, preferably around 200 CAD Do you need an RTCP capable handset? If I knew what that was... :) Do you need a handset that support 802.11p for QoS? No. Will that help narrow things down? -Stephen- Polycom 301 or 501 (probably 501 since you need speakerphone and the 301 has a great speaker but no mic) Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Loudspeaker
Ok. What do you think about this? http://www.astawerks.com/-p-504.html Astawerks VoIP Hardware sales and consulting http://www.astawerks.com 614-495-1400 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Sunday, April 15, 2007 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Loudspeaker This is what I want. Do you have any URLs to such a device as I cannot find any. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cb Sent: Monday, 16 April 2007 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudspeaker On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? Do you already have the loud speaker? If not, I know there are various vendors of extension phone bells that do nothing more than plug into an analog line and ring the nice loud bell when a ring signal is received. You could easily combine one of those with a cheap ATA with FXS port. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/761 - Release Date: 4/14/2007 9:36 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/761 - Release Date: 4/14/2007 9:36 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE205P and channelbank
Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? In my experience, I have never had an issue with faxes and Digium's cards, but I'm sure many people will beg to differ. also, would I need a crossover to the channelbank or is it a patch lead like the connection to the PRI You will most likely need a crossover cable. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Job listing on cisco.com for Asterisk...?
Stephen Bosch wrote: Salvatore Giudice wrote: Roflol. The chance of that happening are slim to none. And Slim just left town. -Stephen- Guess who is still in town? http://www.mapquest.com/directions/main.adp?go=1do=nwrmm=1un=mcl=ENqq=1ADqpk24ofBjOvyL2FGh5cLj2cVnsS0Ywgct2G85dgsy9E%252bpX4AYuZ4SpDuv%252bzmFrBWeM8pXsBKt22m6iCRj%252brQBZbZ1N2e7VoyNfA%252fYWJAfxH3EJdQJu2QYiJyhiFzpBVdkIBHF9ASj%252bOwVDXdrfHT0dcutJ8N0a4dFxoYt%252fw8tI5Se6r5Qjegux%252byHVk0dHCUkQWbKErctEeO4EOgo6o97Rom0O9V0AYkxlowpmzbKNOFtXryhrYokMvav80gV2cj2cc7uNWJlGsxFaycR%252bVqEwQkGaG%252bqyKPXaADkVr46EALxgUyZ6VP3PaEfdRLbct=NArsres=11y=US1ffi=1l=1g=1pl=1v=1n=1pn=1a=150+W+Park+Loop+Nw1c=Huntsville1s=AL1z=35806-17602y=US2ffi=2l=2g=2pl=2v=2n=2pn=2a=901+EXPLORER+BLVD+NW2c=HUNTSVILLE2s=AL2z=35806-2807panelbtn=2 Less than 3 miles away no less (actually less ;-) Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE205P and channelbank
[EMAIL PROTECTED] wrote: Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the PRI any hints / tips greatfully received thanks to Steve Totaro for the nudge in the right direction! Phil I think you will have good luck. Just make sure you use no echo can when bridged. Also, you will need a T1 Crossover cable. I have used channel banks quite a bit but never for the amount of faxing you are looking at. Maybe someone has better input but I would certainly try it myself and feel good about it going in. I bought a 24 port FXS TSU 600 for $200 used on Ebay for the lab, so you may want to go the Used route before buying something new. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loudspeaker
Great price for a great unit. Bookmarked your site. Thanks, Steve Astawerks wrote: Ok. What do you think about this? http://www.astawerks.com/-p-504.html Astawerks VoIP Hardware sales and consulting http://www.astawerks.com 614-495-1400 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Sunday, April 15, 2007 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Loudspeaker This is what I want. Do you have any URLs to such a device as I cannot find any. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cb Sent: Monday, 16 April 2007 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudspeaker On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? Do you already have the loud speaker? If not, I know there are various vendors of extension phone bells that do nothing more than plug into an analog line and ring the nice loud bell when a ring signal is received. You could easily combine one of those with a cheap ATA with FXS port. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Loudspeaker
Cool Thanks. I am adding more Paging Amp's right now. Astawerks VoIP Hardware sales and consulting http://www.astawerks.com 614-495-1400 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Sunday, April 15, 2007 11:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudspeaker Great price for a great unit. Bookmarked your site. Thanks, Steve Astawerks wrote: Ok. What do you think about this? http://www.astawerks.com/-p-504.html Astawerks VoIP Hardware sales and consulting http://www.astawerks.com 614-495-1400 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Sunday, April 15, 2007 10:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Loudspeaker This is what I want. Do you have any URLs to such a device as I cannot find any. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cb Sent: Monday, 16 April 2007 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudspeaker On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? Do you already have the loud speaker? If not, I know there are various vendors of extension phone bells that do nothing more than plug into an analog line and ring the nice loud bell when a ring signal is received. You could easily combine one of those with a cheap ATA with FXS port. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/761 - Release Date: 4/14/2007 9:36 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 269.4.0/761 - Release Date: 4/14/2007 9:36 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
From: Steve Totaro [EMAIL PROTECTED] Date: Sun, 15 Apr 2007 22:36:15 -0400 Stephen Bosch wrote: Steve Totaro wrote: You could try to get it working but it may never be 100%. If your needs are 100% then I suggest using a standard fax and get an analog line and do it the old fashioned way. If you need Hylafax type features then buy a modem that is compatible with Hylafax and run it on a different box. It's not entirely clear to me why people continue to cling to the idea that Asterisk should handle faxing also. What's the benefit? Hylafax is great, and you can even use it on the same machine. On same machine is a bit exaggerated, considering there is a Zaptel card on it. (But if Zaptel and Hylafax can share an X100P driver ...) Yuan Liu -Stephen- I could have sworn that is what I just said. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] FreePBX - Vicidial Integration
Hello, I've got actually near 10 Call Centers which works fine with FreePbx and Vicidial. Its right that if you use the FreePbx Dial Plan with macro it's very slow but you can use all freePbx stuff to create and manage Extension and Standard Pabx functions; and for vicidial you can create other dial plan with minimal things. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Matt Florell Envoyé : vendredi 13 avril 2007 18:13 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] FreePBX - Vicidial Integration On 4/13/07, Diego Quintana Cruz [EMAIL PROTECTED] wrote: Hi all, I am trying to install Vicidial in an existent FreePBX installation (I'm using Xorcom packages for Debian Etch), but I didn't find any documentation, I found only this guide [0], but is for trixbox only, do you think it will work on FreePBX on Etch? [0] http://iptn.org/vicidial/index.html Regards, Diego Quintana Cruz Hello, I would not recommend using FreePBX with VICIDIAL, mostly for efficiency and ease-of-use issues. The FreePBX calling path can contain dozens of steps, all slowing down and causing problems for VICIDIAL calls that are trying to go out. Not to mention the CallerID control issues that will cause you problems with a stock FreePBX/VICIDIAL system I usually recommend getting a separate server that goes to your FreePBX server over IAX if you will be using it in production. The VICIDIAL server would only have the sample VICIDIAL conf files and the changes needed to get your IAX trunk working. MATT--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE205P and channelbank
[EMAIL PROTECTED] wrote: Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the PRI any hints / tips greatfully received Many people have no problems faxing, but that has not been my experience. Some telcos will let you mix B-Channels on your PRI with standard CAS channels. If you had a voice T-1 (non-PRI) they would be CAS channels. The telco some of my clients use, XFone/Louisiana, allows this. In USA/Canada a PRI is just a T-1 (24 channels) with each channel specially configured. There is no *technical* reason your carrier can't make channels 1-12 B-Channels, channel 24 as your D-Channel and channels 13-14 as CAS/FXO channels. One thing you might consider is this: PRI - Channel Bank - TE110P/Asterisk I don't know if the Adtran 624 has a 2nd T-1 port on it, but at least the 850s do, and maybe the 750s. You will want to check this. You could have the channel bank pass thru all your B-Channels and your D-Channel out the 2nd T-1 port to Asterisk and to connect the CAS channel(s) for faxing out the analog FXS ports on the channel bank. This also means that even if Asterisk is down you have 1 or more standard analog channels coming off the channel bank for faxing and emergency phones. It also means that you can buy a 1-port Digium (or compatible) card instead of a 2-port card. XFone does not charge any more for a CAS/FXO channel. The only disadvantages to this are you cannot have your fax number route into Asterisk to be accepted by RXFax and then e-mailed to someone. I just don't like this idea for the main fax number. You can set up combined voice/fax numbers pretty easily and people can accept faxes on their DID and get the fax in E-mail. If this failes then the sender can just send it to the main fax number for the company. The other disadvantage is that the fax lines are dedicated to faxing and cannot be shared with voice calls into Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
On Sun, Apr 15, 2007 at 10:10:34PM -0700, Yuan LIU wrote: (But if Zaptel and Hylafax can share an X100P driver ...) Where can you find a modem driver for a X100P? I recently asked about it in the linmodemds.org mailing list, and aparantly none is available. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
