Re: [asterisk-users] * as VM for legacy PBX?
As others have said, this is certainly possible. Our old NEC phone system had us in the same boat. It triggered voicemail by ringing the VM extension(s) and sending a DTMF burst of the extension to record VM for within 1.5 seconds. In our case, when any call came it in went to the voicemail system to play the main menu and allow the person to dial an extension. With that we were able to move a small set of power users to SIP phones for testing before we decided on a final phone and moved the whole campus. Daniel On Jul 1, 2009, at 8:16 AM, Ken D'Ambrosio wrote: Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm wrong, a six pack of their favorite $BEVERAGE will magically appear at their door. Thanks much! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Normalize Voicemail Volume?
I use the following script to perform compression and normalization on e-mailed voicemails. I put the script in as /usr/local/bin/sox and pre-pend /usr/local/bin to the PATH before asterisk runs in the startup script. The values for the compressor are not scientific, I monkeyed with them until I thought it sounded like a good volume, YMMV. Daniel #!/bin/sh # # $1 = -v # $2 = number # $3 = inFile # $4 = outFile # REALSOX=/usr/bin/sox if [ $1 != -v ]; then $REALSOX $* exit $? fi INFILE=$3 OUTFILE=$4 # # Perform the gain adjustment. # $REALSOX $INFILE $OUTFILE compand 0.1,0.3 -60,-60,-30,-15,-20,-12,-4,-8,-2,-7 0 0 0.2 On Jun 26, 2009, at 7:44 AM, Adam Moffett wrote: We generally get our voicemails emailed to us from asterisk, but some people's messages are extraordinarily loud or quiet. I don't suppose there is any feature to even out the volume level is there? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snom mass deploy help
On Jun 18, 2009, at 2:57 PM, Philipp Kempgen wrote: I think I would prefer this method, but I can't find where to set asterisk to listen to the multicast address nor where to program the notify reply I have already told you that Asterisk is not involved in the process of configuring the phone. In order to use Snom's PnP configuration method you have to write a daemon which opens a socket on 224.0.1.75 (sip.mcast.net), join the multicast group, read packets and send appropriate ua-profile notification events. Have a look at the code I mentioned to get the idea. As Philipp said, you don't. However it would make a great 3rd party module that could be added to Asterisk. I use a combination of the PnP and web redirects (early V6 versions did not support the PnP, but they do automatically request a file from the DHCP web server) and MySQL databases. It is now set where we just add the MAC address to the database and plug the phone it. It auto-configures the rest (along with firmware updates). Daniel Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US Caller ID
On May 1, 2009, at 7:18 AM, Andrew Joakimsen wrote: The *BEST* solution would be to have Verizon switch you over to a PRI. Assuming they were able to do this without a dramatic increase to our bill, would the same hardware still work? Sorry I don't have the exact model number but it is Digium's 2-port digital interface card with HW echo canceler, using the wct4xxp driver. Daniel On Wed, Apr 29, 2009 at 17:29, Daniel Hazelbaker dan...@highdesertchurch.com wrote: Okay, I can't find what might be causing this. Here is what I got: Asterisk server hooked up to a digital T1 line (full 24-channel) via a Digium card. Verizon has turned on caller ID on the first line (I can guarantee it is on as I can hear the FSK tones on this line but not the others). Using zttool an ZapScan() I have determined the following: 1) The RxB/RxD bits toggle from 1 to 0 signaling a ring. 2) A short time later, via ZapScan() I can hear the FSK tone. 3) About the same time I hear the FSK tone I see the Starting simple switch line in the Asterisk console. 4) Next I see the second ring trigger in zttool and then Asterisk say ss_thread: Got event 18 (Ring Begin). Caller ID never shows up. I have tried cranking the rxgain up thinking maybe it was too quiet for Asterisk to detect but that did not help. My caller id settings in zapata.conf are: usecallerid=yes callerid=asreceived cidsignalling=bell cidstart=ring signalling=fxs_ks Is there any existing debug options I can turn on, or do I need to add some to try and figure out what is going on; or does somebody have an instant answer for me? Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] US Caller ID
Okay, I can't find what might be causing this. Here is what I got: Asterisk server hooked up to a digital T1 line (full 24-channel) via a Digium card. Verizon has turned on caller ID on the first line (I can guarantee it is on as I can hear the FSK tones on this line but not the others). Using zttool an ZapScan() I have determined the following: 1) The RxB/RxD bits toggle from 1 to 0 signaling a ring. 2) A short time later, via ZapScan() I can hear the FSK tone. 3) About the same time I hear the FSK tone I see the Starting simple switch line in the Asterisk console. 4) Next I see the second ring trigger in zttool and then Asterisk say ss_thread: Got event 18 (Ring Begin). Caller ID never shows up. I have tried cranking the rxgain up thinking maybe it was too quiet for Asterisk to detect but that did not help. My caller id settings in zapata.conf are: usecallerid=yes callerid=asreceived cidsignalling=bell cidstart=ring signalling=fxs_ks Is there any existing debug options I can turn on, or do I need to add some to try and figure out what is going on; or does somebody have an instant answer for me? Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change Termination of Read Command
On Apr 27, 2009, at 10:29 AM, Danny Nicholas wrote:
Greetings all,
This is a “just-for-fun” question. I was
reading the support forum and a fellow there wanted Read() to stop
on * instead of #. I thought that changing app_read.c would resolve
this
current
if (tmp[x-1] == '#') {
tmp[x-1] = '\0';
break;
new
}if (tmp[x-1] == '*') {
tmp[x-1] = '\0';
break;
}
He applied and recompiled, but no joy. Any ideas why?
Without knowing where in the file this came from I can't say for sure,
but that code looks to me like the code that would run after the
digits are received and is stripping off the # character at the end,
if it is there. Further up (or somewhere else entirely) there is
probably a spot that actually terminates the read command when # is
pressed.
Daniel
Danny Nicholas
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Re: [asterisk-users] Good phone near $125
On Mar 16, 2009, at 3:53 PM, SIP wrote: David Ruggles wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 da...@safedatausa.com I believe SNOM 300s do PoE (might have to check that, though) and are around $100. We've little experience with them, but we use an office full of Snom 320s, and we're nothing but pleased with them. Good speaker, good handset, lots of excellent options. And reasonably priced. They do, we have a bunch of 300's (and 320's) deployed as PoE. Daniel N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Continue in dialplan on hangup
Is there a way to force a channel to continue in the dialplan after the remote end hangs up? Specifically, I am trying to play around with setting up a fax server. I can receive the fax, but sometimes the sending fax hangs up before my System command for printing can run and the fax never prints. I know I can work around by setting up a custom context and use the 'h' extension, but I am hoping for a more simple method. Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue in dialplan on hangup
On Feb 27, 2009, at 1:35 PM, Anthony Messina wrote: On Friday 27 February 2009 14:03:19 Doug Lytle wrote: Daniel Hazelbaker wrote: Specifically, I am trying to play around with setting up a fax server. I can receive the fax, but sometimes the sending fax hangs up If your looking into setting up a reliable fax server and your not doing it over IP, then your best results will be using HylaFAX+ and iaxmodem with Asterisk. HylaFAX+ handles the printing/re-faxing/fax2email of all inbound/outbound faxes via it's FaxDispatch script. It's a 'Set and forget (tm)' package. I absolutely love it. Doug Or, if you're using Asterisk 1.6 and looking to try something new, take a look at http://messinet.com/AsteriskFAXGateway I'll take a look at both packages. I hadn't given HylaFAX(+) any thought as when I searched initially I found just the old version of HylaFAX that last had a release in 2007, which makes me a bit nervous. :) Daniel Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 300 vs Grandstream gxp
On Jan 16, 2009, at 7:52 AM, Julian Lyndon-Smith wrote: Can anyone who has used both comment on the pros and cons ? Need to buy about 30 of these, for a small company with limited IT support. We recently deployed 85 phones to our office. We tested the Grandstream GXP2000, GXP2020, Linksys SPA941, Snom 300 320, and a Polycom 430 (I think that was the series). As an IT department we expected everybody to prefer the Grandstream because it is simple to use. We figured everybody would have the Snom because it is complex to use (though super easy on the IT side to administer). We had the opposite result. Everybody hated the Grandstream because they sounded bad, felt clunky, were difficult to do simple things on (like park a call, can't do it with one button). Nobody really cared for either the Linksys or Polycom. They were just too limited. We ended up rolling out a mixture of the Snom 300 and 320s and couldn't be happier (We looked at the 360, but it really doesn't offer anything except a bigger display, which isn't really utilized). With a simple MySQL database and a few PHP scripts all we had to do was type the MAC address of the phone into the MySQL database (with the login information) and then plug the phones in. No setup on the phone. Phone automatically upgrades the firmware to whatever version we currently use, gets its settings from the server, etc. If a phone has trouble (out of the 85 we had 2 that were a bit finicky and got replaced), we go into the database and change the MAC address and then plug in the new phone. Again, no setup. If you go Snom I would be happy to share these scripts, I just haven't gotten around to building up a nice package and posting them. If your choices are either Snom or Grandstream, I would so go Snom. I spent 2 days trying to configure the GXP's to do the few simple things we wanted and couldn't pull it off (call parking, BLF one-touch dial [does not fully work], etc). I spent 30 minutes on the Snom an had it perfectly configured. Julian Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experiences with grandstream GXW 4024 FXS?
I use the GXW-4008 and have never had any problems with it. Right now it runs 3 analog phones, but we were using it to link our old NEC phone system to the new Asterisk system, so it was used quite a bit and never once had an issue. Daniel On Dec 24, 2008, at 5:30 AM, Hector Quiroz wrote: HI all, does anyone already implemented the GXW-4024 FXS? Some distributors doesn't recommend it for high volume operations. regards, Hector. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dailplan code for holiday detection?
We chose to use a mySQL database to store the holiday information.
When a call is answered we query the database to see if there is a
holiday greeting recorded, if so we play the indicated greeting,
otherwise play the default menu greeting. (We do our dialplans in AEL)
context checkHoliday {
s =
{
begin:
MYSQL(Connect temp communicator username password
asterisk);
MYSQL(Query resultid ${temp} SELECT greeting FROM
menuGreetings WHERE startTime=FROM_UNIXTIME(${EPOCH}) AND
endTime=FROM_UNIXTIME(${EPOCH}) LIMIT 1);
MYSQL(Fetch foundRow ${resultid} sqlGreeting);
MYSQL(Clear ${resultid});
MYSQL(Disconnect ${temp});
if (${foundRow}==1)
{
Background(custom/mainMenu/${sqlGreeting});
goto mainMenu,s,begin;
}
else
{
goto checkTime,s,begin;
}
}
includes
{
mainMenu;
tempGreeting;
voicemail;
publicExt;
}
};
The 'checkTime' context simply checks if we are open or closed and
plays the appropriate greeting (if no holiday greeting is found).
Daniel
On Dec 23, 2008, at 1:14 PM, Scott L. Lykens wrote:
Not the most elegant but since I have a generic context for my IVRs I
simple check the date there.
exten = s,n,GotoIfTime(*|*|1|jan?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|10|apr?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|25|may?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|3|jul?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|7|sep?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|26|nov?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|27|nov?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|25|dec?closed-holiday|1)
exten = s,n,GotoIfTime(*|*|26|dec?closed-holiday|1)
exten =
closed-holiday,1,Background(ivr-closed-holiday-${AUTOATTENDANT}||)
exten = closed-holiday,n,Hangup
This is next year's holidays for us but with this year's Christmas
days
in it.
sl
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Re: [asterisk-users] CDR Design
On Dec 2, 2008, at 7:01 AM, Grey Man wrote: On Mon, Dec 1, 2008 at 3:26 PM, Steve Murphy [EMAIL PROTECTED] wrote: Everyone-- I've just made some major changes to the CDRfix2.rfc.txt file in http://svn.digium.com/svn/asterisk/team/murf/RFCs to accommodate the Leg approach instead of a channel-based approach. Hi murf, I've got a couple of points (as always) from the new design. First one would be the generation of CDRs when putting a call on hold. I don't think that should occur. When a call is put on hold Asterisk never changes the endpoints of a call all it does is possibly change the media to one or both of the call ends. CDRs are about call endpoints not about media transitions. In SIP terms putting a call on hold is no different to changing codecs both operations are re-INVITES and are irrelevant as far as CDRs and billing go. While I agree with your reasoning, I really like the idea of the CDR showing HOLD states. It allows me to generate a report on how often people are on hold. If I see that the incoming calls to my receptionist spend 15% of the time on hold, that means something to me. If someone doesn't care to know the hold states, they (or their script) can just ignore the HOLD CDR records. I don't see that it would impact any final numbers to just skip them, you still get the total call duration between point A and point B. Daniel Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Desgin
On Dec 1, 2008, at 9:07 AM, JD wrote: Steve Murphy wrote: Freddi-- Very interesting. Brian Degenhardt had some code we just gave some thought to, wherein we determine if the last channel involved in a linkedID set has been closed. If so, then the entire set is finished. We can use this facility to get you a closing attribute, that could be added to the last CDR emmitted for that set; OR, we could just emit another CDR with type CLOSE or FINAL or something, that signals the end of the chain. murf Just thinking out loud: how about a feature wherein, after the FINAL is sent, asterisk can 1. create a temp text file with just those entries, and 2. launch a user-made script. cdr_manager.conf [general] legparsecmd=/usr/local/bin/my_parser.pl wherein the linkedID is passed as the first parameter and the text file namepath as the second Ignore this suggestion if it horribly complicates things. Hmm.. While I normally like having this kind of instant notification, I could see this as a very big problem for larger installations. Most OS's are not so great at launching new tasks, and on a heavily loaded system that could easily be a number of tasks launched every second, each doing a lot of database queries. Perhaps a different approach would be to have a field that can be set to show that the record(s) have been parsed into whatever standard CDR format you want. This may or may not make more sense as a separate table with just a list of linkedid's that have been parsed. Daniel John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using MAC or extension number as SIP identifier
It will auto-complete if you hit tab, just like the shell. But I would recommend against it. I can't really think of a good reason to do it. 'sip show peer 268' I can remember to see that status of extension 268 when somebody calls and says I can't dial 268. Whereas 'sip show peer 00147...wtf was his MAC address again?', I have to lookup the extension somewhere and find the MAC address. Any reason you want to use the MAC address? If it is just for easy provisioning, I just put a MAC address field in the realtime SIP table and use a php script to take the phone's MAC address and feed it the login information it needs. Daniel Hi, For a long time, I was wondering if I should use MAC address instead of Extension number to identify SIP endpoints (as I'm mostly not using softphones). Before diving into this, I wondered how people using MAC address are using CLI as it seems more natural and simple to type sip show peer 4566 as opposed to sip show peer 00147F784512. Is there something obvious I'm missing (auto-completion ? aliasing ? ...) ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using MAC or extension number as SIP identifier
On Nov 20, 2008, at 9:02 AM, Olivier wrote: 2008/11/20 Daniel Hazelbaker [EMAIL PROTECTED] Any reason you want to use the MAC address? If it is just for easy provisioning, I just put a MAC address field in the realtime SIP table and use a php script to take the phone's MAC address and feed it the login information it needs. provisioning is the first reason. I also thought it could help to separate devices, users and other resources. What I currently do to separate devices (fax machines, modems, etc.) is give them actual names. I.e. I have northFax and southFax defined (so I would type 'sip show peer northFax'). In my mind, and particularly in my use, anything that a person dial's as an extension is going to have a person on the other end. Other things can have names because end users won't be dialing them as extensions. The fax machines are tied to a dedicated phone number so Asterisk dial's it internally. as you obviously cannot tie MAC address to a dialing string, this forces you to query a database somewhere for every call ... I'm not fully convinced of this, anyway, but when I thought about it, I felt frightened about loosing things I'm used to ... Correct. We setup a macro that uses a MySQL database to handle our extension dialing, we don't dial by MAC address but if you were so inclined, I suppose you could. As far as speed goes, we query the database about 4-6 times for every call. 85 users, 9 telco lines, Dell 2950 server, and we peak at about 0.2% cpu usage. Again for simplicity, having all the front-scene stuff match what the end-user is talking about is very nice. There is no reason you couldn't do some naming convention like 'userExtension#', 'deviceExtension#', 'otherExtension#'. That might help in your separation and wouldn't be too hard to figure out. Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Old mantis e-mails
I am suddenly getting a bunch of OLD (as in 3-9 months old) e-mails from mantis saying things like a note has been added to an issue etc., and yet the issue has not been touched in months and the new note it is referring to is also months old. Consequently, I never received these e-mails before either. The e-mail itself shows that carolina.digium.com received the message back in the day but the next hop (my server) shows todays date. Is it just me or has mantis been holding onto old e-mail and finally sending it? Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Decent Voip Phones for enterprise
On Oct 28, 2008, at 5:13 PM, Kev Szaszvari wrote: Hi there Can anyone reccomend any voip phones( Cisco, Polycom, SNOM ) that have * Central Management for all the phones (We dont mind if we have to buy the software to manage them) * Programable shortcut buttons, So i can program in on certian phones quick dials to queues. * Optional but bonus, The ability to have a shared address book accross the phones. We just rolled out Snom phones and it was the easiest thing in the world. 1) Yes, you can centrally manage your phones. If you realtime SIP with a database then you can do a complete plugplay setup. We use a few scripts to do this here. a) Script to respond to the Snom plugplay request (SIP broadcast message), redirects to PHP script (b). b) A few PHP scripts that update the firmware, provision the phones (via the database), define all the standard buttons, and allow overrides based upon extension number. 2) The Snom's let you program every single button. If you want to re- program the conference button to be a hold button, *shrug* go for it. You can program a button to function as a BLF, speed-dial and call- pickup button all at the same time. (Current 7.3.7 has a bug that only lets you speed-dial and call-pickup when the phone is on-hook, latest beta fixes that). 3) Nearly perfect support for LDAP directory. I say nearly because if you enable the number lookup feature (in addition to the name lookup) then anytime you dial it will immediately match by name and not let you see the number you are dialing. It basically forces you to dial-by- directory, kind of annoying. I got a bug report in on that. In regards to 1b, the PHP script gets the MAC address from the phone (via the URL requested), queries the database, sends back an XML file with all the registration information. With SIP realtime, what this means is that you get a new phone, put in the registration information in your database along with the MAC address of the phone, then plug the phone into the network. Come back 10 minutes later and it has updated itself to the latest firmware and is ready to make receive calls. If you want more specific information on this I would be happy to give you the scripts. As others have noted, the Linksys may be able to do what you want. But if you do end up switching I recommend the Snom if you want the best bang for your buck. Cisco Polycom are good phones, but getting a big enough phone that has programmable buttons etc. gets really pricy. Grandstream is okay, but after comparing them with the rest of the phones they audio quality just isn't there. And for us, the Snom is the only phone I could successfully program to do single-button call parking, which was a major requirement. Daniel Thanks in advance Regards, Kev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Oct 29, 2008, at 8:21 AM, Alex Balashov wrote: In my experience most of the serious QoS issues arise in relation to the Internet pipe (if the provider is IP, and outside the network), not the LAN. Of course, LANs can be heavily contended, but are not in most organisations, especially as gigabit cores are getting increasingly common even in smaller mid-size and small organisations. I would pay most attention to the router(s), unless your PSTN connectivity is TDM and on-premise. I would agree with this as long as you have a decent LAN. We have about 60 computer workstations and 85 phones on our network. The entire thing is Gigabit. Each phone (with a few exceptions that we are running new cable to rectify) has a dedicated ethernet port, no sharing. We are NOT however separating the data/voice networks. They are on one VLAN. We may segment later, but only if the need arises. Right now we have no problems. I should point out that all of our switches have 2+ gigabit links back to the master switch. We've never had a problem with the phones other than related to the outside world (telco side). I won't argue that best practice would probably be to VLAN off the phones, but if you don't have a massive network and are fully gigabit smart switches etc with good cabling, then keeping the two networks merged should not be a problem. I do wholly recommend multiple drops per workstation though. In a day when I can buy CAT 6 cable for 10 cents a foot, there is really just no reason not to be doing multiple drops in new installs. Daniel -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
On Oct 29, 2008, at 10:10 AM, Darrick Hartman wrote: David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. EVER? What about Gigabit networks with 10/100 phones? While some Gigabit phones are available, gigabit POE switches are not cheap, while non-POE gigabit switches are pretty cheap and most business class desktops these days come with gigabit network connections. In a new wiring install I almost always insist on two jacks per location rather than relying on pass-thru connectors on phones. Try giving a few users gigabit access to an Exchange server, then taking it away. They will certainly not be happy! I always considered myself to be rather tight on budget, but maybe I have more money available than most. We use the SGE2000P LinkSys Gigabit, Managed, PoE switches and they work great. I get them for about $800, which is just under $200 more than the non-PoE version. I don't find that to be an excessive price since most decent managed non- PoE switches are in the $500-$600 range (I'm sorry, I just can't bring myself to buy a D-Link or NetGear Gigabit managed switch for $300 to run my entire network on, maybe they are fine but they always struck me as a small player so to speak). Daniel Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax / t38 gateway
On Oct 24, 2008, at 9:49 AM, Wilton Helm wrote: I've been following this thread and trying to sort out what is wanted, what is available, and why. Comments to the following would be appreciated and might be useful to others. 1. Why would anyone originate a FAX via VoIP? If it has to go through a bunch of translation steps at both ends, it would seem better to simply scan the document (assuming it isn't in electronic form to begin with) and attach it to an E-Mail. 2. Why would anyone terminate a FAX call coming through Asterisk in a FAX machine? Isn't there a way to capture it electronically? If so, it seems that putting the electronic documents in a queue where people can open them, save them, and if they wish, print them would be much more useful (and planet friendly, since a lot aren't worth putting on paper). I can answer both of those with a single point. We just switched (entirely) to Asterisk a few weeks ago. We looked, very briefly, at various ways to get rid of the physical, analog, fax machines. They all ended with the answer People can't figure out e-mail as it is, they aren't going to figure out how to fax via e-mail.. What we need is a pure VoIP fax machine. Daniel Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Audio Problem
Might be a stretch, but does the Asterisk log show that the call was answered? I had this problem when interfacing * with an NEC system to do call parking pickup. The NEC would never give a dialtone (nor did it give answer supervision) so * never knew the call got picked up so audio only worked one way. I ended up rigging * to force the line to be considered answered with a patch. Daniel On Oct 13, 2008, at 8:57 AM, GNUbie wrote: Hello Steve, On Mon, Oct 13, 2008 at 10:59 PM, Steve Totaro [EMAIL PROTECTED] wrote: First, drop firewall/iptables/selinux and try again. I already turned off the firewall and I don't have SELinux on my system and the problem is still there. Regards, GNUbie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer/Park Question.
On Oct 10, 2008, at 1:00 PM, Brent Davidson wrote: Doug Lytle wrote: I don't remember where I got it (Might have been the bug tracker) that works fine under the current 1.4.x. I had to do a minor change to get it to apply. Copy into Asterisk source directory patch -p0 *.patch rm *.patch make make install Doug Ok, the patch is working great. Any idea what would make the one step parking not work? I've tried several DTMF combinations in features.conf and none of them seem to work when manually dialed or when bound as a DTMF code to a key. So far I've tried the following under [featuremap] in features.conf: parkcall = *5 parkcall = #72 parkcall = *9 parkcall = #75 I don't even see any acknowledgment of the DTMF tones showing up on the console. You won't. The patch I sent you off-list is incomplete, this one is better. I forgot I fixed the parked has timed out option in another patch before I fixed this part. Anyway, make sure when you dial you put k in the dial options (K too if you want both sides to park). It used to be tied to the t option I believe and then got moved out to k at some point. Other than that, it should work. Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer/Park Question.
On Oct 9, 2008, at 2:59 PM, Brent Davidson wrote: I've got a situation where I need to use a transfer to the parking lot as hold, but am not going to use BLF indicators on the phone to pick up the parked calls so I need to hear the 3-digit extension after the transfer. I'm using Snom 300 phones and have tried setting a programmable button to Key Event F_TRANSFER 700, which successfully does the transfer but cuts off audio so you don't hear the extension to dial. Same with setting a Park Orbit. I can use the DTMF button type to send the transfer command and then the extension but then the person doing the parking hears all of the tones, which is annoying. Is there any way to set up the transfer silently and still get the parking slot extension back? Short answer: currently no. Medium answer: I just rolled out 60+ Snom phones (300s and 320s) and we do call parking with DTMF. People were used to just hitting PARK and their phone displaying the park extension (old NEC system). I didn't tell anybody anything except it will speak the extension back to you and nobody has complained about hearing the DTMF digits. We chose a 3 digit code (#92 I believe) to try an alleviate the possibility of somebody accidently parking a call while filling out a DTMF based form/menu system, but in theory you could assign just * to park and only deal with 1 tone. Just be aware that if the user needs to hit * for anything else, they won't be able to use it. Long answer: Snom phones support text messages to the phone that automatically display. I am looking for a way to use that in conjunction with Snom's ParkOrbit feature (which does work, you just don't hear the extension). Basically Asterisk would do a normal park and then trigger a SIP NOTIFY message to the parkING phone that says Parked: 701. The message can be cleared by the user by pressing X, or ideally Asterisk would auto-clear the message after 10 seconds (or whatever). In theory I can do the long answer now with a Manager application, but I don't like the idea of relying on an external application. If it crashes or locks up for whatever reason then suddenly people get parked and nobody knows where. Also be aware that in 1.2.x and 1.4.x, if you park a call and then pick it up, you can't park it again. At least not with the DTMF method. I borrowed a patch from the 1.6 branch that fixes this and made it applicable to 1.4.20.1, well I borrowed part of it. The entire patch let you configure who could park etc., I wanted both sides to always park so I just took the 2 or 3 lines that were needed for that. If you are interested I can e-mail it to you directly. Regards, Daniel Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
On Oct 7, 2008, at 4:19 AM, Chris Bagnall wrote: I recently purchased a few SRW208P switches. They work fine. If you run Windows. Granted a lot of people run windows instead of Mac or Linux, but be aware (to those looking) that the SRW line of switches REQUIRE Internet Explorer on Windows. The support site says it is recommended, but even the login page does not work properly on anything but IE on Windows. For me, as a Mac user, it is enough to not buy any more of those ever again. That's very strange, I've used FF2 and 3 under Linux plenty of times to configure the SRW224P units. I'd have thought the web interfaces would be pretty similar between the models. I have not personally tried using FF under Linux with these, though I ran across a number of posts that say it doesn't work. I know FF2 and the latest FF3 don't work under Mac (don't work for the SRW that is) and I know they don't work on Windows. (Linksys' official statement is to use that ietab plugin that embeds IE in a firefox tab). I would expect FF to behave the same as far as what works and doesn't in all 3 environments, but maybe not. I'll install FF3 on my Linux server and try as that would be more convenient than firing up Parallels everytime I need to change a config option in the switch. On Win/Mac it lets you log in but the main menu screen is blank, nothing to click on, just the background template. *shrug* Seeing as we already need more than the 8 ports I think I'll stick to the 24/48 port versions anyway. Regards, Chris Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE switch recommendations?
On Oct 6, 2008, at 4:31 PM, Andrew Joakimsen wrote: As for the larger switches I've used Linksys SRW224P. I have a few running for a few years without issues. They have GB uplink but the individual ports are 100M. I recently purchased a few SRW208P switches. They work fine. If you run Windows. Granted a lot of people run windows instead of Mac or Linux, but be aware (to those looking) that the SRW line of switches REQUIRE Internet Explorer on Windows. The support site says it is recommended, but even the login page does not work properly on anything but IE on Windows. For me, as a Mac user, it is enough to not buy any more of those ever again. On the other side, We have a dozen switches in the SGE2000, SGE2000P and SGE2010P series that all work perfectly and with any browser I have tried. Some may wonder why I would buy a 24/48-port fully gigabit switch. It is because I don't want to have to think, or even keep track, of which port on the wall is PoE and which is Gigabit. I just want to plug it in and work. I want to be able to tell my staff Just plug your phone in and it will work, don't worry about trying to find a power adapter. The extra money is worth not trying to keep track of which is which. The SGE2000 switches we bought before the SGE2000P came down in price (it used to be like 4 times the non-PoE version). Now, at a $220 difference ($880 verses $660) there is no question. Beyond that, they work great. VLAN setup and use is simple. Link Aggregation works perfectly. STP works like a charm (no more running around trying to figure out what idiot patched their wall jack into another wall jack). The ability to transfer the switches configuration to a TFTP server (and HTTP in the 2010 version, 2000 is using old firmware) makes it easy to backup the configuration and restore it to a new switch in the event of complete failure. Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zap destroy
On Oct 2, 2008, at 9:10 AM, Jeff Peeler wrote: - Tzafrir Cohen [EMAIL PROTECTED] wrote: Yes, the new changes will be in 1.4.22. I continually have to remind myself that users aren't running the most up to date code. Once 1.4.22 comes out I will report if I am still having those issues. Daniel Jeff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zap destroy
On Oct 1, 2008, at 11:39 AM, Jeff Peeler wrote: Nope, that's the best you can do without restarting Asterisk. Is requiring two restarts reproducible? I'd really like to see console output with verbosity and debug set to 4 on chan_dahdi, preferably while only using zap channels. For me, yes. Every single time I do a zap restart I have to do it twice. If I execute them REALLY fast I have to do it 3 times. I am using 1.4.20.1 with chan_zap still, but I will try to produce you a copy of the log this weekend (doing some phone maintenance anyway). I have experienced this for as long as I can remember, and I know bad form, but just have never gotten around to filing a bug report on it. One of those things you never remember until you do 'zap restart' and it fails and then you do it again and go whew, that was close. Daniel Jeff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect Asterisk PBX to Traditional PBX and retain functionality
If I understand you, then yes you can. I do this now. All our telco lines come through our analog NEC phone switch and then through FXO/ FXS ports to my Asterisk. Asterisk handles voicemail and the menu system so when somebody dials 6 to get my extension the asterisk does the following: Flash(); Wait(0.4); SendDTMF(268); Hangup(); I added the Wait(0.4) as I found that under heavy load the NEC would not catch the first DTMF digit after the Flash. This solution has worked for us for over a year now. Some bonus information that may or may not be relevant to what you are doing: We have a few SIP phones that we needed to be able to do the same kind of thing. We couldn't flash transfer to the Asterisk, but in the NEC I setup a outgoing trunk line (dial 8 to access) that goes to the Asterisk box. Then I setup a forward all calls on extension 268 (when I have my SIP phone active) to dial out to 8268. That way when somebody calls my extension it automatically forwards then to extension 268 on the Asterisk box. Daniel On Jul 23, 2008, at 3:57 PM, Ricardo Melendez wrote: Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have the feature to transfer calls (Incoming call - Answer - FLASH - Dial Number to transfer - Answer - FLASH+4) and the call is transferred, but I have the need to implement an internal ACD using Asterisk as the PBX, the trunks connected to my Asterisk FXO ports are Extensions of my MAINPBX (ex., 5437, 5440 etc), all features work fine, but I have the need to make asterisk act as a normal telephone when transferring calls, I need to release the line (FXO port in my Asterisk) and make the transfer via the MAINPBX feature. Otherwise I will use 2 lines of my Asterisk PBX to make the transfer and it reduce the incoming lines available for my ACD. It’s possible send the commands FLASH, FLASH+4 using the incoming line to my MAINPBX via Asterisk like a normal telephone? Thanks in Advance. Ricardo Melendez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection (Resolved)
On Jul 11, 2008, at 12:58 PM, Daniel Hazelbaker wrote: I may have figured out the problem this morning, but I won't be able to test for a few days (again, aggravating that the only T1 line I have to test with is the live one). I noticed this morning while telneted into the Adtran that when I hangup on our normal incoming lines the Receive A bit toggles. I then noticed that two of the lines do NOT toggle the RA bit during hangup. These happen to the be last two lines in the rotary so I would not normally get incoming calls and complaints on them. They also happen to be the lines I was using to do my testing with. Grrr. Just to close out this thread for anybody interested, last night I hooked up the T1 line again and verified that this was indeed the problem. Out of the 12 lines in use on the T1, 4 of them do not provide the disconnect supervision. So I have called and updated my trouble ticket to include all 4 of those channels. Thanks again everybody for the suggestions and bits of information that helped me track down this problem. Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk cant play sounds from AGI
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
Hi! I am a newbie using Asterisk. I am developing an IVR using perl
from AGI and Cepstral as voices
The AGI is this
[snip]
My problem is that i cant hear anything when play the file sound
using $AGI-stream_file($filename);
I put asterisk in verbose mode but just see that it plays the sound
but I cant hear anything.
I thought maybe was the codec but asterisk can play .wav
But this works
$AGI-say_number('9865');
If Asterisk says it is playing the file, then I would suspect the file
itself has nothing to say. Try copying the file to your computer and
playing it. If it does indeed play locally on your computer with
audio, double check to make sure it is in the right format. I use AGI
to play files all the time. Actually, I use an AGI script as my whole
menu and dialing system to replace having to do it in AEL (so much
nicer to add a single MySQL record and suddenly have voicemail and
direct dial work instantly).
Daniel
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*-Developer DataBase
*-JQ Microsistemas
*-809-849-8087
* Si deseas lograr cosas excepcionales debes de hacer cosas fuera
de lo comun
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Re: [asterisk-users] Tracking Call Time While in Dial()
On Jul 11, 2008, at 10:08 AM, Douglas Garstang wrote: I want to track call duration while the call is in progress. To accomplish what? Are you wanting to beep the channel every 10 seconds? Are you wanting to play a you have 60 seconds left message when they approach some quota? Are you wanting to limit the call to 5 minutes and 23 seconds? Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
On Jul 11, 2008, at 12:09 PM, Jay R. Ashworth wrote: On Tue, Jul 08, 2008 at 11:13:02AM -0700, Daniel Hazelbaker wrote: D-Marc that terminates the 25-pair analog line coming in (this does not just contain our lines as I can tap into other peoples lines and hear there conversations, love security). The T-1's aren't on that, though, right? ... Really? You have an RJ-21X block that contains both analog AND T1 wires? That's really uncommon. I hope they at least put the red special service caps on the T1 wires. Yup. I thought that pretty funny myself. 10 year old analog wires running a digital T1. :) And they do have some caps on them, I think it was red but not 100% sure. I may have figured out the problem this morning, but I won't be able to test for a few days (again, aggravating that the only T1 line I have to test with is the live one). I noticed this morning while telneted into the Adtran that when I hangup on our normal incoming lines the Receive A bit toggles. I then noticed that two of the lines do NOT toggle the RA bit during hangup. These happen to the be last two lines in the rotary so I would not normally get incoming calls and complaints on them. They also happen to be the lines I was using to do my testing with. Grrr. I called Verizon and opened a ticket for why those 2 lines are behaving differently and that sounds like the problem, but I won't know for sure until I can test and try calling on one of the lines that does toggle the RA bit. As soon as I get that tested I will report that, though I expect that should fix the hangup issue. Thanks, Daniel Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http:// baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk cant play sounds from AGI
On Jul 11, 2008, at 1:31 PM, Edwin Quijada wrote: vm-debian#file tts-hello example.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Other than the filename being wrong which I would assume is the result of a copy and paste from the original e-mail, that looks right. Can you paste the asterisk log section around where it is playing the file, including the line that shows it playing? Something in the log may give a clue. Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
Another update on the latest hookup attempt. I can make it work reasonably well with callprogress=yes, it detects the hangup but only after about 7-9 seconds. My config files are the same as the last time I posted (apparently last time I wasn't waiting long enough for callprogress to kick in). If I turn callprogress=off then it never hangs up, if I turn it on, again after 7-9 seconds, it will hangup. My current Adtran unit hangs up almost instantly, so I think it is getting and detecting the disconnect supervision correctly. I am going to try again in a few days, but I want to make some modifications to the zaptel driver (add in some debug code) and revert to Asterisk 1.4, not that I expect that to make a difference but I'll try anything at this point. Before I do a question: I am using esf,b8zs signalling on a true(/inband) T1 line. Does the Zaptel driver use the rxwink timing to detect a hangup by the disconnect supervision? If not, what does it use as it must be able to tell the difference (and I am using analog terms here) between a hangup and a flash for 3-way calling. Thanks again. If I can't figure this out I will have to call them out, but I don't think they will do anything other than say yep it works on our side, fix your own equipment. Daniel On Jul 8, 2008, at 1:11 PM, Daniel Hazelbaker wrote: Just an update for the information I got from Verizon: It is a true T1, not a PRI for sure. b8zs and esf signalling. It is loop start with disconnect supervision (kewlstart as I understand it). I know I had already tried kewlstart before, but I suppose it is possible that some other configuration option was making it not work. Since the only T1 line I have coming in is our live phone lines I will have to test this again late some night this week, so if somebody has an idea of some things to check while I am doing that I will certainly give it a shot, otherwise I will report back afterwords if I had success. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
Date: Mon, 7 Jul 2008 16:48:00 -0400 From: Jason Aarons \(US\) [EMAIL PROTECTED] Digital ISDN used Q931 messages. You should get a disconnect message from telco on the d-channel 23. I am pretty sure it is a T1 and not a PRI. I did try configuring it as a PRI and it started spewing all kinds of errors and completely stopped working. Date: Mon, 07 Jul 2008 16:55:27 -0400 From: Doug Lytle [EMAIL PROTECTED] Daniel Hazelbaker wrote: We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other A T1 or a PRI? Just make sure we're on the same page. Also, show us your zaptel and zapata.conf Again, I am pretty sure T1. It is a Verizon Flex-Grow package, which they list as expandable up to 24 voice channels. That and I tried configuring as a PRI and it harfed. The Adtran box we use now is configured as: Timing Mode Network Format ESF Line Code B8ZS Equalization0 dB CSU LpbkEnable Rx Sensitivity Auto Right now with Asterisk mostly working (it answers calls, dials out, etc. just doesn't detect hangup) my /etc/zaptel.conf is: # # Span Configuration # ~~ span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs # # Channel Configuration # ~ fxsks=1-24 fxoks=25-48 loadzone = us defaultzone=us --CUT-- /etc/asterisk/zapata.conf: [channels] usecallerid=yes callerid=asreceived cidsignalling=bell cidstart=ring callprogress=yes# I have turned this off too ;- ; ; Define telco channels in rotary, these should be answered ; like a normal incoming call. ; context=bridgeNEC usecallerid=yes signalling=fxs_ks group=1 ; Part of ZAP group 1 channel = 1-9 context=incoming channel = 12 ;- ; ; Telco line, computer dialup, needs to be routed to output line. ; group=2 usecallerid=no channel = 10 ; PSTN attached to Span1:Port10 ;- ; ; Telco line, construction trailer fax, needs to be routed. ; group=3 usecallerid=no channel = 11 ; PSTN attached to Span1:Port11 ;- ; ; ADTran lines, used for outgoing to analog devices ; context=incoming group=4 usecallerid=no signalling=fxo_ks channel = 25-36 --CUT-- For context, the bridgeNEC context just dials out one of the ADTran lines to our existing NEC system, but the incoming context starts our menu-system, which was also not detecting hangups. I have also tried using loopstart and groundstart signalling, doesn't seem to make a difference. I am pretty well stumped myself. I need to call the telco about the caller id not working to verify that it is still turned on, but I figure I might as well wait so that if I need to ask them about the signalling I can know all the questions to ask at the same time. Thanks, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
On Jul 8, 2008, at 10:45 AM, Jay R. Ashworth wrote: The Flex-grows I've seen were indeed T1, ESF as I recall the lights on the Adit 600 they terminated them into. Daniel: did Verizontal supply you with a shelf? Or just the smartjack? Uhhh... :) I have in my server room these things: D-Marc that terminates the 25-pair analog line coming in (this does not just contain our lines as I can tap into other peoples lines and hear there conversations, love security). Next to that is a box with 4 slots for T1 cards, we used to have a T1 internet connection and its card is still in there. Slot 2 has the flex-grow T1 card in it. One of the pairs from the D-Marc goes into this T1 card and it provides a RJ-45 connection for the T1 line that runs either to the Adtran or to our Digium T1 card. I hope that answers the question, as I am not entirely sure what a shelf or smartjack are. Though I will feel really stupid if you say a shelf is something you store stuff on. I will be calling Verizon this afternoon to try and get some information on our line. Hopefully I will get somebody that knows what they are talking about (and what I am talking about) as I am going to ask about everything I need to know to hook this up to our new digital PBX. (esf, b8zs, loop start, ground start, kewl start, forward disconnect, etc.) I'm hoping that will at least give me enough information to make some headway and maybe one of you guys enough information to tell me what I am doing wrong (or heaven forbid what Verizon is doing wrong) :). I'll post again after I get something from them. Thanks again, Daniel -- j -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http:// baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] US T1 Hangup Detection
Just an update for the information I got from Verizon: It is a true T1, not a PRI for sure. b8zs and esf signalling. It is loop start with disconnect supervision (kewlstart as I understand it). I know I had already tried kewlstart before, but I suppose it is possible that some other configuration option was making it not work. Since the only T1 line I have coming in is our live phone lines I will have to test this again late some night this week, so if somebody has an idea of some things to check while I am doing that I will certainly give it a shot, otherwise I will report back afterwords if I had success. Daniel On Jul 8, 2008, at 11:13 AM, Daniel Hazelbaker wrote: On Jul 8, 2008, at 10:45 AM, Jay R. Ashworth wrote: The Flex-grows I've seen were indeed T1, ESF as I recall the lights on the Adit 600 they terminated them into. Daniel: did Verizontal supply you with a shelf? Or just the smartjack? Uhhh... :) I have in my server room these things: D-Marc that terminates the 25-pair analog line coming in (this does not just contain our lines as I can tap into other peoples lines and hear there conversations, love security). Next to that is a box with 4 slots for T1 cards, we used to have a T1 internet connection and its card is still in there. Slot 2 has the flex-grow T1 card in it. One of the pairs from the D-Marc goes into this T1 card and it provides a RJ-45 connection for the T1 line that runs either to the Adtran or to our Digium T1 card. I hope that answers the question, as I am not entirely sure what a shelf or smartjack are. Though I will feel really stupid if you say a shelf is something you store stuff on. I will be calling Verizon this afternoon to try and get some information on our line. Hopefully I will get somebody that knows what they are talking about (and what I am talking about) as I am going to ask about everything I need to know to hook this up to our new digital PBX. (esf, b8zs, loop start, ground start, kewl start, forward disconnect, etc.) I'm hoping that will at least give me enough information to make some headway and maybe one of you guys enough information to tell me what I am doing wrong (or heaven forbid what Verizon is doing wrong) :). I'll post again after I get something from them. Thanks again, Daniel -- j -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http:// baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] US T1 Hangup Detection
We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled it. I simply do not see _anything_ related to caller ID going on. (not major, I am not even sure the phone company has it setup properly so I need to talk to them first, Verizon) 2) Asterisk is not detecting the far end hangup. Through the Adtran it does, but direct digital it does not. I bridged an incoming call to an analog phone and listened as I hung up the far end (cell-call). I hear a audible click, silence, and then after maybe a half second I hear dialtone. I tried turning on hanguponpolarity switch, tried turning it off, tried turning callprogress on and off, still does not detect the hangup. Am I missing something obvious in Asterisk (this is my first digital hookup)? I read somewhere that Asterisk is already suppose to detect dialtone to know that the far-end hungup. Do I need to call my phone company and get details on exactly how they are triggering the hangup, though I would think with digital it just happens). Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume
Hi Daniel, I'm intrigued by this and wanted to try it out - but I'm wondering how you get Asterisk to call sox at all during Voicemail()? Our server doesn't even have sox installed, so I'm not sure how to go about tricking Asterisk into running a different one. To do anything useful you would have to get sox installed on your server. But to get asterisk to run a different/fake sox, just install whatever you want to run as /usr/local/bin/sox and then edit your safe_asterisk script as I mentioned below. Asterisk runs the program 'sox' using the first match in your $PATH, so by updating the $PATH before asterisk runs you can direct it to run a different sox program. Be aware that this could pose a security issue as some systems allow regular users to modify /usr/local/bin. So people could install other programs that asterisk runs into that directory as well to get elevated privileges. For me it is not a concern as the machine is used only for Asterisk and only accessed by our IT department. Daniel CP Daniel Hazelbaker wrote: On Apr 1, 2008, at 5:22 PM, [EMAIL PROTECTED] wrote: Can the volume of the recorded voice mail message be changed? If so, what I am doing wrong? Any input would be greatly appreciated. Thanks. I had a similar problem in our setup where we e-mail the recorded messages to e-mail retrieval. But this also helps standard phone retrieval too. What I did was edit the /usr/sbin/safe_asterisk script and add: PATH=/usr/local/bin:$PATH At the top of the script. This would let me override the default sox implementation that Asterisk uses. Then I loaded in a script (called sox) that would compress and normalize the recorded audio (It compresses to deal with the spikes of the noise of the handset being hung up, etc.). It works pretty well for us and makes the volume pretty good so we don't have to crank up the volume on our computers or phones to listen to voicemail messages. And we can't adjust the rxgain as it is already a good volume for normal calls. Daniel --CUT-- #!/bin/sh # # $1 = -v # $2 = number # $3 = inFile # $4 = outFile # REALSOX=/usr/bin/sox if [ $1 != -v ]; then $REALSOX $* exit $? fi INFILE=$3 OUTFILE=$4 # # Perform the gain adjustment. # $REALSOX $INFILE $OUTFILE compand 0.1,0.3 -60,-60,-30,-15,-20,-12,-4,-8,-2,-7 0 0 0.2 --CUT-- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume
On Jun 30, 2008, at 1:04 PM, [EMAIL PROTECTED] wrote: But to get asterisk to run a different/fake sox, just install whatever you want to run as /usr/local/bin/sox and then edit your safe_asterisk script as I mentioned below. I think this is a bad approach. It's going to be a big gotcha down the road for somebody :) Agreed. :) It would be nice if there was something similar to externnotify that was not just a notification but a pipe that you could modify/filter etc the audio file before it continues on its merry way. Asterisk runs the program 'sox' using the first match in your $PATH, so by updating the $PATH before asterisk runs you can direct it to run a different sox program. A quick grep through the Asterisk (1.2.28) sources shows res_monitor using soxmix if the channel variable MONITOR_EXEC is not defined -- but nothing in app_voicemail. Am I missing something? It is possible this is a 1.4.x feature, and specifically it is for the e-mail sending system which as I am thinking about it is a 1.4 feature. I don't use 1.2.x and never have, I started with 1.4. The other option, again I don't know if this is available in 1.2, is to use the externnotify option. When a voicemail is left this script/ program is called with the context, extension and number of new voicemail messages. With a little bit of shell scripting you could walk the list of all messages and process any left (modified) within the last 20 seconds via sox. It is a little more iffy and prone to race conditions, but it should work. Unfortunately I couldn't give you a specific example of doing it this way as I use the e-mail style. Daniel Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume
On Apr 1, 2008, at 5:22 PM, [EMAIL PROTECTED] wrote: Can the volume of the recorded voice mail message be changed? If so, what I am doing wrong? Any input would be greatly appreciated. Thanks. I had a similar problem in our setup where we e-mail the recorded messages to e-mail retrieval. But this also helps standard phone retrieval too. What I did was edit the /usr/sbin/safe_asterisk script and add: PATH=/usr/local/bin:$PATH At the top of the script. This would let me override the default sox implementation that Asterisk uses. Then I loaded in a script (called sox) that would compress and normalize the recorded audio (It compresses to deal with the spikes of the noise of the handset being hung up, etc.). It works pretty well for us and makes the volume pretty good so we don't have to crank up the volume on our computers or phones to listen to voicemail messages. And we can't adjust the rxgain as it is already a good volume for normal calls. Daniel --CUT-- #!/bin/sh # # $1 = -v # $2 = number # $3 = inFile # $4 = outFile # REALSOX=/usr/bin/sox if [ $1 != -v ]; then $REALSOX $* exit $? fi INFILE=$3 OUTFILE=$4 # # Perform the gain adjustment. # $REALSOX $INFILE $OUTFILE compand 0.1,0.3 -60,-60,-30,-15,-20,-12,-4,-8,-2,-7 0 0 0.2 --CUT-- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using * in extension name
I am trying to setup an extension of *7XXX that will allow me to dial *7 and then any extension and use the Pickup application to pickup a ringing phone. Ideally it will also check if the phone is ringing somehow and then either dial the extension or pick it up if it is ringing. But I can't get that far. If I use *7268 specially it works fine, but as soon as I introduce any wild card char (X, N, Z, !, .) it responds with 404 and nothing is logged. Can * be used in this manner in a dialplan? If so then any suggestions on what I can check to see why it is doing this? If not, does anybody have a better suggestion for me? I'd rather not use a regular digit as the begin code. Daniel Hazelbaker ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using * in extension name
(Hope you don't mind me replying to the list) Okay, time for me to feel stupid. Yes I was forgetting to start with an underscore. Somehow while I was looking at all the examples it never clicked that that should be there. :P Daniel On Dec 19, 2007, at 11:34 AM, Martin Smith wrote: If you could show us what you ARE using (the full line of the dialplan), I might be able to offer more suggestions. For example, with the full line, I might notice if you're starting the pattern with the underscore, which I believe is required to pattern match *at all*. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Hazelbaker Sent: Wednesday, December 19, 2007 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Using * in extension name I am trying to setup an extension of *7XXX that will allow me to dial *7 and then any extension and use the Pickup application to pickup a ringing phone. Ideally it will also check if the phone is ringing somehow and then either dial the extension or pick it up if it is ringing. But I can't get that far. If I use *7268 specially it works fine, but as soon as I introduce any wild card char (X, N, Z, !, .) it responds with 404 and nothing is logged. Can * be used in this manner in a dialplan? If so then any suggestions on what I can check to see why it is doing this? If not, does anybody have a better suggestion for me? I'd rather not use a regular digit as the begin code. Daniel Hazelbaker ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using * in extension name
Just to finish off this thread for anybody else who wants the same
functionality:
As Philipp coded out, what I was originally doing was something akin
to the following:
_*7. = {
Pickup(${EXTEN:[EMAIL PROTECTED]);
Dial(Local/${EXTEN:2});
}
Somebody else wondered why I would do this as I wouldn't want to talk
to the original callee. Pickup() will never return if it successfully
picks up the call. So if it does return then we can assume (for now)
that the phone is not ringing and then proceed to dial the number.
As I said, this did work perfectly. However, I discovered that the
Grandstream phones we use automatically prepend ** to the monitored
extension. So if the phone is not ringing it will dial 268, if the
phone is ringing it will dial **268. So in the end I ended up with:
_**. = { Pickup(${EXTEN:[EMAIL PROTECTED]); }
And all works perfectly. Obviously for other phones I would have to
come up with something else like the above, but with the grandstreams
it seems to work great.
Daniel
On Dec 19, 2007, at 11:48 AM, Daniel Hazelbaker wrote:
(Hope you don't mind me replying to the list)
Okay, time for me to feel stupid. Yes I was forgetting to start
with an underscore. Somehow while I was looking at all the examples
it never clicked that that should be there. :P
Daniel
On Dec 19, 2007, at 11:34 AM, Martin Smith wrote:
If you could show us what you ARE using (the full line of the
dialplan),
I might be able to offer more suggestions. For example, with the full
line, I might notice if you're starting the pattern with the
underscore,
which I believe is required to pattern match *at all*.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Hazelbaker
Sent: Wednesday, December 19, 2007 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Using * in extension name
I am trying to setup an extension of *7XXX that will allow me
to dial
*7 and then any extension and use the Pickup application to pickup a
ringing phone. Ideally it will also check if the phone is ringing
somehow and then either dial the extension or pick it up if it is
ringing. But I can't get that far. If I use *7268 specially
it works
fine, but as soon as I introduce any wild card char (X, N, Z,
!, .) it
responds with 404 and nothing is logged.
Can * be used in this manner in a dialplan? If so then any
suggestions on what I can check to see why it is doing this?
If not,
does anybody have a better suggestion for me? I'd rather not use a
regular digit as the begin code.
Daniel Hazelbaker
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Re: [asterisk-users] Realtime SIP BLF
On Nov 28, 2007, at 8:24 PM, [EMAIL PROTECTED] wrote: From memory - 'rtcachefriends=yes' should do the trick. PaulH Sorry for the late response, wanted to make sure everything else was still working. This did indeed solve the problem. The only side affect I have noticed is that changed I make to the realtime database don't get picked up immediately. Not sure what the cache timeout is but I am able to flush it manually so for the moment I don't care. :) Thanks, Daniel On Wed, 2007-11-28 at 16:56 -0800, Daniel Hazelbaker wrote: I am trying to get the presence/hints/BLF working along with Realtime SIP but I never get any busy notification. core show hints always shows the realtime sip user as idle. I have tried setting call-limit to various values, including 1 but nothing seems to help. I have tried limitonpeers both yes and no. Anybody got any other ideas? I do know the hinting is working as I can hint a Zap channel and it works fine. Daniel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime SIP BLF
I am trying to get the presence/hints/BLF working along with Realtime SIP but I never get any busy notification. core show hints always shows the realtime sip user as idle. I have tried setting call-limit to various values, including 1 but nothing seems to help. I have tried limitonpeers both yes and no. Anybody got any other ideas? I do know the hinting is working as I can hint a Zap channel and it works fine. Daniel ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-site / Multi-server coordination
Okay, so we are planning for the future here where I work so we are trying to do testing ahead of time as we might be setting up a satellite campus that would need its own Asterisk phone system but still tied into our main campus phone system. This much we have accomplished. We have a central database that routes calls to the correct server before dialing the final destination SIP phone (or Zap, IAX, whatever). Our last big question is this: How, if at all, does asterisk deal with multi-server voicemail? i.e. User A is at site A and has a voicemail in his mailbox. He knows it really belongs to user B (who is at site B) so after listening to the voicemail he forwards it to user B. Is there a way for Asterisk to realize this box belongs on a different machine and communicate the voicemail over the network to that machine, or is this something I need to write? :) Daniel Hazelbaker Information Technology Director High Desert Church ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Random Double Digits
We have a Asterisk box acting as a voicemail system and greeting/ call director for our phone system (NEC system). The problem we are having is that randomly (though most especially with cell phones) asterisk thinks it is getting a double digit. For example, somebody will enter 269 and asterisk will read 2269. I believe the core problem is the NEC system's volume as we've had problem with volumes for over a year, but short term I would like to find a solution while we try to solve the cause. Is there a way to tweak the zaptel settings so that Asterisk (or zaptel or whatever) better handles our situation? I realize there is probably not a single switch to turn on and I might have to do trial and error stuff, but we are willing to spend some time tweaking to find the best solution. My theory (though hard to prove since it is a random problem) is that the doubled-digit is being detected twice due to a drop out in either volume or (in the case of a cell-phone) poor connection. I have to this point never seen unexpected digits (somebody dials 269 and it read 249 or something like that), but we have seen both double-digits and missing digits (they dial 269 and it only reads one of the digits). Daniel ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring/Off-hook in strange state 6
Alex, I had this problem with a new TDM2400 card that we purchased. Specifically I would get that message and then it would pick up the ringing line AND the line next to it. Basically, lines 1 2 had been cross-linked somehow. After a few weeks of trouble-shooting with Digium tech support they cross-shipped me a new card and the problem (and that message) went away. Daniel Hazelbaker High Desert Church On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote: HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to no, as well as busydetect=no. This is a major problem since this box only has 1 other line, but at least it works. I can't seem to find much info on this issue. I can't believe others haven't run into it. I started a ticket with digium, but I guess they are pretty backed up. Here is what I am getting in the CLI: Thanks for any help -Alex -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 -- SIP/4125-09559118 is ringing ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nic aliases not working
Have you quit and relaunched Asterisk? (not a reload, but a full quit process and restart) I know in the past when I have a process already listening to 0.0.0.0 it will not always pick up a newly added NIC alias address without re-binding. Daniel On Apr 11, 2006, at 12:21 PM, Michael George wrote: I have an * box that I need to chang the IP address on. My hope was that I could add an alias to the interface with a different IP address, have * bind to all addresses, change DNS and when no more hits come on the old address. However, IAX registrations coming in to the alias don't seem to get acknowledged by *. Even with iax2 debug on, I don't see any attempts. We can ssh in on both IP addresses and I have bindaddr=0.0.0.0 in iax.conf. Is this not possible for some reason? Maybe multiple IP addresses work but nic aliases do not? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --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] Networld Interop, Vegas 2006
Does anybody know how big a presence Asterisk and/or Digium will make at Networld Interop this year? I have a part-time guy that is building an Asterisk system for us (in a proof of concept fashion before we do a full switch to it) that I would like to take, but I don't want to waste his time if it is going to just be a yeah, try our product booth and not something he can spend time talking to them about what it can/can't do, see it in use, etc. Daniel ___ --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] Receptionist Phones
I've been quiet on this discussion for a few days and reading everybody's thoughts. But since I brought this subject up in the first place I thought I would bring it to a close with the results summarized. 1. The polycom phones do support attendant stations, but there is some incompatibility between their phone and asterisk that currently limits the number of monitored stations to 7. They are aware of this limitation and are currently working on fixing it. 2. The Snom phones work and do not seem to suffer from this limit. 3. Snom phones give you up to 54 monitored lines currently, but there is indication on their website (use of the word currently) that they are trying to find a way to bring this number up (possibly linking more than one add-on module together). 4. Old style switchboards (such as DSS attendant stations) are a thing of the past and will likely die off in the future, but the world is not there yet, people still want them. 5. Computer based, on-screen monitoring systems are the future for large organizations as they are easy to change and easy to customize to fit exactly what you want. But there are some things to resolve before the average user can use them without a dedicated computer screen to reduce frustration. I have no doubt that as more companies move to VoIP systems these old DSS units will become a thing of the past, we just are not quite there yet. 6. Most people (not all, but most) who are given the option of a new way to monitor line status vs. the old way will probably eventually use the new way (some kind of FOP display). But those same people will refuse to like the new way if the are forced to use it or nothing, just the way people are. I don't doubt that we could slowly train our receptionists to use the new system while they have access to the old and simply wait until they say, you know what, we don't use that old box anymore, it just takes time. :) So my original question was answered long ago, but I have enjoyed the thoughts and opinions of everybody that has contributed to this discussion. Regards, Daniel Hazelbaker ___ --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] Receptionist Phones
What I read on snom's website was the _currently_ only one sidecar can be hooked up at a time. It sounds like they are working on getting multiple sidecars chained together but have not got all of the bugs worked out. I am kind of in the same boat. Our current system offers 60 buttons on the sidecar. It is full but they already don't have everybody. Talking to the receptionists (we have a split office setup, one on each side of the building) they figure it would not be hard for them to remove the extensions that are not used very much to get the number down to the 54 currently allowed on the snom phone. Particularly since it is a short term solution. I expect either snom to get multiple sidecars working fairly soon or polycom to get the issue with its 7-button limit figured out (or Asterisk, as the case may be), and then be able to upgrade their phones to an unlimited button phone. And the price of the snom setup is not bad at all. $235 for the phone and $140 for the expansion module = $375. Not a hard sell to say that they may have to toss a $140 expansion module if they end up going with a different solution later. The phone would still be perfectly good. Daniel On Mar 28, 2006, at 8:12 AM, Bob McDowell wrote: Very true. I am currently debating whether or not to offer it as an option for my employer's system. As it currently stands, we do not have everyone's extensions on a button. With the snom 360 plus the expansion we still don't have them all. While I'm sure it would be 'better than nothing' from my own point of view, it might also be setting up the receptionist for a disappoint. As this system is new, I'm working hard to portray it as the 'limitless future', as opposed to the proprietary and very limited system we were on before. The receptionist not having a sidecar is present my fault, due to lack of finding a good one. Bob McDowell ___ --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] Receptionist Phones
For those of us that only need a small handful of these receptionist phones (for me it is 2), it should not be nearly as much of a problem, correct? For example I only need 2 phones with 60 (well, I can get 54 atm, but would like to expand even more). Assuming everybody picked up their phone at the same time that would only be 180 (60 * 2, plus I am assuming some message to the phone that was picked up) messages. I can't imagine putting a sidecar on every single phone. If average joe really wants to know if somebody is on the phone they can log into a web page that will tell them the status of a phone. Daniel - Good to hear that people from the manufacturing companies traffic these lists! On Mar 28, 2006, at 6:29 PM, Christian Stredicke wrote: Well the problem with the sidecar is simple. Just try to light all lights three times within one second. If you have 50 keys there is already hell breaking loose. If you cascade side cars and say have 100 LED, this is a real Xmas tree. The CPU drowns in XML notifications. We already had trouble, and we don't want to double it at this time. Good work, IETF. BTW this is not only a problem if the phone. If the PBX has to supply 50 phones with 50 LED and e.g. they are going off hook at the same time, we are talking about a burst of 50 * 50 = 2500 messages which will have some impact of the PBX CPU as well. We need to do something about this first before we can start having 100 or 150 LED on a device. Christian - yes I am from snom. ___ --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] Receptionist Phones (was 3Com Phones)
Thanks for all the comments on the 3Com phones. Thankfully, there is a large number of phones out there to dig through looking for the right solution. What I have not been able to find, after spending all weekend looking, is a good solution for an attendant console. We have 2 receptionists that need to be able to view all 60+ phones (we could probably weed it down a bit if we had to, but would like to be able to cover all the phones) and see who is on the phone already. I would like to avoid a software solution as those tend to be confusing and hard for non-computer savvy people to deal with. I have seen that the polycom setup (601+sidecar) works but only for up to 7 phones. Does anybody have a recommendation for a solution for this? I find it hard to believe that nobody makes a compatible phone (or add-on) that is compatible with Asterisk. It seems like such a common thing. Daniel Hazelbaker ___ --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] Receptionist Phones (was 3Com Phones)
We may end up using a software solution, but there are two main issues with a software solution (for us at least): 1) For us in particular, our receptionists have ALWAYS (for the past 15 years at least) used a physical switchboard style for routing and seeing availability. From past hardware-software changes we know that it will be very frustrating for them. For us, it is much more worth it to spend $1,000 to buy each of the two receptionists a really nice phone that supports these features rather than get a cheap software (though very nice) solution. 2) Having a software solution can cause grief and frustration to an already overworked receptionist. Just a few examples (these are not as uncommon as one might think): User quits web browser after finishing looking something up on-line, doesn't realize they just closed out their switchboard until they need it and it is not there. User gets lost trying to find the right window while trying to not sound like an idiot to the person on the phone. Computer has frozen, or otherwise has problems, and must be rebooted. I do like the look of Asternic, it is very old-style and easy to get used to, but we would still prefer a hardware solution if possible. We may end up having to say, sorry but you need to deal with this for a while until some bugs in the system are resolved (i.e. the 7 line problem), but as soon as a hardware solution is available we will switch you back to it. Hopefully we can find something before we switch, but if not it is good to know that software solutions are a viable alternative. Have you looked that the flash operator panel? http://www.asternic.org/demo.html I know you mentioned not wanting a software solution because of confusion but I think that would be pretty easy to understand. Curt ___ --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] Receptionist Phones
Yes, I keep reading on the mailing list archives and the wikis that (wether or not it is indeed a Asterisk issue) Polycom keeps saying that an issue with Asterisk prevents you from monitoring more than 7 total (not per sidecar) extensions. Daniel On Mar 27, 2006, at 12:08 PM, Justin Moore wrote: On 3/27/06, Daniel Hazelbaker [EMAIL PROTECTED] wrote: I have seen that the polycom setup (601+sidecar) works but only for up to 7 phones From what I've seen, each sidecar supports up to 14 additional stations. Three of those along with the 5 buttons on the 601 comes up to 47 on my calculator. Is there a known problem with the 601+sidecars and * that prevents the user from being able to monitor more than 7 extensions? Just curious as I've been leaning toward this for our receptionist as well (only 12 extensions to monitor...) -- Justin Moore aka wantmoore --- www.wantmoore.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Receptionist Phones
Hmm, which phone from Snom are you using for this? I've looked around their website and I can only find 3 VoIP phones, the 300, 320 and 360. The 360 by the looks of it only has 12 buttons you can assign to different extensions; am I missing something or is that the phone and you just do 12 per phone? Daniel On Mar 27, 2006, at 2:28 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yes - set up about 10 of them at a business last year. Monitoring is fine - picking up calls is a bit iffy at the best of times. (that is, picking up a ringing call by pushing the extension button. *8 works fine) Paul Hales Technical Manager AsteriskIT ___ --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] 3Com Phones
Drat, because the 3Com phones looked pretty good for the price. :) Is there somewhere that has a compatibility list for Asterisk with all the phones that are known to work/not work with Asterisk; since apparently VoIP phone companies incorrectly state that they support the SIP protocol (I don't consider, we support SIP as long as it only talks to our server because we tweaked it just a bit to be supported). I am looking for a good 60 phones. We are upgrading our entire phone system (and *old* NEC PBX). We don't need anything fancy on most of the phones, just the usual mid-size business features. Speakerphone, Hold, Park, Transfer, Voicemail; and we need at least 2 attendant stations that can see all in-use phone lines. We are trying to keep the costs (relatively) down, hence using Asterisk instead of a full commercial solution. It is very disconcerting to know the providers are essentially lying about what their phones support. (3Com states their phones are SIP compatible, not 3Com's version of SIP compatibile). Thanks for the info, hopefully somebody will have some recommendations for a good phone brand that actually IS Asterisk compatible. Daniel On Mar 26, 2006, at 12:01 AM, Jared Valentine wrote: I would not recommend the 3Com phones for use with Asterisk. 3Com 3100 series phones do not support SIP with non-3Com systems. They have a basic boot loader which must download code from a 3Com NBX or a 3Com VCX system. If you don't have either of these, then you won't get runtime code on the phone, thereby making it impossible to use the thing with Asterisk. I've heard rumors that the 3103 phones have enough storage space on the phone to store a SIP image, but I don't have any more information than that. As far as 3Com licensing is concerned, it's not per year, it's per- seat (one-time charge), just like any other commercial VoIP PBX vendor (Cisco, Avaya, Shoretel, etc.) Jared Valentine [EMAIL PROTECTED] ___ --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] 3Com Phones
Greetings, We are looking at installing a VoIP system with Asterisk and are currently looking at the line of 3Com phones. Has anybody had success with using the following phones? We need to buy a lot and we don't want to end up with phones that don't work properly with asterisk. 3Com 3101 (model with speakerphone) 3Com 3102 Business Phone 3Com 3103 Manager Phone 3Com 3105 Attendant Console (if these don't work, can somebody recommend another receptionist alternative?) Daniel Hazelbaker ___ --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
