Re: [Asterisk-Users] number that starts with star on PAP2
Yes, the dialplan field in the PAP (not asterisk's dialplan) was the problem. The dialplan used to have *xx in it (as well as lots of other stuff which we left alone), we changed that to *xxx (leaving the double *'s in all of the vertical service activation codes) and it now works. thanks Philippe Lindheimer wrote: You need to modify the dialplan within the PAP2 unit to allow that as a valid number or it won't pass it on. Take a look at the following, it is not specifically for the PAP2 but all the dialplan information should apply: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf From: Time Bandit [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 5 May 2006 09:48:41 -0400 Subject: Re: [Asterisk-Users] number that starts with star on PAP2 Why I did to mine is modify all the internal Vertical Service Activation Codes to be **x instead of *x. There is probably a better way, but this worked for me. We tried that, but users report they are still having the same problem (the site is located in a different country so I can't check myself). Sorry, I don't have my PAP2 under hand, but this is all I did, changed every *xx to **xx and it worked. Something that may help you is http://www.netphonedirectory.com/pap2_dialplan.htm Philippe Lindheimer wrote: Yes - that's your problem. You need to porgram the dialpan in the PAP2 appropriately, disable functions you don't want, etc. We were trying to dial *100, and there wasn't anything in either of the Codes section that started with *1. Do we have to disable every function that starts with a star to get anything to work? Also, is a function disabled by clearing it? I didn't try that so I don't know. Just make sure that you changed every single Vertical Service Activation Codes to a double *. If you still can't fix it, let me know and I will get back my PAP2 and try to help you ___ --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] number that starts with star on PAP2
I wrote: In the PAP2's setup there are all of these Vertical Service Activation Codes that start with star and Outbound Call Codec Selection Codes, also the setup menu is accessed by pressing star four times, could they be intefering with dialing numbers that start with a star? And is there any way to get *8 and *XXX to dial? Time Bandit wrote: Why I did to mine is modify all the internal Vertical Service Activation Codes to be **x instead of *x. There is probably a better way, but this worked for me. We tried that, but users report they are still having the same problem (the site is located in a different country so I can't check myself). Philippe Lindheimer wrote: Yes - that's your problem. You need to porgram the dialpan in the PAP2 appropriately, disable functions you don't want, etc. We were trying to dial *100, and there wasn't anything in either of the Codes section that started with *1. Do we have to disable every function that starts with a star to get anything to work? Also, is a function disabled by clearing it? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] number that starts with star on PAP2
We have some extensions in our dialplan that start with a star. We can dial them from Zap phones and SIP phones, but not from phones connected to a PAP2. After the user presses star follwed by two digits (our extensions are dialed with star followed by three digits) he hears a fast-busy that comes from the PAP2, not from Asterisk. This also happens with the builtin *8 (call pickup). In the PAP2's setup there are all of these Vertical Service Activation Codes that start with star and Outbound Call Codec Selection Codes, also the setup menu is accessed by pressing star four times, could they be intefering with dialing numbers that start with a star? And is there any way to get *8 and *XXX to dial? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PAP2 won't make two g729 calls at the same time
I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729 licenses, and no others were in use at the times this happened, but even if we didn't have enough, how would the PAP2 know that? Here's a good, and a bad INVITE message, from the log file with sip debug enabled. Has anyone seen anything like this? INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3e521aa From: PAP 220 sip:[EMAIL PROTECTED];tag=6b66e68deef168b2o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: PAP 220 sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 246 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 261305180 261305180 IN IP4 192.168.254.44 s=- c=IN IP4 192.168.254.44 t=0 0 m=audio 16392 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15 From: PAP 220 sip:[EMAIL PROTECTED];tag=b8b86be991749af5o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: PAP 220 sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Linksys/PAP2-3.1.3(LS) Content-Length: 267 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 261589835 261589835 IN IP4 192.168.254.44 s=- c=IN IP4 192.168.254.44 t=0 0 m=audio 16400 RTP/AVP 0 8 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ___ --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] Two FXOs getting bridged?
Dan Elder wrote: Hey all, I've been seeing this repeatedly and am wondering if anyone has a clue what's causing it.. at least once a day I see two zap fxo channels being bridged, and hanging..now, these two channels should never bridge, but they keep doing it.. any leads on where to look for what's causing it? here's what I see when I 'show channels' Zap/37-1 [EMAIL PROTECTED]:1Up Bridged Call(Zap/38-1) Zap/38-1 [EMAIL PROTECTED] Up Dial(ZAP/g0/16509524400) 37 38 are the 1st two FXOs on the system, connected to a channel bank (CAC ABI) with POTS lines. When I spy on the channels, there is no activity (audio) any leads as to where I should be looking to try to track this down? I'm unable to actually get an outbound trunk via the incoming lines, so I don't think it's a dialplan error..but who knows?.. using AAH 2.0 (* 1.2.1) Could someone have managed to transfer one outside call to another? That's happened a few times on my system, something like this: user calls first number, hangs up very briefly and picks up again (which is interpreted as a flash, so now the first outside call is on hold) calls the second number, eventually hangs up and now the two calls are bridged. Here's how it looks on the console. In this example I bridged a Zap FXS to a Sip phone but it looks the same with outside calls two Zap FXO's (I don't have any outside lines connected to my test system at the moment). I don't know how to stop this from happening, either (without disabling call transfer, which I don't want to do) other than telling everyone to make sure that they don't flash when they really wanted to hang up. I dialed 102, which dials Zap/2 -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, Zap/2) in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 here's where I flashed, putting Zap/2 on hold -- Starting simple switch on 'Zap/1-2' -- Started three way call on channel 1 -- Started music on hold, class 'default', on channel 'Zap/2-1' I dialed 139, which dials Sip/139 -- Executing Dial(Zap/1-2, Sip/139) in new stack -- Called 139 -- SIP/139-9fde is ringing -- SIP/139-9fde answered Zap/1-2 here's where I hung up, bridging Zap/2 and Sip/139 -- Stopped music on hold on Zap/2-1 -- Hungup 'SIP/139-9fdeMASQ' == Spawn extension (internal, 102, 1) exited non-zero on 'Zap/1-2 == Spawn extension (internal, 139, 1) exited non-zero on 'Zap/1-2' -- Hungup 'Zap/1-2' badger*CLI show channels Channel Location State Application(Data) Zap/2-1 [EMAIL PROTECTED]:1 Up Bridged Call(SIP/139-9fde) SIP/139-9fde [EMAIL PROTECTED]:1 Up Dial(Zap/2) 2 active channels 1 active call The show channels looks like yours, which is why I think it might be the same thing. Look in the log file if the lines that show how this happened have scrolled off your screen. One additional mystery is that I don't know why these calls persist. When I hang up either of the bridged extension on my test system, the bridged call ends. When a single outside call is hung up on the other side, asterisk notices. I don't have enough phone lines and cellphones to test if this works when two outside lines are bridged. Does external hangup detection work on your system? ___ --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] Problem calling out
I see these from time to time, I think it means that packets got lost, or received out of sequence. It looks to me like asterisk manages to deal with this, so unless your calls have also stopped working, I wouldn't worry. (If we should be worrying, I expect someone will let us know). [EMAIL PROTECTED] wrote: Hi All, I installed Asterisk recently and it was working from 2 weeks without a problem until today. Today it started showing strange error Feb 28 03:14:08 WARNING[31430]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED]' Whatever number I call it displays this, please tell how can I fix this? I have no idea what is happening and the cause of this error? ___ --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] user places two calls, hangs up, they get connected to one another
Leo Ann Boon wrote: Warren Burstein wrote: I've observed a situation on my production system, and have managed to recreate it on my test system (both running 1.2.4). I pick up a phone connected to a TDM400B's FXS line. I dial a number (in my tests, it was another local phone, but in production it was an outside call), and that call is answered. I flash, hear a stutter dialtone, and dial another number, which also is answered. Asterisk is treating it as call transfer. When I hang up, my two local calls are now connected. But on the production system, I think the two outside calls are connected. I don't know why they don't hang up. Perhaps the production system isn't detecting hangup (I'm going to test for this). Or maybe both calls are to someone else's switchboard, which have placed the calls into a queue, and we're tying up two outside lines to bridge the your call is important to us messages to each other. Is there some way I can forbid bridging of calls like this, but still allow a call to be bridged to a different local phone? thanks Can I make asterisk refuse to transfer one outside call to another (or ask for confirmation first) without disabling transfer to an internal number? The problem is that the user had no idea he had flashed, (he hung up quickly and thought he had disconnected the first call and didn't notice the stutter dialtone) and wasn't trying to bridge the calls, so he didn't realize he had tied up two outside lines. Once someone did this on two overseas calls. ___ --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] user places two calls, hangs up, they get connected to one another
I've observed a situation on my production system, and have managed to recreate it on my test system (both running 1.2.4). I pick up a phone connected to a TDM400B's FXS line. I dial a number (in my tests, it was another local phone, but in production it was an outside call), and that call is answered. I flash, hear a stutter dialtone, and dial another number, which also is answered. When I hang up, my two local calls are now connected. But on the production system, I think the two outside calls are connected. I don't know why they don't hang up. Perhaps the production system isn't detecting hangup (I'm going to test for this). Or maybe both calls are to someone else's switchboard, which have placed the calls into a queue, and we're tying up two outside lines to bridge the your call is important to us messages to each other. Is there some way I can forbid bridging of calls like this, but still allow a call to be bridged to a different local phone? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk logger - urgent!!!
I agree that systems should be well-adminstered, I also prefer programs that don't run amok even when there are lapses in administration. Since detecting which file caused the SIGFSZ is impractical, how about if we do this. a) don't rotate logs on SIGFSZ if it's done it recently. b) when it does rotate files on SIGFSZ,. it should rotate the csv file, too, and any other files that are written to (maybe only of they are larger than the file size limit) Kevin P. Fleming wrote: Warren Burstein wrote: How about if it would set a global variable before each disk write so the SIGFSZ handler would know which file caused it? Ha! Signals are asynchronous. This global variable would to be lock-protected, would require copying (possibly long) paths for every write, and would not necessarily be correct when the signal arrived. Sorry, this is not a solution. There is no solution, other than paying attention to your server and making sure that files don't get ridiculously large. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk logger - urgent!!!
Kevin P. Fleming wrote: Dov Bigio wrote: Any way, if any developers are reading this, I don't think that rotating asterisk logs is the best way to handle this problem! Maybe a more user-friendly message could be logged, infoming which file reached the 2.0GB. Unfortunately when we receive SIGFSZ from the kernel, we have no way to know which file caused it. The assumption in Asterisk is that the only files we write to that will ever reach that size are log files. If any other file does, there will be trouble, as you have seen. How about if it would set a global variable before each disk write so the SIGFSZ handler would know which file caused it? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
I took a look at the asterisk-1.2.3 Makefile, seems to me that the WARNING is just a list of all the .so files found in the modules directory that aren't also found in a subdirectory, it isn't checking that they were built with the current version. So it's going to complain about the modules that come from asterisk-addons every time make install is run in asterisk, no matter what. Not a big problem once you learn to ignore the message, but people are probably going to keep asking what it means. Julian Lyndon-Smith wrote: Warren, You may only use cdr_addon_mysql.so, but I believe that * normally automatically loads all modules it finds (see modules.conf for autoload=yes). The following modules were found in your modules directory, and 1.2.3 of * did not like them, because you got a warning after compile. In the case of app_rxfax.so and app_txfax.so these must of been compiled with a previous version of *, otherwise it would not have complained about them (I know this, because I had a similar issue). If you have kept the previous version of *, check your makefile for app_txfax and app_rxfax, make the same mods to your 1.2.3 makefile and recompile. * will then not complain about the *fax* modules. You may also need to recompile the asterisk-addons, simply because header files and or libraries may have changed in the core asterisk files. I guess what I am saying is that 1.2.3 of * may work with 1.2.1 of asterisk-addons (that is the latest version as you say), but asterisk-addons would need recompiling as well. If you make cleam;make and make install the asterisk-addons, do you get the same error when you make install asterisk ? Julian. app_addon_sql_mysql.so app_rxfax.so app_saycountpl.so app_striplsd.so app_substring.so app_txfax.so cdr_addon_mysql.so chan_modem_aopen.so chan_modem_bestdata.so chan_modem_i4l.so chan_modem.so format_mp3.so res_config_mysql.so Warren Burstein wrote: Julian Lyndon-Smith wrote: These modules are not part of the standard 1.2.3 release - did you also install the 1.2.3 release of the asterisk-addons package ? The lastest asterisk-addons I found at http://ftp.digium.com/pub/asterisk/ is 1.2.1. The only module I use is cdr_addon_mysql.so. I've been using it with 1.2.2 and 1.2.3 without any problem other than the message during make install, which I just ignore. Is there a need for an update to asterisk-addons? ___ --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] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
Julian Lyndon-Smith wrote: These modules are not part of the standard 1.2.3 release - did you also install the 1.2.3 release of the asterisk-addons package ? The lastest asterisk-addons I found at http://ftp.digium.com/pub/asterisk/ is 1.2.1. The only module I use is cdr_addon_mysql.so. I've been using it with 1.2.2 and 1.2.3 without any problem other than the message during make install, which I just ignore. Is there a need for an update to asterisk-addons? ___ --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] WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38
I didn't find that exact message in the RFC's, but I did find something similar in RFC 3407 (http://www.rfc-archive.org/getrfc.php?rfc=3407), a=cdsc: 4 image udptl t38 Which means that the sender is capable of sending T.38 fax over UDP. I wouldn't worry about it unless you were trying to receive a T.38 fax over UDP, or it causes some other problem. If you need to get further into this, run sip debug from the console so you can see the entire SIP message in which this line appears. Giorgio Incantalupo wrote: Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo ___ --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] DTMF not working on overseas cellphone calls
I thought I sent this earlier this week, but I didn't see it. If I missed it, I apologize for the resend. We are running asterisk 1.2.2 with a TDM04B connected to PSTN lines. On incoming calls from cellphones located overseas, DTMF is not recognized - we have many single-digit choices in our menu so the problem isn't that some digits aren't working, it's not listening at all. Works fine from domestic landlines and cellphones and from overseas landlines. I know the cellphones don't have a problem with DTMF, they work with other IVRs. I've placed overseas calls (I'm currently in a different country from the asterisk machine) from both landlines and cellphones, and can't hear a difference in quality. Could playing with rxgain help? Is there any chance that I could cause the calls that don't have a problem to either be too loud or get distorted due to clipping? ___ --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] DTMF not recognized on overseas call from cellphone
We have PSTN lines connected to FXO lines of a TDM400B. I just got a complaint that overseas callers who are using cellphones sometimes find that DTMF digits aren't working - they press digits and the menu goes on as if they hadn't pressed anything. Since it sometimes works, and other IVRs work over the same cellphones, it's not that the cellphone isn't sending the digits. I asked if they had similar problems from landlines overseas and they did not. Any ideas? If I play around with rxgain could I overload something for callers who don't currently have any problems? ___ --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] outgoing calls that last an unreasonably long time
I'm running asterisk 1.0.9 with TDM400B's for both internal and external lines. I put in the macro that dials outside lines an AbsoluteTimeout(36000), never expecting it to happen. But it does, a few times a month. I've noticed an odd thing, it seems that it usually happens twice in a row from the same internal phone (connected to a TDM400B, not an IP phone) as if someone dialed a number, something went wrong, they flashed and dialed again. What happened next I don't know. If they left the phone offhook for the rest of the day, that could explain how they managed to keep two outside lines busy. What is frustrating is that the cdr file shows the dst as T rather than as the phone number dialed. I realize that AbsoluteTimout causes it to jump to the T extension, but it would help to know who the user dialed (asking a week later isn't going to get any useful information out of the user). It's not in the log file, either - would increasing the log level help here? I'm reluctant to decrease the abosolute timeout because someone is going to come to me saying I was on hold for five hours for tech support and before I finally got a human, I was disconnected and had to wait all over again. Is there something I could run from the console that would show how long each channel has been connected, and to who? That way we might be able to catch the next one of these as it happens instead of much later. ___ --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] outgoing calls that last an unreasonably long time
Simone Cittadini wrote: Warren Burstein ha scritto: What is frustrating is that the cdr file shows the dst as T rather than as the phone number dialed. I realize that AbsoluteTimout causes it to jump to the T extension, but it would help to know who the user dialed (asking a week later isn't going to get any useful information out of the user). It's not in the log file, either - would increasing the log level help here? I don't know how this AbsolutTimeout works, anyway I put all the info I need in variables before the actual Dial, then in the h extension I call SetUserField() (or whatever is called), helps me keeping track of reasons for non-terminated calls ... I am not using the userfield for anything so that sounds like a good idea. It's SetCDRUserField by the way. ___ --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] One Touch Record in 1.2
Tim Litwiller wrote: Well, I'd like them to drop in my voicemail when done recording - maybe in a separate recordings folder but I'd like to use the same interface to play them back. I would like that, too. Is anyone working on it? If not, I will put it on my TODO list. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call wating and call transfer
Recently I put callwaiting=yes in zapata.conf because customers want to speak to the operator in person, not leave her a voicemail, when she's busy with another caller. But now she can't transfer either of the calls (which she can do when there's only a single call). The operator has an analog phone connected to a TDM400B FXS line. The calls are coming from PSTN lines connected to FXO lines on the TDM400B. I searched the wiki and the list archives, found a message from this January, http://lists.digium.com/pipermail/asterisk-users/2005-January/082367.html, about the same issue, saying you can try using # as a way of transfering the call, but that's a blind transfer meaning that you will be prompted an extension number and the call will be transfered and that's it It wasn't clear to me what sort of channel they were talking about, but that would be fine, our dialplan will let the caller get back to the operator if the transferred extension is busy or doesn't answer. I haven't used this feature before (I told the users to transfer calls by flashing the switchhook) I tried on my test system in my office (the production system is in another country) to dial one Zap FXS from another and hit # and nothing happened. Does the # transfer only work when there is more than one call arriving on a channel? Or does something need to be done to enable this? I searched the wiki and list for enable unattended transfer and enable blind transfer but didn't find the answer. I did find in http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20zap%20transfer, How to transfer a call on a phone connected to a ZAP channel, a remark, You can try # as well instead of flash but nothing about how to enable # transfers. Is there a difference between 1.0.9 (the production system) and 1.2.0-beta1 (the test system)? Yes, I know I shouldn't be testing on a different version ... Here's the zapata.conf on my test system. I didn't enable callwaiting there - does that affect # transfers? [channels] language = en musiconhold = default signalling = fxo_ks context = internal threewaycalling = yes transfer = yes group = 1 pickupgroup = 1 callerid = 101 echocancel = yes channel = 1 callerid = 102 channel = 2 signalling = fxs_ks context = pstn group = 2 busydetect = yes channel = 3-4 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions beginning with *
Arik Funke wrote: can anybody tell me how to create an extension that starts with a *? The expression matching works well if * is embedded in numbers but if the extension starts with *, it is not executed but extension s instead. Is there another way besides using a lot of if statements in the s extension? This works for me on asterisk-1.0.9 dialong on an analog phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 x TDM400P in one PC ??
Joerg Wleklik wrote: Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? We've got four TDM400P cards in one PC running without any trouble, since January. It's not a fair comparison between new cards that come from Digium with support and a second-hand channel bank from Ebay. If an analog phone interface doesn't work, I can figure out if the problem is the card, the wiring, or the phone with a voltmeter, and unless the entire card gets fried, it's just one phone that's out, and even so, unless the entire computer is fried it's only four. If a T1 interface, technology with which I have no experience, fails, the entire system is out, and I don't have a clue how to find out what the problem is, is it the wiring? The channel bank? Is tab A plugged into slot B? Did the channel bank turn up on Ebay because someone else couldn't get it to work either? Lastly, I was told that I would have to bring in a technician to wire up a channel bank, I didn't have to do any wiring with the TDM400P's - I plugged in the cards, took the RJ-11's out of the PBX I was replacing, and plugged them in to the cards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 x TDM400P in one PC ??
C F wrote: I wish you would know what you are talking about. We've got four TDM400P cards in one PC running without any trouble, since January. Good for you, most people don't have it this way. It's not a fair comparison between new cards that come from Digium with support and a second-hand channel bank from Ebay. If an analog phone Really Digium cards in such a setup *do have problems* (ask Digium the will tell you the same), while a working channel bank if it works *does not* have problems. interface doesn't work, I can figure out if the problem is the card, the wiring, or the phone with a voltmeter, and unless the entire card gets fried, it's just one phone that's out, and even so, unless the entire computer is fried it's only four. If a T1 interface, technology with So why do phone companies not stick to a pair a copper for each line? why do they use fiber lines? which I have no experience, fails, the entire system is out, and I don't So get the experience, by crying I don't have the experience you will never get it. have a clue how to find out what the problem is, is it the wiring? The Well it's actualy easier with a T1, since there is only one line to check, while with your 3 TDMs you have to first find the pair. channel bank? Is tab A plugged into slot B? Hmm, so you telling me that for wiring the TDMs you use pre crimmped wires even when using 12 runs? you outa go to wiring school as well, why don't you use a 66 or 110, it's much neater, and easier to expand, move, or change. Did the channel bank turn up on Ebay because someone else couldn't get it to work either? Looks like you don't know eBay (you even mispelled it). Lastly, I was told that I would have to bring in a technician to wire up a channel bank, I didn't have to do any wiring with the TDM400P's - I plugged in the cards, took the RJ-11's out of the PBX I was replacing, and plugged them in to the cards. Exactly, because you havn't got a clue about wiring, you shouldn't be touching phone systems at all. ___ I'll apologize to the users and offer to put back the mechanical PBX and 4-line phones that they were using before I committed my outrage. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: some questions about busy detection
Warren Burstein wrote: I'm going to be hooking FXS lines on a TDM400 to a PBX which doesn't drop line voltage at the end of a call, so I'm going to have to use busy detection. A few questions - The tones are taken from the tones specified by the zone in zaptel.conf, right? Which tones cause hangup? The PBX may not use the national standard tones. Does anyone have any suggestion for how I can determine what tones it uses if the information is not in the PBX manual? Back when I used to use Dialogic cards, there was a program called PBXpert (or something like that) which would measure all the PBX's tones. But I think all I need is to record a tone and get a program that would tell me what tones/cadence are in the sample. Cadence I could probably figure out from any graphical sound editor, tones too if it's single-frequency, but what if it's dual-frequency? Is there a sound editor (free, or at least with an evaluation version that has this feature) that would do this for me? Well I wound up answering my own question. It turned out that asterisk was able to hang up when the PBX ended a call, but a separate IVR system connected to the PBX wasn't disconnecting when asterisk ended a call. So I dialed into the IVR from a local phone, an external phone, and from asterisk, went into voicemail, and hung up. Now I had wave files with all three termination tones - the IVR left a few cycles of the tones it got on local and remote hangups in the voicemail (because it recognized those tones and stopped recording, but fortunately left what it had already heard so far in the file) and quite a lot of the asterisk tone (because it didn't recognize it, good thing there is a limit on voicemail messages in the IVR or I would still be waiting for it to finish). I remembered that I used to use an editor on Windows named Cool Edit, that did frequency analysis. The current version turns out to be a rather expensive program for something I don't need to run all that often (there is also a time-limited demo version of the current product), but I found an old shareware version at http://www.threechords.com/hammerhead/cool_edit_96.shtml. Highlight a section of the sound, and it tells you how long it lasts, and hit alt-Z and it performs a frequency analysis on the highlighted section. It turned out that a disconnected internal call left a fast-busy, 400/500, 0/500, an external call left the same but both parts were 250 msec, and a call from asterisk ended with 480+620/250, 0/250, which is ZT_TONE_CONGESTION in zaptel's zonedata.c (I am using the US tones). The easiest thing was to just change the congestion tone there to 400/250,0/250, restart everything, and it now works. Now I'm wondering if asterisk detected the fast and slow busy tones (maybe not, because they're not the US busy tones), or if the PBX dropped voltage. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callers who don't press any keys
I've noticed that some callers listen to our main menu and don't press any keys. I have it set up to restart the menu a few times and eventually hang up. I'm wondering if these are wrong numbers (in that case, why don't they hang up) or they really want to speak to someone here but don't understand the menu (what's so hard about for the operator, press zero?). They couldn't still have rotary phones (or phones set to pulse dial), could they? I've been thinking of changing the menu so that if they don't press any keys, the eventually get the operator. Does anyone have any experience with this? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI on Zap analog phone not lighting
We are using Bellsouth 8867 phones on our TDM400B FXS lines (asterisk-1.0.3). It has a Voicemail light, which appears to be MWI (according to the manual it works with voicemail from the telco that sends a FSK signal). The dialtone stutters when a line has voicemail, so I know that I have the mailbox setting right in zapata.conf, but the light doesn't go on. I am also getting caller-id on the phone's display, so I guess that shows FSK works from the card to the phone. I did some searching before posting, found http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20zapata.conf which says On supported hardware, the message waiting light http://www.voip-info.org/wiki-PBX+Message+Waiting+Indicator will also be activated this probably requires that you also set adsi=yes. Update: This option does NOT require ADSI. It will send a standard FSK tone down the line that lights up the MWI on any capable analog phone. That looks like it should be working. I didn't find anything on this list that I recognized as the answer to this problem. I am not sure if the phone needs batteries to do CID and MWI or not, but just to be safe I put in batteries. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] not sharing IRQ's
I'm not having any trouble with interrupts, but here's my /proc/interrupts on Fedora Core 2 on a hyper-threading CPU and using the SMP kernel (2.6.5-1.138). I don't think I need to worry about uhci_hcd, nothing is plugged into USB, but libata is the disk driver. How do I get libata and wctdm to use different interrupts? $ cat /proc/interrupts CPU0 CPU1 0:39957053931405IO-APIC-edge timer 1:530489IO-APIC-edge i8042 2: 0 0 XT-PIC cascade 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 12: 56 0IO-APIC-edge i8042 15:489 0IO-APIC-edge ide1 169:51072365082420 IO-APIC-level libata, uhci_hcd, wctdm 177:2136633 0 IO-APIC-level eth0, Intel ICH5 185:10019076889735 IO-APIC-level uhci_hcd, wctdm 193: 0 0 IO-APIC-level uhci_hcd 201: 0 0 IO-APIC-level ehci_hcd 217:59781561900756 IO-APIC-level wctdm 225:19173325960110 IO-APIC-level wctdm NMI: 0 0 LOC:79268527926712 ERR: 0 MIS: 0 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] not sharing IRQ's
Michael Welter wrote that I should be worried about the usb module. Would rmmod uhci_hcd be enough, or should I disable it in the BIOS like Shoval said? Also, after the rmmod, I still have the conflict with libata on 169 CPU0 CPU1 0:73110067252568IO-APIC-edge timer 1:530489IO-APIC-edge i8042 2: 0 0 XT-PIC cascade 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 12: 56 0IO-APIC-edge i8042 15:489 1326IO-APIC-edge ide1 169:84365328405830 IO-APIC-level libata, wctdm 177:3896993 0 IO-APIC-level eth0, Intel ICH5 185:1001907 13525492 IO-APIC-level wctdm 201: 0 0 IO-APIC-level ehci_hcd 217: 118384162607961 IO-APIC-level wctdm 225:2687600 11826296 IO-APIC-level wctdm -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dead line (no LED) on a TDM400B?
I moved my TDM400B cards (first two cards are 40's, third is a 31, last is an 04) from one computer to another, copied all the config files, and now the LED on the line 11 - third line of the third card doesn't go on (it used to on the previous computer). I can get by telling * not to use this line for now but does anyone have any suggestions for getting it to work? Unseat and reseat the daughterboard? Call Digium for support? /etc/zaptel.conf contains: fxoks=1-11 fxsks=12-16 loadzone = us defaultzone=us -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Russian characters showing up on safe_asterisk console in RedHat 9 and Fedora Core 2
Here's a strange one - when I run safe_asterisk on either of these distros, words that are colored blue or violet (but not red) turn up in Russian (and some other languages, I think). If I run asterisk with the same arguments (-vvvg -c) as safe_asterisk does, from the console, it's OK. If I run it in a Putty window it's OK. If I run asterisk -r from another console or from Putty it's OK. So I ran asterisk in 'script' and cut a line containing some blue text (Registered application 'Exec', the word Exec is in blue) and sent it to all my virtual consoles. It looks OK on tty1 thru tty8, but on tty9 and up it's both blue and Russian. So I just changed TTY from 9 to 8 in safe_asterisk, but this is the sort of trivial problem that keeps me up at night, does anyone know if there's something different about tty9 and up? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can the dialtone be changed after pressing 9?
extensions.conf has
ignorepat = 9
exten = _9X.,1,Dial(Zap/G2/${EXTEN:1})
The first user to try it asked if instead of keeping the same dialtone
after pressing 9, if I could play a different dialtone. Can this be
done? I'm running asterisk 1.0.0 in case that matters.
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[Asterisk-Users] Subject: Re: Dial with no phone line connected
Rich Adamson wrote: How old of code are you looking at? The wcfxs driver was renamed to wctdm some time ago. Current cvs doesn't include it. I was using zaptel-1.0.0, but I took a look at 1.0.3 and it's still wcfxs there. I'm about to bring this online, would rather stick with releases. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] disable ringback of held call on zap channel
One Zap FXS channel has dialed to another. Zapata.conf has transfer = yes and threewaycalling = yes. I flash on one of the phones, the other gets the music on hold. If I hang up the flashed phone, it rings back and I am reconnected to the other phone. Is there some way (with flash, not with #) that I could leave the other phone on hold for a longer time? Preferably without having to dial something after the flash. thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Subject: Re: Dial with no phone line connected
I have more FXO ports on TDM400's than I have PSTN lines available for
testing. When all the lines were used up (the FXO ports are all in
zap group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial
succeeded even though there is neither line voltage nor dial tone.
Can at least the lack of voltage be detected? It would be good in
case one of the phone wires fell out that it would just move on to the
next outgoing line.
Yes, the chip set on the TDM card does provide flags for indicating no
voltage (disconnected), low voltage (something is off hook), and normal
pstn voltage (on-hook).
About three months ago, Mark added code that detected when a pstn line was
unavailable (eg, rj11 disconnected, damaged cable, someone disconnected
the wrong pstn line). The code created a problem for someone (I don't
remember the details), and he changed the code to be a compile-time config
option.
I don't have any past references to that other then from memory. Maybe
someone that can read code can find that option for you.
I read code, so I looked for this code in the sources, I remembered that
fxstest stats prints the voltage, found that it did an ioctl
WCFXS_GET_STATS, and searched for this in both asterisk and zaptel (version
1.0.0) of each, and found the only place that this constant appears is in
fxstest and the wcfxs driver. Is there a different way to test for a
disconnected cable other than this ioctl?
Could it have been taken out entirely and not just ifdef'd out?
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[Asterisk-Users] Sending call to analog then to Vmail after timeout?
I came across the same problem today. Phase one of our * project is to replace the PBX in one of our offices with *, and one of the extensions will be sent over VOIP to service representatives at a different location. But as a fallback, we want to dial directly if VOIP doesn't work (maybe the network is down) and if that doesn't work, send the caller to voicemail. And the same problems are happening. I'm not sure if shmaltz's solution helps us here, during working hours at the place where the service reps sit, it will go into a PBX (not *), I think it will ring on the rep's desk, using Pavlovian techniques (e.g. send the ring voltage to their chair if they don't press a key), they can be trained to hit a key when they pick up the phone. But when the help desk isn't staffed, we want to ring someone's cellphone. Which is fine if he answers, but if he misses the call, or is already taking a call, we want it to go to his cellphone's voicemail (which is more accessable to him outside the office than his voicemail on the * server). Maybe we can put a DTMF digit into the cellphone voicemail greeting. On the other hand, maybe hitting any key while recording the greeting will stop recording. Then I was thinking about phase two of our project. In our main office, we currently have a PBX which we're not planning to replace, but we do want to replace our IVR/voicemail system that runs on Windows using a Dialogic card with an * system, the dialogic picks up incoming calls, transfers them (sending the PBX a flash followed by a local extension) to people, and detects if they answer or not. The Dialogic card knows how to detect ring, busy, no answer, no dialtone, congestion tone, and so on, so if the callee does not answer the call, it goes to voicemail (or to a different person, or back to the main menu so the caller can try someone else). It looks to me like we would have to have everyone in the office press a key every time any phone rings, I'm not sure if that's acceptable. Would call progress be any help? From what I've read, it can sometimes cause calls to be disconnected. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial with no phone line connected
I have more FXO ports on TDM400's than I have PSTN lines available for
testing. When all the lines were used up (the FXO ports are all in zap
group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial succeeded
even though there is neither line voltage nor dial tone. Can at least the
lack of voltage be detected? It would be good in case one of the phone
wires fell out that it would just move on to the next outgoing line.
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[Asterisk-Users] Can I tell if it hung up due to busydetect or disconnect supervision?
The FXO lines on my TDM400 are connected to PSTN lines in Israel. As far as I know, the lines around here have disconnect supervision (I've seen some other Israelis on this list, anyone know for sure?), because it's worked on Dialogic cards, which reported hangup, not busy detect (while when I connect a Dialogic card to a PBX, I have to measure the busy signal's frequency/cadence or it doesn't disconnect). I couldn't tell for sure - I don't have a phone with a light, and the only voltmeters I have are digital, no way to see if there is a drop to zero between the several volts when the line is offhook to the 47 volts when it is onhook. But * would only recognize hangup if I unplugged the PSTN line. Otherwise it would play my voicemail menu over and over if I hung up without choosing an extension, good thing I put in a counter and did Hangup after a few tries. So, I turned on busydetect, and now disconnect works (I was surprised that it worked right away, I didn't even need to find out where the busy signal is defined, maybe the busy here is the same or close enough to the US tone), but I was wondering, is there any way I can get the console to let me know whether it disconnected due to voltage drop or busy detection? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Hook/Flash, Hold, Call Waiting, Three Way Calling
Do you have threewaycalling and transfer set in your zapata.conf? Here's mine (four TDM400's, seems to be working so far). I didn't do anything in my extensions.conf for any of these features (what confused me at first is the t and T options of the Dial application in extensions.conf are for transfers via the # key), when you flash you get another dialtone that works just like the first one. I haven't tried three way calling yet, I only have two hands, but transfer and hold (with music) work. zaptel.conf --- fxoks=1-11 fxsks=12-16 loadzone = us defaultzone=us zapata.conf --- [channels] language = en musiconhold = default signalling = fxo_ks context = internal threewaycalling = yes transfer = yes group = 1 channel = 1-11 signalling = fxs_ks context = pstn group = 2 busydetect = yes channel = 12-16 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] music on hold without sound card
http://lists.digium.com/pipermail/asterisk-users/2004-September/061677.html says If you don't have a soundcard then try just loading the sound module, that might just be enough. I don't know what that means. What sound module? In Asterisk? In Linux? Yes, I have mpg123 0.59r ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfer: hookflash vs #
I think Ive managed to figure out that there are two ways to transfer a Zap call, using hookflash (defined in zapata.conf) or the # key (the t and T options of the Dial command in the dialplan), but not why there are two ways to do this, nor what the difference is between them. Is there something that explains this? thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] does a TDM04B (all FXOs) need a power connector?
Is the power connector on the TDM400P only needed for line and dial voltage, or do you also need it if it has all FXO lines? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gap in priorities - what happens
When I first saw the priority numbers in extensions.conf, I thought BASIC, if a number is missing, * will fall thru to the next number. I learned that this is not so, if you have nothing between 1 and 3, you don't ever get to 3. But I'm wondering what does happen? Hangup and wait for next offhook? Undefined? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Voice Prompt Info
Ariel Batista wrote: Warren Burstein wrote: One more thing about prompts, it's better to say for sales press 5 than press 5 for sales, because by the time you hear sales you've already forgotten what number it was. If you add the sounds all you need is For Sales recorded the new sounds have press # already. So you don't need to get any additional recorded items except the one that says For Sales by Allison. If you want have her record Press as an additional recorded item. She already did a message containing just press, sounds/vm-press.gsm in asterisk-1.0.0 (not asterisk-sounds). I was thinking about a project I have to bring up very soon won't be time to wait for new recordings, so we are going to record the messages ourselves, but yeah, if we're making messages for everyone to use, we already have press and press 1. Anyway, the departments we need are support, sales, projects. systems, and operator. There already is %for-tech-support.gsm%For technical support %to-reach-operator.gsm%to reach an operator so all we would like to add is sales, projects, and systems. Maybe so all the messages are similar, add for an operator as well. Although we might decide to stick with our own recordings, in case later we need to change a message ASAP. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] can a TDM400P FXS drop voltage on hangup?
I thought I had posted this, but I didnt see it in the archives, so I guess I hadnt. Ive got FXS lines going to a legacy IVR. When I Dial into one of these lines and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I would like the IVR to hang up sooner. I could do this by either making the IVR recognize the standard Congestion tone, or changing the Congestion tone to be one that the IVR already recognizes (by the way, I was surprised to find that Zap tones are compiled in, not in indications.conf any thought of changing this (with backward compatability, of course)? I might be able to do this myself). But if I could get the FXS to drop voltage instead of play Congestion (or a second of Congestion in case a person is listening, and then drop voltage) that would be even simpler. But can I make that happen, and how? thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Voice Prompt Info
One more thing about prompts, it's better to say for sales press 5 than press 5 for sales, because by the time you hear sales you've already forgotten what number it was. So record for sales press and the digits (you could use the digits that come with *, but a sentence in two voices sounds very funny, I know, the user directory on an old IVR of ours works that way). That way when you need to change the numbers the menu you can do it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to make asterisk drop battery on a FXS?
I connected two plain old telephones to FXS lines of a TDM400P (defined as fxoks in zaptel.conf), one of them dials the other with Dial(Zap/2), I talk to myself for a while, hang up either of the phones, and the phone that remains off-hook gets the congestion tone until it goes on-hook (at least as long as I've cared to wait). I don't have a voltmeter on the line, but if I'm hearing the congestion tone, there's battery, right? The reason I want to drop battery is that I want to connect some FXS lines to an old IVR system, and if I dial the IVR and hang up, the IVR doesn't notice. Another possibility is to see if I can change either the IVR or Zaptel (why doesn't the Zap channel get its tones from indicators.conf, like other channels do, instead of having them compiled in?) I spent a while playing with ) so that the IVR recognizes the congestion tone and hangs up, but battery drop, if I can do it, sounds like a better way to do it. thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ringing() doesn't play sound while phone is ringing
I have: RedHat 9.0 TDM40B asterisk-0.9.0 compiled from sources zaptel-0.9.1 likewise /etc/zaptel.conf contains fxoks=1-4 loadzone = us defaultzone=us loaded modules zaptel and wcfxs /etc/askterisk/zapata.conf contains [channels] language = en signalling = fxo_ks context = phones channel = 1-4 /etc/askterisk/extensions.conf contains [general] static=yes writeprotect=yes [phones] exten = 101,1,Ringing() exten = 101,2,Dial(Zap/1,10) exten = 101,3,Congestion I also uncommented the noload = chan_oss.so in /etc/asterisk/modules.conf because I dont have a sound card. Other than that, all conf files are the originals from make samples. But when I dial 101 from Zap/2, Zap/1 rings (and if I pick it up, I can have a conversation with myself), but I dont hear a ringing tone out of Zap/2. I commented out the Dial and Congestion, and then I heard a two ringing tones, a click, and a congestion tone, while the console said: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in context 'phones' Im guessing that Dial stops Ringing. How do I tell Ringing to continue while Dial is working, and if it isnt stopped by Dial, not to time out after two rings? show application ringing doesnt describe any parameters to Ringing() Thanks.
RE: [Asterisk-Users] Ringing() doesn't play sound while phone is ringing
That did it. I thought Ringing() did that, but I guess it's just for when you want to fake a ringing tone. I'll add a comment to http://www.voip-info.org/wiki-Asterisk+cmd+Ringing. thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bart Coppens Sent: Wednesday, August 11, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Ringing() doesn't play sound while phone is ringing try exten = 101,2,Dial(Zap/1,10,r) instead of exten = 101,2,Dial(Zap/1,10) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
