[asterisk-users] 1.4 sounds long space before and after prompt
Is anyone else finding in the new audio files that the longer space at the beginning and end of the files tends to be extremely irritating? An excellent example is when going into voicemail and Allison says how many messages you have, the space between the files is annoyingly long: you have .. four .. old .. messages ..and.. first .. message .. received . July . twenty .. second Under the old sound files, this continuity was still a little long, but workable. The new sound files make these positively sound like a computer playing individual files rather than a continuous sentence. If I release these sound files as they are to my users, they are going to revolt. They already complain about the old Octel VM system prompts being played back too slowly and these are much slower than that. I mentioned this a while back when the new sounds were in beta, but haven't seen anything more about it. So either this says something about my and my users' level of patience, I'm missing something that changed between 1.2 and 1.4 that could fix this, or the focus has been on lower-level issues with 1.4 than on the sound files. With the new higher-quality sound files, I could manually edit all the offending files (there are lots of them) and correct what I perceive to be a problem. However, if this is a common enough complaint, maybe others would want to help as well, and we could get the fixed files put back into core Asterisk. Note that this doesn't appear to be a problem with the speed of the sound files as some others have experienced. The tempo is probably okay, and the pitch is fine. It's the spacing between files that's the issue I'm talking about. Thanks in advance for any feedback. --- Gil Kloepfer [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] Inconsistency with ANI and channel callerid
I've recently noticed some oddities in my CDR records. In some cases
the original CallerID that I've set in the .conf file for the extension
showed-up in the CDR as the originating extension (on Zap/ devices on
the channel bank), and in other places it was the one that I set
using Set(CALLERID(num)=something) (SIP/ devices).
Digging around a little in the source (and doing some printing of stuff
in the dialplan), it appears as that:
1. The CDR prefers to use the ANI as the from extension, and if
that doesn't exist, then use the CALLERID(num)
2. If you specify callerid= in the {sip,iax,mgcp,zaptel}.conf
file, the ANI only gets set for Zap/, and I think MGCP/ and vpb/
devices.
So my question: Is this a bug or a feature? Am I missing something?
It would seem that the right behavior would be one of consistency -- if
someone specifies the callerid= option in any of the channel .conf files,
then it should either set or not set ANI, but not behave differently for
different channels.
The reason this problem came to light is that we have both DID and non-DID
internal numbers. The non-DID numbers can make outgoing calls to the
PSTN, and their outbound CallerID when hitting the PSTN is set to
our main reception desk. What I'd LIKE to do is maintain the
originating number in the CDR so that we have a record of the
actual non-DID extension making outgoing calls, but set the reception
desk as the outbound CallerID on the PSTN. What's happening now is
that this is working for Zap/ devices, but not for SIP/ devices because
of the problem noted above.
A workaround I could use would be to make a Macro that is something
like this:
[macro-ChangeCIDnum]
exten = s,1,GotoIf(${ISNULL(${CALLERID(ANI)})}?:aniset)
; If ANI isn't set, then set it to the current CALLERID(num) before
; we change the CALLERID(num)
exten = s,n,Set(CALLERID(ANI)=${CALLERID(num)})
; Now change the CALLERID(num)
exten = s,n(aniset),Set(CALLERID(num)=${ARG1})
exten = s,n,MacroExit
and use this every time I would normally Set(CALLERID(num)=something).
But this seems like the wrong thing to do.
Thanks in advance for any feedback.
---
Gil Kloepfer
[EMAIL PROTECTED]
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[Asterisk-Users] VMauthenticate always asks for mailbox
I've been trying to use the VMAuthenticate function in 1.2+. This function is supposed to behave[s] the same way as the Authenticate application, but the passwords are taken from voicemail.conf. The problem is that it always gives the comedian mail prompt and requests the mailbox number, even though I provide the mailbox number already. The upshot is that VMAuthenticate is not working like Authenticate and makes the user think they are connecting to voicemail. My motivation to use VMAuthenticate was to provide for a single password (the voicemail password seemed like a good idea) for users to do various things. Am I missing the point of VMAuthenticate or is this a bug? If it's a bug, I know where it is...but I wanted to ask before going down the bugpath road. (Techie note: In function vm_authenticate() in app_voicemail.c, when skipuser is set to 0, the mailbox number is always requested, and that's the way it is called in the VMAuthenticate app.) Thanks! --- Gil Kloepfer [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
Re: [Asterisk-Users] VMauthenticate always asks for mailbox
On 1/10/06, Gil Kloepfer [EMAIL PROTECTED] wrote: The problem is that it always gives the comedian mail prompt and requests the mailbox number, even though I provide the mailbox number already. On Tue, Jan 10, 2006 at 10:00:44AM -0500, C F wrote: Are you supplying the context? Actually I wasn't (according to the help it is optional). However, even with the context it is behaving the same way. Is it correctly working for you? If so, would you mind including an example of what you're doing so I can try that? Here's a trace on my PBX: -- Executing Goto(SIP/3771-9210, Features|11945485|1) in new stack -- Goto (Features,11945485,1) -- Executing Macro(SIP/3771-9210, TestFeatureEnabled|5485|CallFwdUncond) in new stack -- Executing GotoIf(SIP/3771-9210, 1?:featnotenb) in new stack -- Executing GotoIf(SIP/3771-9210, 1?:featnotenb) in new stack -- Executing NoOp(SIP/3771-9210, CallFwdUncondIsEnabled) in new stack -- Executing Answer(SIP/3771-9210, ) in new stack -- Executing VMAuthenticate(SIP/3771-9210, [EMAIL PROTECTED]) in new stack -- Playing 'vm-login' (language 'en') Jan 10 09:24:33 WARNING[8797]: app_voicemail.c:4912 vm_authenticate: Couldn't read username == Spawn extension (Features, 11945485, 3) exited non-zero on 'SIP/3771-9210' (I hung-up on it after it asked for the mailbox) Note that [EMAIL PROTECTED] exists and has a password. --- Gil Kloepfer [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
Re: [Asterisk-Users] format_mp3 uninstalling mpg123
On Tue, Dec 13, 2005 at 05:12:44PM -0200, Dov Bigio wrote: For that, the wiki says Be sure to remove mpg123 from your system (this may attribute to 'Request to schedule in the past!?!?!' messages). Now you are set! How do I uninstall mpg123? How did you install mpg123? If you installed it with the package management system, then use the package management system on your OS to remove it. If you installed it manually, you'll need to remove it manually. To actually allow format_mp3 to work you also need to change musiconhold.conf from mode=quietmp3 to mode=files. Hope that helps --- Gil Kloepfer [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
Re: [Asterisk-Users] format_mp3 uninstalling mpg123
On Tue, Dec 13, 2005 at 05:47:47PM -0200, Dov Bigio wrote: Actually I did it manually (tar -xvzf)... but I am not sure which files I have to delete manually.. is there an explanation somehere? I couldn't find it on Google... If you do a tar -tvzf of the same tar archive, those will be the files you will need to delete. Essentially, you'll be uninstalling mpg123 entirely. To actually allow format_mp3 to work you also need to change musiconhold.conf from mode=quietmp3 to mode=files. This is new for me... I didn't find any information on this mode parameter... Should it be put under [classes] or [moh_files] in musiconhold.conf??? I made a bad assumption that you were running 1.2.*. If you're running 1.0, then there are other parameters you'll need to change but I can't recall what they are right now (and unfortunately I don't have any of my old config files around to find out). Sorry about that. --- Gil Kloepfer [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
Re: [Asterisk-Users] Bad FCS nightmare to Nortel SL100 with TE410P
On Tue, Sep 27, 2005 at 01:22:48AM -0500, I wrote: I have an * box connected to a Nortel SL100 through a PRI (US) using the Digium TE410P (quad-span T1 card). I don't have access to the SL100 - it is handled by another group. The span comes up OK (timing, framing fine). However, as soon as the D channel comes up, I get endless HDLC Bad FCS errors. We have resolved the issue - it turned out that there was something wrong at the SL100 side of the connection. The bad news is that I can't get anyone who manages that SL100 to tell me exactly what the problem actually was. The best I was able to get was, It was some kind of cable or configuration problem. The only private feedback I got were recommendations to try the Sangoma cards (it looks like they keep more in the way of statistics). I also have no idea why the TE410P complained, but the Cisco AS5350 didn't. Thanks to everyone who provided feedback. --- Gil Kloepfer [EMAIL PROTECTED] ___ --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] Bad FCS nightmare to Nortel SL100 with TE410P
I have an * box connected to a Nortel SL100 through a PRI (US) using the Digium TE410P (quad-span T1 card). I don't have access to the SL100 - it is handled by another group. The span comes up OK (timing, framing fine). However, as soon as the D channel comes up, I get endless HDLC Bad FCS errors. I modified logger.conf to get rid of the messages (so I could see what else was going on), and noticed that the B-channel restart was going horribly slow, and the D channel was essentially flapping up and down. I could sometimes squeeze a call in while the D channel was up, but it would only last a few seconds. I also get short write errors as well (unfortunately I don't have a log of these and can't get at the PRI at the moment to get the exact error message). I've had the physical circuit tested and there are no issues with it. In fact, it was working fine to the same switch as an EM digital trunk up until we tried to change it to a PRI. I've tried 3 different TE410Ps on three different * versions (based on things I've seen in previous posts). All behave exactly the same. The versions are 1.0.5, 1.0.9, and a CVS version of 1.2.0-beta1 pulled down at the end of August. In all cases, the systems are Dell PowerEdge 1750s (using RAID, no IDE drives involved) on Debian / kernel 2.4.27. I see no indication of problematic interrupts. In one test, there were 3 other PRIs running on the TE410P (in production) and there are no problems with any other PRIs. Ditto the configuration (I've checked and am doing the exact same thing with all my PRIs, just on different channels). Before I start providing configuration excerpts - has anyone had this problem connecting to an older Nortel Meridian switch and if so, what did you do to fix it? I suspect that there is a subtle configuration option on the SL100 that is wrong, but since I don't have access to it I can't confirm that. Can the wrong switch type cause FCS errors? Is there anything specific I can look at? For those who speak SL100, do you know of any specific parameter I can point the SL100 guy to? One more data point: I threw the PRI from the SL100 onto a spare port on a Cisco AS5350 and the AS5350 isn't complaining (no frame slips, no problem with the D channel). I'm pulling my hair out with this. Any help or pointers to info would be helpful. I will post a summary to the list if I get any useful private e-mail about this. Thanks! --- Gil Kloepfer [EMAIL PROTECTED] ___ --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] SIP Video Conferencing System to PRI
I have a Polycom VX7000 video conferencing system (VTC) on loan that
has SIP capability. What I'm trying to do is:
ISDN PRI -(Digium Quad T1)- Asterisk -(net)- VX7000
If someone calls on the number for the video conferencing system,
the Asterisk server forwards the call over SIP to the VX7000.
I also have another Polycom VTC that uses 3 ISDN BRIs and am trying to
call the VX7000 from that VTC.
It doesn't work. What happens is that the VX7000 appears to answer
the SIP call, but the BRI-based VTC doesn't ever see the call as
completed successfully.
When I do a pri debug on the span from the Asterisk box, there is a
User Information Layer 1 of type Unknown (4)...and also, there is an
ISDN message sent that the call is no longer end-to-end ISDN. I think
both of these may have something to do with the problem.
I suspect that the problem here is somehow translating the ISDN PRI
information type 4 to the proper SIP RTP stream type, but I don't know
for sure.
Has anyone done this successfully before and can give me some pointers
as to how this would get configured? For reference, here is the
pertinent config file stuff:
(in sip.conf)
[4910]
type=friend
secret={hidden}
host={also-hidden}
nat=no
canreinvite=no
allow=all
videosupport=yes
(in zapata.conf)
signalling=pri_cpe
switchtype=ni1
usecallerid=yes
hidecallerid=no
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=no
busydetect=no
immediate=no
context=sbc-0074-incoming
group=1
channel = 1-23
(in extensions.conf)
[sbc-0074-incoming]
exten = {number hidden},1,Dial(SIP/4910)
Thanks in advance for your help. If you'd like to take the discussion
off the list, you can contact me at [EMAIL PROTECTED] (temporary e-mail
address).
---
Gil Kloepfer
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