[asterisk-users] Problem with Read() ?
All,
I have a weird situation here and haven't been able to turn up any useful
information in searches, so I thought I'd post to the list.
Essentially, I have a customer who wants us to forward some of their calls
to various cell phones. Normally, I'd use FollowMe() for this (that's how
most of our customers are set up.) However, this particular customer has
some requirements that make it impractical to use FollowMe() as far as I
could tell, so I had to engineer a dialplan-based solution instead. Of
note, they DO want the call recipient (on a cell phone) to have to press a
key to confirm the call, so that calls don't end up in personal voice mail
boxes.
So, this is what I did:
I setup a custom context within their configuration to send these forwarded
calls to. Can't remember off the top of my head why I had to use a custom
context as opposed to sending it to the SIP provider directly, but I'm
pretty sure there was a reason. Here's the custom context:
[hosted-1004-outcall]
exten =>
_NXXNXX,1,Dial(SIP/sip-provider/+1${EXTEN},60,rU(hosted-1004-ccall))
same => n,Busy()
Now, whenever I want to forward a call to a cell phone (or more), I use:
Dial(Local/551212@hosted-1004-outcall
&Local/551213@hosted-1004-outcall)
This sends the call to both phones simultaneously, using the above Dial()
code. As you can see, I'm using the U option to call a subroutine on the
*callee* (cell phone user) when they answer. The code for that is:
[hosted-1004-ccall]
exten => s,1,Read(MOT,pbx/1004/ccall,1,in,3,5)
same => n,GotoIf($["${MOT}" == "1"]?10)
same => n,Set(GOSUB_RESULT=CONTINUE)
same => n,Return()
same => 10,Set(GOSUB_RESULT=)
same => n,Return()
exten => h,1,Set(GOSUB_RESULT=CONTINUE)
same => n,Return()
exten => i,1,Set(GOSUB_RESULT=CONTINUE)
same => n,Return()
The idea here being that "pbx/1004/ccall" is played to the cell phone user
(basically letting them know to press 1 to take the call.) If they press 1,
we return with no result and the call gets connected. Otherwise we return
with CONTINUE and the call is rejected (by the local channel) as busy, and
the next step in the dial plan continues (go to office voice mail -- which
isn't listed here.)
Anyway, this has all worked fine and dandy for quite some time. However, in
the past few weeks I've been getting complaints that whenever they press 1,
nothing happens, and the menu is simply repeated. Eventually it exceeds the
timeout/maximum tries and disconnects.
I've tried testing this with my cell phone, with my cable company phone, IP
phones, etc. Same result.
I've even upgraded from 11.7.0 to 11.11.0 just today (in case there was a
bugfix for it) -- same problem.
What's weird is that I can set up a DID to call Read() and it works fine.
It just seems to not hear (for lack of a better term) the digits when being
called as the U parameter to dial.
FWIW, I have also tested NOT using the Local/NXXNXX dialing and
actually using a Dial to our upstream carrier, still with the U option, and
same results. So, I honestly don't think it is an issue with using Local
channels.
Any thoughts? Am I just blatantly missing something? Or is something broke?
Thanks in advance. :)
--
Jeremy Gault, KD4NED
Senior Engineer, Voiceopia
office: (423) 509-8000 x 102
toll-free: (866) 928-4078 x 102
fax: (423) 244-0565
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Re: [Asterisk-Users] DID on analog line
George Pajari wrote: Depends on your ILEC/CLEC. Here is Vancouver they are the same price as non-DID trunks with DID numbers $2/ea in quantities < 1000 (from at least one CLEC). I have heard of CLECs in the US where DIDs are a tenth of this cost. Yep. We're using US LEC here (with a PRI) and they charge us ~$4/month for a block of 20 DIDs. I'm not sure if they would do any of the analog DID stuff, though. Actually, I don't think you can purchase straight analog lines from them (unless you co-lo at their switch.) Instead, if you need analog, they'll bring in a T1 and setup a channel bank for you. I've always wondered why some places charge so much for DIDs, though. Jeremy -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 Want a free GMail invite? E-Mail me and let me know! ___ --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] Busy not jumping n + 101 anymore
[EMAIL PROTECTED] wrote:
On 10/14/2005, "William M. Sandiford" <[EMAIL PROTECTED]>
wrote:
Ever since this upgrade, the system is jumping n+101 if it gets a busy
on a Dial command, it is now proceeding to the next priority (n+1)
Has something changed with this? Is there a way to change it back?
So glad to see you read the documentation...
Try scanning UPGRADE.txt
A lot has changed.
We've had the same problem here ever since we upgraded to CVS-HEAD.
When someone placed a call to a number that was busy, they would just
receive the "call cannot be completed" recording we have setup at n+1.
Not to sound nitpicky or hateful, but I just reviewed UPGRADE.txt again
here and I don't see anything about it. If it is in there, could you
please point it out to me? (Seriously, as I didn't see it.) If it
isn't, someone with CVS access should probably add it in.
Now, I will say that I'm assuming (from the new behavior and the "show
application dial" output) that one should now be using the ${DIALSTATUS}
variable to handle these conditions. (i.e. from your dial, make n+1 be a
Goto(s-${DIALSTATUS}) command, and create s-BUSY, s-CONGESTION, etc. in
the same context.) Once I get around to updating our dialplans, that's
what I plan on doing.
Someone please correct me if I am wrong. *dons asbestos armor, just in
case*
Jeremy
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Member: Bradley County ACS/RACES/SkyWarn
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Re: [Asterisk-Users] Don't know what to do if second ROSE component is of type 0x6
FWIW, we are also seeing this message each time we receive a call. I also went the Google route and found only questions, not answers. We are running a PRI from US LEC (channels 1-10 are B-channels, with channel 24 being the D-channel, and we are only running voice on the PRI.) The PRI is connected directly to our Digium TE110P card, and obviously we are using the zaptel drivers. We did not see this message when running */zaptel/libpri 1.0.9. However, after upgrading (we went to 1.2.0-beta1 and now to CVS) we started seeing the message. (I don't remember exactly if we saw it in the beta, but we do in the CVS.) In our case, it does not seem to affect the stability of our * machine. (However, bear in mind that you may be using parts of * that we do not, and the problem could lie in those parts.) We're handling all PSTN calls via the PRI, except outbound to toll-free which are handed off to an IAX gateway on the Internet. Our employees' desks are connected via the LAN (using Polycom 500/501 SIP phones.) I have a remote extension at home (also SIP) using a Sipura SPA-2000. We did have some stability issues (Asterisk would segfault) when we first moved to CVS. Of course, safe_asterisk handled this and a couple of days later we updated again from CVS and it seemed to fix the stability issue we were having. If you are using CVS (but not the latest one) you may want to try upgrading. I wouldn't worry about that message, though. However, I would also be interested in knowing what it means/what causes it. :) Jeremy Tom Rymes wrote: Has anyone figured out what this message means: Don't know what to do if second ROSE component is of type 0x6 We are running a PRI through a Sangoma card that is handling the D-channel natively at this point, but we go the error when zaptel was handling the D-channel, too. I have googled, but all of the messages are like this one with no answers that I can find. It's probably a non-issue, but we have been having issues with stability of our * install and I'd like to figure this out! Tom ___ --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 -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 Want a free GMail invite? E-Mail me and let me know! ___ --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] Silence suppression /RTP VAD and Asterisk? Dropped calls on IP-500
david.j as wrote:
problems. They tell me that I need to disable silence suppression on
the IP-500 because Asterisk won't support it and either Asterisk or
the MG thinks the call is dropped in abstence of RTP packets.
However I can't find any reference to this in the Polycom manual.
Does anyone know how to turn VAD/Silence suppression on the Polycom --
or if it even supports it? Is there a work around with Asterisk?
We have it disabled here. It is in the ipmid.cfg file. Open up your
ipmid.cfg file and do a search for VAD. We have a line like this:
To be more specific, look in your ipmid.cfg under the section.
That is where you should find it. Most likely, voice.vadEnable="1" in
your installation. Change this to 0, save the file, and reboot the
phones. If VAD (Voice Activity Detetion) is indeed the cause of your
problem, it should go away after updating the config and rebooting.
Jeremy
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Re: [Asterisk-Users] kill a .call file
If I understand correctly what you are wanting to do, I think you can simply delete the call file from the Asterisk outgoing spool directory and it will stop it from trying again. If I'm wrong, someone correct me on this. :) Jeremy jltaylor wrote: From my CLI: Attempting call on SIP/gw/19857749166 for [EMAIL PROTECTED]:1 (Retry 114) Attempting call on SIP/gw/19037747603 for [EMAIL PROTECTED]:1 (Retry 83) Attempting call on SIP/gw/19857747603 for [EMAIL PROTECTED]:1 (Retry 80) I want to stop it from any future attempts. Any idea about a command to kill or where the data is stored? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of trixter http://www.0xdecafbad.com Sent: Monday, September 19, 2005 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] kill a .call file On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote: Any means of killing a .call file that is in progress? You mean once the call has begun? You prolly want to hangup the call ... asterisk -rx "soft hangup " Or is there something else that you wanted? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --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 -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [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] Round-robin with Queue
List,
Okay, here's one that has me stumped, and it might just be something simple.
Currently, we are setup so that when someone calls in and tries to reach
the operator / front desk, it rings several different phones in
sequence. (i.e. it rings the front desk for 15 seconds, then a guy down
the hall from it for 15 seconds, then my desk for 15 seconds, and as a
last resort, my cordless for 15 seconds.)
We're doing this with Dial statements (most of which ring a single
device, except the last two which ring two devices each.) If all that
fails, they go to voicemail.
However, I'd like to use the Queue app to do this instead. This would
keep one person from having to deal with two calls at a time. (Plus it
provides a nicer interface for the caller.)
So, I setup a queue like this: (I'm including just the relevant parts)
strategy=roundrobin
member=SIP/100 ; fromt desk
member=SIP/112 ; the other guy
member=SIP/102 ; me
member=Zap/28 ; cordless
Here's what happens. Say caller #1 calls in, hits 0, gets put into the
queue, and is answered by SIP/112 (the 2nd member.) A few minutes
later, caller #2 calls in, hits 0, and rings directly to SIP/102 (me)
without ringing SIP/100 first. (To me, that sounds like what rrmemory
should do, not roundrobin.)
Is something broken with the queue app, or am I setting this up wrong?
I did try adding penalties for the members (i.e. no penalty for SIP/100,
a penalty of 1 for SIP/112, 2 for SIP/102, etc.) That just resulted in
only SIP/100 being rung.
So, what am I missing/doing wrong here? :)
Jeremy
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Re: [Asterisk-Users] Caller Name: Asterisk reading too fast
As was already suggested, Wait() is your friend. We had the same problem when our PRI was installed. It was supposed to include Caller ID Name delivery, but it seemed to be hit-or-miss as to if it would work. This is what I found: When people call our auto-attendant and dial an extension, the name would work fine. However, if they dialed one of our DIDs (each extension has its own DID) then the name did not work. Inserting a Wait() command on the DIDs for extensions seemed to do the trick, like this: exten => 423303,1,Wait(0.75) exten => 423303,2,Dial(SIP/xxx) Here's my take on what happened: A call comes in to the auto-attendant, which immediately answers and begins the greeting. Shortly after, the telco sends the name down the PRI, and Asterisk associates it to that call. Then the caller punches in the extension and it goes through with name. (Keep in mind it only takes a fraction of a second for the name to show up on the PRI, so by the time a normal human realizes the auto-attendant has answered and dials an extension, plenty of time has passed for the name to come in.) However, the DIDs were spwaning Dial() *immediately* when the call came in. Of course, at that point the name was not present yet. (The telco was still doing the lookup.) But, Dial() has to send CID immediately when invoked, so it sent the number twice. Adding the 0.75 second delay gave time for the name to arrive before spawning Dial(). I've found 0.75 to be a fairly reliable delay. It misses a name here or there, but not often enough to be a problem. Most callers won't really notice the delay. If I increased this to one full second, it would be more noticeable but would probably fix the few ones we miss. You may have to play with this value to find the optimum setting for your setup. Another test you should be able to do: Have a call come in (when it shows only the number twice) and answer it. Then login to the Asterisk console, do "show channels" and find out which Zap channel it's on. Do a "show channel Zap/whatever" (whichever channel it is) and see if the name has shown up. IIRC, before I added the Wait, the "show channel" would still see the name, even though the phone didn't. Good luck! Jeremy J Thomas wrote: I asked my telco to release caller name on the PRI. Earlier they were releasing only the phone number. I still did not see the name, but only the number in caller id. Actually I now see number twice. When I inquired with them this is the response I got: "I ran a trace on your TG. I see that your switch is picking up the call so fast that it is not able to pick up the name. The name is being sent, but I suspect after it is too late. This is something that will need to be corrected in your switch. I have attached a sample call out of the trace I performed this morning." They have sent me the trace file. Is there a way as it is in Asterisk so that it reads the caller name properly? Thanks, -- jt ___ --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 -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [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
Re: [Asterisk-Users] All Page ??
I have (sort of) made this work in our environment. There was an AGI script (Google for polycom allcall.agi) written in Perl that would implement a hack to do this. Basically, you can set the Alert-Info SIP header and cause the Polycom phones to auto-answer. The general idea behind allcall.agi (and the associated dialplan that's documented with it) is that you dial a paging extension, allcall.agi creates callfiles for each extension (with Alert-Info set for auto-answer), puts them in /var/spool/asterusk/outgoing and causes Asterisk to call each extension. When those extensions answer, it puts them into a MeetMe conference (in listen-only mode.) Then the caller goes into the same conference in talk-only mode. There's a hard time limit after which the call is terminated. I had some issues with the Perl script, so I wrote my own version in PHP. I also made some modifications to the conference flags (making the person paging a marked user, and having all the pagees be disconnected when the marked user hangs up, which takes away the time limit on pages and even the need for it.) The PHP script itself worked well. However, I ran into another issue: I had an extra call file that would basically dial Local/[EMAIL PROTECTED] (I setup an internal-special context, with a pagebeep extension that would wait a second, play a beep, and hang up.) The idea was to tell people to wait until after the beep (which would give the phones all time to sync up) before speaking. That worked great when paging from an analog extension, but when doing it from a SIP extension, Asterisk complained about codecs and formats not matching. I tried it with MeetMe2 and it worked, but MeetMe2 does not support marked users and such (which I want to have.) I could dispose of the beep and just tell people to wait a few seconds, and that would work, and would probably work for you. Jeremy -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ 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] Overriding Caller ID
I may be wrong here, so if anyone else here knows contrary, please feel
free to jump in and correct me. ::dons his asbestos armor::
When we first deployed * we were coming from an analog channel bank
setup (hooked into our old PBX as analog lines.) I was able to connect
* to the T1 and use E&M Wink signaling to make things work. However, we
couldn't control our caller ID. The number that appeared would depend
on which channel the call took.
Not long thereafter, we migrated to a PRI. Once we were on the PRI, we
were able to have control of the CID.
As far as I know, you can't control your outbound CID on a T1 setup the
way yours is. You probably need to switch to a PRI instead if you want
this ability. But again, that's based on my knowledge and experience,
so I could be wrong. If so, hopefully someone else here will clear it
up for both of us.
Jeremy
Waldo Rubinstein wrote:
Hello list,
We have some kind of a problem with our Asterisk installation. We
want to be able to publish different caller id when placing outbound
calls through the PSTN. We have Asterisk with TE410P and T1 from FDN
Communications. The problem is that all our outbound calls show our
main number, regardless of what we set with SetCallerID, even using
CallingPres with all possible combinations. When speaking with FDN,
they say they have set their T1 to show our main number for outbound
calls, but that we should be able to override that with no problem.
As I said, I have tried all possible combinations, yet, nothing seems
to work. Below are snippets of some of our configs:
extensions.conf
;
; Local calls
;
exten => _NXXNXX,1,CallingPres(32)
exten => _NXXNXX,2,SetCallerID(2125551234)
exten => _NXXNXX,3,Dial(${TRUNK_LO}/${EXTEN})
zapata.conf
[channels]
usecallerid=yes
cidsignalling=bell
cidstart=ring
hidecallerid=no
restrictcid=no
usecallingpres=yes
callerid=asreceived
switchtype = dms100
signalling = em_w
group = 1
context=inbound
callerid=asreceived
channel => 1-24
Does anyone have any suggestions?
Thanks,
Waldo
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Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain
It sounds to me like an issue of transmitting DTMF tones from the SIP phones. There are several methods that can be used to accomplish DTMF from SIP phones. Of course, you may ask why it isn't just sent as audio (like a regular POTS phone would.) What happens if you are using a SIP phone, hold down the number 4 button for two seconds (so it sends 2 seconds worth of DTMF on the audio stream) and there is some packet loss during that time? You'll have an audio dropout (thus, tone followed by brief silence and tone again.) The remote end will see this as two tones, not one, which obviously can cause undesired results (and is why it's not a good idea to send DTMF in the audio stream.) That being said, look in your sip.conf for a dtmfmode parameter. You can use inband (in the audio stream, not recommended), RFC2833, or SIP INFO. Your SIP phone should also allow you to set how DTMF is sent (although it may not support all of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated. 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 -- Jeremy Gault, KD4NED<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 msn msgr: [EMAIL PROTECTED] ___ 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] Polycom SoundPoint 501 power adapter
Same here on my PolyCom 501. Also, polarity is positive on the center pin, negative on the outside. Jeremy Melanson wrote: All of my 500's and 501's are 12V 400ma as well. On Thu, 2005-08-18 at 14:00 -0500, Alan Bunch wrote: Mine is 12V 400 ma alabun Paul Belanger wrote: Can somebody who has a SoundPoint 501 please confirm the power adapter input / output settings: Input: 120V AC 60HZ 20W Output: 24V DC 500mA PB ___ 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 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 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 -- Jeremy Gault<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 url: http://www.winworld.cc/ ___ 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] TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 url: http://www.winworld.cc/ ___ 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] Weird issues with TDM400P
We have a TDM400P installed here with four FXS modules. It works well except for a couple of issues: First, I have a Panasonic KX-TG2431 telephone (so others can reach me when I am in o ther parts of the building) hooked up to one of the FXS ports. When the other end hangs up, I get the usual CPC disconnect signal. After the CPC, sometimes it will go to a dialtone, and other times a reorder. There doesn't seem to be any rhyme or reason behind what it does, it's random. That in and of itself is not a problem, but it did seem odd. However, sometimes during the reorder (after about 3 seconds) the reorder stops and I get an FSK data burst (assumedly the VMWI.) My understanding is that the data burst isn't supposed to happen until *after* my phone goes on-hook. Any idea what's up here? Second, one of our ports has static on it. A tenant moved in and we configured their fax for our fourth (until then unused) FXS port. They had numerous fax problems, and when I connected a buttset up to their wall jack, there were minor static issues. I swapped their fax connection to a different port on the TDM400P, and the static went away. I put a different line (to elsewhere in the building) into their usual fax port on the TDM400P, and the static showed up there. So, the problem follows the FXS port, and not the line itself. For now, I have given them the FXS port I was using for my Panasonic, and put my Panasonic on the noisy port. It's not all that noticeable, but of course the fax doesn't like any noise at all. Any idea what this could be? I haven't done much troubleshooting yet, but I plan on taking down the * box and re-seating the FXS module. If that doesn't work, I'll swap it and another FXS module on the board around to see if the problem follows the module or the socket. Other than that, any recommendations? Third (minor) issue, which affects all the ports, but I am using my phone as an example: I can call from my desk phone (Polycom IP 501) to my analog phone and answer. Then I can hit flash on the analog phone, and I'll hear the usual double-click on the Polycom as the analog phone flashes. However, between those two clicks, there's some sort of weird noise. It's not a radio station or anything. It's more like a whine of sorts. The same noise can be heard by hooking up a buttset to one of the ports and putting it in monitor mode (where it monitors the line, but doesn't go offhook.) Is it normal for Digium cards to generate this noise, or does this indicate something is wrong with my card? (Our Comdial PBX at church does a similar type of thing when flashing on an analog extension, but that still doesn't tell me if it's normal or not.) Anyone else see this same type of thing? Fourth: I get this message in the log (related to our analog FXS ports) Aug 11 16:46:17 WARNING[23855]: zt hook failed: Device or resource busy I think this may have something to do with getting a dialtone instead of reorder after hangup (the first thing I mentioned.) Not 100% sure though. Anyone have any ideas on any of these? If you can share I'd appreciate it. TIA. Jeremy -- Jeremy Gault<[EMAIL PROTECTED]> Network Administrator, WinWorld Corporation voice: +1.423.473.8084 fax: +1.423.472.9465 fwd: 461771 url: http://www.winworld.cc/ ___ 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] Re: call does not hangup after client quits
Steve, If I am understanding your situation correctly (i.e. you are using a SIP client and then forcibly disconnecting/shutting it off during a call) you may want to look at your sip.conf for a setting called rtptimeout. This may do exactly what you want. When on a SIP call, and you disconnect/shut off your client (without properly hanging up first) then (obviously) * does not receive a SIP message saying the call has ended. However, the RTP (audio) stream will stop. The rtptimeout setting lets you define a time period that after seconds of no audio packets, it's assumed the SIP client has gone away and the call should be terminated. Jeremy Stephen J. Wilcox wrote: Hello, can anyone help with my problem below, searching doesnt show any results.. thanks Steve On Wed, 3 Aug 2005, Stephen J. Wilcox wrote: Hi, I'm seeing a problem where if I place a call, then forcibly quit or turn off the client the call stays active. The frames counters stop so its apparent the client has gone away but the call remains active. Asterisk is CVS-HEAD 23-Jun-05 What is supposed to happen in this scenario? thanks Steve ___ 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 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
