[Asterisk-Users] Outbound paging dialplan example?

[Patrick Friedel]

Due to changes at the office, I'm finally getting around to setting up 
an AA to deal with incoming calls.  One of the big changes is that we're 
dropping the old alphanumeric pager and will just send pages to our 
phones.  I've got the outbound greeting message working in a test 
context no problem right now, but I'm kind of stuck on how to capture a 
DTMF sequence from a user and doing anything with it.


Right now the pertinent DP features look like this:

exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,2
exten => s,5,Background(greeting)

exten => 1,1,Voicemail(u100) ; Press 1 to leave a message.

exten => 2,1,Voicemail(u6003) ; Press 2 to send an emergency page

exten => t,1,Dial(SIP/person,30,t) ; Ring my extension on timeout

Obviously extension 2 needs to be changed, right now it just leaves a 
message in my mailbox.  I'm figuring I'll add a new message that says 
"Please enter your callback number, followed by the pound sign." and put 
that in as a Background() message.  The tricky bit that I can't figure 
out (without sample dialplans in voip-info) is how to capture the DTMF 
the caller provides and send it out via a System() call to an external 
application to page the oncall person.  As the oncall person will 
conceivably change on a regular basis, we can't just hand it out to 
customers, unfortunately/thankfully.  Thanks for any assistance!

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Re: [Asterisk-Users] Asterisk 1.0.8 and TDM stop acking inbound calls?

[Patrick Friedel]

Cirelle Enterprises wrote:



we also experienced this with asterisk 1.0.9 and rev H of the tdm with 
4 fxo modules


we were restarting asterisk every night via cron and this still happened

in our case, 3 out of 4 fxo modules (2,3,4) crapped out and stopped 
ack'ing incoming

calls (outgoing calls were fine)

it took a reboot of the server to get the card operational again and 
answering calls


 As a certain Zippy would say: Yow!  I assume you reached that point 
because unloading and reloading the wctdm modules didn't do anything?


 Do the digital interfaces have these sorts of problems?  Is there an 
alternate FXO solution?  I've heard nothing but trouble with the TDM, 
and I know that's probably because the 99% of satisfied users are 
generally quiet but still...


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Re: [Asterisk-Users] Asterisk 1.0.8 and TDM stop acking inbound calls?

[Patrick Friedel]

Rich Adamson wrote:

My office has been running Asterisk 1.0.8 and a TDM04B for a few months 
now without too much trouble.  After a while we discovered that after a 
certain period (about a month), asterisk stopped acknowledging inbound 
calls.  Since I was out of the office the first time it happened, 
another admin rebooted the whole box which solved the problem.  The 
second time it happened I discovered that just restarting gracefully 
solved the problem, so I put that into my cron and forgot about it.  (I 
know, it's not "right", but debugging something that happens 
unpredictably once a month could go on for way too long to be acceptable..)
   


Check the revision of the TDM card. If rev E/F, call digium support to
get it replaced. Known problem with early versions of the card.
 

 The rev is labelled on the itty bitty xilinx chip and not under the 
modules, right?  Dang, rev F.  Okiedoke, off to digium I go.  Thanks!

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Re: [Asterisk-Users] Asterisk 1.0.8 and TDM stop acking inbound calls?

[Patrick Friedel]

Patrick Friedel wrote:

I couldn't find a changelog for 1.0.9 to see if it's worth the 
off-hours maintenance window, and we're too dependant on the phones to 
try 1.2.  Should I try the next step up in the probably unnecessary 
preventative maintenance and unload/reload the wctdm module during the 
asterisk restart?  Is there any way to have asterisk notify that it's 
running low on/out of resources?  We don't typically ever tie up all 
of our zap channels except for really particularly exciting days, so 
if they are all in use it would be cause for concern..



 In the interim, and completely on a whim, I've put a couple of 
splitters and added another FXO device onto the line, a good old 
fashioned analog phone that chirps once until asterisk picks up the 
line.  With any luck, the next time asterisk takes a dive, that phone 
will continue to ring and we'll catch on faster than the first customer 
that decides to email us wondering why nobody is picking up. :)


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[Asterisk-Users] Asterisk 1.0.8 and TDM stop acking inbound calls?

[Patrick Friedel]

My office has been running Asterisk 1.0.8 and a TDM04B for a few months 
now without too much trouble.  After a while we discovered that after a 
certain period (about a month), asterisk stopped acknowledging inbound 
calls.  Since I was out of the office the first time it happened, 
another admin rebooted the whole box which solved the problem.  The 
second time it happened I discovered that just restarting gracefully 
solved the problem, so I put that into my cron and forgot about it.  (I 
know, it's not "right", but debugging something that happens 
unpredictably once a month could go on for way too long to be acceptable..)


 Now, less than a week since I did that, asterisk stopped ringing our 
extensions on inbound calls again.  "sip show peers" showed that 
asterisk knew about all of the extensions.  I forgot to check "zap show 
channels" when * was ignoring inbound calls, is it possible that * 
thinks all the lines are still off hook?  Is there anything else I 
should do to figure out what's causing trouble?  Unfortunately it's 
usually something of a panic situation, so I'm not allowed the chance to 
troubleshoot as thoroughly as I'd like.


 Speaking of, I've fiddled and tweaked left and right to get hangup 
detection working better to no real avail.  Asterisk eventually decides 
the far side hung up about 10 seconds after the fact.  Am I 
understanding right that call progress is still something of a black art 
for analog FXO devices?  Not getting 10 second dead air voicemails when 
people hang up would be sweet. :)


 I couldn't find a changelog for 1.0.9 to see if it's worth the 
off-hours maintenance window, and we're too dependant on the phones to 
try 1.2.  Should I try the next step up in the probably unnecessary 
preventative maintenance and unload/reload the wctdm module during the 
asterisk restart?  Is there any way to have asterisk notify that it's 
running low on/out of resources?  We don't typically ever tie up all of 
our zap channels except for really particularly exciting days, so if 
they are all in use it would be cause for concern..

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Re: [Asterisk-Users] Snom 360 record button?

[Patrick Friedel]

Nils Ohlmeier wrote:


On Thursday 28 July 2005 17:12, Patrick Friedel wrote:
 


 Yeah, that was in the middle of a call - the only other SIP debug
information is the normal call build up and tear down.  I can generate
it if you want, but it's nothing exciting, just the usual handshaking.
But that was kind of what I was thinking would be a solution - Asterisk
sees the Record INFO packet, and conferences the call to a local
extension with Monitor() going.  I'm not 100% sure whether or not the
Snom 360 expects anything _else_ (other than a simple acknowledgement)
back from the PBX, as there doesn't appear to be a whitepaper for it it
on Snom's site.  I would assume it would be fairly straightforward in
   



Sorry for the missing whitepaper ;-)
The snom phone is not expecting anything else then any reply for the INFO 
request.
 

 Heh, figured that would draw you out of the woodwork.  It's good to 
know that information.

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Re: [Asterisk-Users] Snom 360 record button?

[Patrick Friedel]

Maik Schmitt wrote:


Is this during a call? Can you please send me a full SIP DEBUG of the call?

Brainstorming, maybe we could treat this as a transfer to a local
extension somehow and turn monitor on in the dial plan that way...
   



Hmm IMO the automon-feature would be better for this. It does exactly what we 
want (start recording during a call) and is configurable via Dial-Options. 
The only thing I don't know is how to activate it without sending the DTMF 
sequence.
 

 Hmm, you're right, I wasn't aware of the automon feature - I don't 
know if it was in the original SIP trace, but the 360 is sending _some_ 
DTMF signalling, but I don't know what it's actually sending.  I'm 
currently at 9 for verbosity, let me amp that up and see if it will ever 
actually display the tones it's receiving.



voip*CLI> sip debug peer pjf
SIP Debugging Enabled for IP: 10.0.1.213:2051
voip*CLI> set verbose 255
Verbosity was 9 and is now 255
voip*CLI>
[boring build up that isn't new to anyone]

Sip read:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-aytv2p1rs5ih;rport
From: "Patrick" ;tag=4rbnk4yyxd
To: ;tag=as3b19835c
Call-ID: [EMAIL PROTECTED]
CSeq: 3 INFO
Max-Forwards: 70
Contact: 
User-Agent: snom360/3.60r
Record: on
Content-Length: 0


11 headers, 0 lines
Receiving DTMF!
Jul 28 14:12:15 WARNING[26025]: chan_sip.c:6166 receive_info: Unable to 
parse INFO message from [EMAIL PROTECTED] Content

Transmitting (no NAT):
SIP/2.0 415 Unsupported media type
Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-aytv2p1rs5ih
From: "Patrick" ;tag=4rbnk4yyxd
To: ;tag=as3b19835c
Call-ID: [EMAIL PROTECTED]
CSeq: 3 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0

[teardown]

Hmm.  It's getting _something_, but the verbosity won't reveal it.  
Checking chan_sip.c, I don't see a mechanism for revealing the data, 
it'd be around line 7716.  It we could reveal it (or Nils just tells us. 
:), it might be as simple as editing features.conf and setting automon 
to whatever DTMF the Snom sends when you hit the record button.

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Re: [Asterisk-Users] Snom 360 record button?

[Patrick Friedel]

Olle E. Johansson wrote:


Patrick Friedel wrote:
 


Sorry if this is an obvious question, but I haven't seen an obvious
answer on the wiki that I remember.  Has anyone managed to make the
record button on the snom 360 fire off the Monitor() application?  I
don't see a bounty, and googling for "snom 360 record button asterisk"
returns tons of product specification pages. (Joy!) I don't see a bounty
for it, and the only mention I _see_ on the wiki is "one touch RECORD
button usuable only with special PBX support via SIP INFO method" which
isn't much of an answer.

   


Is this during a call? Can you please send me a full SIP DEBUG of the call?

Brainstorming, maybe we could treat this as a transfer to a local
extension somehow and turn monitor on in the dial plan that way...
 

 Yeah, that was in the middle of a call - the only other SIP debug 
information is the normal call build up and tear down.  I can generate 
it if you want, but it's nothing exciting, just the usual handshaking.  
But that was kind of what I was thinking would be a solution - Asterisk 
sees the Record INFO packet, and conferences the call to a local 
extension with Monitor() going.  I'm not 100% sure whether or not the 
Snom 360 expects anything _else_ (other than a simple acknowledgement) 
back from the PBX, as there doesn't appear to be a whitepaper for it it 
on Snom's site.  I would assume it would be fairly straightforward in 
chan_sip.c to bang in a new method, but whether there are any 
ramifications, well...

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[Asterisk-Users] Snom 360 record button?

[Patrick Friedel]

Sorry if this is an obvious question, but I haven't seen an obvious 
answer on the wiki that I remember.  Has anyone managed to make the 
record button on the snom 360 fire off the Monitor() application?  I 
don't see a bounty, and googling for "snom 360 record button asterisk" 
returns tons of product specification pages. (Joy!) I don't see a bounty 
for it, and the only mention I _see_ on the wiki is "one touch RECORD 
button usuable only with special PBX support via SIP INFO method" which 
isn't much of an answer.


I assume the answer is no because of this, but I'm hoping against hope 
this is just because I don't have anything set up for it:


SIP Debugging Enabled
voip*CLI>
*[I hit the record button at this point]*
Sip read:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-q6cqmwneki97;rport
From: "Patrick" ;tag=1pvw4rlq7s
To: ;tag=as51ba7d7b
Call-ID: [EMAIL PROTECTED]
CSeq: 4 INFO
Max-Forwards: 70
Contact: 
User-Agent: snom360/3.60r
Record: on
Content-Length: 0


11 headers, 0 lines
Receiving DTMF!
Jul 27 17:24:52 WARNING[26025]: chan_sip.c:6166 receive_info: Unable to 
parse INFO message from [EMAIL PROTECTED] Content

Transmitting (no NAT):
SIP/2.0 415 Unsupported media type
Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-q6cqmwneki97
From: "Patrick" ;tag=1pvw4rlq7s
To: ;tag=as51ba7d7b
Call-ID: [EMAIL PROTECTED]
CSeq: 4 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


to 10.0.1.213:2051

 I assume the first one is the snom requesting the PBX to start 
recording, and the second is asterisk reminding the snom that it doesn't 
allow the INFO method and to get bent?



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Re: [Asterisk-Users] ATXFER discussion, what's your opinion ?

[Patrick Friedel]

[EMAIL PROTECTED] wrote:


That's exactly my opinion: isn't ironic that the only function "joe
sixpack" will use in a pbx is the worst implemented ?
 



 


Maybe because most asterisk PBX's are implemented using business class
softphones rather than analogue phones? Most business class SIP phones
(and even grandstream phones) allow attended/non-attended transfers with
asterisk-stable and/or asterisk-head...
   



I think that's mostly right, but it should also be a "native" xfer function working the 
same way regarding of the user agent, some sort of common ground we can "trust"  for 
installation with mixed devices.

By the way: anyone got experience in attended trasfer with snom ? :)

Alessio Focardi
 



 Oh, you mean the completely natural feeling "put them on hold, dial 
new party, tell them you have a transfer, hit transfer"?  I want some of 
whatever kool-aid the person who thought that one up had.  I still feel 
like I'm losing a call every time I do an attended transfer.


 Is there a _technical_ (e.g. SIP) limitation on this?
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Re: [Asterisk-Users] Any suggestions for an IP phone?

[Patrick Friedel]

Pavel Jezek wrote:

according to this debate, I would like to try snom 360 still more 
(features, opensource support, linux based)  ;-)
any good or bad experience with support from snom? or reliability of 
snom phones?

PJ

 I've been fiddling with a set of Snom 360's for a while now and really 
the worst I can say about them are:


1)  The buttons feel.  Odd.  They seem to be wearing in, though.  When I 
first got them they were a bit stiff/unresponsive, but my main testing 
phone is nicely broken in by now.


2) Occasionally my phone's display has gotten garbled.  But I'm fast and 
crazy and running with the beta firmware.  Some of the stuff in 3.60k  
beta made subscribe/notify seem to work better, but both 3.60k and q 
both garbled the screen.  3.60l seems stable, though.  The worst is the 
occasional inexplicable screen clearing events.  It _seems_ like the 
phone is still fine, but it has forgotten about the screen entirely.  
Again, I think that's a beta firmware issue.


3) Related, Snom releases new firmwares for free on a fairly regular 
basis.  Which is good and bad.  Read it as you will.


4) Snom seems to pay attention to this mailing list, they've answered at 
least one of my questions already.


5) The screen seems..  Underutilized.  I mean, right now I have 4 button 
labels, a big analog clock and date, my line appearance and a slightly 
goofy snom.com logo.  Incoming calls do a little song and dance, but it 
seems like you could do more with the display and rely less on the hard 
lights.  OTOH there would be an application break from the 190 firmware 
for doing this.


6) There are still the odd little corners where there's some polish 
missing - e.g. usually the display indicates which button mode you're in 
(abc, ABC or 123), but you find yourself in places where it doesn't.  
Usually after you've changed modes to deal with the occasional password 
issue.


7) Odd personal complaint, but snom hasn't learned the trick of tucking 
a pound of iron away in the base of the phone to make it seem more 
sturdy that I like out of telecomm products.


8) Memory?  I've started seeing low memory warnings with 2 line 
appearances and under 30 phone book entries.  (fortunately project 
Ghetto Queue failed to work and I went back to a single line..)


   For the most part I'm really happy with them, though.  There's a 
learning curve, but what doesn't have one?  I say this as someone who 
hasn't touched any other hardphones, though, so take it with a grain of 
salt.

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[Asterisk-Users] SNOM 360 and parking

[Patrick Friedel]

OK, last showstopper that I just can't puzzle my way through - parking 
calls with the snom phones.  I get the two phones connected, I hit 
transfer on one, the other phone goes to MOH and the first phone gives 
me DT, so I dial 700 and hit the OK button.  Call transferred, the SNOM 
hangs up before I have a chance to hear which extension it parked to.  
Is there a way to make the SNOM phones stay off hook until you 
explicitly hang up during a transfer?  (my only complaint about these 
phones - occasionally they're just too darn smart for their own good.)


I can live without actual snom-style orbits at this time (handy though 
they might be), since the current system involves parking the call on an 
external line and walking over to another office to say that they have a 
call.  I imagine that down the road it'll usually just be an attended 
transfer, but we do park calls around phones a fair bit as we brainstorm 
issues.


(Actually, I can't get attended transfers working, either.  All 
transfers are blind.  Related?)


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Re: [Asterisk-Users] Odd MOH problem...

[Patrick Friedel]

There don't appear to be any running after asterisk dies, while asterisk 
is alive I have this:


voip:/etc/asterisk# ps aux | grep mpg
root 15785  0.0  0.2  2316 1032 pts/1S+   16:55   0:00 /bin/sh 
/usr/local/bin/mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 -z [a whole 
load of the right filenames here]


 Upon investigation, though, I find /usr/local/bin/mpg123 to be, in 
fact, this:


voip:/etc/asterisk# cat /usr/local/bin/mpg123
#!/bin/sh

cat /home/ats/test.wav

 Hmm.  *sigh*  Thanks for making me use my brain muscle.


Bryce Chidester wrote:


When you restarted Asterisk, did you kill the mpg123 processes?


-Bryce
[EMAIL PROTECTED]


NOTICE: The views expressed in this e-mail do not neccesarily reflect  
those of my employer. This is a personal e-mail and as such, the  
opinions expressed are my own.


On Jul 12, 2005, at 11:52, Patrick Friedel wrote:

So I decided, for the formal asterisk rollout, to change over to  
less commercially-infringing MOH than the prior admin had thrown on  
the server.  (plus: it was blown out and nasty sounding over the  
phones.  Ew.)  I changed the files in /var/lib/asterisk/mohmp3 to  
something else (can't dig up the link, but it was from the voip- info 
wiki).  My musiconhold.conf looks like this:


;
; Music on hold class definitions
;
[classes]
default => quietmp3:/var/lib/asterisk/mohmp3,-z

 When I put a phone on hold, I get this in the console:

   -- Started music on hold, class 'default', on SIP/pjf-51af

 And yet, the MOH is still the same old song from before when I put  
a caller on hold.  Asterisk has restarted, the phones (snom 360s)  
don't have their personalized SIP MOH settings set, the offending  
file has been deleted from the filesystem, I can't find anything  
else that sets the MOH to a different class, umm.  Any ideas?


 Out of curiousity, I tried setting up this:

exten => 6101,1,Answer
exten => 6101,2,MusicOnHold(default)

 Same results.
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Re: [Asterisk-Users] Pushing new firmware to Snom 190

[Patrick Friedel]

Bob Goddard wrote:


There are 2 problems here, the first is if you click on "memory" and
the connection count is not 0, then you will be unable to reboot the
phone, all you can do then is power cycle it.

Secondly, to update the phone, you have to create 2 files, the first
is entered into the "Setting URL:" and should at a minimum consist
of something like:



firmware_status: http://81.187.187.52/snom190/snom190-firmware.htm



You then have to create a second file with reference to the above:



firmware: http://www.snom.com/download/share/snom190-3.60i-SIP-j.bin



It is a pathetically stupid way of doing things, but it is the Snom way.
 

And you may need to set firmware_interval (that might be a 360 only 
option), which is scaled in (minutes? seconds? Seconds with a minimum of 
60?  Whatever it is, it's _incredibly_ sticky.).  I guess for me the 
next question is whether there is a more user-friendly way of forcing 
the updates?  I've got the chain set up, it rebooted and found the new 
firmware nicely enough, but the only change it did was indicate on the 
phone that there was a new firmware available on the status line - it 
didn't actually load the new firmware until the EU asked for it.  No 
biggie in my SOHO (3 whole phones!), but in a huge rollout I could see 
that being really annoying.


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[Asterisk-Users] Odd MOH problem...

[Patrick Friedel]

So I decided, for the formal asterisk rollout, to change over to less 
commercially-infringing MOH than the prior admin had thrown on the 
server.  (plus: it was blown out and nasty sounding over the phones.  
Ew.)  I changed the files in /var/lib/asterisk/mohmp3 to something else 
(can't dig up the link, but it was from the voip-info wiki).  My 
musiconhold.conf looks like this:


;
; Music on hold class definitions
;
[classes]
default => quietmp3:/var/lib/asterisk/mohmp3,-z

 When I put a phone on hold, I get this in the console:

   -- Started music on hold, class 'default', on SIP/pjf-51af

 And yet, the MOH is still the same old song from before when I put a 
caller on hold.  Asterisk has restarted, the phones (snom 360s) don't 
have their personalized SIP MOH settings set, the offending file has 
been deleted from the filesystem, I can't find anything else that sets 
the MOH to a different class, umm.  Any ideas?


 Out of curiousity, I tried setting up this:

exten => 6101,1,Answer
exten => 6101,2,MusicOnHold(default)

 Same results.
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[Asterisk-Users] Snom 360 NOTIFY syntax

[Patrick Friedel]

I'm rolling out an installation with snom 360s in the near future.  
Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a 
snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002.  I 
have the 360's set up to subscribe and notify for the line use lights, 
which works like a charm for interoffice calling (between the 360's, 
anyway.  The IAXy, 200 and, softphone will be used by less phone 
dependant types) but what I can't figure out from the Wiki is if it's 
possible to have the ZAP lines notify for the outbound lines so we can 
see how many lines are in use.


 My configuration looks something like this:

sip.conf:
[mjg]
type=friend
username=mjg
context=sip
callerid="Masuo" <6001>
secret=
host=dynamic
defaultip=199.242.227.227
canreinvite=no
mailbox=6001
subscribecontext=sip

[pjf]
type=friend
username=pjf
context=sip
callerid="Patrick" <6003>
secret=
host=dynamic
defaultip=199.242.227.227
canreinvite=no
mailbox=6003
subscribecontext=sip

360 configuration:
fkey6!: dest 
fkey7!: dest 

extensions.conf:
[macro-oneline]
exten => s,1,Dial(${ARG1},20,t)
exten => s,2,Voicemail(u${MACRO_EXTEN})
exten => s,3,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup

exten => 6001,hint,SIP/mjg
exten => 6001,1,Macro(oneline,${MJG})

exten => 6003,hint,SIP/pjf
exten => 6003,1,Macro(oneline,${PJF})

 Is there any convenient way to monitor the status of my FXO lines from 
the phones?  Or do I have to set up the interested parties with gastman?


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