Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

2011-01-09 Thread Jim Dickenson
I am running version 1.4.x. Where do I get PRICAUSE? I tried making a call that 
was not answered and I did not see any more information. The dumpchan of 
DADHI/23-1 did not happen as that is in a macro that only gets called for an 
answered call.

I only see this:


Executing [91112223...@empl:8] Dial(SIP/mine-0521, 
Dahdi/G1/111222|60|gM(out-dial)) in new stack
DEBUG[4907]: dsp.c:1682 ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/111222
DEBUG[3188]: chan_dahdi.c:10135 pri_dchannel: Queuing frame from 
PRI_EVENT_PROCEEDING on channel 0/23 span 1
-- DAHDI/23-1 is proceeding passing it to SIP/mine-0521
-- DAHDI/23-1 is ringing
DEBUG[3188]: chan_dahdi.c:1790 dahdi_enable_ec: Echo cancellation already on
-- DAHDI/23-1 answered SIP/mine-0521
-- Executing [...@macro-out-dial:1] DumpChan(DAHDI/23-1, ) in new stack
Dumping Info For Channel: DAHDI/23-1:

Info:
Name=   DAHDI/23-1
Type=   DAHDI
UniqueID=   sys.domain.com-1294514614.2630
CallerID=   9111222
CallerIDName=   (N/A)
DNIDDigits= (N/A)
RDNIS=  (N/A)
State=  Up (6)
Rings=  0
NativeFormat=   0x4 (ulaw)
WriteFormat=0x4 (ulaw)
ReadFormat= 0x4 (ulaw)
1stFileDescriptor=  35
Framesin=   189 
Framesout=  176 
TimetoHangup=   0
ElapsedTime=0h0m4s
Context=macro-out-dial
Extension=  s
Priority=   1
CallGroup=  
PickupGroup=
Application=DumpChan
Data=   (Empty)
Blocking_in=(Not Blocking)

Variables:
MACRO_DEPTH=1
MACRO_PRIORITY=1
MACRO_CONTEXT=from-outside
MACRO_EXTEN=
DIALEDPEERNUMBER=G1/111222
TRANSFERCAPABILITY=SPEECH

DEBUG[4907]: app_macro.c:379 _macro_exec: Executed application: DumpChan
DEBUG[4907]: app_dial.c:1927 dial_exec_full: Macro exited with status 0
DEBUG[4907]: chan_dahdi.c:3464 dahdi_setoption: Set option AUDIO MODE, value: 
ON(1) on DAHDI/23-1
DEBUG[4907]: chan_dahdi.c:3092 dahdi_hangup: Not yet hungup...  Calling hangup 
once with icause, and clearing call
DEBUG[4907]: chan_dahdi.c:3460 dahdi_setoption: Set option AUDIO MODE, value: 
OFF(0) on DAHDI/23-1
-- Hungup 'DAHDI/23-1'
  == Spawn extension (empl, 9111222, 8) exited non-zero on 
'SIP/mine-0521'
-- Executing [...@empl:1] Verbose(SIP/mine-0521, 2|Hangup 
SIP/mine-0521 with cause 16) in new stack
  == Hangup SIP/mine-0521 with cause 16
-- Executing [...@empl:2] DumpChan(SIP/mine-0521, ) in new stack
Dumping Info For Channel: SIP/mine-0521:

Info:
Name=   SIP/mine-0521
Type=   SIP
UniqueID=   sys.domain.com-1294514614.2629
CallerID=   444555
CallerIDName=   Jim Dickenson
DNIDDigits= 9111222
RDNIS=  (N/A)
State=  Up (6)
Rings=  0
NativeFormat=   0x2 (gsm)
WriteFormat=0x2 (gsm)
ReadFormat= 0x2 (gsm)
1stFileDescriptor=  65
Framesin=   248 
Framesout=  253 
TimetoHangup=   0
ElapsedTime=0h0m0s
Context=empl
Extension=  h
Priority=   2
CallGroup=  
PickupGroup=
Application=DumpChan
Data=   (Empty)
Blocking_in=(Not Blocking)

Variables:
DIALSTATUS=ANSWER
DIALEDTIME=5
ANSWEREDTIME=1
RTPAUDIOQOS=ssrc=671389293;themssrc=651772178;lp=0;rxjitter=0.001217;rxcount=248;txjitter=0.00;txcount=252;rlp=0;rtt=0.00
BRIDGEPEER=DAHDI/23-1
DIALEDPEERNUMBER=G1/111222
DIALEDPEERNAME=DAHDI/23-1
MACRO_DEPTH=0
RCStatus=0
MyChan=SIP
sipcallid=0b69233cd5469...@192.168.0.16
SIPUSERAGENT=Grandstream GXP2000 1.2.2.6
SIPDOMAIN=sys.domain.com
SIPURI=sip:m...@00.00.000.000:5064

   -- Executing [...@empl:3] ExecIf(SIP/mine-0521, 
0|Set|DB(conf//haveadmin)=no) in new stack

-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jan 7, 2011, at 12:44 PM, C F wrote:

 PRICAUSE will give you lots of info on why a call was hungup on. Not
 sure if SIP will give you the same.
 
 On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
 Does Asterisk, currently using version 1.4, get any more information about 
 the result of an outbound call made over a PRI line compared to a call via a 
 SIP trunk?
 
 As an example, in a PRI call there is this message that shows up on the 
 console:
 
 [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network.
 
 for a call to a fax machine. Does asterisk set anything that a dialplan can 
 access that can know the call was to a fax machine?
 

Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

2011-01-09 Thread Doug Lytle

Jim Dickenson wrote:

I am running version 1.4.x. Where do I get PRICAUSE?


NoOP(Hangup Cause: ${HANGUPCAUSE})

Doug


--
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

2011-01-08 Thread C F
PRICAUSE will give you lots of info on why a call was hungup on. Not
sure if SIP will give you the same.

On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
 Does Asterisk, currently using version 1.4, get any more information about 
 the result of an outbound call made over a PRI line compared to a call via a 
 SIP trunk?

 As an example, in a PRI call there is this message that shows up on the 
 console:

 [2011-01-05 14:59:02]     -- Channel 23 detected a CED tone from the network.

 for a call to a fax machine. Does asterisk set anything that a dialplan can 
 access that can know the call was to a fax machine?

 If a call is placed to a number that is disconnected so a special information 
 tone is played can either a PRI call or a SIP call know this without 
 analyzing the audio stream?

 Are there reasons to prefer the use of PRI over SIP or SIP over PRI?

 I would like people's opinions as to if one form is better than the other in 
 any meaningful way.

 Thanks for you feed-back.
 --
 Jim Dickenson
 mailto:dicken...@cfmc.com

 CfMC
 http://www.cfmc.com/




 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

2011-01-08 Thread Tom Rymes

On Jan 6, 2011, at 8:08 PM, Joel Maslak wrote:

 On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
 Are there reasons to prefer the use of PRI over SIP or SIP over PRI?

[snip]

 I run the PBX for my organization which has about 160 extensions.  I
 wouldn't even think of doing anything but PRI for the main lines
 because (A) for our size organization where we are located, we're
 talking a couple hundred dollars a month difference between PRI and
 SIP in cost so it's nearly break-even in cost which means cost
 difference isn't a huge motivator, (B) it supports FAX, modems, and
 TTYs - perfectly, (C) Quality is 100% consistent.  In addition, the
 reliability is good enough that I'm willing to use it for 911.

[snip]

I have to agree with most of what Joel said in his message. For me, the main 
problem with many sip implementations is that your phone service will be only 
as reliable as your internet service. If you have a dedicated internet line 
that is highly reliable, that's not a big deal, but DSL, Cable, and the like 
aren't reliable enough for our needs.

Having said that, one downside of a PRI is that you are paying for all of those 
channels, even when you aren't using them. Companies like Paetec and most other 
large telcos are offering SIP trunks over an MPLS circuit, running on a T1 
loop. This covers the reliability problem, as you are running over the same 
type of circuit as your PRI, and it allows you to take advantage of unused 
channels as data bandwidth. This is especially helpful for folks who have a 
data T1 and a PRI, as they can get higher bandwidth for data when there isn't 
much voice traffic. Because they use G.729, you can also fit more calls on the 
same circuit. That choice of codec eliminates the ability to send/receive 
faxes, though, and it's likely expensive when compared to other SIP solutions, 
but it does appear to be pretty slick. 

Another benefit of SIP is that it doesn't require a Digium, Sangoma, or similar 
interface card in the server, simplifying migrations and reducing cost in many 
scenarios. 

Tom
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] Benefit of PRI vs SIP trunk calls

2011-01-07 Thread Joel Maslak
On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
 Are there reasons to prefer the use of PRI over SIP or SIP over PRI?

Assuming you are talking to connect a PBX to the PSTN...

PRI advantages:

1. Relatively little equipment between the PTSN and the PBX.  Less to
break or go wrong.

2. Simple to set up.  No need for QoS, routing, authentication, etc.
Of course if you only know IP, SIP is easier, but if you learn both,
ISDN is easier.

3. If compared to SIP over internet, PRI has guaranteed quality.
Granted, SIP *can* have just as good (and better) quality, just not
guaranteed if done over the internet (it can be guaranteed over a
private circuit).

4. Less latency/delay so there is less talk-over.

5. FAX, high speed modem, TTY, etc, pass-through actually works.  (it
*can* work over SIP, but Asterisk just isn't quite there yet)

I run the PBX for my organization which has about 160 extensions.  I
wouldn't even think of doing anything but PRI for the main lines
because (A) for our size organization where we are located, we're
talking a couple hundred dollars a month difference between PRI and
SIP in cost so it's nearly break-even in cost which means cost
difference isn't a huge motivator, (B) it supports FAX, modems, and
TTYs - perfectly, (C) Quality is 100% consistent.  In addition, the
reliability is good enough that I'm willing to use it for 911.

Of course if this installation wasn't in downtown Denver, where ISDN
PRI is very cheap (a full CLEC 23-channel ISDN PRI costs roughly what
6 or 7 ILEC POTS lines cost), then SIP would be interested.

SIP advantages:

1. Cheap (at least SIP-over-internet)

2. Easy and quick to scale if you have bandwidth.

3. Great for disaster recovery if using SIP over internet

4. Very cheap to get local numbers from all around the world.

5. If using SIP over internet, easy to compare providers

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] Benefit of PRI vs SIP trunk calls

2011-01-06 Thread Jim Dickenson
Does Asterisk, currently using version 1.4, get any more information about the 
result of an outbound call made over a PRI line compared to a call via a SIP 
trunk?

As an example, in a PRI call there is this message that shows up on the console:

[2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network.

for a call to a fax machine. Does asterisk set anything that a dialplan can 
access that can know the call was to a fax machine?

If a call is placed to a number that is disconnected so a special information 
tone is played can either a PRI call or a SIP call know this without analyzing 
the audio stream?

Are there reasons to prefer the use of PRI over SIP or SIP over PRI?

I would like people's opinions as to if one form is better than the other in 
any meaningful way.

Thanks for you feed-back.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/




--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users