[asterisk-users] asterisk 10.4.2 realtime callgroup missing

[Stephen Collier]

I am using asterisk 10.4.2 and everything is working correctly except
callgroup is not working correctly. I have callgroup field in mysql but
if I run sip show peer  on the asterisk cli I get the callgroup
field as blank all other fields correctly populated including
pickupgroup.

I'm not sure where to look to debug the realtime sip connection

Cheers
Stephen Collier


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Re: [asterisk-users] Dahdi-2.4.0+2.4.0 means ??

[Shaun Ruffell]

On Fri, Jun 22, 2012 at 09:34:34AM +0530, upendra wrote:
> hi,
> 
> in the documents its says about  dahdi-linux + asterisk change, but there
> is no explanation about it .If any know clearly about dahdi versioning then
> please let us know .

Hi. Which document does it say that it's dahdi-linux + asterisk? 

As Patrick said (and you can see at http://www.asterisk.org/downloads ),
x.y.z+a.b.c is a tarball that contains version x.y.z of DAHDI-Linux, and
version a.b.c of DAHDI-Tools. It's intended to be a convenience since on Linux
you will typically need both of these packages to operate.

DAHDI-Linux and DAHDI-Tools released as separate packages since it's possible
for DAHDI-Tools to be used on other platforms such as FreeBSD.

The general scheme of the version numbers is normal semantic versioning first
described here: 
http://lists.digium.com/pipermail/asterisk-dev/2008-December/035673.html

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Re: [asterisk-users] low success rate for ReceiveFax

[Steve Underwood]

On 06/22/2012 11:58 AM, Roi Stork wrote:

Hi,

Im able to send faxes with no errors, but the success rate for the
receiving side is less than 50%.

Asterisk usually returns records these errors as partial fax and fax
protocol error.

A lot of the error values returned by FAXOPT are 3RD_T2_TIMEOUT and T2_TIMEOUT.

Any suggestions on how to improve the fax receiving rate?


"I have a problem. Can you fix it?" is not really a meaningful question.

Steve


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Re: [asterisk-users] Dahdi-2.4.0+2.4.0 means ??

[upendra]

hi,


in the documents its says about  dahdi-linux + asterisk change, but there
is no explanation about it .If any know clearly about dahdi versioning then
please let us know .


regards
Upendra

On Wed, Apr 11, 2012 at 6:11 PM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 04/11/2012 01:39 PM, upendra wrote:
>
>> Hi,
>>
>> can anyone tell me what does that 2.4.0+2.4.0 means in dahdi release
>> numbering ??? 2.4.0?
>>
>
> A combination of dahdi-linux 2.4.0 and dahdi-tools 2.4.0.
> See http://www.asterisk.org/**downloads
>
> I recommend you use the latest version of both (2.6.0).
>
> Regards,
> Patrick
>
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[asterisk-users] low success rate for ReceiveFax

[Roi Stork]

Hi,

Im able to send faxes with no errors, but the success rate for the
receiving side is less than 50%.

Asterisk usually returns records these errors as partial fax and fax
protocol error.

A lot of the error values returned by FAXOPT are 3RD_T2_TIMEOUT and T2_TIMEOUT.

Any suggestions on how to improve the fax receiving rate?

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Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk

[Steve Edwards]

On Wed, 20 Jun 2012, Tim Nelson wrote:


Have a look at VQmonitor:

http://www.manageengine.com/products/vqmanager/

It works very well.


...it worked well when you could buy it. Apparently it is EOL now [1]. 
Sorry for the noise. These aren't the droids you're looking for.


You give up too easily :)

http://archives.manageengine.com/vqmanager/7011/ManageEngine_VQManager.bin

Reverts to 'free' version after 30 days.

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Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] Timeout for Huntgroup

[eherr]

I have not found a solution so I am checking with the Masses here.

 

I have a client who has a old 5 line key system without voicemail.

 

Currently, I can set up a huntgroup and ring each line for 15 seconds and after 
the 5th line has reached its limit, the call goes to
voicemail.

 

The problem with this is that the caller will hear roughly 1m15s worth of 
ringing.

 

I trying to give them the same ability to always ring the huntgroup, starting 
with line 1, and hunting only if its busy or the
timeout is reached.

 

The modification needed is to only ring the huntgroup for 25 seconds ( roughly 
5 rings ) and then pull the call back and send it to
voicemail if the line was not answered.

 

Has anyone seen this feature created and implemented and if so, how?

 

Thanks,

--eherr 

 

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Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk

[Valer Nur]

You can try PBXMate. It is more of speech improvement software (i.e. noise 
removal etc.)  but it also gives you speech quality statistics.
It is not a free tool but I think there is a free evaluation version.
http://www.solicall.com/products.html#PBXMate




 From: Stefan at WPF 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
 
Sent: Wednesday, June 20, 2012 9:25 PM
Subject: Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk
 

Yeah, I noted that too, but besides that it seems like it is exactly what I am 
looking for. I am especially confused that there's no hint like "hey, buy our 
new product", just EOL. So let's say I am looking for an alternative to this. 
And unfortunately I have to add it's for private use and I therefore need a 
free solution, which probably restricts the selection ): Well, anything better 
than checking logs by hand would be already a good start :-)--
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Re: [asterisk-users] Spandsp supports T.38?

[Steve Underwood]

On 06/22/2012 12:49 AM, Ahmed Munir wrote:

Hi,

I would like to know whether SpanDSP supports T.38 for Asterisk 10? 
Because as far as using Fax for Asterisk, I'm getting some issues 
using T.38
Only spandsp fully supports T.38 in Asterisk 10. The Digium module 
cannot work in gateway mode.


Steve


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[asterisk-users] Spandsp supports T.38?

[Ahmed Munir]

Hi,

I would like to know whether SpanDSP supports T.38 for Asterisk 10? Because
as far as using Fax for Asterisk, I'm getting some issues using T.38

-- 
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Ahmed Munir Chohan
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Re: [asterisk-users] Asterisk 10/1.6.1 and Dahdi/Libpri compatilities in BRI /PtmP

[Richard Mudgett]

> My previous message was incomplete.
> 
> 
> On thing to note is I had to forbid hfcmulti in modprobe.d in the
> second box to comply with a warning from dahdi. Without that, I could
> see this line in the output of lsmod:
> mISDN-core  hfcmulti
> 
> 
> 1. What is the root cause that makes a board change its sync source ?
> How can I check this ?

I would think layer 1 going down.  Many European telcos for BRI PTMP lines
drop layer 2 and then layer 1 to conserve power.

Is the switching of clock sources causing a problem?

> 2.  How can I get rid of these alarms ?

See the chan_dahdi.conf.sample file about the following options.

You could use the layer1_presence option to make Asterisk ignore those
alarms.

You could use the layer2_persistence option to keep layer 2 up.  To use
this option however, requires using libpri SVN 1.4 branch code as current
released versions do not support the option.  Using the layer2_persistence
option restores behavior that was removed for better Q.921 conformance for
PTMP after libpri v1.4.10.2 and is why you are seeing a behavior difference
between versions.

> 3. Shall I report this ?

It is normal with BRI PTMP lines.  It is also the reason for the
layer1_presence and layer2_persistence options.

> 4. Waht would you suggest ?
> 
> Regards
> 
> 
> 
> 2012/6/21, Olivier :
> > Hi,
> >
> > After an upgrade, I discovered yesterday strange things I would
> > like
> > to share here.
> >
> > Basically, I'me comparing platforms:
> > The first one is a 2.6.26 (Debian Lenny) platform, with Asterisk
> > 1.6.1.18, Libpri 1.4.10.2, Dahdi revision 8853 (must be between 2.3
> > and 2.5, I think).
> > The second one is a 2.6.32 (Debian Squeeze) platform, with Asterisk
> > 10.5.1, Libpri 1.4.12, Dahdi 2.6.1.
> > Both are connected to telco BRI lines in TE/PtmP mode through a
> > Junghanns QuadBRI board (wcb4xxp driver).
> > Both handle incoming and outgoing calls correctly, as far as I can
> > tell.
> >
> > But on the second one, though working fine, Dahdi keeps showing
> > alarm
> > messages such as:
> > [71765.784120] wcb4xxp :01:0e.0: new card sync source: port 1
> > [71767.484151] wcb4xxp :01:0e.0: new card sync source: port 1
> > [71771.184119] wcb4xxp :01:0e.0: new card sync source: port 2
> > [71794.184164] wcb4xxp :01:0e.0: new card sync source: port 1
> >
> > and "pri show spans" mostly (but not always) report worrying
> > status:
> > PRI span 1/0: Down, Active
> > PRI span 2/0: In Alarm, Down, Active
> >
> > On the first box "pri show spans" constantly reports the line is
> > up.
> >
> > On thing to note is I had to forbid hfcmulti in modprobe.d in the
> > second box to comply with a warning from dahdi. Without that, I
> > could
> > see this line in the output of lsmod:
> > mISDN-core

Both mISDN and DAHDI have drivers for your BRI card.  Only one of them
should be loaded.  Since you are using DAHDI and not mISDN, you should
load the DAHDI version.

Richard

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[asterisk-users] Reliable method for FoIP

[Ahmed Munir]

Hi,

I'm looking for a method to setup FoIP i.e. using T.38 protocol with no
PSTN lines.

I tested T.38 feature for Asterisk but the problem I'm getting is unable to
send more than 2 pages but getting timeout error.

Past couple of years I also configured and tested hylafax + iaxmodem for
T.30 faxing but I would like to know whether it  also supports T.38
protocol or not?

Is there any other reliable method available for FoIP? If it is, please
share your views.


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Re: [asterisk-users] Fax setup T.38 Help needed

[Matthew Jordan]

- Original Message - 

> From: "Thorben Jensen" 
> To: "asterisk-users" 
> Sent: Wednesday, June 20, 2012 8:25:28 PM
> Subject: [asterisk-users] Fax setup T.38 Help needed

> Hi,

> I'm looking for someone who can help us setup Fax with T. 38 on
> asterisk 10.x.x - We need to be able to do FoIP (Fax over IP) as we
> have no pstn lines available.

> Do you know how to setup a reliable fax system, then we will pay you
> to help us do this.

If you're looking for consultants, you may want to try the asterisk-biz
mailing list.

--
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Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk

[Marek Cervenka]

Dne 21.6.2012 9:52, Ishfaq Malik napsal(a):

On Wed, 2012-06-20 at 20:04 +0200, Stefan at WPF wrote:

Hello,

1) I am wondering what is the best practice to monitor if there are or
were problems with SIP calls on my Asterisk box. E.g. how about a
software that extracts all calls from the /var/log/asterisk/full (I
have permanently enabled verbose 10 and sip debug) log and tells me on
which of them were problems? Checking the logs manually is very hard,
but as SIP is a standardized protocoll, there should be tools doing
that for you? As an example, a person calling me recently got a 488
Not acceptable error as reply from my Asterisk box. Nothing came
through to my SIP phone, so I didn't know anything about the call or
the problems (which were on his phone btw). I would like to be
informed about such cases, know that there was a call to my Asterisk
box that made problems.

2) How about monitoring speech quality? E.g. sometimes it seems like a
packet is missing (I then have a short pause during the call), how to
monitor such things and create statistics out of this data?

So basically I want to monitor my Asterisk installation proactively
for reliability/problems and (speech) quality.


check asterisk testsuite
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation

thereis scenarios for console sip client pjsua(from pjproject) which can 
perform speech quality measurement


marek cervenka


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[asterisk-users] Dahdi and one port HFC cards

[Olivier]

Hi,

Is this http://www.voip-info.org/wiki/view/Asterisk+vzaphfc page data
still up to date ?
In other words, is it possible to use One port BRI cards with Dahdi ?

Regards

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Re: [asterisk-users] Asterisk 10/1.6.1 and Dahdi/Libpri compatilities in BRI /PtmP

[Olivier]

My previous message was incomplete.


On thing to note is I had to forbid hfcmulti in modprobe.d in the
second box to comply with a warning from dahdi. Without that, I could
see this line in the output of lsmod:
mISDN-core  hfcmulti


1. What is the root cause that makes a board change its sync source ?
How can I check this ?
2.  How can I get rid of these alarms ?
3. Shall I report this ?
4. Waht would you suggest ?

Regards



2012/6/21, Olivier :
> Hi,
>
> After an upgrade, I discovered yesterday strange things I would like
> to share here.
>
> Basically, I'me comparing platforms:
> The first one is a 2.6.26 (Debian Lenny) platform, with Asterisk
> 1.6.1.18, Libpri 1.4.10.2, Dahdi revision 8853 (must be between 2.3
> and 2.5, I think).
> The second one is a 2.6.32 (Debian Squeeze) platform, with Asterisk
> 10.5.1, Libpri 1.4.12, Dahdi 2.6.1.
> Both are connected to telco BRI lines in TE/PtmP mode through a
> Junghanns QuadBRI board (wcb4xxp driver).
> Both handle incoming and outgoing calls correctly, as far as I can tell.
>
> But on the second one, though working fine, Dahdi keeps showing alarm
> messages such as:
> [71765.784120] wcb4xxp :01:0e.0: new card sync source: port 1
> [71767.484151] wcb4xxp :01:0e.0: new card sync source: port 1
> [71771.184119] wcb4xxp :01:0e.0: new card sync source: port 2
> [71794.184164] wcb4xxp :01:0e.0: new card sync source: port 1
>
> and "pri show spans" mostly (but not always) report worrying status:
> PRI span 1/0: Down, Active
> PRI span 2/0: In Alarm, Down, Active
>
> On the first box "pri show spans" constantly reports the line is up.
>
> On thing to note is I had to forbid hfcmulti in modprobe.d in the
> second box to comply with a warning from dahdi. Without that, I could
> see this line in the output of lsmod:
> mISDN-core
>

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[asterisk-users] Tools for Load testing

[upendra]

Hi,


Which are the tools for testing the load test for dahdi/Asterisk .

- Call load test .
- Stress test.


Regards
Upendra
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Re: [asterisk-users] asterisk with ss7 voice broadcast

[[Digital^Dude] ®]

Asterisk 1.8.7.1 built by root on a x86_64 running Linux.
CentOS release 5.5 (Final)
RAM: 4 GB
CPU: Dual Xeon 2.66 GHz

Asterisk is running as root

data seg size   (kbytes, -d) unlimited
file size   (blocks, -f) unlimited
pending signals (-i) 38912
max locked memory   (kbytes, -l) 32
max memory size (kbytes, -m) unlimited
open files  (-n) 4096
pipe size(512 bytes, -p) 8
POSIX message queues (bytes, -q) 819200
stack size  (kbytes, -s) 10240
cpu time   (seconds, -t) unlimited
max user processes  (-u) 38912
virtual memory  (kbytes, -v) unlimited
file locks  (-x) unlimited

The changes in ulimit apparently don't get reflected when I run a broadcast
on asterisk.
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[asterisk-users] Asterisk 10/1.6.1 and Dahdi/Libpri compatilities in BRI /PtmP

[Olivier]

Hi,

After an upgrade, I discovered yesterday strange things I would like
to share here.

Basically, I'me comparing platforms:
The first one is a 2.6.26 (Debian Lenny) platform, with Asterisk
1.6.1.18, Libpri 1.4.10.2, Dahdi revision 8853 (must be between 2.3
and 2.5, I think).
The second one is a 2.6.32 (Debian Squeeze) platform, with Asterisk
10.5.1, Libpri 1.4.12, Dahdi 2.6.1.
Both are connected to telco BRI lines in TE/PtmP mode through a
Junghanns QuadBRI board (wcb4xxp driver).
Both handle incoming and outgoing calls correctly, as far as I can tell.

But on the second one, though working fine, Dahdi keeps showing alarm
messages such as:
[71765.784120] wcb4xxp :01:0e.0: new card sync source: port 1
[71767.484151] wcb4xxp :01:0e.0: new card sync source: port 1
[71771.184119] wcb4xxp :01:0e.0: new card sync source: port 2
[71794.184164] wcb4xxp :01:0e.0: new card sync source: port 1

and "pri show spans" mostly (but not always) report worrying status:
PRI span 1/0: Down, Active
PRI span 2/0: In Alarm, Down, Active

On the first box "pri show spans" constantly reports the line is up.

On thing to note is I had to forbid hfcmulti in modprobe.d in the
second box to comply with a warning from dahdi. Without that, I could
see this line in the output of lsmod:
mISDN-core

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[asterisk-users] Strange behavior - Can't figure out

[Zohair Raza]

Hi,

I have two asterisk boxes, one with asterisk 1.8.12.0 and the other
with asterisk 1.8.9.2

Sip show settings of both boxes have no difference and also the peers

I am generating a call using call file with following details:
Channel: SIP/1028
Account: 9164421122< -- this is the accountcode of 1028
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: default
Extension: 1031
Priority: 1
CallerID: "Zohair Raza"<1031>  < -- I want to see this caller id at
dialing peer (1028) and "Test" <1028>  (originiating caller id) at
dialed peer

On asterisk 1.8.9.2 I get results as expected and debug output is as below

-- Executing [1031@default:1] AGI("SIP/1028-3897", "agi.php")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
AGI Tx >> agi_request: agi.php
AGI Tx >> agi_channel: SIP/1028-3897
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: a-1340263981.14503
AGI Tx >> agi_version: 1.8.9.2
AGI Tx >> agi_callerid: 1028
AGI Tx >> agi_calleridname: Test  <--
caller id of 1028
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: unknown
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: default
AGI Tx >> agi_extension: 1031
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode: 9164421122
  < -- accountcode of 1028 here
AGI Tx >> agi_threadid: 1095772480
AGI Tx >>
AGI Rx << GET VARIABLE CDR(clid)
AGI Tx >> 200 result=1 ("Test" <1028>)



Same I am trying on another box with these details

Channel: SIP/5405
Account: 6167531316< -- this is the accountcode of 5405
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: default
Extension: 5050
Priority: 1
CallerID: "Test 2"<5050>  < -- I want to see this caller id at dialing
peer (5405) and "Test" <5050>  (originiating caller id) at dialed peer

But, for some reason it is showing Test 2 <5050 on both phones.

On Cli Debug, the behavior is also different

-- Executing [5050@default:1] AGI("SIP/5405-01f7", "agi.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
AGI Tx >> agi_request: agi.php
AGI Tx >> agi_channel: SIP/5405-01f7
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: TT-1340270088.522
AGI Tx >> agi_version: 1.8.12.0
AGI Tx >> agi_callerid: 5050
AGI Tx >> agi_calleridname: Test 2< -- here
it's callerid of 5050 instead of 5405
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: unknown
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: default
AGI Tx >> agi_extension: 5050
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode: 6167531316< --
account code of 5405
AGI Tx >> agi_threadid: 1084270912
AGI Tx >>
AGI Rx << GET VARIABLE CDR(clid)
AGI Tx >> 200 result=1 ("Test 2" <5050>)


Can anybody help me on figuring this out please.

Thanks

Regards,
Zohair Raza

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Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk

[Ishfaq Malik]

On Wed, 2012-06-20 at 20:04 +0200, Stefan at WPF wrote:
> Hello,
> 
> 1) I am wondering what is the best practice to monitor if there are or
> were problems with SIP calls on my Asterisk box. E.g. how about a
> software that extracts all calls from the /var/log/asterisk/full (I
> have permanently enabled verbose 10 and sip debug) log and tells me on
> which of them were problems? Checking the logs manually is very hard,
> but as SIP is a standardized protocoll, there should be tools doing
> that for you? As an example, a person calling me recently got a 488
> Not acceptable error as reply from my Asterisk box. Nothing came
> through to my SIP phone, so I didn't know anything about the call or
> the problems (which were on his phone btw). I would like to be
> informed about such cases, know that there was a call to my Asterisk
> box that made problems.
> 
> 2) How about monitoring speech quality? E.g. sometimes it seems like a
> packet is missing (I then have a short pause during the call), how to
> monitor such things and create statistics out of this data?
> 
> So basically I want to monitor my Asterisk installation proactively
> for reliability/problems and (speech) quality.
> 
> Thanks for any hints!
> 
> Best regards
> Stefan
> --

I've not used this myself but had a look at the site and I think it's
pretty much what you're after...

http://www.voipmonitor.org/

Ish

-- 
Ishfaq Malik 
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
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COMPANY REG NO. 04920552


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