[asterisk-users] iax trunking on OpenBSD

[Sebastian Reitenbach]

Hi,

do I have a chance to use iax trunking on OpenBSD where there is no zaptel 
driver or ztdummy available? Do I can use sth. else as timing source?

kind regards
Sebastian

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Re: [asterisk-users] Slow list

[Remco Post]

Philipp Kempgen wrote:
> Wow. My message made it to the list after more than 3 hours.
> 
> 
>   Philipp
> 

I noticed similar delays, no wonder we get a lot of 'me too'-s to the
list (sorry list for my bitching).

-- 

Remco Post

"I didn't write all this code, and I can't even pretend that all of it
makes sense." -- Glen Hattrup
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Re: [asterisk-users] Scaling Asterisk: High volume benchmarks (0 to 450 calls)

[Remco Post]

Matthew J. Roth wrote:
> List users,
> 
> This post contains the benchmarks for Asterisk at high call volumes on a
> 4 CPU, dual-core (8 cores total) server.  It's a continuation of the
> posts titled "Scaling Asterisk: Dual-Core CPUs not yielding gains at
> high call volumes".  They contain a fair amount of information,
> including details about our servers and the software on them.  I'm happy
> to answer any questions you might have, but please take a moment to
> review those posts to make sure they don't contain the information
> you're seeking.

I guess that if I read these stats correctly, the bottleneck for * is
not so much cpu power, it's the cpu cache. As I see it, the cpu cache
becomes far less efficient for larger call volumes, eg. the cache is
unable to keep the most frequently used code and data in cache, due to
the sheer amount of call date going through the cpu. I guess that you do
have some gain from going from single core to dual-core but is dwarfed
by the very limited effect on the cache.

But that is just a guess. Maybe for pure voip solutions cpu's with a
huge cache like eg Power5+ would perform much better that ia32/x64 cpu's.

> 
> Thank you,
> 
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
> 
> 
> Conclusions
> ---
> Once again, I'm presenting the conclusions first. Scroll down if you're
> more interested in the raw data.
> 
>  1. Asterisk scales quite well up to a certain number of calls.  At this
> point, the cost in CPU cycles per call starts to increase more
> drastically.  A graph of the Avg Used% values can be used to demonstrate
> this.  It can be described as consisting of two roughly linear
> segments.  The first segment is from 0 to 110 calls.  The rest of the
> graph is a second, steeper segment.  This is not entirely true, as in
> fact each new call costs a little more than the last, but it is a useful
> simplification.
>  2. Even at very high call volumes, Asterisk uses less than 512 KB of
> memory.  2 GB of RAM would probably avoid swapping and excessive disk
> activity on most Asterisk installations.
>  3. Future benchmarks should be based on the number of active channels,
> not active calls.
> 
> I'm relying on you to point out my mistakes and omissions, so please
> take a look at the data and respond with your own analysis and conclusions.
> 
> Benchmarking Methodology
> 
> The benchmarks are based on data I collected over the period of
> 5/12/2007 to 05/30/2007 from two production servers used in our inbound
> call center.  The servers are identical 8-core Dell PowerEdge 6850s as
> documented in my prior posts.  They are meant to be used as a
> primary/backup pair, but both were used in production in order to rule
> out a hardware failure as the cause of our scaling issues.
> 
> The data was collected by a bash script executed from cron every 2
> minutes.  This script utilizes some basic Linux tools (such as sar,
> free, df, and the proc filesystem) to record information about the
> system, and 'asterisk -rx "show channels"' to record information about
> the number of active calls and channels within Asterisk.
> 
> Unfortunately, the sample sizes this produced are relatively small for
> the 300-450 call range.  This is due to two factors:
> 
>  1. The majority of the time we don't operate at such high call volumes.
>  2. Asterisk intermittently fails to report call and channel statistics
> when the CPU idle is low.
> 
> This means that the benchmarking results are somewhat erratic for the
> 300-450 call range.  The good news is that they are pretty consistent
> for 0 to 300 calls, and I'd imagine that covers the range most people
> are interested in.
> 
> Keep in mind that the impetus behind this benchmarking was the lack of a
> performance boost on the dual-core server at high call volumes, so the
> high call range may also be skewed by whatever bottleneck is being hit
> on the 8-core servers.  In the near future, we will be adding one of our
> 4-core PowerEdge 6850s to our production environment.  I'll collect and
> analyze the same data, which I believe will show similar performance (as
> defined by cumulative idle CPU percentage) at around 200-300 calls.
> 
> In the end, I hope to understand this problem well enough to overcome it
> or determine what the optimal point is for achieving the highest call
> volume without over-dimensioning the hardware.
> 
> Call Types and a Note on Channels
> -
> All of the calls are SIP-to-SIP using the uLaw codec.  The vast majority
> of the calls are either in queue or connected to an agent, but there are
> also a small number of regular outbound calls and transfers.  Every call
> that is connected to an agent is recorded via the Monitor() application
> in PCM format to a RAM disk.  In short, there was no transcoding,
> protocol bridging, or TDM hardware involved on the servers being
> benchmarked.
> 
> At any given time, t

[asterisk-users] SIP buddy watch

[clive.chan\(Alpha Trilogies Networks\)]

Hi all, 

Have some one advice me on the following needs?

1)   Clients need to use 50 Hard IP Phone, and 20 Soft-Phone. 

2)   Clients have 20 Hard IP phone and 10 Soft-phone within Operational
Department.

3)   The Soft-phone mostly is division head, and they wish to monitor
the online status of the existing hard-phone and soft-phone through the
soft-phone. The status like "On-line", "Busy", "Away"  

 

Any recommendation for those needs? 

 

 

 

 

>From Clive Chan

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Re: [asterisk-users] meetme realtime

[ram]

>
>  is this possible ?
>
   You can only do it with realtime static.



how can i do that, any document URL to achieve that

ram
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Re: [asterisk-users] Phantom calls: Detecting hangup quicker

[Stephen Bosch]

Gavin Henry wrote:
> Dear all,
> 
> We seem to be getting phantom calls when a inbound caller via the
> legacy pbx hangups before
> the SIP handsets have answered. The extensions also seem to hear
> ringing on the lines too sometimes.
> 
>   SIP Inbound  >
>   |
> legacy pbx (analogue) <-> (sangoma a400d) asterisk <-> SIP phones
> 
> Basically if a user hangups before the call has bridged, I think.
> 
> Is there anything we can do about this?

Yet another "call progress detection" issue.

Analog lines are problematic this way. Search the archives for "call
progress detection" or "disconnect supervision".

-Stephen-
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Re: [asterisk-users] Noise on FXS ports (Sangoma)

[Stephen Bosch]

John Novack wrote:
> Since the OP said the noise was on FXS ports, Jorge's answer isn't
> relevant.
> After listening to  a wav file of the noise, it sounds to my old ears
> like a background hiss or so called "comfort" noise, except for a couple
> of short pops which I assume to be an open microphone.

Yeah, that's correct.

Sometimes the background noise warbles a bit. This is why I am wondering
about ground.

The background noise is too inconsistent to my virgin ears to feel like
deliberate comfort noise. I wasn't aware that these modules were
supposed to generate that kind of comfort noise. Usually sidetone is
enough to make the set feel live to the user.

> Assuming this isn't generated by the POTS set connected to the port, I
> guess you'll have to dig into the configuration.
> Some POTS sets generate a similar though more erratic noise, caused by
> carbon transmitters and defective varistors in the network, but that is
> ancient history.
> More modern POTS sets with electret transmitters could certainly
> generate a similar noise.
> I am assuming that several different sets have been tried with similar
> results.

I have tried:

a cheap AT&T set;
a refurbished Northern Telecom "Symphony";
a classic ITT Touch Tone set

All of them sound great when connected directly to the PSTN (okay, maybe
the cheap AT&T set sounds a bit tinny), so I'm confident the sound is
not coming from the sets themselves.

> My experience with the Sangoma A200, and in fact Digium 400 and Adtran
> channel bank FXS circuits all are very quiet, so this is a bit of a puzzle.

Well, that's reassuring, because it means a solution must exist.

I don't know what configuration changes I would make to solve this. My
gut feeling is that it's an electrical problem somewhere within the
system, but perhaps I'm reaching for that too soon; in any case, I
wouldn't know where to start even if we accept that it is an electrical
problem.

Is a dirty ground not a possibility?

Is there a way to test this card in another server without having to
install Asterisk on it? I want to see if the noise is coming off the PCI
bus.

-Stephen-

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[asterisk-users] Solved: [SetAccount in extensions.conf]

[Matt]

> I'm using Asterisk 1.4 and I'm wanting to set an
> account code for incoming calls.  In the
> extensions.conf file I have the following:
> 
> exten => s,1,SetAccount(1234)
> exten => s,n,Dial(SIP/1234)
> 
> Then when I dial the extension the following error
> message pops up in the CLI:
> 
> [Jun  6 19:12:40] WARNING[28167]: pbx.c:1783
> pbx_extension_helper: No application 'SetAccount'
> for
> extension (inbound, XX, 1)
> 
> I already have the accountcode set up in sip.conf
> for
> that SIP phone and there are no problems with that. 
> Does anyone know what the problem is?  On
> voip-info.org, another person had the same problem
> in
> the comments section, but seemed to think it was a
> syntax difference and that SetAccount() would work
> for
> *1.2.
> 
> Any info will be greatly appreciated.
> Thank You,
> -Matt

An easy way to set an account code for the above
example for extensions.conf is the following:

exten => s,1,Set(CDR(accountCode)=1234)
exten => s,n,Dial(SIP/1234)

Hope this helps others.  I was consulting some other
people and they solved the problem before the
digium-users could.

-Matt


   

Be a better Heartthrob. Get better relationship answers from someone who knows. 
Yahoo! Answers - Check it out. 
http://answers.yahoo.com/dir/?link=list&sid=396545433
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Re: [asterisk-users] Noise on FXS ports (Sangoma)

[Stephen Bosch]

shadowym wrote:
> All I know is that your rx gain, at 0db is probably way too low.  My tx gain
> is typically set to -3db.

I set it 10 as per your previous post and ended up with audio so loud
there was clipping and distortion. I don't think that's the cause.

Sangoma has expressly told me to *increase* the txgain and leave the
rxgain alone.

Don't get me wrong -- I appreciate the help -- but I think we're barking
up the wrong tree here.

> Yes, I was thinking sip phones when I made the
> amplifier comment.  If you hear the noise as soon as you take the analog
> phone off hook without dialing the PSTN then I don't see how it could have
> anything to do with the telco card.

But that's just it -- I wasn't referring to the telco card. I was
referring to the FXS module which is providing dial tone to internal
analog phones. They're the only devices that exhibit the noise.

Do these gain settings have any impact on FXS modules in the first place?

> Is it a feedback type noise or
> background hiss?  Feedback type buzz would lead me to believe it's ground
> loop as someone else mentioned.  Ground loop hum is a PITA to try deal with.

No, it's like a background hiss. I have a recording of it if you're
interested in hearing it.

-Stephen-

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"Me Too" mails DOSsing the list (was: Re: [asterisk-users] Voip-info.org)

[Brian Capouch]

Compnet Bobby wrote:

Same in southern cali!



Folks, at the risk of sounding mean, once there are a couple of "Me Too" 
emails on a given topic, everyone who reads the list is clued in, and we 
don't really need another hundred or so one- or two-word emails adding 
to the already-bad load of traffic the list generates.


Very early on, we heard from the site maintainer that they were being 
DOSsed.


I still have many hours of list traffic ahead of me to read, and I 
despair of how many more of them are likely to be just like this 
one--and I'm not picking on you, Bobby; you just happened to post the 
25th one of these I've read today.


Thanks.

b.

--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.

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Re: [asterisk-users] Voicemail marking messages as Old

[John Novack]

Jason Parker wrote:
Yes, that is what Asterisk does.  I personally have never used a 
voicemail system that had any behavior other than that.  I certainly 
wouldn't expect it to be any different - however, it's possible that 
somebody would be willing to write a patch to allow that as an option.
If the listener disconnects before the message is finished playing, MANY 
VM systems keep that message marked as new. The listener has to finish 
the message OR press a digit to mark as delete or save.
The VM system in Asterisk is limited in many respects. Wasn't there a 
rewrite of the system in progress??


John Novack



- Original Message -
From: "Adrian A" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, June 6, 2007 2:50:00 PM (GMT-0600) America/Chicago
Subject: [asterisk-users] Voicemail marking messages as Old

It seems to me that simply listening to a new voicemail message will 
move the message to the Old folder, without any other user 
interaction. I'm working on a voicemail callback queue script and I 
have wrongly assumed that messages remain in INBOX unless the user 
actually saves or deletes them.

I'm running an older 1.2 version of Asterisk.
Is anyone able to confirm the same behavior in newer versions? Is 
there a way for Asterisk voicemail to behave like regular voicemail 
where a message remains "New" until the caller does something to it 
(other than simply listening to it) ?


Thanks.


--
Jason Parker
Digium


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Solved: [asterisk-users] Subscriptions to Agent hints stopped working after a changing the numbering

[Alexander Topolanek]

"Hint-priorities" alone in the dialplan do not work, there has to be at
least one real (numbered) priority for this extension, whith whatever
action behind:

exten => 41,hint,Agent/9761
exten => 41,1, Playback(silence)

PS.: I'm not perfectly sure, but Grandstreams with a "stable" firmware
seem to have a limitation of two digit numbers for the BLS's.


Am Donnerstag, den 31.05.2007, 09:10 +0200 schrieb Alexander Topolanek:
> Hi,
> 
> I had a working solution to display an agent status on the BLF of a
> Grandstream. Then I made some minor changes to the numbering (changed
> the extensions from 8X to 6X, and the Agent numbers from 978X to 976X),
> and out of the blue grandstreams do not register for an Angent status.
> 
> This is the relevant line in my extensions.conf:
> 
> exten => 41,hint,Agent/9761
> 
> and I see the hints, but no subscription:


best regards
Alexander

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[asterisk-users] Best Codec

[Davis Sylvester III]

We are evaluating starting a small VoIP consumer based platform.
What is the best codec to use with customers using primarily DSL as 
internet connectivity?


I know that g729 is the king-all, but I want to know what the rest of 
the professional are using out there.  g729 has a cost involved, so does 
the cost really offset the performance?  Or is it better to go with g711 
to start off?


We plan on using Linksys SPA921 as the primary phone and asterisk open 
source as the softswitch.  Any information you can pass would be 
appreciated.


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Re: [asterisk-users] Queue problem

[BJ Weschke]

On 6/6/07, Elmar Haneke <[EMAIL PROTECTED]> wrote:

Hi,

On asterisk 1.4.4 I have an strange effect on agents answering queue calls:

If an agent does set current call on hold the phone
immediately gets connected to the next incoming call.

What might cause this effect?
How can it be removed?



Set your core debug level to greater than 2 and then try this process
again and the go into your log that you've redirect DEBUG messages to
and search on " changed to state ".

I think you'll find that by doing what you're doing above, the
caller's channel state will change to some kind of ONHOLD state (if
SIP - or other channel tech that supports that dev state) but I would
think the agent channel itself might go to NOT IN USE.

Let us know what you find. This might be worth opening a bug on
Mantis based on what you come up with as I would think if you placed a
queue caller on hold you wouldn't then want to receive additional
calls in from queue.

BJ


--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [asterisk-users] 1.4 Zaptel/Sangoma Issues on CentOS

[Bruce Reeves]

Which CentOS version? You might try, if you have not already the beta
wanpipe drivers, they have:

Support for 2.6.20 kernel
s which include CentOS 5



On 6/6/07, Steve Totaro <[EMAIL PROTECTED]> wrote:


Any ideas?  Sangoma support is closed for the evening.

I have the latest Sangoma drivers and Asterisk 1.4 everything installed.

When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up"
repeated over and over.  The B channels never come up.  There are no
errors in any of the logs, zttool, or the wanpipe tools.

Intense pri debug output:
< Unnumbered frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended
) ]
< 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement

> [ 02 01 73 ]

> Unnumbered frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
> 0 bytes of data
-- Restarting T203 counter
q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:664 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Restarting T203 counter
  == Primary D-Channel on span 1 up
pri intense debug span 1
< [ 02 01 7f ]

< Unnumbered frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended
) ]
< 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement

> [ 02 01 73 ]

> Unnumbered frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
> 0 bytes of data
-- Restarting T203 counter
q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:664 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Restarting T203 counter
  == Primary D-Channel on span 1 up
< [ 02 01 7f ]

< Unnumbered frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended
) ]
< 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement

> [ 02 01 73 ]

> Unnumbered frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
> 0 bytes of data
-- Restarting T203 counter
q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:664 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Restarting T203 counter
  == Primary D-Channel on span 1 up

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB



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--
Bruce Reeves
Nortex Networks
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Re: [asterisk-users] 1.4 Zaptel/Sangoma Issues on CentOS

[Bruce Reeves]

Which CentOS version? You might try, if you have not already the beta
wanpipe drivers, they have:

Support for 2.6.20 kernels which include CentOS 5



On 6/6/07, Steve Totaro <[EMAIL PROTECTED]> wrote:


Any ideas?  Sangoma support is closed for the evening.

I have the latest Sangoma drivers and Asterisk 1.4 everything installed.

When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up"
repeated over and over.  The B channels never come up.  There are no
errors in any of the logs, zttool, or the wanpipe tools.

Intense pri debug output:
< Unnumbered frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended
) ]
< 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement

> [ 02 01 73 ]

> Unnumbered frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
> 0 bytes of data
-- Restarting T203 counter
q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:664 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Restarting T203 counter
  == Primary D-Channel on span 1 up
pri intense debug span 1
< [ 02 01 7f ]

< Unnumbered frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended
) ]
< 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement

> [ 02 01 73 ]

> Unnumbered frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
> 0 bytes of data
-- Restarting T203 counter
q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:664 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Restarting T203 counter
  == Primary D-Channel on span 1 up
< [ 02 01 7f ]

< Unnumbered frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended
) ]
< 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement

> [ 02 01 73 ]

> Unnumbered frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
> 0 bytes of data
-- Restarting T203 counter
q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:664 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Restarting T203 counter
  == Primary D-Channel on span 1 up

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB



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--
Bruce Reeves
Nortex Networks
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Re: [asterisk-users] Asterisk call quality detection

[Andrew Kohlsmith]

On Wednesday 06 June 2007 3:33 pm, Jared Smith wrote:
> Hopefully in the future we'll have the RTCP reports logged (either as
> part of the CDR records, or in a Call Quality log of some kind).
> Until then, I'm pretty sure you can listen for RTCP events through the
> Asterisk Manager Interface, and log them yourself.

Yes, you can.  The patch is in bug 8613.  Unfortunately, the manager events do 
NOT give you any information which you can link to a particular callno.  I've 
been meaning to add that to the patch, but haven't got to it just yet.

-A.
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Re: [asterisk-users] meetme realtime

[Carlos Chavez]

On Wed, 2007-06-06 at 23:42 +0530, ram wrote:
> Hi
>  
> iam using 1.2.17
>  
> does any one have information meetme in realtime
> and store in mysql i dont see any document
>  
> could some one help me
>  
>  is this possible ?
>  
You can only do it with realtime static.

> 
-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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[asterisk-users] 1.4 Zaptel/Sangoma Issues on CentOS

[Steve Totaro]

Any ideas?  Sangoma support is closed for the evening.

I have the latest Sangoma drivers and Asterisk 1.4 everything installed.

When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up"
repeated over and over.  The B channels never come up.  There are no
errors in any of the logs, zttool, or the wanpipe tools.

Intense pri debug output:
< Unnumbered frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended
) ]
< 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement

> [ 02 01 73 ]

> Unnumbered frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
> 0 bytes of data
-- Restarting T203 counter
q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:664 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Restarting T203 counter
  == Primary D-Channel on span 1 up
pri intense debug span 1
< [ 02 01 7f ]

< Unnumbered frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended
) ]
< 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement

> [ 02 01 73 ]

> Unnumbered frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
> 0 bytes of data
-- Restarting T203 counter
q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:664 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Restarting T203 counter
  == Primary D-Channel on span 1 up
< [ 02 01 7f ]

< Unnumbered frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000EA: 1
<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended
) ]
< 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement

> [ 02 01 73 ]

> Unnumbered frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000EA: 1
>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
> 0 bytes of data
-- Restarting T203 counter
q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED
q921.c:664 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED
-- Restarting T203 counter
  == Primary D-Channel on span 1 up

Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
 


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[asterisk-users] SetAccount in extensions.conf

[Matt]

I'm using Asterisk 1.4 and I'm wanting to set an
account code for incoming calls.  In the
extensions.conf file I have the following:

exten => s,1,SetAccount(1234)
exten => s,n,Dial(SIP/1234)

Then when I dial the extension the following error
message pops up in the CLI:

[Jun  6 19:12:40] WARNING[28167]: pbx.c:1783
pbx_extension_helper: No application 'SetAccount' for
extension (inbound, XX, 1)

I already have the accountcode set up in sip.conf for
that SIP phone and there are no problems with that. 
Does anyone know what the problem is?  On
voip-info.org, another person had the same problem in
the comments section, but seemed to think it was a
syntax difference and that SetAccount() would work for
*1.2.

Any info will be greatly appreciated.
Thank You,
-Matt


 

The fish are biting. 
Get more visitors on your site using Yahoo! Search Marketing.
http://searchmarketing.yahoo.com/arp/sponsoredsearch_v2.php
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Re: [asterisk-users] Stanaphone/Asterisk issue: No Audio with SIP to only one provider when switching servers

[Hadar Pedhazur]

Alejandro Lengua wrote:

Hello,
did you got your issue solved?
I am suffering of the same issue


Hi. I had it off for a few weeks, and then decided to try again, 
and it "just worked". I didn't change a single thing, only 
uncommented the register statement that I had previously 
commented. It's been "reliable" now for the past 2 weeks since I 
turned it back on.


I didn't bother to report here because I didn't have a "solution".

I guessed that they changed something on their side, since I did 
report the problem to them when it first happened (though they 
didn't respond), but, if you're having the problem, perhaps I just 
"got lucky".


Sorry to hear you're having the problem, I know how frustrating it is!
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[asterisk-users] Slow list

[Philipp Kempgen]

Wow. My message made it to the list after more than 3 hours.


  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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[asterisk-users] Sending multiline SMS

[Patrick Zwahlen]

Hi everyone,

How do you send multiline SMSs using smsq or .call files ?

smsq --motx-channel="mISDN/g:bri/" 078 "line1 line2"

How can I have a carriage return between line1 and line2 ? I have tried
the regular \n and \r. No success.

Thanks for any help. - Patrick -
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Re: [asterisk-users] Re: Verizon Interconnection

[carl]

I've connected to Verizon BRI circuits and had major echo issues. Moved to a 
Paetec PRI and bing all calls now work great.
  - Original Message - 
  From: Klaverstyn, David C 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Wednesday, June 06, 2007 1:47 PM
  Subject: RE: [asterisk-users] Re: Verizon Interconnection


  I have connected with a PRI service with Verizon but not SIP.  What is their 
SIP product as I am not familiar with it?

   


--

  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt
  Sent: Wednesday, 6 June 2007 9:15 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and 
Business-Oriented Asterisk Discussion
  Subject: [asterisk-users] Re: Verizon Interconnection

   

  So absolutely no one here was interconnected with Verizon?  I am going to 
shoot this over to asterisk-biz, also, in hopes someone may have missed it that 
is on the biz list.  The question again is:

  Has anyone on this list connected with Verizon's SIP product?  We are 
currently undergoing interop testing with Verizon, and honestly, it seems like 
the most convoluted process.   I'd be interested in talking with someone else 
who has gone through this and run a few things past you. 



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Re: [asterisk-users] Voicemail marking messages as Old

[Jason Parker]

Yes, that is what Asterisk does. I personally have never used a voicemail 
system that had any behavior other than that. I certainly wouldn't expect it to 
be any different - however, it's possible that somebody would be willing to 
write a patch to allow that as an option. 

- Original Message - 
From: "Adrian A" <[EMAIL PROTECTED]> 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
 
Sent: Wednesday, June 6, 2007 2:50:00 PM (GMT-0600) America/Chicago 
Subject: [asterisk-users] Voicemail marking messages as Old 

It seems to me that simply listening to a new voicemail message will move the 
message to the Old folder, without any other user interaction. I'm working on a 
voicemail callback queue script and I have wrongly assumed that messages remain 
in INBOX unless the user actually saves or deletes them. 
I'm running an older 1.2 version of Asterisk. 
Is anyone able to confirm the same behavior in newer versions? Is there a way 
for Asterisk voicemail to behave like regular voicemail where a message remains 
"New" until the caller does something to it (other than simply listening to it) 
? 

Thanks. 


-- 
Jason Parker 
Digium 
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[asterisk-users] Reload in 1.4 clears regexten

[Douglas Garstang]

Ok, I could have sworn this was fixed in Asterisk 1.2, but it seems in
Asterisk 1.4.4, that doing a reload, or even an 'extensions reload' will
clear any extensions that have been created by regexten. This is VERY
bad!

 

Doug.

 

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[asterisk-users] Console duplicate output problem

[Barton Fisher]

This is really strange.  Every message to the (VGA) console is written 
twice to the screen, but not on the SSH connection.

Any clues how to stop this behavior?

   -- Executing BackGround("Zap/216-1", "custom/3566/91_|m|") in 
new stack
   -- Executing BackGround("Zap/216-1", "custom/3566/91_|m|") in 
new stack


Bart
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Re: [asterisk-users] X100P Clone

[John Novack]

Henry Cobb wrote:

On 6/5/07, Jared Smith <[EMAIL PROTECTED]> wrote:

Most of the clone cards don't support far-end disconnect supervision,
so you'll have problems where Asterisk can't tell that the other party
has hung up the phone.  You'd be better off to buy a modern Asterisk
telephony card.


Why would anybody plug a telephone line into an X100P clone?

???
What else would one plug into it?
The X100 isn't all that bad, at least for a starter card. It can have 
issues  with long loops, doesn't work well in the UK.

I support several users with VERY SHORT loops where it simply works.
What do you want for 20 bucks anyway??


And when will Digium offer affordable one-port cards again?

Probably never.


John Novack



-HJC
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Re: [asterisk-users] Asterisk call quality detection

[carl]

I've noticed that there is the odd vnak message displayed in my asterisk syslog 
traces. Would have to alert on those i'd assume..
  - Original Message - 
  From: Matt 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Wednesday, June 06, 2007 1:44 PM
  Subject: Re: [asterisk-users] Asterisk call quality detection


I chart VNAKs per hour.


  Would you care to share how you accomplish this?   What programs do you use? 



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Re: [asterisk-users] PRI Partial Re-Rounting

[Anthony Francis]

You are right. It would be nice to "redirect" the calling party so that 
it connects the calling party to the forwarded number instead of using 
multiple channels.


Matthew Fredrickson wrote:
I think what they're talking about is forwarding the call before the 
call is established.  If I remember correctly, it's call CF[U,B,NR] 
for call forward on unavailable, busy, and no response.  Unfortunately 
though, none of the switchtypes support this variant of this 
function.  However, if 2BCT is acceptable, we have a working 
implementation for DMS100 switchtype in 1.4.


Matthew Fredrickson

On Jun 6, 2007, at 9:53 AM, Eric "ManxPower" Wieling wrote:


Jon Schøpzinsky wrote:

Hello List
We are trying to redirect calls directly, instead of opening a new 
channel and dialing out.

Etc:
A calls B on our asterisk, and is directly redirected to C
We have been told that this feature should be available on a PRI 
level, and is called Partial re-routing.

Anybody has an idea of whether this is supported in Asterisk?


It is called 2BCT.  It is supported on AT&T 5ESS PRI lines.  I don't 
think it is supported on NI2 or non-AT&T switches.  I've never used it.


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[asterisk-users] Polycoms lose registration and won't re-register

[ewr]

For the last few months we have intermittently been experiencing some very
strange registration problems with certain polycom phones.

Here is some background information:
I have about 150 Polycom Soundpoint IP 600s, 601s, and 650s spread between 8
servers at different locations.  Each phone is on the same network (and
subnet) as the server it connects to.  There is no NAT or anything else
strange that should be messing with the connection between the phones and
asterisk.  We are using Netgear FSM7328P and FSM7352 POE switches, with both
the server and phones directly connected to the same switch.  We have been
experiencing this issue intermittently for several months with all 3 types
of Polycom phones that we own. (600, 601, and 650)  It has happened with
phone firmware versions ranging from 1.6.6 to 2.1.1.  Asterisk versions have
been since at least 1.2.15, and it is still occurring with 1.2.18.

Seemingly randomly, a single phone will stop registering with asterisk.  All
of the other phones continue to work fine.  A "sip show peers" will show the
non-registered extension as:

Name/username  HostDyn Nat ACL Port Status
3310/3310  (Unspecified)D  0UNKNOWN

The phone can still make outgoing calls, but any calls to it will go
straight to voicemail.

Rebooting the phone (by the keypad, or by removing power) will not cause it
to re-register, nor will stopping asterisk and restarting it.

This has happened on phones that use realtime, and on ones that are manually
set up in the sip.conf.

The phones are provisioned via ftp, and if I take a different phone than the
misbehaving one and rename the .cfg's from the misbehaving phone
to the new phone, the new phone will always work fine.  The phone that
stopped working will continue not to register even if it is moved to a
different extension.  We have also tried 'touch'ing all of the config files
for a phone that won't register in order to update the timestamps.  It did
not make a difference.

If the phone that refuses to register is moved to a completely different
location and server, it will begin working again fine.  It can then be moved
back to the original location/server and will be fine.

We always start with the stock sip.cfg/phone.cfg/etc for whatever firmware
version we are using, and then make a few very minor changes.

Below are excerpts from the registration section from the polycom
"phone3310.cfg" and "sip.cfg" for the 3310 extension I am currently
fighting.  It is using the 2.1.1 firmware and connecting to asterisk 1.2.18.
These excerpts are from the normal config files we usually use, but I have
also tried changing the transport to "UDPOnly" and explicitly setting the
registration expiration and overlap in the configs to shorter values than
the default of 3600/60.

Over the last few months, this has happened several times to about 7 phones
on 5 different servers.  Since I have a phone that is currently doing this,
I would be happy to capture any sort of debug output that may help determine
the cause of the problem.  Any help or suggestions would be greatly
appreciated!

Thanks, 
Eric

Excerpt from phone3310.cfg:
   

Excerpt from sip.cfg:
  


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Re: [asterisk-users] Voip-info.org

[Remco Post]

Compnet Bobby wrote:
> Same in southern cali!
> 

ok, it's down, we all know it, use the mirrors, stop spamming, thanks :)

-- 

Remco Post

"I didn't write all this code, and I can't even pretend that all of it
makes sense." -- Glen Hattrup
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Re: [asterisk-users] cepstral TTS and app_swift

[Julian Lyndon-Smith]

Thanks for the input - that's what we've ended up doing. I was concerned 
at the impact on system performance, but it seems negligible.


I tested it with 30 simultaneous calls (1 calls in total) using sipp 
and it didn't crash once.


Julian

Mojo with Horan & Company, LLC wrote:

Have you tried something along the lines of:

System("swift blah blah blah -o blah.wav")
Playback("blah.wav")

It does have an inherent delay for the generation step but maybe swift 
binary segfaults less?  I've only used cepstral via swift binary, and it 
has never segfaulted for me. My swift and voice are version 4.2.0.


I doubt different voices behave differently, but just in case, I use the 
$7 Damien voice.


Moj

Julian Lyndon-Smith wrote:
We are having some major problems with app_swift since we went live. 
It is regularly segfaulting.


I don't know if this is my fault or not, but here's the story:

Installed the cepstral voices (at the time, 4.0) on our test system 
(2.6.9-42.0.10.ELsmp)
and later added some extra voices (now 4.2). All worked fine - we 
stress tested (20+ simultaneous calls).


Move to live ( 2.6.9-22.0.1.ELsmp) . Installed the cepstral voices 
(only 4.2).


Started having problems with only 5 calls: swift by itself on the 
command line was fine (it worked) but app_swift complained that it 
couldn't find any voices.


Looking into /opt/swift/lib, I saw that it was different to my test 
system.


On live I had (snipped)

-rwxrwxrwx  1 root root 139612 Jun  1 23:10 libceplang_en.so
=rwxrwxrwx  1 root root 139612 Jun  1 23:11 libceplang_en.so.4
-rwxr-xr-x  1 root root 139612 Jun  1 07:09 libceplang_en.so.4.2
-rwxrwxrwx  1 root root 547624 Jun  1 23:11 libceplex_uk.so
-rwxrwxrwx  1 root root 547624 Jun  1 23:11 libceplex_uk.so.4
-rwxr-xr-x  1 root root 547624 Jun  1 07:09 libceplex_uk.so.4.2

on test I had

lrwxrwxrwx  1 999 20202  20 Apr 24 16:17 libceplang_en.so -> 
libceplang_en.so.4.2
lrwxrwxrwx  1 999 20202  20 Apr 24 16:17 libceplang_en.so.4 -> 
libceplang_en.so.4.2

-rwxrwxr-x  1 999 20202  315933 Aug 17  2006 libceplang_en.so.4.1
-rwxrwxr-x  1 999 20202  139612 Mar 15 18:21 libceplang_en.so.4.2
lrwxrwxrwx  1 999 20202  19 Apr 24 16:17 libceplex_uk.so -> 
libceplex_uk.so.4.2
lrwxrwxrwx  1 999 20202  19 Apr 24 16:17 libceplex_uk.so.4 -> 
libceplex_uk.so.4.2

-rwxrwxr-x  1 999 20202  591033 Aug 17  2006 libceplex_uk.so.4.1
-rwxrwxr-x  1 999 20202  547624 Mar 15 18:20 libceplex_uk.so.4.2

I then removed all the non 4.2 libs and created a symbolic link to the 
4.2 libs to match test.


fired it all up, and app_swift then worked. Or so I thought. segfault 
- but not on every call.


what I would like to know is:

A) has anybody got a later version of app_swift (0.9.1)
B) does anyone else use cepstral, and how ?
C) what is the story with the cepstral libraries ?

many thanks

Julian


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Re: [asterisk-users] Needed changes in Asterisk to change the SIP port to 5062

[Jaswinder Singh]

In sip.conf it should be bindport=5062

On 06/06/07, Crazy Boy <[EMAIL PROTECTED]> wrote:

Hi Friends,
I want to use 5062 port for SIP protocol. I made the below modifications in
my server to use 5062 port.
Polycom phone: port=5062
 Trunk settings: port=5062
 sip.conf: bindaddr=5062
 Extension configuration details: 5062
Our VoIP provider told me that they are allowing the SIP traffic through
5060 to 5064. I observed on my server console that my server is registered
with our VoIP provider with 5062 port. But, I am unable to make outgoing
calls.
Do I need to modify any other settings in Asterisk?
Look forward to your response. Thank you.
Regards,
 Chandra.

 
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[asterisk-users] Phantom calls: Detecting hangup quicker

[Gavin Henry]

Dear all,

We seem to be getting phantom calls when a inbound caller via the
legacy pbx hangups before
the SIP handsets have answered. The extensions also seem to hear
ringing on the lines too sometimes.

  SIP Inbound  >
  |
legacy pbx (analogue) <-> (sangoma a400d) asterisk <-> SIP phones

Basically if a user hangups before the call has bridged, I think.

Is there anything we can do about this?

Thanks,

Gavin.
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Re: [asterisk-users] Voip-info.org

[Jaswinder Singh]

Its up and working now .

On 06/06/07, Compnet Bobby <[EMAIL PROTECTED]> wrote:

Same in southern cali!




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voip-info.org

Yep its down for me tooo .

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
>
>
>
>
> Is anyone else having trouble going into voip-info.org today?
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Re: [asterisk-users] zaptel make problem

[Tzafrir Cohen]

On Wed, Jun 06, 2007 at 12:08:48PM -0500, Malcom Kemp wrote:
> I am installing asterisk on a second box with OpenSuSE 10.2.  I have
> installed libpri, run menuselect/configure and then make.  The make
> stops at the last line shown below.  Looking at the processes, the
> current process is running sed.  Not sure from where.  Any ideas?
> 
>  
> ...
> 
> checking for ar... /usr/bin/ar
> 
> checking for cp... /bin/cp
> 
> checking for ln... /bin/ln
> 
> checking for mkdir... /bin/mkdir
> 
> checking for nroff... /usr/bin/nroff
> 
> checking for rm... /bin/rm
> 
> checking for strdup... yes
> 
> checking for vsnprintf... yes
> 
> configure: creating ./config.status
> 
> config.status: creating Makefile


Hmmm You left out the error message, which was probably something of
the sort of: "please run ./configure before make" .

Next time rune:

  ./configure
  make

This time make ran ./configure for you .

Now shouldn't autoconf generate the makefile to prevent such errors?


-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Asterisk call quality detection

[carl]

Any explaination as to what it is, does it work and how to setup?
Is the vnak found in the logs and is it only represented for iax calls?

- Original Message - 
From: "Henry Cobb" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, June 06, 2007 12:07 PM
Subject: Re: [asterisk-users] Asterisk call quality detection



On 6/6/07, carl Lougher <[EMAIL PROTECTED]> wrote:

Hi Chaps,
Is there a way to detect/highlight poor quality voice
calls going through an asterisk server?

Was thinking of picking up a cdr record or some other
variable showing poor quality on calls when the
internet is having issues.

Is there any qos or poor audio quality variables
available?


I chart VNAKs per hour.

-HJC
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Re: Re[2]: [asterisk-users] Re: Verizon Interconnection

[laurent schweizer]

I'm using racoon for the VPN.

to redirect the traffic, I'm using openser with rtpproxy.

Laurent


2007/6/6, Matt <[EMAIL PROTECTED]>:


James,
What did you use for the VPN?  Cisco/Racoon, etc?   We are inter-opting
just for DIDs and termination.

RIght now we are able to get phase-1 up, but are having an issue with
phase-2 of the IPSEC tunnel.I'm also wondering what you did to get
Asterisk to redirect the traffic from the VPN IP to Asterisk's actual
external IP.

On 6/6/07, James Coberly <[EMAIL PROTECTED]> wrote:
>
> We did the US interconnect with Verizon about a year ago to Asterisk.
> Yes it is a convoluted process,  but if you can get a day to focus on it
> only,  you can complete 95% of the interop,  then schedule the last bit
> with VZ to finish up.
> It is generally solid though once it is up.
>
> Are you inter-oping for the conference features also, as they add
> another layer of PITA processes.
>
>
> On Wed, 2007-06-06 at 16:05 +0200, Gunnar Schaller wrote:
> > Hello,
> > Switzerland is good :-)
> > My company in Switzerland has a contract with Verizon for wholesale in
> > pstn and we are thinking about SIP. Can you please tell me your
> > experience with their SIP product?
> >
> > Regards,
> > Gunnar Schaller
> >
> >
> >
> >
> > Wednesday, June 6, 2007, 2:55:54 PM, you wrote:
> >
> > > HI,
> >
> > > yes we are interconnected with Verizon in SIP, but we are in europe
> > > (Switzerland) so I don't know If it is the same process in USA ...
> >
> > > Laurent
> >
> >
> > > 2007/6/6, Matt <[EMAIL PROTECTED]>:
> > >>
> > >> So absolutely no one here was interconnected with Verizon?  I am
> going to
> > >> shoot this over to asterisk-biz, also, in hopes someone may have
> missed it
> > >> that is on the biz list.  The question again is:
> > >>
> > >> Has anyone on this list connected with Verizon's SIP product?  We
> are
> > >> currently undergoing interop testing with Verizon, and honestly, it
> seems
> > >> like the most convoluted process.   I'd be interested in talking
> with
> > >> someone else who has gone through this and run a few things past
> you.
> > >>
> > >>
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Re: [asterisk-users] Needed changes in Asterisk to change the SIP port to 5062

[~Russell]

if it is properly into Trunk, I mean into extensions.conf then it shud work
properly

On 6/6/07, Crazy Boy <[EMAIL PROTECTED]> wrote:


Hi Friends,
I want to use 5062 port for SIP protocol. I made the below modifications
in my server to use 5062 port.
Polycom phone: port=5062
Trunk settings: port=5062
sip.conf: bindaddr=5062
Extension configuration details: 5062
Our VoIP provider told me that they are allowing the SIP traffic through
5060 to 5064. I observed on my server console that my server is registered
with our VoIP provider with 5062 port. But, I am unable to make outgoing
calls.
Do I need to modify any other settings in Asterisk?
Look forward to your response. Thank you.
Regards,
Chandra.

--
Need a vacation? Get great deals to amazing places
on
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Re: [asterisk-users] PRI Partial Re-Rounting

[Matthew Fredrickson]

I think what they're talking about is forwarding the call before the 
call is established.  If I remember correctly, it's call CF[U,B,NR] for 
call forward on unavailable, busy, and no response.  Unfortunately 
though, none of the switchtypes support this variant of this function.  
However, if 2BCT is acceptable, we have a working implementation for 
DMS100 switchtype in 1.4.


Matthew Fredrickson

On Jun 6, 2007, at 9:53 AM, Eric "ManxPower" Wieling wrote:


Jon Schøpzinsky wrote:

Hello List
We are trying to redirect calls directly, instead of opening a new 
channel and dialing out.

Etc:
A calls B on our asterisk, and is directly redirected to C
We have been told that this feature should be available on a PRI 
level, and is called Partial re-routing.

Anybody has an idea of whether this is supported in Asterisk?


It is called 2BCT.  It is supported on AT&T 5ESS PRI lines.  I don't 
think it is supported on NI2 or non-AT&T switches.  I've never used 
it.


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Re: [asterisk-users] Needed changes in Asterisk to change the SIP port to 5062

[Philipp Kempgen]

Crazy Boy wrote:

>  sip.conf: bindaddr=5062

bindaddr=0.0.0.0  ; IP address to bind to (0.0.0.0 binds to all)
bindport=5062 ; UDP Port to bind to (SIP standard port is 5060)


  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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[asterisk-users] Voicemail marking messages as Old

[Adrian A]

It seems to me that simply listening to a new voicemail message will move
the message to the Old folder, without any other user interaction. I'm
working on a voicemail callback queue script and I have wrongly assumed that
messages remain in INBOX unless the user actually saves or deletes them.
I'm running an older 1.2 version of Asterisk.
Is anyone able to confirm the same behavior in newer versions? Is there a
way for Asterisk voicemail to behave like regular voicemail where a message
remains "New" until the caller does something to it (other than simply
listening to it) ?

Thanks.
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Re: [asterisk-users] Noise on FXS ports (Sangoma)

[John Novack]

Since the OP said the noise was on FXS ports, Jorge's answer isn't relevant.
After listening to  a wav file of the noise, it sounds to my old ears 
like a background hiss or so called "comfort" noise, except for a couple 
of short pops which I assume to be an open microphone.


Assuming this isn't generated by the POTS set connected to the port, I 
guess you'll have to dig into the configuration.
Some POTS sets generate a similar though more erratic noise, caused by 
carbon transmitters and defective varistors in the network, but that is 
ancient history.
More modern POTS sets with electret transmitters could certainly 
generate a similar noise.
I am assuming that several different sets have been tried with similar 
results.


My experience with the Sangoma A200, and in fact Digium 400 and Adtran 
channel bank FXS circuits all are very quiet, so this is a bit of a puzzle.


John Novack


Jorge Mendoza wrote:

Stephen Bosch wrote:

Hi, Jorge:

Jorge Mendoza wrote:
 


Never experienced with FXS modules on a PC with  Asterisk. However we
have experienced that kind of problems on legacy PBX without a good
ground. If you replace the system with a analogue set and have not
noise, then a ground current is generated in your system, probably
originated at FXO side. Have you tested the PC isolated, with not lines
and not switches? just the FXS calling the voicemail?



No, I haven't gone that far yet, but it might be worth trying.

One question I have: if this turned out to be the cause, what could I do
to clean up the ground? There are so many elements -- the power supply
ground, the telephone lines, the network cable ground, etc.

-Stephen-
  

Stephen,

Good question. Finding ground problems is an art.
First thing I should do: measuring your ground with respect of CO 
ground. With a voltmeter between the tip wire of your CO line (0 VDC ) 
and your local ground, voltage should be less than 5 VDC.


Jorge
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Re: [asterisk-users] Stanaphone/Asterisk issue: No Audio with SIP to only one provider when switching servers

[Alejandro Lengua]

Hello,
did you got your issue solved?
I am suffering of the same issue

On 4/28/07, Hadar Pedhazur <[EMAIL PROTECTED]> wrote:


I snipped all of the previous data, as I'm trying to "boil down"
this problem to its essence...

I turned off the firewall for a few seconds, and still got no
audio. For those that will be suspicious, the commands were:

shorewall stop
shorewall clear

tested connection, no audio

shorewall start

I also have a SIPPhone number, which (obviously), connects via
SIP. I called that number from the outside, using one of their
"Access Numbers", and my phone rang and I heard audio in both
directions (this with the firewall back on), so SIP definitely
works, just not with StanaPhone.

Then I connected from another server that I run, which is behind a
NAT router. That server is running 1.2.18 (as is the one that
isn't working, but is on a public IP). Audio works perfectly with
this one.

To my knowledge the only difference between them is that the two
servers that work are both Red Hat 9, with Asterisk 1.2.18 built
from source. The one that fails is CentOS 5.0, with Asterisk
1.2.18 built from source. Here is a dump of the active channel
from the NAT'ed server, which _works_:

   * SIP Call
   Direction:  Incoming
   Call-ID:
[EMAIL PROTECTED]
   Our Codec Capability:   1822
   Non-Codec Capability:   1
   Their Codec Capability:   262
   Joint Codec Capability:   262
   Format  ulaw
   Theoretical Address:204.147.183.18:5060
   Received Address:   204.147.183.18:5060
   NAT Support:RFC3581
   Audio IP:   XX.XX.XX.XX (local)
   Our Tag:as78cfb201
   Their Tag:  da6aae9eb017f29b6c9de270fb85c352
   SIP User agent: Sippy
   Original uri:   sip:204.147.183.55:1024
   Caller-ID:  XX
   Need Destroy:   0
   Last Message:   Rx: ACK
   Promiscuous Redir:  No
   Route:
sip:204.147.183.18;ftag=da6aae9eb017f29b6c9de270fb85c352;lr=on
   DTMF Mode:  rfc2833
   SIP Options:(none)

The only things edited above are the Audio IP, which is my correct
"local" (before NAT) server address, and my Caller-ID. Everything
else is unchanged.

Here is the channel with dead audio:

   * SIP Call
   Direction:  Incoming
   Call-ID:
[EMAIL PROTECTED]
   Our Codec Capability:   1542
   Non-Codec Capability:   1
   Their Codec Capability:   262
   Joint Codec Capability:   6
   Format  ulaw
   Theoretical Address:204.147.183.18:5060
   Received Address:   204.147.183.18:5060
   NAT Support:RFC3581
   Audio IP:   XX.XX.XX.XX (local)
   Our Tag:as45dbcfef
   Their Tag:  420bab62c5da9eae42686897ae65a385
   SIP User agent: Sippy
   Original uri:   sip:204.147.183.55:1024
   Caller-ID:  XX
   Need Destroy:   0
   Last Message:   Rx: ACK
   Promiscuous Redir:  No
   Route:
sip:204.147.183.18;ftag=420bab62c5da9eae42686897ae65a385;lr=on
   DTMF Mode:  rfc2833
   SIP Options:(none)


The same two fields are edited above, and both were correct.

To my eye, these are identical. Both are selecting ulaw,
correctly. I'm stumped. I guess that I didn't do any packet
tracing, but I'm not sure what the value of that would be given
that it's not a firewall problem...

Suggestions welcome!
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[asterisk-users] Scaling Asterisk: High volume benchmarks (0 to 450 calls)

[Matthew J. Roth]

List users,

This post contains the benchmarks for Asterisk at high call volumes on a 
4 CPU, dual-core (8 cores total) server.  It's a continuation of the 
posts titled "Scaling Asterisk: Dual-Core CPUs not yielding gains at 
high call volumes".  They contain a fair amount of information, 
including details about our servers and the software on them.  I'm happy 
to answer any questions you might have, but please take a moment to 
review those posts to make sure they don't contain the information 
you're seeking.


Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer


Conclusions
---
Once again, I'm presenting the conclusions first. Scroll down if you're 
more interested in the raw data.


 1. Asterisk scales quite well up to a certain number of calls.  At 
this point, the cost in CPU cycles per call starts to increase more 
drastically.  A graph of the Avg Used% values can be used to demonstrate 
this.  It can be described as consisting of two roughly linear 
segments.  The first segment is from 0 to 110 calls.  The rest of the 
graph is a second, steeper segment.  This is not entirely true, as in 
fact each new call costs a little more than the last, but it is a useful 
simplification.
 2. Even at very high call volumes, Asterisk uses less than 512 KB of 
memory.  2 GB of RAM would probably avoid swapping and excessive disk 
activity on most Asterisk installations.
 3. Future benchmarks should be based on the number of active channels, 
not active calls.


I'm relying on you to point out my mistakes and omissions, so please 
take a look at the data and respond with your own analysis and conclusions.


Benchmarking Methodology

The benchmarks are based on data I collected over the period of 
5/12/2007 to 05/30/2007 from two production servers used in our inbound 
call center.  The servers are identical 8-core Dell PowerEdge 6850s as 
documented in my prior posts.  They are meant to be used as a 
primary/backup pair, but both were used in production in order to rule 
out a hardware failure as the cause of our scaling issues.


The data was collected by a bash script executed from cron every 2 
minutes.  This script utilizes some basic Linux tools (such as sar, 
free, df, and the proc filesystem) to record information about the 
system, and 'asterisk -rx "show channels"' to record information about 
the number of active calls and channels within Asterisk.


Unfortunately, the sample sizes this produced are relatively small for 
the 300-450 call range.  This is due to two factors:


 1. The majority of the time we don't operate at such high call volumes.
 2. Asterisk intermittently fails to report call and channel statistics 
when the CPU idle is low.


This means that the benchmarking results are somewhat erratic for the 
300-450 call range.  The good news is that they are pretty consistent 
for 0 to 300 calls, and I'd imagine that covers the range most people 
are interested in.


Keep in mind that the impetus behind this benchmarking was the lack of a 
performance boost on the dual-core server at high call volumes, so the 
high call range may also be skewed by whatever bottleneck is being hit 
on the 8-core servers.  In the near future, we will be adding one of our 
4-core PowerEdge 6850s to our production environment.  I'll collect and 
analyze the same data, which I believe will show similar performance (as 
defined by cumulative idle CPU percentage) at around 200-300 calls.


In the end, I hope to understand this problem well enough to overcome it 
or determine what the optimal point is for achieving the highest call 
volume without over-dimensioning the hardware.


Call Types and a Note on Channels
-
All of the calls are SIP-to-SIP using the uLaw codec.  The vast majority 
of the calls are either in queue or connected to an agent, but there are 
also a small number of regular outbound calls and transfers.  Every call 
that is connected to an agent is recorded via the Monitor() application 
in PCM format to a RAM disk.  In short, there was no transcoding, 
protocol bridging, or TDM hardware involved on the servers being 
benchmarked.


At any given time, the makeup of the calls varied (ie. calls in queue 
vs. calls connected to agents).  The calls connected to agents involve 
bridging two SIP channels, so they are more resource intensive.  This 
means that the number of active channels is probably a better base 
benchmarking unit than the number of active calls.  Fortunately, the 
ratio of calls to channels is somewhat consistent so this round of 
benchmarking still produced useful results.


Future Tests

I'm aware that using a live environment isn't ideal for testing.  I have 
some ideas for setting up more controlled tests using SIPp, VICIDIAL, or 
call files.  I think I have the necessary hardware, but I haven't had 
the time to do much research, let alone implement anything.


I

Re: [asterisk-users] Asterisk call quality detection

[Jared Smith]

On 6/6/07, Mike Lynchfield <[EMAIL PROTECTED]> wrote:

also note vnaks are iax i think


For SIP traffic, you might want to look into RTCP (realtime transport
control protocol).  If you're getting RTCP reports, they'll help you
identify problems with the RTP such as jitter and lost packets.

Hopefully in the future we'll have the RTCP reports logged (either as
part of the CDR records, or in a Call Quality log of some kind).
Until then, I'm pretty sure you can listen for RTCP events through the
Asterisk Manager Interface, and log them yourself.

-Jared
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Re: [asterisk-users] TCP<->UDP SIP proxy?

[Jaswinder Singh]

I think there is a patch for sip over tcp in asterisk but not sure if
its stable or not

try this http://bugs.digium.com/view.php?id=4903

You can also install openser as sip proxy . it supports sip over tcp .

On Wed,  6 Jun 2007 19:14 +0300, Yehavi Bourvine +972-8-9489444
<[EMAIL PROTECTED]> wrote:

Hello,

   One of our faculties have Microsoft's LCS and would like to connect it to
our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS
talks SIP over TCP with TLS. Anyone can recommend a gateway between these two
protocols?

  Thanks! __Yehavi:
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[asterisk-users] Record CDR in a Oracle database

[Everton Goularth]

Hello All,

How can I do to record my asterisk's CDR in a Oracle database?

I have to use unixODBC?

Can anybody send me a step to step to do this configuration?

Thank's All

Everton Goularth




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Re: [asterisk-users] blades?

[Jon Pounder]

Quoting Mike Lynchfield <[EMAIL PROTECTED]>:

does the piggyback actually require the type T chassis or can it go in  
the type H chassis ? (I have never been able to see a picture of the  
pci module so don't know where the card backplanes are - are they out  
the front of the blade for external connections ?) you wouldn't by  
chance have a photo of one would you ?


Also - we normally use the scsi piggybacks on our blades, can you have  
a pci and a scsi on there at the same time or is it just one or the  
other ?


the pci module docs talk about internal redundancy vaguely - is the  
pci module actually connected to anything internally on the midplane  
besides power or just the blade itself and however the pci card  
backplanes are exposed ??







yes we have several of these...

you need to use a PCI expensiont slot.. its an addon that piggy backs to a
blade and takes 1 u ..so total blade will take 2 u's...

but you can hook 2 PCIS on it.. sangoma or whatever...

This way we can Redundantly failover 2 PRIS on each other with each blade
have 2 cards A102D's that or cross linked to each seperate PRI Circuit..
So Blade A has 2 A102d's and so to B
Each 102's has 2 ports..
A
1a BTN1
1b TF1
2a BTN2
2b TF2

B
1a BTN1
1b TF1
2a BTN2
2b TF2
---
! A  1a !
! 1b !
! 2a !
! 2b !
---
---
! B  1a !
! 1b !
! 2a !
! 2b !


then you put a T1 switch module.. then its all automatic ..


PRI DROP X ( Locals) failvoer on carrier side to Y

PRI DROP Y (toll frees) failover on carrier dide to X


On 6/6/07, Jon Pounder <[EMAIL PROTECTED]> wrote:


Quoting Dean Collins <[EMAIL PROTECTED]>:


http://www.theregister.co.uk/2007/06/06/sun_thinner_blades/



this article got me thinking - is anyone running asterisk on blade
servers?




We have a bunch of ibm blades, but the issue at least with the H
series cabinets we have is that there is no where to put any pci cards
of any sort so you would be limited to purely a voip setup.

There is a T series cabinet that allows pci cards for just such
purposes as asterisk (T is telephony), but the information out there
about just what pieces you need is pretty vague. Anyone have a no bs
description of how the bits actually work together in that setup ?






Any lessons for us to learn?







Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).










Jon Pounder

   _/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
_/_/_/  _/  _/ _/_/_/  _/  _/_/
   _/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


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Re: [asterisk-users] background dialing

[Rizwan Hisham]

in that case you can do this:

exten=> 1,1,Dial(SIP/123,15,m(default))
exten=> 1,2,Gotoif($["${DIALSTATUS}"="ANSWER"]?6:)
exten=> 1,3,Dial(SIP/456,15,m(default))
exten=> 1,4,Gotoif($["${DIALSTATUS}"="ANSWER"]?6:)
exten=> 1,5,Dial(SIP/789,15,m(default))
exten=> 1,6,Hangup

Caller might notice a little bit of delay between 2 dial commands. i have
checked this dialplan. but hopefully this will work for u.

On 6/5/07, Thomas Stein <[EMAIL PROTECTED]> wrote:


On Tuesday 05 June 2007, Rizwan Hisham wrote:
> i didnt see you were alreadyusing the m option. sorry about that. you
just
> need to make a new MOH class which play the msg instead of music. you
dont
> have to use the local dial option.
>
> On 6/5/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote:
> > You can use the small 'm' option in the dial command like this:
> >
> > exten=>1,1,Dial(SIP/123&SIP/456&SIP/789,,m(default))   ;default is the
> > MOH class name
> > This will play music on hold. To play specific msg instead of MOH, you
> > can create a different MOH class where you can specify to play that
> > specif msg instead of music on hld.

Your right but with your example SIP/123&SIP/456&SIP/789 are ringing at
the
same time. Thats not what i want. I was hoping there is a possibilty to
call
the phones one after another while playing a music oder soundfile.

regards
t.
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Re: [asterisk-users] TCP<->UDP SIP proxy?

[Kristian Kielhofner]

On Wed,  6 Jun 2007 19:14 +0300, Yehavi Bourvine +972-8-9489444
<[EMAIL PROTECTED]> wrote:

Hello,

   One of our faculties have Microsoft's LCS and would like to connect it to
our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS
talks SIP over TCP with TLS. Anyone can recommend a gateway between these two
protocols?

  Thanks! __Yehavi:


SER/OpenSER can relay between TCP and UDP no problem!


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Re: [asterisk-users] TCP<->UDP SIP proxy?

[Remco Post]

Yehavi Bourvine +972-8-9489444 wrote:
> Hello,
> 
>One of our faculties have Microsoft's LCS and would like to connect it to
> our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS
> talks SIP over TCP with TLS. Anyone can recommend a gateway between these two
> protocols?
> 

(open)ser

>   Thanks! __Yehavi:
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Remco Post

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makes sense." -- Glen Hattrup
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Re: [asterisk-users] Verizon Interconnection

[Jared Smith]

On 6/5/07, Matt <[EMAIL PROTECTED]> wrote:

Has anyone on this list connected with Verizon's SIP product?


Which SIP product.  I know of at least two SIP products that Verizon
offers (2-way traffic and Carrier IP Termination), and there are
probably more.  I helped a client of mine go through the interop
testing on both of the products listed above.  I also know that a
friend of mine has done the same thing.


We are currently undergoing interop testing with Verizon, and honestly, it seems
like the most convoluted process.   I'd be interested in talking with
someone else who has gone through this and run a few things past you.


Yes, it's very in-depth testing, which isn't a bad thing to do with
your upstream carrier.  If you understand SIP and Asterisk, it isn't
terribly difficult to do.  Mostly you just need to make sure you
understand what they want in the P-Assert-Identity header and make
sure your phone numbers are formatted correctly (sometimes E.164,
other times not).

What specific questions did you have?  I can't guarantee I remember
the answers, but go ahead and throw out the questions.

-Jared
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Re: [asterisk-users] TCP<->UDP SIP proxy?

[Alex Robar]

SIP Express Router (SER -  http://www.iptel.org/ser/) is fairly common
solution for this problem.

AR

On Wed, 6 Jun 2007 19:14 +0300, Yehavi Bourvine +972-8-9489444 <
[EMAIL PROTECTED]> wrote:


Hello,

   One of our faculties have Microsoft's LCS and would like to connect it
to
our Asterisk system. the problem is that Asterisk talks SIP over UDP while
LCS
talks SIP over TCP with TLS. Anyone can recommend a gateway between these
two
protocols?

  Thanks! __Yehavi:
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[asterisk-users] Queue Job

[Jason Adams]

We have a job that requires extensive knowledge of asterisk queues.  The
work can be done remotely.  Our customer is looking to completely
overhaul their current queue structure.  Please contact me offlist if
you are interested or need more details.
 
 - Jason  
 
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Re: [asterisk-users] TCP<->UDP SIP proxy?

[Victor Toofic]

El Wed, Jun 06 de 2007 a las 19:14 +0300, Yehavi Bourvine +972-8-9489444 
comentaba:
> Hello,
> 
>One of our faculties have Microsoft's LCS and would like to connect it to
> our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS
> talks SIP over TCP with TLS. Anyone can recommend a gateway between these two
> protocols?

OpenSER?

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[asterisk-users] meetme realtime

[ram]

Hi

iam using 1.2.17

does any one have information meetme in realtime
and store in mysql i dont see any document

could some one help me

is this possible ?

ram
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RE: [asterisk-users] PRI Partial Re-Rounting

[John Treble]

Jon,

Google "TBCT + Asterisk".


John Treble
Ottawa, Canada


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric "ManxPower" Wieling
> Sent: June 6, 2007 10:53 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] PRI Partial Re-Rounting
> 
> Jon Schøpzinsky wrote:
> > Hello List
> >
> > We are trying to redirect calls directly, instead of opening a new
> channel and dialing out.
> > Etc:
> >
> > A calls B on our asterisk, and is directly redirected to C
> >
> >
> > We have been told that this feature should be available on a PRI level,
> and is called Partial re-routing.
> >
> > Anybody has an idea of whether this is supported in Asterisk?
> 
> It is called 2BCT.  It is supported on AT&T 5ESS PRI lines.  I don't
> think it is supported on NI2 or non-AT&T switches.  I've never used it.
> 
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RE: [asterisk-users] Voip-info.org

[Compnet Bobby]

Same in southern cali!




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voip-info.org

Yep its down for me tooo .

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
>
>
>
>
> Is anyone else having trouble going into voip-info.org today?
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Re: [asterisk-users] cepstral TTS and app_swift

[Mike Lynchfield]

what versions of asterisk on both systems ?



On 6/5/07, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote:


Have you tried something along the lines of:

System("swift blah blah blah -o blah.wav")
Playback("blah.wav")

It does have an inherent delay for the generation step but maybe swift
binary segfaults less?  I've only used cepstral via swift binary, and it
has never segfaulted for me. My swift and voice are version 4.2.0.

I doubt different voices behave differently, but just in case, I use the
$7 Damien voice.

Moj

Julian Lyndon-Smith wrote:
> We are having some major problems with app_swift since we went live. It
> is regularly segfaulting.
>
> I don't know if this is my fault or not, but here's the story:
>
> Installed the cepstral voices (at the time, 4.0) on our test system
> (2.6.9-42.0.10.ELsmp)
> and later added some extra voices (now 4.2). All worked fine - we stress
> tested (20+ simultaneous calls).
>
> Move to live ( 2.6.9-22.0.1.ELsmp) . Installed the cepstral voices (only
> 4.2).
>
> Started having problems with only 5 calls: swift by itself on the
> command line was fine (it worked) but app_swift complained that it
> couldn't find any voices.
>
> Looking into /opt/swift/lib, I saw that it was different to my test
system.
>
> On live I had (snipped)
>
> -rwxrwxrwx  1 root root 139612 Jun  1 23:10 libceplang_en.so
> =rwxrwxrwx  1 root root 139612 Jun  1 23:11 libceplang_en.so.4
> -rwxr-xr-x  1 root root 139612 Jun  1 07:09 libceplang_en.so.4.2
> -rwxrwxrwx  1 root root 547624 Jun  1 23:11 libceplex_uk.so
> -rwxrwxrwx  1 root root 547624 Jun  1 23:11 libceplex_uk.so.4
> -rwxr-xr-x  1 root root 547624 Jun  1 07:09 libceplex_uk.so.4.2
>
> on test I had
>
> lrwxrwxrwx  1 999 20202  20 Apr 24 16:17 libceplang_en.so ->
> libceplang_en.so.4.2
> lrwxrwxrwx  1 999 20202  20 Apr 24 16:17 libceplang_en.so.4 ->
> libceplang_en.so.4.2
> -rwxrwxr-x  1 999 20202  315933 Aug 17  2006 libceplang_en.so.4.1
> -rwxrwxr-x  1 999 20202  139612 Mar 15 18:21 libceplang_en.so.4.2
> lrwxrwxrwx  1 999 20202  19 Apr 24 16:17 libceplex_uk.so ->
> libceplex_uk.so.4.2
> lrwxrwxrwx  1 999 20202  19 Apr 24 16:17 libceplex_uk.so.4 ->
> libceplex_uk.so.4.2
> -rwxrwxr-x  1 999 20202  591033 Aug 17  2006 libceplex_uk.so.4.1
> -rwxrwxr-x  1 999 20202  547624 Mar 15 18:20 libceplex_uk.so.4.2
>
> I then removed all the non 4.2 libs and created a symbolic link to the
> 4.2 libs to match test.
>
> fired it all up, and app_swift then worked. Or so I thought. segfault -
> but not on every call.
>
> what I would like to know is:
>
> A) has anybody got a later version of app_swift (0.9.1)
> B) does anyone else use cepstral, and how ?
> C) what is the story with the cepstral libraries ?
>
> many thanks
>
> Julian
>
>
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Re: Re[2]: [asterisk-users] Re: Verizon Interconnection

[Matt]

James,
What did you use for the VPN?  Cisco/Racoon, etc?   We are inter-opting just
for DIDs and termination.

RIght now we are able to get phase-1 up, but are having an issue with
phase-2 of the IPSEC tunnel.I'm also wondering what you did to get
Asterisk to redirect the traffic from the VPN IP to Asterisk's actual
external IP.

On 6/6/07, James Coberly <[EMAIL PROTECTED]> wrote:


We did the US interconnect with Verizon about a year ago to Asterisk.
Yes it is a convoluted process,  but if you can get a day to focus on it
only,  you can complete 95% of the interop,  then schedule the last bit
with VZ to finish up.
It is generally solid though once it is up.

Are you inter-oping for the conference features also, as they add
another layer of PITA processes.


On Wed, 2007-06-06 at 16:05 +0200, Gunnar Schaller wrote:
> Hello,
> Switzerland is good :-)
> My company in Switzerland has a contract with Verizon for wholesale in
> pstn and we are thinking about SIP. Can you please tell me your
> experience with their SIP product?
>
> Regards,
> Gunnar Schaller
>
>
>
>
> Wednesday, June 6, 2007, 2:55:54 PM, you wrote:
>
> > HI,
>
> > yes we are interconnected with Verizon in SIP, but we are in europe
> > (Switzerland) so I don't know If it is the same process in USA ...
>
> > Laurent
>
>
> > 2007/6/6, Matt <[EMAIL PROTECTED]>:
> >>
> >> So absolutely no one here was interconnected with Verizon?  I am
going to
> >> shoot this over to asterisk-biz, also, in hopes someone may have
missed it
> >> that is on the biz list.  The question again is:
> >>
> >> Has anyone on this list connected with Verizon's SIP product?  We are
> >> currently undergoing interop testing with Verizon, and honestly, it
seems
> >> like the most convoluted process.   I'd be interested in talking with
> >> someone else who has gone through this and run a few things past you.
> >>
> >>
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> >>
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Re: [asterisk-users] Asterisk call quality detection

[Willy Wouters]

Henry,

I have never seen VNAK in any of the asterisk logs.  What version of 
asterisk are you running?


Henry Cobb wrote:
grep VNAK /var/log/asterisk/full | cut -d ' ' -f 4 | cut -d : -f 1 | 
uniq -c


Needs a bit of an adjustment between the 1-9th and 10th-31st of the
month so I'm looking for something to chomp this automatically.


--
Willy Wouters, PhD
Asterisk Telephony
Web Applications
MAGU ENTERPRISES
Tel: 713 474-1534
Fax: 501 665-1544

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[asterisk-users] zaptel make problem

[Malcom Kemp]

I am installing asterisk on a second box with OpenSuSE 10.2.  I have
installed libpri, run menuselect/configure and then make.  The make
stops at the last line shown below.  Looking at the processes, the
current process is running sed.  Not sure from where.  Any ideas?

 
...

checking for ar... /usr/bin/ar

checking for cp... /bin/cp

checking for ln... /bin/ln

checking for mkdir... /bin/mkdir

checking for nroff... /usr/bin/nroff

checking for rm... /bin/rm

checking for strdup... yes

checking for vsnprintf... yes

configure: creating ./config.status

config.status: creating Makefile



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Re: [asterisk-users] Re: Verizon Interconnection

[laurent schweizer]

no you can have swiss number.

they also can do number portability from swiss number.

SIP proxy are not in switzerland but in belgium and england

Laurent


2007/6/6, Matt <[EMAIL PROTECTED]>:


Verizon has phone service in Switzerland?  Or are you getting US numbers?

On 6/6/07, laurent schweizer < [EMAIL PROTECTED]> wrote:
>
> HI,
>
> yes we are interconnected with Verizon in SIP, but we are in europe
> (Switzerland) so I don't know If it is the same process in USA ...
>
> Laurent
>
>
> 2007/6/6, Matt <[EMAIL PROTECTED]>:
> >
> > So absolutely no one here was interconnected with Verizon?  I am going
> > to shoot this over to asterisk-biz, also, in hopes someone may have missed
> > it that is on the biz list.  The question again is:
> >
> > Has anyone on this list connected with Verizon's SIP product?  We are
> > currently undergoing interop testing with Verizon, and honestly, it seems
> > like the most convoluted process.   I'd be interested in talking with
> > someone else who has gone through this and run a few things past you.
> >
> >
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Re: [asterisk-users] Voip-info.org

[Guillermo Salas M.]

On Wed, 2007-06-06 at 11:21 -0400, Justin Moore wrote:
> On 6/6/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
> > Is anyone else having trouble going into voip-info.org today?
> 
> Yep. Dead for me too.
> 

Dead from Ecuador too.

Regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
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[asterisk-users] Needed changes in Asterisk to change the SIP port to 5062

[Crazy Boy]

Hi Friends,
 I want to use 5062 port for SIP protocol. I made the below modifications in my 
server to use 5062 port.
 Polycom phone: port=5062
 Trunk settings: port=5062
 sip.conf: bindaddr=5062
 Extension configuration details: 5062
 Our VoIP provider told me that they are allowing the SIP traffic through 5060 
to 5064. I observed on my server console that my server is registered with our 
VoIP provider with 5062 port. But, I am unable to make outgoing calls.
 Do I need to modify any other settings in Asterisk?
 Look forward to your response. Thank you.
 Regards,
 Chandra.

   
-
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RE: [asterisk-users] Noise on FXS ports (Sangoma)

[shadowym]

All I know is that your rx gain, at 0db is probably way too low.  My tx gain
is typically set to -3db.  Yes, I was thinking sip phones when I made the
amplifier comment.  If you hear the noise as soon as you take the analog
phone off hook without dialing the PSTN then I don't see how it could have
anything to do with the telco card.  Is it a feedback type noise or
background hiss?  Feedback type buzz would lead me to believe it's ground
loop as someone else mentioned.  Ground loop hum is a PITA to try deal with.

Yes, ztmonitor works with Sangoma cards.  It's fxotune that doesn't work and
is not needed on Sangoma cards.

-Original Message-
From: Stephen Bosch [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, June 05, 2007 11:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Noise on FXS ports (Sangoma)

shadowym wrote:
>  
> Every Sangoma A200 card I have ever connected to the PSTN required a 
> rx gain of at least 10.  Yours is commented out which I believe would 
> make it default to 0?

The noise is present on FXS ports only and is audible the moment the
receiver is lifted.

I mention this because you say "Every Sangoma A200 card I have ever
connected to the PSTN". The affected ports aren't connected to the PSTN.

> I am guessing that because the rx gain is so low the users are 
> cranking up the phone volume all the way and maybe your hearing 
> amplifier background noise??

Well, these are simple, analog sets. There's no amplifier or gain control on
the set. They have the traditional varistor for amplitude balancing. If what
you say is true, the varistor might be compensating so aggressively that
it's making the noise audible.

This explanation seems inadequate, though, as I don't hear this sound on any
of the SIP phones; only the analog phones connected to FXS ports.

A Sangoma engineer told me to increase the txgain, but as I mentioned, that
didn't help.

> Your should run ztmonitor and adjust your gains.

I didn't know it was appropriate to use ztmonitor with Sangoma hardware.

-Stephen-



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Re: [asterisk-users] Asterisk call quality detection

[Mike Lynchfield]

also note vnaks are iax i think

On 6/6/07, Henry Cobb <[EMAIL PROTECTED]> wrote:


On 6/6/07, Matt <[EMAIL PROTECTED]> wrote:
> > I chart VNAKs per hour.
>
> Would you care to share how you accomplish this?   What programs do you
use?

grep VNAK /var/log/asterisk/full | cut -d ' ' -f 4 | cut -d : -f 1 | uniq
-c

Needs a bit of an adjustment between the 1-9th and 10th-31st of the
month so I'm looking for something to chomp this automatically.

-HJC
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--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Re: [asterisk-users] Voip-info.org

[Mojo with Horan & Company, LLC]

I guess the problem with these mirrors is you can't see the mirror links 
unless you can GET to voip-info.org so here's mine:


http://voip-info.sitkavoip.com/

And the mirrored list of mirrors, to choose one closer:

http://voip-info.sitkavoip.com/wiki/view/Voip-Info+Mirrors.html

Moj

Ed Nuñez wrote:

Is anyone else having trouble going into voip-info.org today?




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Re: [asterisk-users] Voip-info.org

[Mike Lynchfield]

wget -q -O - --connect-timeout=5 http://www.voip-info.org |grep '149461'

gives me the string..

its up for now.. could of been just rebooted.


On 6/6/07, Roger Schreiter <[EMAIL PROTECTED]> wrote:


Ed Nuñez schrieb:
> Is anyone else having trouble going into voip-info.org today?

Yes. Me.


Roger.





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Re: [asterisk-users] Dial plan inquiry using GotoIf()

[Mike Lynchfield]

yes on home pbx i love the s/CALLERID..

maybe you should

f($[${CALLERID(number)} = "15552221313"]?15:5)

try to isolate string to strings.

this is not good i think

you need qhotes on the callerid part too if you evaluate to the "1555xxx"

f($["${CALLERID(number)}" = "15552221313"]?15:5)

maybe im wrong need another cofee

On 6/6/07, Steve Murphy <[EMAIL PROTECTED]> wrote:


On Wed, 2007-05-30 at 20:05 -0400, Steve Finkelstein wrote:
> Thanks for the help on this thread all.
>
> It would make sense if I write an AGI and incorporate a DB backend to
> check against numbers I want explicitly dropped. If anyone has such a
> utility, I'd love to -not- reinvent the wheel. Otherwise, I'll go whip
> it up and probably provide a web frontend for adding/removing numbers.
>

You can still use the dialplan with the DB func to check incoming
CID info. Also, the Dial() app has several options for call screening
and
privacy; these would be performed when dialing your extension.

You can have Dial keep a DB of callers, and remember whether to always
just patch them right thru, play them a polite "go away and don't come
back",
or send them off to torture scripts, or just route them straight to VM.
And, Dial() will ask you what you want to do, on the first call. Read
thru the Dial doc you get with "core show application dial". There's
an option to store an intro from each caller, where it records in  a
sound file, who they say they are. I have several hundreds of these, and
play them as the
call comes in, so we know who's calling without having to run to a CID
display.
For those who have poor to no vision, this can be a cool feature.

murf


> - sf
>
> C F wrote:
> > It fails because the right function is ${CALLERID(num)}
> >
> > On 5/30/07, Steve Finkelstein <[EMAIL PROTECTED]> wrote:
> >> Hi all,
> >>
> >> I'm looking for some rudimentary insight on GotoIf() which seems to
be
> >> failing on me in my dial plan. All I basically wish to do is block a
> >> particular caller. Sounds easy enough, but my ternary operator/plan
> >> currently is not properly being implemented. Can anyone spot where
I'm
> >> being a momo?
> >>
> >> All extensions get forwarded to the following macro:
> >>
> >> [macro-forward]
> >> ; arg1 = phone number
> >> ; arg2 = timeout
> >> ; arg3 = extension (voicemail)
> >> ; arg4 = mobile number
> >> exten => s,1,Zapateller(answer|nocallerid)
> >> exten => s,2,PrivacyManager
> >> exten => s,3,Wait(1)
> >> exten => s,4,GotoIf($[${CALLERID(number)} = "15552221313"]?15:5)
> >> exten => s,5,GotoIf($[${LEN(${CALLERID(number)})} = 4]?6:8)
> >> exten => s,6,AGI(didextlookup.agi|${CALLERID(number)})
> >> exten => s,7,Set(CALLERID(number)=${didlookup})
> >> exten => s,8,GotoIf($[${LEN(${CALLERID(number)})} = 10]?9:10)
> >> exten => s,9,Set(CALLERID(number)=1${CALLERID(number)})
> >> exten => s,10,Dial(${ARG1},${ARG2})
> >> exten => s,11,GotoIf($[${EXISTS(${ARG4})}]?11:12)
> >> exten => s,12,Dial(${ARG4},${ARG2})
> >> exten => s,13,Voicemail(u${ARG3})
> >> exten => s,14,Playback(vm-goodbye)
> >> exten => s,15,HangUp
> >> exten => s,105,HangUp
> >>
> >> As you can tell, exten => s,4,GotoIf($[${CALLERID(number)} =
> >> "15552221313"]?15:5)  is what I recently added.
> >>
> >> Here's what I see in the CLI logs:
> >>
> >> -- Executing [EMAIL PROTECTED]:1] Macro("IAX2/lime-3",
> >> "forward|SIP/5609|15|5609|IAX2/[EMAIL PROTECTED]/15164766201") in new
stack
> >> -- Executing [EMAIL PROTECTED]:1] Zapateller("IAX2/lime-3",
> >> "answer|nocallerid") in new stack
> >> -- Executing [EMAIL PROTECTED]:2] PrivacyManager("IAX2/lime-3",
"")
> >> in new stack
> >> -- CallerID Present: Skipping
> >> -- Executing [EMAIL PROTECTED]:3] Wait("IAX2/lime-3", "1") in new
> >> stack
> >> -- Executing [EMAIL PROTECTED]:4] GotoIf("IAX2/lime-3", "0?15:5")
in
> >> new stack
> >> -- Goto (macro-forward,s,5)
> >>
> >> It evaluates to false, hence goes to s,5. I keep dialing from that
> >> particular number (the one in the example is clearly masked as a
false
> >> CID), and verified it's showing up as that number on callerID.
> >>
> >> Also one last question. Say I need to add more numbers to block in
the
> >> future, is there an easier way to do this than renumbering my entire
> >> macro? Renumbering everything is just begging for a typo which can
> >> effectively render my dial plan broken.
> >>
> >> Thank you kindly, everyone!
> >>
> >> - sf
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> >> To UNSUBSCRIBE or update options visit:
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> >>
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> >
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Re: [asterisk-users] shorting flash time

[Mojo with Horan & Company, LLC]

I don't think OP meant "the phone generates a flash even though the 
phone wasn't picked up" I think they meant "the flash generated when the 
phone's flash feature is activated is not picked up by asterisk"


am I right?

I believe the setting is in zapata.conf, in the section pertinent to the 
FXS port the phone is hooked up to.  Here's the voip-info info :)


http://voip-info.sitkavoip.com/wiki/view/Asterisk+config+zapata.html#TimingParameters

Mojo

Eric "ManxPower" Wieling wrote:

Steve Kennedy wrote:

Is there anyway to change the "flash" time on a TDM400 phone port (a
user has a phone that seems to generate a short flash which isn't being
picked up).


I suspect the phone us going off hook every once in a while to check if 
there is a stutter dialtone.  If there is, it can light it's message 
waiting light.


Don't you love the analog world?
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Re: [asterisk-users] Outlook dialing

[Mike Lynchfield]

http://outcall.sourceforge.net/

we use outcall

and modded the source directly for our apps.. 0$ fee.. 100% flexibility..
Works like a charm !



On 6/6/07, Martin Smith <[EMAIL PROTECTED]> wrote:


We've been using SIPTAPI and love it for our call center. We originally
used ASTTAPI, but liked the idea of not running AstManProxy.

http://siptapi.sourceforge.net/ - website for SIPTAPI

http://projects.bebr.ufl.edu/wiki/AsteriskTAPI - our external
documentation, for the outside world in case it helps :)


Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221






From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent: Tuesday, June 05, 2007 11:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Outlook dialing



The bar is getting raised yet again
http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack
et8+Virtual+Office.aspx



I personally use Snapanumber $30 or there abouts (after trialing
a few other TAPI solutions and finding them sub-par) and think it's a
great product but interesting to see how more people are expecting
desktop/phone integration applications.



Does anyone else have a favorite Outlook autodial application
they use and love?









Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).





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Re: [asterisk-users] blades?

[Mike Lynchfield]

yes we have several of these...

you need to use a PCI expensiont slot.. its an addon that piggy backs to a
blade and takes 1 u ..so total blade will take 2 u's...

but you can hook 2 PCIS on it.. sangoma or whatever...

This way we can Redundantly failover 2 PRIS on each other with each blade
have 2 cards A102D's that or cross linked to each seperate PRI Circuit..
So Blade A has 2 A102d's and so to B
Each 102's has 2 ports..
A
1a BTN1
1b TF1
2a BTN2
2b TF2

B
1a BTN1
1b TF1
2a BTN2
2b TF2
---
! A  1a !
! 1b !
! 2a !
! 2b !
---
---
! B  1a !
! 1b !
! 2a !
! 2b !


then you put a T1 switch module.. then its all automatic ..


PRI DROP X ( Locals) failvoer on carrier side to Y

PRI DROP Y (toll frees) failover on carrier dide to X


On 6/6/07, Jon Pounder <[EMAIL PROTECTED]> wrote:


Quoting Dean Collins <[EMAIL PROTECTED]>:

> http://www.theregister.co.uk/2007/06/06/sun_thinner_blades/
>
>
>
> this article got me thinking - is anyone running asterisk on blade
> servers?
>
>

We have a bunch of ibm blades, but the issue at least with the H
series cabinets we have is that there is no where to put any pci cards
of any sort so you would be limited to purely a voip setup.

There is a T series cabinet that allows pci cards for just such
purposes as asterisk (T is telephony), but the information out there
about just what pieces you need is pretty vague. Anyone have a no bs
description of how the bits actually work together in that setup ?




>
> Any lessons for us to learn?
>
>
>
>
>
>
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED]
>  +1-212-203-4357 Ph
> +61-2-9016-5642 (Sydney in-dial).
>
>
>
>
>
>



Jon Pounder

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_/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


This message was sent using IMP, the Internet Messaging Program.


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Re: [asterisk-users] Voip-info.org

[Atis]

On 6/6/07, Roger Schreiter <[EMAIL PROTECTED]> wrote:

Ed Nuñez schrieb:
> Is anyone else having trouble going into voip-info.org today?

Yes. Me.


Sometimes works,  sometimes don't. Just keep trying (or use
web.archive.org or google's cache)

Regards,
Atis
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Re: [asterisk-users] any codec passthru mode

[Jaswinder Singh]

Yes it might be dumb but since asterisk is a pbx and not a sip proxy
it has to perform many other functions as well .  But i do think that
asterisk should act little smart in this case


SIP wrote:
> That just seems really, REALLY dumb for a program of this magnitude.
>
> I know this has been patched here and there by one person or another,
> but does anyone know if any of these patches to make CODEC negotiation
> actually, you know, negotiate a CODEC will ever make it into the core
> src?
>
>
> Jaswinder Singh wrote:
>> Asterisk by default uses the codec preferred by other device/client  .
>> Asterisk 1.2 ( dunno abt 1.4 specifically)  is not intelligent enough
>> to check if it can avoid transcoding by forcing same codec on other
>> side of conversation . If both sides prefer g729 then asterisk does
>> not do transcoding but if one side prefer gsm and other prefers g729
>> and the gsm side can also support g729 still asterisk will transcode .
>> Someone posted a patch to this in mantis bug tracking system at digium
>> for 1.2 .. google for it and maybe you can find  .
>>
>> On 31/05/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote:
>>> Does anybody has any documentation on codec negotiation within
>>> asterisk?
>>>
>>> Well im using free g729 codec for testing purposes. i mentioned g729
>>> just as
>>> an example. whatever codec is mentioned in user perefernce, asterisk
>>> uses
>>> ulaw to throw out the call.
>>>
>>>
>>> On 5/30/07, Marco Mouta <[EMAIL PROTECTED]> wrote:
>>> > so you r sure you have g729 licences installed and ur * is
>>> transcoding
>>> your RTP streaming?
>>> >
>>> > Test the work flow with disallow=all and allow=g729, can be my
>>> mistake but
>>> I remember to read somewhere on the net any issue about codec
>>> negotiating
>>> precedence when you use allow=all.
>>> >
>>> > good luck
>>> >
>>> >
>>> >
>>> > On 5/30/07, Rizwan Hisham < [EMAIL PROTECTED]> wrote:
>>> > >
>>> > > Hi all,
>>> > > My configuration is:
>>> > > USER (connects to)> ASTERISK---(connects to)--->CARRIER-OUT
>>> > >
>>> > > i want the user preffered codec to pass thru asterisk to
>>> carrier-out.
>>> what i mean is:
>>> > > USER (user uses g729)> ASTERISK---(asterisk should use
>>> g729 for
>>> dialing out)--->CARRIER-OUT
>>> > >
>>> > > instead, this is what happens
>>> > > USER (user uses g729)> ASTERISK---(asterisk uses
>>> g711u)--->CARRIER-OUT
>>> > >
>>> > > How can i force asterisk to use user preffered codec for dialing
>>> out so
>>> that my asterisk machine saves time by no conversion
>>> > > USER PREFERENCE IS
>>> > > disallow=all
>>> > > allow=g729
>>> > >
>>> > > CARRIER PREFERENCE IS
>>> > > allow=all
>>> > >
>>> > > Anybody who can help?
>>> > >
>>> > > --
>>> > > Rizwan Hisham
>>> > > Software Engineer
>>> > > AXVOICE Inc.
>>> > > ___
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>>> > >
>>> > > asterisk-users mailing list
>>> > > To UNSUBSCRIBE or update options visit:
>>> > >
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>> > >
>>> > >
>>> >
>>> >
>>> >
>>> > --
>>> > Esta mensagem (incluindo quaisquer anexos) pode conter informação
>>> confidencial para uso exclusivo do destinatário. Se não for o
>>> destinatário
>>> pretendido, não deverá usar, distribuir ou copiar este e-mail. Se
>>> recebeu
>>> esta mensagem por engano, por favor informe o emissor e elimine-a
>>> imediatamente. Obrigado.
>>> >
>>> > This e-mail message is intended only for individual(s) to whom it is
>>> addressed and may contain information that is privileged, confidential,
>>> proprietary, or otherwise exempt from disclosure under applicable
>>> law. If
>>> you believe you have received this message in error, please advise the
>>> sender by return e-mail and delete it from your mailbox. Thank you.
>>> > ___
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>>> > asterisk-users mailing list
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>>> >
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Rizwan Hisham
>>> Software Engineer
>>> AXVOICE Inc.
>>> ___
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[asterisk-users] TCP<->UDP SIP proxy?

[Yehavi Bourvine +972-8-9489444]

Hello,

   One of our faculties have Microsoft's LCS and would like to connect it to
our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS
talks SIP over TCP with TLS. Anyone can recommend a gateway between these two
protocols?

  Thanks! __Yehavi:
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Re: [asterisk-users] Voip-info.org

[Luis Morales]

yep! 


On Wed, 2007-06-06 at 16:51 +0200, Roger Schreiter wrote:
> Ed Nuñez schrieb:
> > Is anyone else having trouble going into voip-info.org today?
> 
> Yes. Me.
> 
> 
> Roger.
> 
> 
> 
> 
> 
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Re: [asterisk-users] Voip-info.org

[Dave Bour]

Yes

Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]

For those who just want it to work...
Giving you complete IT peace of mind. 

(Sent via Blackberry - hence message may be shorter than my usual verbose 
responses)
PIN 4cc364db (as of March 24, 2007)  

- Original Message -
From: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 

Sent: Wed Jun 06 09:46:04 2007
Subject: [asterisk-users] Voip-info.org

Is anyone else having trouble going into voip-info.org today? 

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Re: [asterisk-users] Voip-info.org

[Paul]

Ed Nuñez wrote:

> Is anyone else having trouble going into voip-info.org today?
>
I didn't have trouble "going there" a few hours ago but they were
delivering blank pages.

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Re: [asterisk-users] Voip-info.org

[James H Thompson]

The colo where voip-info.org is hosted is suffering a DOS attack.
They hope to recover soon.  In the meantime the voip-info.org mirrors are 
available

http://72.14.253.104/search?q=cache:8E6ozIeVoSkJ:www.voip-info.org/wiki/view/Voip-Info%2BMirrors+voip-info+mirrors&hl=en&lr=lang_en&strip=1


[EMAIL PROTECTED]

- Original Message - 
From: Ed Nuñez 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Sent: Wednesday, June 06, 2007 3:46 AM
Subject: [asterisk-users] Voip-info.org


Is anyone else having trouble going into voip-info.org today? 






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Re: [asterisk-users] g729

[Jaswinder Singh]

Just read somewhere that you can use extension as g729 even in
mixmonitor so it will record g729 stream and later you can convert it
to mp3 or wav using sox . If this fails then try monitor application .


On 06/06/07, Jaswinder Singh <[EMAIL PROTECTED]> wrote:

I think asterisk first converts audio stream to slin for recording to
a wav file . Since you are using hardware g729 transcoder i think this
is what is causing the problem . Is the calla actually being recorded
?  I suggest that you use monitor application since it can directly
record g729 audio stream and run some cron script with sox mixing the
IN and OUT files in 1 file .

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
> Yes
>
> This is my extensions.conf entry.
>
> exten => _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
> exten =>
> _1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
> RID}-${EXTEN}-${TIMESTAMP}-OUT)
> exten =>
> _1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
> OUT)
> exten => _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
> exten => _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
> exten => _1NXXNXX,6,Set(CALLERID(number)=14073844200)
> exten => _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
> exten => _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
> Singh
> Sent: Wednesday, June 06, 2007 4:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] g729
>
> Are you trying to record the conversation as well ?
>
> On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> >
> > I installed a hardware g729 codec card in my asterisk, and I'm getting the
> > following error when calling from a g729 sip extension to a SIP trunk also
> > set to g729.  The call goes through just fine, but these error messages
> keep
> > flying by until I disconnect the call.
> >
> >
> >
> > Any ideas?
> >
> >
> >
> > ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
> > failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> > ___
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> > To UNSUBSCRIBE or update options visit:
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
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>
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Re: [asterisk-users] Voip-info.org

[Jaswinder Singh]

Yep its down for me tooo .

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:





Is anyone else having trouble going into voip-info.org today?
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Re: [asterisk-users] g729

[Jaswinder Singh]

I think asterisk first converts audio stream to slin for recording to
a wav file . Since you are using hardware g729 transcoder i think this
is what is causing the problem . Is the calla actually being recorded
?  I suggest that you use monitor application since it can directly
record g729 audio stream and run some cron script with sox mixing the
IN and OUT files in 1 file .

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:

Yes

This is my extensions.conf entry.

exten => _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
exten =>
_1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID}-${EXTEN}-${TIMESTAMP}-OUT)
exten =>
_1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
OUT)
exten => _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten => _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
exten => _1NXXNXX,6,Set(CALLERID(number)=14073844200)
exten => _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
exten => _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729

Are you trying to record the conversation as well ?

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
>
>
>
>
> I installed a hardware g729 codec card in my asterisk, and I'm getting the
> following error when calling from a g729 sip extension to a SIP trunk also
> set to g729.  The call goes through just fine, but these error messages
keep
> flying by until I disconnect the call.
>
>
>
> Any ideas?
>
>
>
> ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
> failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
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>
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>
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RE: [asterisk-users] Re: Verizon Interconnection

[Mike Hammett]

Now that MCI and Verizon are one, they're probably on legacy MCI.  MCI was
also the one that was doing the wholesale SIP pre-merger.

 

 

-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com

 

 

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, June 06, 2007 9:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Verizon Interconnection

 

Verizon has phone service in Switzerland?  Or are you getting US numbers?

On 6/6/07, laurent schweizer < [EMAIL PROTECTED]
 > wrote:

HI,

 

yes we are interconnected with Verizon in SIP, but we are in europe
(Switzerland) so I don't know If it is the same process in USA ...


Laurent


 

2007/6/6, Matt <[EMAIL PROTECTED]>: 

So absolutely no one here was interconnected with Verizon?  I am going to
shoot this over to asterisk-biz, also, in hopes someone may have missed it
that is on the biz list.  The question again is: 



Has anyone on this list connected with Verizon's SIP product?  We are
currently undergoing interop testing with Verizon, and honestly, it seems
like the most convoluted process.   I'd be interested in talking with
someone else who has gone through this and run a few things past you. 


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[asterisk-users] Polycom 320 messages

[Mike Hammett]

I used this site (and perhaps a couple other Google returned) as well as the
Polycom Admin guide as reference.

http://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+wit
h+Asterisk

 

299 is the extension one dials to access their voicemail (with caller ID
sending to the correct voicemail).  I see the bold on the wiki page telling
me to put in my voicemail context, but I'm not sure where they're talking
about.  Previous to doing this work, the phone said there were two
voicemails when there were none.  Now it doesn't say there are voicemails
when they are there.

 

 

This is the entry in the sip.conf

[rwest200]

type=friend

secret=abc123

context=rwest

host=dynamic

[EMAIL PROTECTED]

callerid=Rob West <200>

username=rwest200

qualify=no

port=5060

nat=no

dtmfmode=rfc2833

canreinvite=no

 

This is the voicemail.conf

[rwest]

200 => 1234,Rob West

201 => 1234,Julia Zeiter

202 => 1234,Larry Sallberg

 

This is the phonex.cfg







  



 



  



 

 

 

 

-
Mike Hammett
Intelligent Computing Solutions
  http://www.ics-il.com

 

 

 

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Re: [asterisk-users] shorting flash time

[Steve Kennedy]

On Wed, Jun 06, 2007 at 08:46:20AM -0500, Eric ManxPower Wieling wrote:

> Steve Kennedy wrote:
> >Is there anyway to change the "flash" time on a TDM400 phone port (a
> >user has a phone that seems to generate a short flash which isn't being
> >picked up).
> I suspect the phone us going off hook every once in a while to check if 
> there is a stutter dialtone.  If there is, it can light it's message 
> waiting light.
> Don't you love the analog world?

No, there's a button which they press which is generating a short break.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: Re[2]: [asterisk-users] Re: Verizon Interconnection

[James Coberly]

We did the US interconnect with Verizon about a year ago to Asterisk.
Yes it is a convoluted process,  but if you can get a day to focus on it
only,  you can complete 95% of the interop,  then schedule the last bit
with VZ to finish up.  
It is generally solid though once it is up.

Are you inter-oping for the conference features also, as they add
another layer of PITA processes.


On Wed, 2007-06-06 at 16:05 +0200, Gunnar Schaller wrote:
> Hello,
> Switzerland is good :-)
> My company in Switzerland has a contract with Verizon for wholesale in
> pstn and we are thinking about SIP. Can you please tell me your
> experience with their SIP product?
> 
> Regards,
> Gunnar Schaller
> 
> 
> 
> 
> Wednesday, June 6, 2007, 2:55:54 PM, you wrote:
> 
> > HI,
> 
> > yes we are interconnected with Verizon in SIP, but we are in europe
> > (Switzerland) so I don't know If it is the same process in USA ...
> 
> > Laurent
> 
> 
> > 2007/6/6, Matt <[EMAIL PROTECTED]>:
> >>
> >> So absolutely no one here was interconnected with Verizon?  I am going to
> >> shoot this over to asterisk-biz, also, in hopes someone may have missed it
> >> that is on the biz list.  The question again is:
> >>
> >> Has anyone on this list connected with Verizon's SIP product?  We are
> >> currently undergoing interop testing with Verizon, and honestly, it seems
> >> like the most convoluted process.   I'd be interested in talking with
> >> someone else who has gone through this and run a few things past you.
> >>
> >>
> >> ___
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> >> --
> >>
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> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
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Re: [asterisk-users] Voip-info.org

[Justin Moore]

On 6/6/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:

Is anyone else having trouble going into voip-info.org today?


Yep. Dead for me too.

--
Justin Moore
aka wantmoore
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Re: [asterisk-users] shorting flash time

[Patrick]

On Wed, 2007-06-06 at 12:24 +0100, Steve Kennedy wrote:
> Is there anyway to change the "flash" time on a TDM400 phone port (a
> user has a phone that seems to generate a short flash which isn't being
> picked up).

This was something I bumped into in 2006. Digium support could not tell
me how to fix this so I had to figure it out myself. Here are the
settings that I use for Dutch phones with short flash times:

maxpulse 80
rxflash 200

In zaptel.h change the ZT_MAXPULSETIME value to maxpulse.
In zaptel.h change the ZT_DEFAULT_RXFLASHTIME value to rxflash.

Hope this helps.

Regards,
Patrick

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Re: [asterisk-users] Where to find Polycom firmware with 330/320 support?

[Jessee J Holmes]

Version C is the latest available and for now is only available  
through your reseller or service provider.


Version C supports the new 320s and 330s. The bootrom got updated  
too, I think version 3.2.3b is the latest out there at this time.


Polycom normally only publicly lists the prior version of the  
firmware only (1.6.7 and 2.0.3b at this time)


Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com/



On Jun 6, 2007, at 8:29 AM, Fuermann, Jason Bryce wrote:

Looks like they haven't worked out all the links yet (they just  
redid their site).
http://www.polycom.com/common/documents/support/downloads/voice/ 
spip_ssip_sip_2_0_3b_sig.zip is one gen behind and can be  
downloaded without a reseller account.


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk- 
[EMAIL PROTECTED] On Behalf Of Mandeep Singh Bhabha

Sent: Wednesday, June 06, 2007 5:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Where to find Polycom firmware with  
330/320 support?


I just copied these files from given link.

On Tue, Jun 05, 2007 at 06:35:15PM -0600, Stephen Bosch wrote:

Fuermann, Jason Bryce wrote:
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/ 
soundpoint_ip330_320.html


This only works if you have a reseller account.

-Stephen-
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--
With Regards,
Mandeep Singh Bhabha


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Re: [asterisk-users] Dial plan inquiry using GotoIf()

[Steve Murphy]

On Wed, 2007-05-30 at 20:05 -0400, Steve Finkelstein wrote:
> Thanks for the help on this thread all.
> 
> It would make sense if I write an AGI and incorporate a DB backend to
> check against numbers I want explicitly dropped. If anyone has such a
> utility, I'd love to -not- reinvent the wheel. Otherwise, I'll go whip
> it up and probably provide a web frontend for adding/removing numbers.
> 

You can still use the dialplan with the DB func to check incoming 
CID info. Also, the Dial() app has several options for call screening
and 
privacy; these would be performed when dialing your extension.

You can have Dial keep a DB of callers, and remember whether to always
just patch them right thru, play them a polite "go away and don't come
back",
or send them off to torture scripts, or just route them straight to VM.
And, Dial() will ask you what you want to do, on the first call. Read
thru the Dial doc you get with "core show application dial". There's
an option to store an intro from each caller, where it records in  a
sound file, who they say they are. I have several hundreds of these, and
play them as the
call comes in, so we know who's calling without having to run to a CID
display.
For those who have poor to no vision, this can be a cool feature.

murf


> - sf
> 
> C F wrote:
> > It fails because the right function is ${CALLERID(num)}
> > 
> > On 5/30/07, Steve Finkelstein <[EMAIL PROTECTED]> wrote:
> >> Hi all,
> >>
> >> I'm looking for some rudimentary insight on GotoIf() which seems to be
> >> failing on me in my dial plan. All I basically wish to do is block a
> >> particular caller. Sounds easy enough, but my ternary operator/plan
> >> currently is not properly being implemented. Can anyone spot where I'm
> >> being a momo?
> >>
> >> All extensions get forwarded to the following macro:
> >>
> >> [macro-forward]
> >> ; arg1 = phone number
> >> ; arg2 = timeout
> >> ; arg3 = extension (voicemail)
> >> ; arg4 = mobile number
> >> exten => s,1,Zapateller(answer|nocallerid)
> >> exten => s,2,PrivacyManager
> >> exten => s,3,Wait(1)
> >> exten => s,4,GotoIf($[${CALLERID(number)} = "15552221313"]?15:5)
> >> exten => s,5,GotoIf($[${LEN(${CALLERID(number)})} = 4]?6:8)
> >> exten => s,6,AGI(didextlookup.agi|${CALLERID(number)})
> >> exten => s,7,Set(CALLERID(number)=${didlookup})
> >> exten => s,8,GotoIf($[${LEN(${CALLERID(number)})} = 10]?9:10)
> >> exten => s,9,Set(CALLERID(number)=1${CALLERID(number)})
> >> exten => s,10,Dial(${ARG1},${ARG2})
> >> exten => s,11,GotoIf($[${EXISTS(${ARG4})}]?11:12)
> >> exten => s,12,Dial(${ARG4},${ARG2})
> >> exten => s,13,Voicemail(u${ARG3})
> >> exten => s,14,Playback(vm-goodbye)
> >> exten => s,15,HangUp
> >> exten => s,105,HangUp
> >>
> >> As you can tell, exten => s,4,GotoIf($[${CALLERID(number)} =
> >> "15552221313"]?15:5)  is what I recently added.
> >>
> >> Here's what I see in the CLI logs:
> >>
> >> -- Executing [EMAIL PROTECTED]:1] Macro("IAX2/lime-3",
> >> "forward|SIP/5609|15|5609|IAX2/[EMAIL PROTECTED]/15164766201") in new stack
> >> -- Executing [EMAIL PROTECTED]:1] Zapateller("IAX2/lime-3",
> >> "answer|nocallerid") in new stack
> >> -- Executing [EMAIL PROTECTED]:2] PrivacyManager("IAX2/lime-3", "")
> >> in new stack
> >> -- CallerID Present: Skipping
> >> -- Executing [EMAIL PROTECTED]:3] Wait("IAX2/lime-3", "1") in new
> >> stack
> >> -- Executing [EMAIL PROTECTED]:4] GotoIf("IAX2/lime-3", "0?15:5") in
> >> new stack
> >> -- Goto (macro-forward,s,5)
> >>
> >> It evaluates to false, hence goes to s,5. I keep dialing from that
> >> particular number (the one in the example is clearly masked as a false
> >> CID), and verified it's showing up as that number on callerID.
> >>
> >> Also one last question. Say I need to add more numbers to block in the
> >> future, is there an easier way to do this than renumbering my entire
> >> macro? Renumbering everything is just begging for a typo which can
> >> effectively render my dial plan broken.
> >>
> >> Thank you kindly, everyone!
> >>
> >> - sf
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> >>
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> > !DSPAM:1020,465db390179485209328925!
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Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *

[Steve Murphy]

On Wed, 2007-05-30 at 07:50 -0400, Matt wrote:
> We still run 1.2.6 on some of our production systems because, so far,
> it has been the only stable release of Asterisk for us.   Other
> versions core dump for no reason and do all kinds of other funky
> things.
> 
> On 5/29/07, Jaswinder Singh <[EMAIL PROTECTED]> wrote:
> What you say might be true for small business or home  pbx
> systems .
> But if you have a production server handling sip/iax
> trunks  over
> internet then you need to upgrade to avoid  security related
> bugs and
> exploits that are released . 

Well, once you've developed the "perfect" asterisk app, I guess you could 
just stick with it forever and never change anything.

Or, you invest some time==money into the process, and keep a separate 
machine to do testing with, file bugs, and push them thru so someday,
you get
an even better "perfect" asterisk app.

A "once a day" crash could never, ever get fixed without users
collecting a backtrace, filing a bug, and helping out with testing!

If asterisk is really involved in your cash flow, I don't see how you
can 
justify grabbing one version and staying with it forever. Such a
business 
model is doomed to extinction. All apps, for that matter, all living and
even 
artificial things follow a predictable sequence that ends in death.
Your 
perfect server will last maybe 5 or so years max, and then the hardware
will
fail or obsolesce, and unless you actively participate in keeping the
software up to date, you will have an expensive, time-consuming, perhaps
fatal ordeal getting your app up and running on newer hardware/software
again. A "sustaining" model has you actively participate in the
evolution of the software you find "critical". Right?

In a closed source model, you'd pay yearly maintenance, and get a voice
in the direction the software evolved. In an open-source model, the
yearly maintenance cost is your own time in filing bugs, and testing,
submitting patches, etc. One way or the other, it costs you something.
Or, like dinosaurs, stick with one version and follow that evolutionary
branch to its end.


murf

-- 
Steve Murphy
Software Developer
Digium


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Re: [asterisk-users] PRI Partial Re-Rounting

[Eric \"ManxPower\" Wieling]

Jon Schøpzinsky wrote:

Hello List

We are trying to redirect calls directly, instead of opening a new channel and 
dialing out.
Etc:

A calls B on our asterisk, and is directly redirected to C


We have been told that this feature should be available on a PRI level, and is 
called Partial re-routing.

Anybody has an idea of whether this is supported in Asterisk?


It is called 2BCT.  It is supported on AT&T 5ESS PRI lines.  I don't 
think it is supported on NI2 or non-AT&T switches.  I've never used it.


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Re: [asterisk-users] Voip-info.org

[Roger Schreiter]

Ed Nuñez schrieb:

Is anyone else having trouble going into voip-info.org today?


Yes. Me.


Roger.





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Re: [asterisk-users] PRI Partial Re-Rounting

[Anthony Francis]

This is of great interest to me as well, it would nice be able to send a 
"Temporarily moved to ''" out the PSTN just like you can do in sip.


Jon Schøpzinsky wrote:

Hello List

We are trying to redirect calls directly, instead of opening a new channel and 
dialing out.
Etc:

A calls B on our asterisk, and is directly redirected to C


We have been told that this feature should be available on a PRI level, and is 
called Partial re-routing.

Anybody has an idea of whether this is supported in Asterisk?

Kind Regards
Jon Schøpzinsky
Detele.

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.472 / Virus Database: 269.8.9/834 - Release Date: 05-06-2007 14:38
 
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RE: [asterisk-users] Outlook dialing

[Martin Smith]

We've been using SIPTAPI and love it for our call center. We originally
used ASTTAPI, but liked the idea of not running AstManProxy.
 
http://siptapi.sourceforge.net/ - website for SIPTAPI
 
http://projects.bebr.ufl.edu/wiki/AsteriskTAPI - our external
documentation, for the outside world in case it helps :)
 

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent: Tuesday, June 05, 2007 11:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Outlook dialing



The bar is getting raised yet again
http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack
et8+Virtual+Office.aspx

 

I personally use Snapanumber $30 or there abouts (after trialing
a few other TAPI solutions and finding them sub-par) and think it's a
great product but interesting to see how more people are expecting
desktop/phone integration applications.

 

Does anyone else have a favorite Outlook autodial application
they use and love?

 

 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).

 

 

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Re: [asterisk-users] Re: Verizon Interconnection

[Matt]

Verizon has phone service in Switzerland?  Or are you getting US numbers?

On 6/6/07, laurent schweizer <[EMAIL PROTECTED]> wrote:


HI,

yes we are interconnected with Verizon in SIP, but we are in europe
(Switzerland) so I don't know If it is the same process in USA ...

Laurent


2007/6/6, Matt <[EMAIL PROTECTED]>:
>
> So absolutely no one here was interconnected with Verizon?  I am going
> to shoot this over to asterisk-biz, also, in hopes someone may have missed
> it that is on the biz list.  The question again is:
>
> Has anyone on this list connected with Verizon's SIP product?  We are
> currently undergoing interop testing with Verizon, and honestly, it seems
> like the most convoluted process.   I'd be interested in talking with
> someone else who has gone through this and run a few things past you.
>
>
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[asterisk-users] asterisk 1.2.18 problems...

[Angel Luis Martinez]

Hi All:

I have experienced some big problems on an asterisk production server
under 1.2.18:

First of all, a very rare message like this... No application Macro ???

   -- Saved useragent "Linksys/SPA922-5.1.7" for peer 1363
Jun  6 15:08:24 WARNING[406]: pbx.c:1720 pbx_extension_helper: No
application 'Macro' for extension (pbx-incoming, 1133, 1)
  == Spawn extension (pbx-incoming, 1133, 1) exited non-zero on
'SIP/1210-081aa708'
Jun  6 15:08:24 WARNING[406]: pbx.c:1720 pbx_extension_helper: No
application 'Macro' for extension (pbx-incoming, h, 1)
  == Spawn extension (pbx-incoming, h, 1) exited non-zero on
'SIP/1210-081aa708'

After rebooting the machine completely, appears to work fine.. but i think
that 1.2.18 is not very stable... because i have the same error 3 times on
the last 15 days...

Any suggestion to rollback to another asterisk more STABLE version ??
(1.2.15 for example) ??

Thanks a lot...
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