Re: [asterisk-users] create table in mysql using asterisk

[James Sharp]

On 01/09/2012 02:44 AM, Eyal wrote:

Hi,
I try to create a new table using MYSQL command in asterisk.
This is what i write:
*Query resultid ${connid} CREATE TABLE IF NOT EXISTS "conference_600"
("id" int(11) NOT NULL auto_increment, "channel_id" varchar(40),
"number_in_line" int(2), PRIMARY KEY("id")")*
and this is the warning that i get in the cli:
*app_addon_sql_mysql.c:383 aMYSQL_query: aMYSQL_query: mysql_query
failed. Error: You have an error in your SQL syntax; check the manual
that corresponds to your MySQL server version for the right syntax to
use near '"conference_600" ("id" int(11) NOT NULL auto_increment,
"channel_id" varchar(40)' at line 1
*
What is the problem do you think?
Do I in the direction or have a completely different way to do this?


That's a MySQL syntax error, not an Asterisk error.  However, the 
solution is to not put quotes around your table and field names.  That 
will make MySQL happy.


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[asterisk-users] create table in mysql using asterisk

[Eyal]

Hi,
I try to create a new table using MYSQL command in asterisk.
This is what i write:
Query resultid ${connid} CREATE TABLE IF NOT EXISTS "conference_600"
("id" int(11) NOT NULL auto_increment, "channel_id" varchar(40),
"number_in_line" int(2), PRIMARY KEY("id")")
and this is the warning that i get in the cli:
app_addon_sql_mysql.c:383 aMYSQL_query: aMYSQL_query: mysql_query
failed. Error: You have an error in your SQL syntax; check the manual
that corresponds to your MySQL server version for the right syntax to
use near '"conference_600" ("id" int(11) NOT NULL auto_increment,
"channel_id" varchar(40)' at line 1

What is the problem do you think?
Do I in the direction or have a completely different way to do this?

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[asterisk-users] video mail is not store

[Durgesh Mishra]

Hi, 

I am facing an issue while testing the video mail service of Asterisk. I have 
two different setup on one setup client being used is Mercuro while on the 
other client is Android based. 

On the Mercuro setup video mail is stored and retrieved properly while with 
Android based setup video mail is not stored (audio is through). 

Both the client use H.264 codec with following sdp information: 



Android Based Client SDP Parameters 

v=0 
o=- 1325786904 1325786904 IN IP4 172.16.130.47 
s=Polycom RealPresence 
c=IN IP4 172.16.130.47 
b=AS:1920 
t=0 0 
a=sendrecv 
m=audio 3230 RTP/AVP 118 115 114 113 0 8 119 
a=rtpmap:118 SIRENLPR/48000 
a=fmtp:118 bitrate=64000 
a=rtpmap:115 G7221/32000 
a=fmtp:115 bitrate=48000 
a=rtpmap:114 G7221/32000 
a=fmtp:114 bitrate=32000 
a=rtpmap:113 G7221/32000 
a=fmtp:113 bitrate=24000 
a=rtpmap:0 PCMU/8000 
a=rtpmap:8 PCMA/8000 
a=rtpmap:119 telephone-event/8000 
a=fmtp:119 0-15 
m=video 3232 RTP/AVP 109 110 
a=rtcp-fb:* ccm fir tmmbr 
a=rtpmap:109 H264/9 
a=fmtp:109 profile-level-id=42800d; max-mbps=108000; max-fs=3840; max-br=1920; 
sar=13 
a=rtpmap:110 H264/9 
a=fmtp:110 profile-level-id=42800d; packetization-mode=1; max-mbps=108000; 
max-fs=3840; max-br=1920; sar=13 
m=application 3236 RTP/AVP 100 
a=sendrecv 
a=rtpmap:100 H224/4800 





MERCURO SDP Parameters 

v=0 
o=- 1234 1235 IN IP4 10.34.77.90 
s=Mercuro IMS Client Session 
t=0 0 
m=audio 31098 RTP/AVP 0 8 101 
c=IN IP4 10.34.77.90 
a=rtpmap:0 PCMU/8000/1 
a=rtpmap:8 PCMA/8000/1 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 
a=silenceSupp:off - - - - 
a=sendrecv 
m=video 34442 RTP/AVP 113 
c=IN IP4 10.34.77.90 
a=rtpmap:113 H264/9 
a=fmtp:113 fmtp:113 profile-level-id=42e00a; packetization-mode=1; max-br=2000; 
max-mbps=11880 
a=sendrecv 



Plz tell me is there any limitation from the Asterisk side i.e. H.264 codec is 
supported only with limited parameters. 

I would like to know what parameters of H.264 codec are supported by Asterisk? 



Your comnments are most welcome. 



Regards, 

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Re: [asterisk-users] Best non polycom SIP conference room phone

[Carlos Alvarez]

On Sun, Jan 8, 2012 at 1:57 PM, Luke Hamburg  wrote:

> Carlos-
> Sorry if this is too much of a digression but this piqued my interest as
> I've been pretty happy with Polycom in my limited experience (haven't used
> the SPAs much, just Yealink & Polycom, and an occasional Snom here and
> there).   If the config files were not the issue for you, then what _were_
> the problems?
>

The first thing that comes to mind is the long boot time for the Polycoms,
which I know has improved in recent models but is still longer than the
SPA.  If we're troubleshooting or experimenting with changes this is
annoying.  The Polycoms pull down a lot of data when they boot, and we've
not figure out a way to prevent that.  We have a lot of customers on WiMAX
connections where the voice runs on a separate VLAN for quality guarantee,
but it has very limited bandwidth because only voice should be on it.  If a
customer with Polycoms and this connection reboots a phone, it floods the
voice VLAN.  We've all found that we prefer to use the SPA's web UI a lot
more than the Polycom.  And the SPA has a very nice and easy to use
encrypted configuration system that is preferable for a hosted service
provider like us where the configs travel the wild internet.

How do you like the Yealink phones?  Is the cheap price worth it?  We
looked at them a long time ago but there were some issues that prevented us
using them (centered around secure config over the internet).


-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller

[Luke Hamburg]

Are we the only 2 people on this list experiencing this issue?  (surprised)

Anyone else have any insights?

My hunch is that this is likely some type of FreePBX issue with how it 
generates the [from-internal-xfer] context.

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen
Sent: Saturday, January 07, 2012 3:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Blind transfers being cancelled by asterisk & 
hanging up on remote caller

 

Oh crap. I just reread the previous post & realized I'm not alone. Hallelujah! 
I'll post back more info soon.

-
Doug Mortensen 
Sent via DroidX2 on Verizon Wireless™



-Original message-

From: Ryan Wagoner 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Sat, Jan 7, 2012 15:59:36 GMT+00:00
Subject: Re: [asterisk-users] Blind transfers being cancelled by asterisk & 
hanging up on remote caller

 

On Sat, Jan 7, 2012 at 5:19 AM, Luke Hamburg  wrote:

Doug:
for what it's worth I am having the exact same nightmare.  Not sure exactly
when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I am
running 1.8.9rc1).  I also have Polycom (335, 550, 650)  and blind transfers
are broken.  All legs of the call are dropped when the xfer is executed.  A
calls B, B xfer to C and (C) blips for a split second like its ringing but
then all calls go dead.  I tried to debug myself using some sip tracing but
I didn't get very far.  I even tried mucking around with a few settings in
my Polycom provisioning I thought might be related e.g.

 voIpProt.SIP.allowTransferOnProceeding
 voIpProt.SIP.connectionReuse.useAlias
 voIpProt.SIP.useContactInReferTo
 voIpProt.SIP.conference.parallelRefer
 voIpProt.SIP.strictLineSeize
 voIpProt.SIP.strictUserValidation
 voIpProt.SIP.strictReplacesHeader
 voIpProt.SIP.useContactInReferTo

and also upgraded to the new 3.3.4 firmware which is out yesterday,  didn't
change a thing.
stuck here for now,  Attended xfers seem to work.I am not sure this is a
Polycom-specific issue because I was seeing this bad behavior even using
some Softphones I set up for testing.

my next recourse is to try rolling back to 1.8.8.0 or earlier and if that
fixes it then I will open a JIRA ticket with more details.

Luke


--
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas
Mortensen
Sent: Thursday, January 05, 2012 3:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Blind transfers being cancelled by asterisk &
hanging up on remote caller


Hello all,

I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5
from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that
blindpreferred=1 (all transfers default as blind transfers). If a customer
calls in & we answer & transfer, everything works fine. But if we call out
to a customer & then transfer to another internal extension, that extension
quickly rings & then the call is immediately gone & hung up. We are using
Polycom firmware 3.3.3.

In troubleshooting this & analyzing the asterisk logs (& asterisk SIP
debug), I am seeing a few interesting items. Any help would be appreciated.

[...]

Thanks,
-
Doug Mortensen


I can't reproduce this on a test system with Asterisk 1.8.8.1 using a Polycom 
335 and 550 running firmware 3.2.6. I called an external number using Vitelity 
then blind transferred to the other phone. I am interested as I have a 
production system with Polycom 335 phones running 1.8.7.0 that works.

Ryan

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Re: [asterisk-users] Best non polycom SIP conference room phone

[Luke Hamburg]

Carlos-
Sorry if this is too much of a digression but this piqued my interest as
I've been pretty happy with Polycom in my limited experience (haven't used
the SPAs much, just Yealink & Polycom, and an occasional Snom here and
there).   If the config files were not the issue for you, then what _were_
the problems?  

Luke

--
From: Carlos Alvarez
Sent: Sunday, January 08, 2012 3:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best non polycom SIP conference room phone

We have no problem with their config files.  They are no worse than anything
else, including the SPA series phones that we greatly prefer over the
Polycom.  The Polycom phones simply are more effort and more time-consuming
than the SPA phones, and some others (though there are worse phones).  We
hate working with them for a wide variety of reasons, but the config files
are certainly not one of them.
 

-- 
Carlos Alvarez
TelEvolve
602-889-3003



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Re: [asterisk-users] Best non polycom SIP conference room phone

[Carlos Alvarez]

On Sun, Jan 8, 2012 at 9:02 AM, C F  wrote:

> I find that the bottom line of all polycom haters is ones inability of
> comprehending the config files and not in its quality.
>

We have no problem with their config files.  They are no worse than
anything else, including the SPA series phones that we greatly prefer over
the Polycom.  The Polycom phones simply are more effort and more
time-consuming than the SPA phones, and some others (though there are worse
phones).  We hate working with them for a wide variety of reasons, but the
config files are certainly not one of them.


> However check out Panasonic. They make a sip conference phone.
>

I didn't know about the conference phone from them.  We sell their wireless
phones and found them extremely annoying to learn to configure, with lots
of quirks and bugs, but once they are working they are good.  Once you get
to know the oddities and have a suitable provisioning server set up,
deploying more is no problem.  Troubleshooting is annoying because the
documentation is poor and there are lots of quirks/bugs/unexpected
"features."

User acceptance on the Panasonic is very good.

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Best non polycom SIP conference room phone

[Ryan Wagoner]

On Sun, Jan 8, 2012 at 12:03 PM,  wrote:

> Thank you for your responses. No where did I say I hate polycom phones. I
> personally do not like their approach to sip as a company. Their audio
> quality  is top notch but for me the rest leaves me wanting. Has anyone
> used the newer snom conference room phone?
>
>
If the Snom conference phone is anything like their deskphone speakerphone
I would stay away. We purchased Snom 360s for the large number of BLF and
VPN capability. However I quickly had complaints about the speakerphone.
Additionally the user interface was laggy. I've tried changing settings and
they still sound like a non duplex speakerphone. I only have a few Snom
phones left and everything else is Polycom. You can't beat their sound
quality and the user interface is responsive. If you keep an eye out on the
clearance deals at telephonydepot.com you can sometimes grab a Polycom
speakerphone for a great price.

Ryan
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Re: [asterisk-users] Answering call from queue, then put back in queue?

[Todd Routhier]

Thanks Jim, I like that idea.


On Sun, Jan 8, 2012 at 12:42 PM, Jim Dickenson  wrote:

> One way to deal with this is to have two queues. Give priority to the
> original queue callers land in. Once answered put the call in to the second
> queue. They will then be in the second queue in the order the agents
> answered the first queue.
> --
> Jim Dickenson
> mailto:dicken...@cfmc.com 
>
> CfMC
> http://www.cfmc.com/
>
>
>
> On Jan 8, 2012, at 10:26 AM, Todd Routhier wrote:
>
> Version: Asterisk 1.8.x
>
> Question: Is it possible for an agent to answer a call from a queue, then
> place the call back in the queue in the same position they were in?
>
>
> Seems that the answer would be yes to the remove from queue, then place
> back in by having the agent just transfer the call back to the queue but is
> there any way to put them back in line where they were?
>
> The idea is that the owner of the queue doesn't want callers waiting on
> hold without first having an agent at least answer the call and ask them to
> please hold. What's the best way to handle this?
>
> Thanks in advance!
>
> --Todd
> --
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>
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Re: [asterisk-users] Answering call from queue, then put back in queue?

[Jim Dickenson]

One way to deal with this is to have two queues. Give priority to the original 
queue callers land in. Once answered put the call in to the second queue. They 
will then be in the second queue in the order the agents answered the first 
queue.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jan 8, 2012, at 10:26 AM, Todd Routhier wrote:

> Version: Asterisk 1.8.x
> 
> Question: Is it possible for an agent to answer a call from a queue, then 
> place the call back in the queue in the same position they were in?
> 
> 
> Seems that the answer would be yes to the remove from queue, then place back 
> in by having the agent just transfer the call back to the queue but is there 
> any way to put them back in line where they were?
> 
> The idea is that the owner of the queue doesn't want callers waiting on hold 
> without first having an agent at least answer the call and ask them to please 
> hold. What's the best way to handle this?
> 
> Thanks in advance!
> 
> --Todd
> --
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[asterisk-users] Answering call from queue, then put back in queue?

[Todd Routhier]

Version: Asterisk 1.8.x

Question: Is it possible for an agent to answer a call from a queue, then
place the call back in the queue in the same position they were in?


Seems that the answer would be yes to the remove from queue, then place
back in by having the agent just transfer the call back to the queue but is
there any way to put them back in line where they were?

The idea is that the owner of the queue doesn't want callers waiting on
hold without first having an agent at least answer the call and ask them to
please hold. What's the best way to handle this?

Thanks in advance!

--Todd
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Re: [asterisk-users] Connecting to an Old Phone System

[Don Kelly]

I've done this with US ISDN PRI. Both with a Digium card (PCI) and an
Astribank (USB). I'd expect it's doable with the several products that
support Euro ISDN.

You can set up a simple VoIP gateway, but you can also do all sorts of magic
things in the Asterisk system for selected calls before or in lieu of
passing them along to the legacy system.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Sunday, January 08, 2012 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Connecting to an Old Phone System

What type of phone system?
And what type of connectivity are you trying to give the old pbx?

On 1/6/12, Dan Journo  wrote:
> Hi,
>
> This is not strictly an asterisk questions, but... ive got a client 
> with an old digital pbx phone systems connected to an isdn30e line.
>
> I've been shown a sip gateway that can connect to asterisk on one 
> side, and also has an ISDN30e socket that the old phone system can 
> connect to. But it's a bit pricey.
>
> Is there such a thing as an ISDN30e PCI card which can be used with a 
> copy of Asterisk, that can act like a voip gateway between the old 
> phone system, and our asterisk box?
> Or can anyone recommend a gateway that isn't too expensive?
> They use 8 channels of the isdn30e
>
> Many thanks
> Dan
>
> Dan Journo
> Kesher Communications (UK)
> Business Phone Systems | Hosted 
> PBX
>
>

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Re: [asterisk-users] Best non polycom SIP conference room phone

[BryantZ]

Thank you for your responses. No where did I say I hate polycom phones. I 
personally do not like their approach to sip as a company. Their audio quality  
is top notch but for me the rest leaves me wanting. Has anyone used the newer 
snom conference room phone?

Bryant Zimmerman 

On Jan 8, 2012, at 10:59 AM, C F  wrote:

> I find that the bottom line of all polycom haters is ones inability of
> comprehending the config files and not in its quality.
> However check out Panasonic. They make a sip conference phone.
> 
> On 1/5/12, Carlos Alvarez  wrote:
>> On Thu, Jan 5, 2012 at 5:10 PM, C F  wrote:
>> 
>>> On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman 
>>> wrote:
 I am looking for a really good SIP conference room phone for use with
 asterisk. I do not like Polycom at all.
>>> 
>>> You have a really bad taste.
>>> 
>> 
>> There was an interesting flamewar one day in the Asterisk IRC channel over
>> Polycom love/hate.  We fall into the hate category here, and hope to never
>> have to deal with them.  If there was an SPA-series conference phone, we'd
>> all rejoice.
>> 
>> --
>> Carlos Alvarez
>> TelEvolve
>> 602-889-3003
>> 
> 
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Re: [asterisk-users] Best non polycom SIP conference room phone

[C F]

I find that the bottom line of all polycom haters is ones inability of
comprehending the config files and not in its quality.
However check out Panasonic. They make a sip conference phone.

On 1/5/12, Carlos Alvarez  wrote:
> On Thu, Jan 5, 2012 at 5:10 PM, C F  wrote:
>
>> On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman 
>> wrote:
>> > I am looking for a really good SIP conference room phone for use with
>> > asterisk. I do not like Polycom at all.
>>
>> You have a really bad taste.
>>
>
> There was an interesting flamewar one day in the Asterisk IRC channel over
> Polycom love/hate.  We fall into the hate category here, and hope to never
> have to deal with them.  If there was an SPA-series conference phone, we'd
> all rejoice.
>
> --
> Carlos Alvarez
> TelEvolve
> 602-889-3003
>

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Re: [asterisk-users] Connecting to an Old Phone System

[C F]

What type of phone system?
And what type of connectivity are you trying to give the old pbx?

On 1/6/12, Dan Journo  wrote:
> Hi,
>
> This is not strictly an asterisk questions, but... ive got a client with an
> old digital pbx phone systems connected to an isdn30e line.
>
> I've been shown a sip gateway that can connect to asterisk on one side, and
> also has an ISDN30e socket that the old phone system can connect to. But
> it's a bit pricey.
>
> Is there such a thing as an ISDN30e PCI card which can be used with a copy
> of Asterisk, that can act like a voip gateway between the old phone system,
> and our asterisk box?
> Or can anyone recommend a gateway that isn't too expensive?
> They use 8 channels of the isdn30e
>
> Many thanks
> Dan
>
> Dan Journo
> Kesher Communications (UK)
> Business Phone Systems | Hosted
> PBX
>
>

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[asterisk-users] cached VMI on manual voicemail update

[Tzafrir Cohen]

Hi

I use a simple script to enable/disable for voicemail on an extension
for testing (attached: test_voicemail). The script only created the txt
file, but I don't suppose this should matter.

I set up a mailbox. The new message shows fine in 'voicemail show users'
(The new user takes 'voicemail reload' for it to be added on the list,
but after it has been initially added , it seems to be updated
immediately (or by running 'voicemail show users'?).

However the voicemail indication to the tested phone (in chan_dahdi, in
this case) did not follow suit: it never updates. Even a full 'dahdi
restart' is not good enough. Nothing short of a restart of asterisk
seems to make it update.

Looking further I see that the MWI values are cached:

kemeny*CLI> event dump cache MWI 
Event Type: MWI
Cache Unique Keys:
--> Mailbox
--> Context

--- Begin Cache Dump ---

Event: MWI
Mailbox: 4004
Context: default
NewMessages: 1
OldMessages: 0
EntityID: 00:1c:c0:fc:f0:ad

--- End Cache Dump ---

kemeny*CLI> voicemail show users 
ContextMbox  User  Zone   NewMsg
default40040
1 voicemail users configured.


Is there any way to tell Asterisk to flush that cache? Is this
considered a bug?


Tested on 1.8 (1.8.7.0 and SVN-branch-1.8-r349672).

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
#!/bin/sh

NUM=${2-:4001}

vm_base="/var/spool/asterisk/voicemail/default"
vm_dir="$vm_base/$NUM/INBOX"

setup() {
sed -i -e "/^$NUM/d" -e "\$a$NUM => ," /etc/asterisk/voicemail.conf
}

on() {
mkdir -p "$vm_dir"
cat <"$vm_dir/msg.txt"
;
; Message Information file
;
[message]
origmailbox=$NUM
context=from-internal
macrocontext=
exten=s
priority=1
callerchan=DAHDI/1-1
callerid="Channel 1" <$NUM>
origdate=`date -R`
origtime=`date +'%s'`
category=
duration=4
EOF
}

off() {
rm -f "$vm_dir/"*
}

case "$1" in
setup | on | off) "$1";;
esac
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