Re: [asterisk-users] SMS with Asterisk

[Per Jessen]

Steve Totaro wrote:

> This link show how to send SMS using HTTP(s) and the format of the
> URL.
> http://www.kannel.org/download/1.4.1/userguide-1.4.1/userguide.html#AEN4201
> 
> The previous link is good news to me.  Now I can do anything by
> hitting a URL. it is so simple.

I've been asterisk's smsq for a couple of years:

/usr/sbin/smsq --motx-channel='mISDN/2/062210' --motx-callerid=211 $1 
"$message"


/Per Jessen, Zürich

-- 
http://www.spamchek.com/ - your spam is our business.


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[asterisk-users] Office timings only work asterisk after that voicemail

[mahesh katta]

Hi,

I have small asterisk pbx. I was made a dialplan like whenever out side
world is dialing to asterisk pbx DID's it was goes to dial his extension
number. if this extension is not pick the call after 30 sec it will dial
his(user) mobile number. i need to create dialplan only office timings(9:00
am to 19:00, sun to Thu days) needs to dial his mobile number after that
will play like "this is not office time" whatever. after office timing it
should be go voicemail .

[default]
exten => 4578901,1,AGI(agi://127.0.0.1:4577/call_log)
exten =>
4578901,2,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten => 4578901,3,Dial(SIP/5001,30,tTo)
exten => 4578901,4,Gotoiftime(09:00-19-00,sun-fri,*,*?default,4578901,5)
exten => 4578901,5,Dial(Zap/g0/0554721368,,tTo)
exten => 4578901,6,VoiceMail(u5001)
exten => 4578901,7,playback(vm-goodbye)
exten => 4578901,8,Hangup
up to hundred did's have same dialplan

currently I am using above dial plan i need to add my question in this
dialplan . is there any way. please help'

Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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[asterisk-users] Question on pause in dialing

[Jerry Geis]

I am using asterisk 1.4.41 and polycom phones.
When I dial long distance I hear a pause on the last 2 digits.

This is the dialout context that matches.

[ Context 'smvoice-dialout' created by 'pbx_config' ]
 '_71XX' => 1. Set(CALLERID(number)=3175551212)   
[pbx_config]
   2. Dial(DAHDI/g1/${EXTEN:1})  
[pbx_config]
 '_7XXX' =>1. Set(CALLERID(number)=3175551212)   
[pbx_config]
   2. Dial(DAHDI/g1/317${EXTEN:1})   
[pbx_config]


So when I click "new call", and dial 7 1 3 1 7 5 0 6 x x then I get a 
pause before I can continue

entering the last two digits x x .

Why is that and how can I remove it.

Thanks,

Jerry


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Re: [asterisk-users] SMS with Asterisk

[Jared Geiger]

I dropped my DC customers for much safer Bethesda customers :)

On Tue, Jun 21, 2011 at 2:11 PM, Robert Huddleston wrote:

> Hahahah Baltimore and SE DC… How about Philly too J
>
> ** **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Totaro
> *Sent:* Tuesday, June 21, 2011 2:08 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] SMS with Asterisk
>
> ** **
>
> ** **
>
> On Mon, Jun 20, 2011 at 2:48 PM, Warren Selby 
> wrote:
>
> On Mon, Jun 20, 2011 at 3:52 AM, Steve Totaro <
> stot...@asteriskhelpdesk.com> wrote:
>
> Two requests, not from me but the community.
>
> 1.  Don't top post
>
>
> *cough*
>  
>
> 2.  When you find your solution, reply to this thread so others will be
> (silver) spoon fed the answers and blindly accept them without trying things
> and going through a learning curve and experimentation when they find your
> post in Google.
>
>
> I hear some people are actually deploying their asterisk solutions in war
> zones and are taking heavy fire while they're looking for answers - seems
> like it would make their life a whole lot easier (and safer!) if people
> posted simple responses on this list when suggestions worked for them...
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 
>
>
> LOL at the haters.
>
> 1.  It was joke for those with senses of humor and know me (Randy got it),
> but I top post when others do.  I bottom post when others do.  I just go
> with the flow.  I am not uptight about it.
>
> 2.  I have never heard that but it may be true.
>
> Personally, I have been shot at on top the Iraqi Government building in the
> IZ from the Red Zone.  I was setting up and troubleshooting the Motorola
> Canopy WiFi system.  Just a few 7.62x39 rounds, nothing I would call heavy
> fire.
>
> The only "Heavy Fire" I took was standing on top of one of the buildings at
> the FOB trying to trace a cable and the ricochets from the firing range were
> landing all over the place.  That happens when 30 guys are training with
> AKs  and a T-Wall as the backstop.
>
> I have deployed Asterisk systems in war zones many times, in West African
> countries, Iraq, Baltimore and South East DC.  I would certainly seek
> shelter/defensive position if there was gun play.  LOL, you can wish
> yourself into a gun fight but you cannot wish yourself out. 
>
>
> It would also be a whole lot easier for someone to physically feed me so my
> hands could be free to work in hostile environments, maybe an LN can bring
> me a portable toilet and make sure it is fresh, that would make everything
> so easy and easy is what we all want.
>
> Heck, I could just set it up at the FOB and then deploy it.
>
> At any rate, I asked the guy to post his success, so I am not sure why you
> posted, but thanks.  It only takes 10% truth to make a legend.
>
> Thanks,
> Steve T 
>
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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[Robert-iPhone]

wow I think someone needs to just spend some time reading and playing. Getting 
these phones working is not rocket science and there are similarities with how 
to do firmware / config pushes.

Not to sound mean but RTFM

Sent from my iPhone

On Jun 21, 2011, at 7:45 PM, Warren Selby  wrote:

> On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad  wrote:
> Dear Warren;
> 
> Please, keep all discussions to the list.  There's no need to email me 
> personally about this. 
> 
> 
>  
> cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone 
> load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP 
> Phone firmware files only. So what is the difference between them "the load 
> and the firmware"?
> 
> The .sgn file is basically just a zip container that the Cisco Call Manager 
> uses.  You'll want to grab the zip file, extract the contents of the file 
> into your tftp root directory.  The latest firmware that I've used was 8.5.2, 
> in which most everything I needed worked for me.  I don't know specifics 
> about the later versions of Cisco's SIP releases.
>  
> Now, when I need to do the upgrade for the Phone, then I have to determine in 
> the xml files the needed firmware?
> 
> You should have, at least with firmware 8.5.2, the following files in your 
> tftproot directory after unzipping the zip file:
> 
> apps41.8-5-2TH1-9.sbn
> cnu41.8-5-2TH1-9.sbn
> cvm41sip.8-5-2TH1-9.sbn
> dsp41.8-5-2TH1-9.sbn
> jar41sip.8-5-2TH1-9.sbn
> SIP41.8-5-2S.loads
> term41.default.loads
> term61.default.loads
> XMLDefault.cnf.xml
> SEP[_MAC-ADDR_].cnf.xml
> 
> I provide samples of the last two files on the blog post mentioned earlier.  
> The last file, that starts with SEP, should contain the actual mac address of 
> the phone you are trying to provision.  So, for example, it would be 
> SEP0003C9DD5624.cnf.xml, if the mac address of your phone was 0003.C9DD.5624. 
>  The example files are pretty much all you need, just go through them and 
> change any location specific variables (such as _USER_, _IPADDR_, or 
> _PASSWD_) to the proper values for your environment.
> 
> Once you've got your tftp server setup properly with all of the appropriate 
> config files, plug your phone in and follow the instructions at the bottom 
> part of my blog post that explain how to get the phone reflashed to the SIP 
> image and registered to your asterisk server.
> 
> 
> -- 
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com
> 
> --
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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[Warren Selby]

On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad  wrote:

> Dear Warren;
>

Please, keep all discussions to the list.  There's no need to email me
personally about this.




> cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP
> Phone load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is
> SIP IP Phone firmware files only. So what is the difference between them
> "the load and the firmware"?
>

The .sgn file is basically just a zip container that the Cisco Call Manager
uses.  You'll want to grab the zip file, extract the contents of the file
into your tftp root directory.  The latest firmware that I've used was
8.5.2, in which most everything I needed worked for me.  I don't know
specifics about the later versions of Cisco's SIP releases.


> Now, when I need to do the upgrade for the Phone, then I have to determine
> in the xml files the needed firmware?
>

You should have, at least with firmware 8.5.2, the following files in your
tftproot directory after unzipping the zip file:

apps41.8-5-2TH1-9.sbn
cnu41.8-5-2TH1-9.sbn
cvm41sip.8-5-2TH1-9.sbn
dsp41.8-5-2TH1-9.sbn
jar41sip.8-5-2TH1-9.sbn
SIP41.8-5-2S.loads
term41.default.loads
term61.default.loads
XMLDefault.cnf.xml
SEP[_MAC-ADDR_].cnf.xml

I provide samples of the last two files on the blog post mentioned earlier.
The last file, that starts with SEP, should contain the actual mac address
of the phone you are trying to provision.  So, for example, it would be
SEP0003C9DD5624.cnf.xml, if the mac address of your phone was
0003.C9DD.5624.  The example files are pretty much all you need, just go
through them and change any location specific variables (such as _USER_,
_IPADDR_, or _PASSWD_) to the proper values for your environment.

Once you've got your tftp server setup properly with all of the appropriate
config files, plug your phone in and follow the instructions at the bottom
part of my blog post that explain how to get the phone reflashed to the SIP
image and registered to your asterisk server.


-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com 
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Re: [asterisk-users] call paging interrupts call when using Mitel 5224

[Terry Brummell]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Tuesday, June 21, 2011 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call paging interrupts call when using Mitel
5224

 

Is anybody using Mitel phones? It appears that when you page a Mitel
phone using asterisk's MeetMe, the paged phone will hang up the call its
on to take the page. Thanks in advance.  

 

This does not happen to me, my call stays up.  Caller with the page gets
a busy signal.

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[asterisk-users] calleridname presentation Asterisk ==> Siemens

[Rafael dos Santos Saraiva]

Hi

I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i can't
show the callerid name in the way Asterisk ==> Siemens. I realized that
Asterisk send calleridname in format "namePresentationAllowedSimple" to
Siemens e Siemens send calleridname in format
"namePresentationAllowedExtended". I need change the format of the
calleridname in asterisk.

How to change?

Thank´s and sorry for my wrong english.
Att,
Rafael Saraiva
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[asterisk-users] Inbound SMS

[ERIC HERRON]

I know Asterisk 1.8 can send out texts via SMS()

 

Can I send Asterisk a text via a DID and it do something?

 

So far I am reading that it cannot but I do not know if there have been
updates.

 

Thanks,

-E

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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[bilal ghayyad]

Dear Warren;

It look like u have a good experience in 791x series and in selecting SIP 
formware, so I am sure you might be able to help in the following:

As u know, there are SIP firmware for Cisco phones to be used with Call Manager 
and other firmware to be used with Generic SIP Server (other than Cisco Unified 
Call Manager). Actually the firmware that start by P0S is that used for Generic 
SIP Server and that start by cmterm is used for Cisco Unified Call Manager.

Can I understand from ur blog, that u can use the files that its name start by 
cmterm to make the Cisco IP Phone to be SIP image that can be used by Asterisk? 

In my case, as the Phones are 7942G, then there are two files are available, 
really I do not know the difference between them (if u can advise):

cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone 
load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP 
Phone firmware files only. So what is the difference between them "the load and 
the firmware"?

Now, when I need to do the upgrade for the Phone, then I have to determine in 
the xml files the needed firmware?

Appreciate your kindly help.

Regards
Bilal




--
 
> > You are "supposed" to go via cisco and support
> contract BUT Google is your
> > friend (JFGI)
> >
> >
> The support contract from Cisco is only US $8.99 on
> CDW
> 
> I really hate to link to my own blog, but I do have a post
> on there that
> details how to setup a 79x1 phone using SIP firmware with
> asterisk.  Click
> the link in my signature and go to the Blog and you should
> be able to easily
> find the relevant post.
> 
> -- 
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 


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Re: [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko

[Cary Fitch]

-- 
I know we've come a long way,
We're changing day to day,
But tell me, where do the children play?

Yusuf Islam, Nov. 1970

AKA CAT STEVENS.

--


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Re: [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko

[Norbert Zawodsky]

Am 21.06.2011 18:28, schrieb Mark Deneen:

On Tue, Jun 21, 2011 at 4:12 AM, randulo  wrote:

On Tue, Jun 21, 2011 at 5:47 AM, Alex Balashov
  wrote:

I nominate this for most imaginative use of Asterisk-users of 2011.

It's already qualified to win in the grammar and spelling categories.

/r


My 3 year old. unfortunately, has sent a few messages like this in the
past.  I guess she watched me unlock the screen enough times to
memorize the key code.

-M

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Reading that, I would nominate this for the "youngest asterisk-users 
user award" ;-)


Norbert

--
I know we've come a long way,
We're changing day to day,
But tell me, where do the children play?

Yusuf Islam, Nov. 1970


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Re: [asterisk-users] how to get on hold events with AMI

[Richard Mudgett]

> I am getting events using asterisk 1.4.41. However, when I place a
> call
> on hold I do not get that event.
>
> Some of the events I am getting I show below. I wish to monitor when
> channels are placed on hold
> and taken off hold.

Set callevents=yes in sip.conf to enable Hold/Unhold AMI call events for SIP 
channels.

Richard

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[asterisk-users] how to get on hold events with AMI

[Jerry Geis]

I am getting events using asterisk 1.4.41. However, when I place a call 
on hold I do not get that event.


Some of the events I am getting I show below. I wish to monitor when 
channels are placed on hold

and taken off hold.

How do I that?

Thanks,

jerry

-


Event: Newstate
Privilege: call,all
Channel: SIP/401-0003
State: Up
CallerID: 401
CallerIDName: 
Uniqueid: 1308689635.3

Event: Newstate
Privilege: call,all
Channel: SIP/404-0002
State: Up
CallerID: 204
CallerIDName: Jerry Geis
Uniqueid: 1308689635.2

Event: Link
Privilege: call,all
Channel1: SIP/404-0002
Channel2: SIP/401-0003
Uniqueid1: 1308689635.2
Uniqueid2: 1308689635.3
CallerID1: 204
CallerID2: 401

Event: PeerStatus
Privilege: system,all
Peer: SIP/404
PeerStatus: Registered

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Re: [asterisk-users] call paging interrupts call when using Mitel 5224

[Terry Brummell]

I have a 5224 and 5220's, I will try it tonight when I get home.



From: vip killa
Sent: Tue 6/21/2011 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call paging interrupts call when using Mitel 5224


Is anybody using Mitel phones? It appears that when you page a Mitel phone 
using asterisk's MeetMe, the paged phone will hang up the call its on to take 
the page. Thanks in advance.  
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Re: [asterisk-users] SMS with Asterisk

[Warren Selby]

On Tue, Jun 21, 2011 at 1:07 PM, Steve Totaro
wrote:

> LOL at the haters.
>
>
Ahh, how often I forget that the subtleties of sarcasm are usually lost in
email...




> It would also be a whole lot easier for someone to physically feed me so my
> hands could be free to work in hostile environments, maybe an LN can bring
> me a portable toilet and make sure it is fresh, that would make everything
> so easy and easy is what we all want.
>

You're right, why should we make things easy for other people?  While we're
at it, let's toss out instructions manuals - those books just make things
easier for people to learn what they're doing, so full of answers to common
problems as they are!  And websites like wiki.asterisk.org, that has a ton
of configuration examples that just makes life easy on those bothersome
people who don't already know all the answers!  And mailing lists!  How dare
helpful contributors such as yourself or myself make things easier for
people by just giving them answers that we had to work hard to figure out
ourselves!

And yes, I'm just being sarcastic.  I don't "hate" on anyone who posts on
this list - we've all been in situations where we've needed help to figure
things out in the past.  Who among us hasn't used google to learn
something?  I may not be as experienced as some people, but I've got my
little niche that I'm good at and when someone posts questions relating to
those topics, I try to be helpful.

And occasionally, I can be a little sarcastic...  :)

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--Warren Selby, dCAP
http://www.SelbyTech.com 
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[asterisk-users] call paging interrupts call when using Mitel 5224

[vip killa]

Is anybody using Mitel phones? It appears that when you page a Mitel phone
using asterisk's MeetMe, the paged phone will hang up the call its on to
take the page. Thanks in advance.
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Re: [asterisk-users] SMS with Asterisk

[jon pounder]

Personally, I have been shot at on top the Iraqi Government building 
in the IZ from the Red Zone.  I was setting up and troubleshooting the 
Motorola Canopy WiFi system.  Just a few 7.62x39 rounds, nothing I 
would call heavy fire.




It was because you were setting up the canopy stuff, not related to 
being in Iraq. Motorola sure goofed on that abortion.





The only "Heavy Fire" I took was standing on top of one of the 
buildings at the FOB trying to trace a cable and the ricochets from 
the firing range were landing all over the place.  That happens when 
30 guys are training with AKs  and a T-Wall as the backstop.


I have deployed Asterisk systems in war zones many times, in West 
African countries, Iraq, Baltimore and South East DC.  I would 
certainly seek shelter/defensive position if there was gun play.  LOL, 
you can wish yourself into a gun fight but you cannot wish yourself out.



It would also be a whole lot easier for someone to physically feed me 
so my hands could be free to work in hostile environments, maybe an LN 
can bring me a portable toilet and make sure it is fresh, that would 
make everything so easy and easy is what we all want.


Heck, I could just set it up at the FOB and then deploy it.

At any rate, I asked the guy to post his success, so I am not sure why 
you posted, but thanks.  It only takes 10% truth to make a legend.


Thanks,
Steve T


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Re: [asterisk-users] SMS with Asterisk

[Robert Huddleston]

Hahahah Baltimore and SE DC. How about Philly too J

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Tuesday, June 21, 2011 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SMS with Asterisk

 

 

On Mon, Jun 20, 2011 at 2:48 PM, Warren Selby  wrote:

On Mon, Jun 20, 2011 at 3:52 AM, Steve Totaro 
wrote:

Two requests, not from me but the community.

1.  Don't top post


*cough*
 

2.  When you find your solution, reply to this thread so others will be
(silver) spoon fed the answers and blindly accept them without trying things
and going through a learning curve and experimentation when they find your
post in Google.


I hear some people are actually deploying their asterisk solutions in war
zones and are taking heavy fire while they're looking for answers - seems
like it would make their life a whole lot easier (and safer!) if people
posted simple responses on this list when suggestions worked for them...

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com


LOL at the haters.

1.  It was joke for those with senses of humor and know me (Randy got it),
but I top post when others do.  I bottom post when others do.  I just go
with the flow.  I am not uptight about it.

2.  I have never heard that but it may be true.  

Personally, I have been shot at on top the Iraqi Government building in the
IZ from the Red Zone.  I was setting up and troubleshooting the Motorola
Canopy WiFi system.  Just a few 7.62x39 rounds, nothing I would call heavy
fire.

The only "Heavy Fire" I took was standing on top of one of the buildings at
the FOB trying to trace a cable and the ricochets from the firing range were
landing all over the place.  That happens when 30 guys are training with AKs
and a T-Wall as the backstop.

I have deployed Asterisk systems in war zones many times, in West African
countries, Iraq, Baltimore and South East DC.  I would certainly seek
shelter/defensive position if there was gun play.  LOL, you can wish
yourself into a gun fight but you cannot wish yourself out. 


It would also be a whole lot easier for someone to physically feed me so my
hands could be free to work in hostile environments, maybe an LN can bring
me a portable toilet and make sure it is fresh, that would make everything
so easy and easy is what we all want.

Heck, I could just set it up at the FOB and then deploy it.

At any rate, I asked the guy to post his success, so I am not sure why you
posted, but thanks.  It only takes 10% truth to make a legend.

Thanks,
Steve T 

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Re: [asterisk-users] SMS with Asterisk

[Steve Totaro]

On Mon, Jun 20, 2011 at 2:48 PM, Warren Selby  wrote:

> On Mon, Jun 20, 2011 at 3:52 AM, Steve Totaro <
> stot...@asteriskhelpdesk.com> wrote:
>
>> Two requests, not from me but the community.
>>
>> 1.  Don't top post
>>
>
> *cough*
>
>
>> 2.  When you find your solution, reply to this thread so others will be
>> (silver) spoon fed the answers and blindly accept them without trying things
>> and going through a learning curve and experimentation when they find your
>> post in Google.
>>
>
> I hear some people are actually deploying their asterisk solutions in war
> zones and are taking heavy fire while they're looking for answers - seems
> like it would make their life a whole lot easier (and safer!) if people
> posted simple responses on this list when suggestions worked for them...
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 
>

LOL at the haters.

1.  It was joke for those with senses of humor and know me (Randy got it),
but I top post when others do.  I bottom post when others do.  I just go
with the flow.  I am not uptight about it.

2.  I have never heard that but it may be true.

Personally, I have been shot at on top the Iraqi Government building in the
IZ from the Red Zone.  I was setting up and troubleshooting the Motorola
Canopy WiFi system.  Just a few 7.62x39 rounds, nothing I would call heavy
fire.

The only "Heavy Fire" I took was standing on top of one of the buildings at
the FOB trying to trace a cable and the ricochets from the firing range were
landing all over the place.  That happens when 30 guys are training with
AKs  and a T-Wall as the backstop.

I have deployed Asterisk systems in war zones many times, in West African
countries, Iraq, Baltimore and South East DC.  I would certainly seek
shelter/defensive position if there was gun play.  LOL, you can wish
yourself into a gun fight but you cannot wish yourself out.

It would also be a whole lot easier for someone to physically feed me so my
hands could be free to work in hostile environments, maybe an LN can bring
me a portable toilet and make sure it is fresh, that would make everything
so easy and easy is what we all want.

Heck, I could just set it up at the FOB and then deploy it.

At any rate, I asked the guy to post his success, so I am not sure why you
posted, but thanks.  It only takes 10% truth to make a legend.

Thanks,
Steve T
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Re: [asterisk-users] DTMF begin ignored

[Marcelo Ellmann Clemente]

I'm not Asterisk expert but I had some DTMF problems in the recent past...

This message is called on main/channel.c everytime a DTMF is received, and, 
afaik, this is not an error, it Asterisk ignores the first milisenconds of the 
DTMF to distinguish between a real DTMF and any sound in the same frequency of 
that DTMF that may come in the call (and is not a real DTMF).

I was having some problems with the E1 card I was using and no DTMF mas 
detected at all. I've fixed it (using another card :p) and I've got a full 
system in production (with a quite heavy load of calls and DTMF inputs) and 
that DTMF begin ignored is quite normal during all key presses.

Can you show a piece of code of the extension/context which is reading this 
DTMF? Maybe that would help!

Good luck,
--- 
Marcelo Ellmann 
Freeddom Tecnologia e Serviços S/A
+55 11 52133200 Ramal 1016




- Original Message -
From: "vip killa" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Tuesday, 21 June, 2011 2:07:37 PM
Subject: [asterisk-users] DTMF begin ignored


we've been getting complaints that DTMF is not working, i checked full log for 
a call that they claimed DTMF didnt work, I noticed this: 
DTMF begin '7' received 
DTMF begin ignored 
DTMF end '7' received 
DTMF end passthrough '7' 


why is the "DTMF begin ignored" called? 
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Re: [asterisk-users] menu issue

[khalid touati]

Hi I wanted to help out with dial plan, but it's not obvious what you want
to achieve, also I do recommend to read the chapter that talks about
contexts and dialplan from future of asterisk book. but if you're in rush
just try to make clear how you want your system to behave and i'll be glad
to help.

On Tue, Jun 21, 2011 at 9:22 AM, salaheddine elharit <
salah.elharit...@gmail.com> wrote:

>  Hello
>
>
>
> I have created the menu below, with this menu when I call 520460XXX I can
> hear the welcome message [home] context and when I press the # I can go to
> the [menu] context and hear the menu message
>
>
>
> When I press 1 in order to go to the [call] I can hear the call message
>
>
> Now I have 2 client sip 222 and 223 with x-lite, how can I do in order to
> receive the call in 223 with [call] and receive a call in 222 with [support]
>
> thanks and best regards
>
>
> exten => 520460XXX,1,Ringing()
> exten => 520460XXX,2,Wait(4)
> exten => 520460XXX,3,Goto(home,s,1)
>
> [home]
> exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
> exten => s,2,Background(${sounds_path}welcome)
> exten => s,3,WaitExten(10)
>
> exten => #,1,Goto(menu,s,1)
> exten => i,1,Playback(${sounds_path}error-key)
> exten => t,1,Goto(home,s,1)
>
> [menu]
> exten => s,1,Background(${sounds_path}menu)
>
> exten => 0,1,Goto(menu,s,1)
> exten => 1,1,Goto(call,s,1)
> exten => 2,1,Goto(support,s,1)
>
> exten => i,1,Playback(${sounds_path}error-key)
> exten => i,2,Goto(call,s,1)
> exten => t,1,Goto(call,s,1)
> [call]
> exten => s,1,Background(${sounds_path}call)
>
> exten => 0,1,Goto(menu,s,1)
> exten => 223,1,Dial(SIP/${EXTEN},20,tr)
>
> exten => i,1,Playback(${sounds_path}error-key)
> exten => t,1,Goto(appel,s,1)
>
>
> [support]
> exten => s,1,GoToIfTime(09:00-17:00|mon-fri|*|*?s,4)
> exten => s,2,Playback(${sounds_path}no-relation-support)
> exten => s,3,Goto(menu,s,1)
> exten => s,4,Playback(${sounds_path}relation-support)
> exten => s,5,Queue(default)
> exten => t,1,Hangup()
>
>
>  2011/6/20 Warren Selby 
>
>>   On Mon, Jun 20, 2011 at 12:17 PM, salaheddine elharit <
>> salah.elharit...@gmail.com> wrote:
>>
>>> [home]
>>> exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
>>> exten => s,2,Background(${sounds_path}welcome)
>>> exten => #,1,Goto(menu,s,1)
>>> exten => i,1,Playback(${sounds_path}error-key)
>>> exten => t,1,Goto(home,s,1)
>>>
>>>
>> You need to add the following to the [home] context:
>>
>> exten => s,3,WaitExten(10)
>>
>> which will cause the call to wait 10 seconds for input, otherwise it will
>> timeout and go to the 't' extension.  The way you currently have it, the
>> call will end after the Background() app finishes playing because it has no
>> additional steps and nothing that will tell it to go to the 't' extension.
>>
>> Also, consider switching your dialplan priorities away from "1,2,3..." and
>> go to "1,n,n,n..." as this reduces headaches in the longrun.
>>
>> --
>> Thanks,
>> --Warren Selby, dCAP
>> http://www.SelbyTech.com 
>>
>>
>> --
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>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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> _
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-- 
Abdullah
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[asterisk-users] DTMF begin ignored

[vip killa]

we've been getting complaints that DTMF is not working, i checked full log
for a call that they claimed DTMF didnt work, I noticed this:
DTMF begin '7' received
DTMF begin ignored
DTMF end '7' received
DTMF end passthrough '7'

why is the "DTMF begin ignored" called?
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Re: [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko

[Mark Deneen]

On Tue, Jun 21, 2011 at 4:12 AM, randulo  wrote:
> On Tue, Jun 21, 2011 at 5:47 AM, Alex Balashov
>  wrote:
>> I nominate this for most imaginative use of Asterisk-users of 2011.
>
> It's already qualified to win in the grammar and spelling categories.
>
> /r


My 3 year old. unfortunately, has sent a few messages like this in the
past.  I guess she watched me unlock the screen enough times to
memorize the key code.

-M

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Re: [asterisk-users] Call to *2*999... : IP-phone configration

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On 06/21/2011 08:37 AM, Jonas Kellens wrote:

> At the moment, I don't really know what I'm looking for. So if anyone
> knows how to do it in a Cisco, Grandstream, Yealink or Snom IP-phone I
> can find out myself what settings to look for in other IP-phones.

On a Polycom phone you'd be looking for the 'digitmap' to make the
adjustments.

Barry

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Version: GnuPG v1.4.5 (GNU/Linux)

iD8DBQFOAMPhCFu3bIiwtTARAtfWAJ0YD3MYaYHCeFabf6QdTvc8oRCnFQCfbgtk
8Mfp7Pmo/G4FTsueqWNpdEc=
=6eo/
-END PGP SIGNATURE-

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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[Dan Austin]

You do not need sccp.conf if you are not using chan_sccp.
It has different features(bugs) than chan_skinny, but yes
it would also reset the phones (if it supports reload, and 
I have no idea if it does).

Also if the phone is in a call it will not reset until after
the user hangs up.  Reloading the channel triggers a soft reset
that causes the phone to request its configuration, which may have
changed.  

Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Tuesday, June 21, 2011 1:15 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

Dear Dan;

I have to do something in the compilation to have chan_sccp? Because, I do not 
have this channel and I have only chan_skinny.

Even in the /usr/lib/asterisk/module/, I did not find chan_sccp.

Maybe that is the reason why I do not have the sccp.conf file?

So, using the sccp channel, will also face the same problem that the phones 
will restarted if I did reload?

Regards
Bilal


--- On Mon, 6/20/11, Dan Austin  wrote:

> From: Dan Austin 
> Subject: RE: Cisco IP Phones and Skinny in asterisk
> To: "bilal ghayyad" 
> Date: Monday, June 20, 2011, 7:09 PM
> It would be best to keep this on the
> list, I just had not
> had a chance to reply yet.
> 
> Your first issue is just how the SCCP protocol works. 
> Every
> keypress is relayed to the server, so the phones must
> maintain
> an active connection to the PBX.  You can avoid this
> by just
> reloading the modules you update and not the whole PBX-
>     ie- sip reload or module reload
> chan_sip
> 
> The second issue is likely a firmware issue on the phone,
> and
> Likely one where the phone software is too new.  You
> might also
> Not have the correct definition in skinny.conf
> 
> I did use chan_sccp years ago, but have not kept up with
> it.
> The configuration should be with the source package for
> that
> channel.  The configuration is similar, but you cannot
> rename
> the files as there are key differences.
> 
> Dan
> 
> -Original Message-
> From: bilal ghayyad [mailto:bilmar...@yahoo.com]
> 
> Sent: Monday, June 20, 2011 3:40 PM
> To: Dan Austin
> Subject: Re: Cisco IP Phones and Skinny in asterisk
> 
> Dear Dan;
> 
> Because you are using skinny with your Cisco IP Phones in
> the office, so I beleive you might help me really to resolve
> my problem (please).
> 
> First of all, are u using skinny channel or sccp channel?
> 
> Actually, I tried skinny and I faced two major problems (so
> if I am going to face same problems in sccp then no need to
> use sccp, so please advise).
> 
> The two problems that I faced them are:
> 
> 1) When I do reload then the skinny channel is reloaded and
> that will cause a restart for the Cisco IP Phones (that are
> registered to skinny channel). Is the same thing happening
> with u when u r using sccp channel?
> 
> 2) When I called the Phone, it is ringing, when we pickup
> the handset to answer the call, we hear
> t and we do not hear what source is
> talking and source does not hear us even .. but if we select
> music on hold, then caller will hear the music. Also, when
> we tried to use the Ciscp IP Phone to place a call, while we
> are dialing, the too tone is always existed and
> it is ringing at destination but no voice (always
> t).
> 
> So if I used sccp then I will not face these problems?
> 
> From the other side, if I need to use sccp (if we assumed
> the above problems are not existed) then can u please help
> for below:
> 
> 1) If i used sccp and I gave the IP Phone the IP address
> TFTP server, and no configuration files were existed on
> TFTP, then it will register on the asterisk sccp channel?
> 
> 2) The sccp.conf file, where I can find it? Is it the same
> as the skinny.conf file?
> 
> 3) To use sccp instead of the skinny channel, all what I
> need is to unload the skinny from the modules.conf file and
> load the sccp channel in the modules.conf, and I can use the
> skinny.conf file for the configuration? About the firmware
> on the Phone, it will stay the same?
> 
> I appreciate the kindly help please.
> Regards
> Bilal
> 

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Re: [asterisk-users] "pickupsound = beep" kills call pickup in Asterisk 1.8.4.2

[Richard Mudgett]

> I have discovered that if I enable pickupsound = beep in
> features.conf,
> if I try to do a pickup with *8, the calling channel keeps on ringing,
> while the phone where I pick-up from shows that the call has been
> answered (I don't know where though). Also, it seems to completely
> bugger up my outgoing IAX trunk (I really can't see the connection, as
> I'm doing pick-up for a SIP channel). I can only shut Asterisk down
> with
> killall asterisk -s9 - nothing else works.
> 
> I've tried starting the console with asterisk -rvv - but there is
> nothing unusual there.
> 
> Could someone please confirm this behaviour on their box, before I go
> and submit a bug - in case I am doing something wrong?
> 
> As soon as I comment out "pickupsound = beep" - everything works just
> fine and I can do call pickup with *8.
> 
> Sebastian

I think this problem is already fixed in the SVN 1.8 branch.
See https://issues.asterisk.org/jira/browse/ASTERISK-17264

Richard

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Re: [asterisk-users] Problem with ReceiveFAX app from FFA

[khalid touati]

@ Bryant: thanks so much for the interesting figure of use.


> Why do so may people think their problems are unique. Many people use FFA
> and spandsp. They all come across this. The issue is widely known, well
> understood, and not at all strange once you think about it.
>
> Steve
>
>
 @ Steve: don't get that mad dude, my impression is only My impression and
it only affects me, so nothing to worry about, i'd rather discuss asterisk
issues instead of discussing my impression, but thanks for your help.

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-- 
Abdullah
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Re: [asterisk-users] Problem with ReceiveFAX app from FFA

[Steve Underwood]

On 06/21/2011 09:12 PM, khalid touati wrote:
Ok, for the variables, I can retrieve some of them like the caller 
number and so on (I would assume that all the variables that last for 
duration of call are there), but I still think that I sould not use 
the h extension to continue after ReceiveFAX use, it's like not a lot 
of people use FFA, moreover very few came accross such an issue which 
is fine.
Why do so may people think their problems are unique. Many people use 
FFA and spandsp. They all come across this. The issue is widely known, 
well understood, and not at all strange once you think about it.


Steve

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Re: [asterisk-users] Asterisk call limitation

[satish patel]

check this out:  
http://comments.gmane.org/gmane.comp.telephony.pbx.asterisk.user/258425 

> From: kche...@xplorium.com
> To: asterisk-users@lists.digium.com
> Date: Tue, 21 Jun 2011 13:25:39 +0300
> Subject: Re: [asterisk-users] Asterisk call limitation
> 
> Any update ?
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
> Chehab
> Sent: Tuesday, June 21, 2011 12:40 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Asterisk call limitation
> 
> 
> The problem remains  even when 
> 
> I add to /etc/init.d/asterisk
> ulimit -n 65536
> 
> [root@localhost ~]# ulimit -a
> core file size  (blocks, -c) 0
> data seg size   (kbytes, -d) unlimited
> scheduling priority (-e) 0
> file size   (blocks, -f) unlimited
> pending signals (-i) 65536
> max locked memory   (kbytes, -l) 32
> max memory size (kbytes, -m) unlimited
> open files  (-n) 1024
> pipe size(512 bytes, -p) 8
> POSIX message queues (bytes, -q) 819200
> real-time priority  (-r) 0
> stack size  (kbytes, -s) 10240
> cpu time   (seconds, -t) unlimited
> max user processes  (-u) 65536
> virtual memory  (kbytes, -v) unlimited
> file locks  (-x) unlimited
> [root@localhost ~]#
> 
> -Original Message-
> From: Khaled W. Chehab [mailto:kche...@xplorium.com]
> Sent: Tuesday, June 21, 2011 12:25 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Asterisk call limitation
> 
> Can  you please specify more 
> 
> 1-how to set the ulimit on
> [root@localhost ~]# ulimit
> unlimited
> [root@localhost ~]# ulimit --help
> -bash: ulimit: --: invalid option
> ulimit: usage: ulimit [-SHacdfilmnpqstuvx] [limit]
> -
> How to set the ulimit command on in  /etc/init.d/asterisk Since there is  no
> parameter for ulimit in the file
> 
> Thanks in advance
> 
> Regards
> 
> 
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel
> Sent: Tuesday, June 21, 2011 12:15 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk call limitation
> 
> Oh! Wait you set ulimit for running shellYou should set ulimit on  
> asterisk. Also you can set ulimit command on asterisk startup file /
> etc/init.d/asterisk and restart asterisk also you can set in limit.conf file
> 
> I had this issue before and I solved that way.
> 
> --
> Sent from my iPhone
> 
> On Jun 20, 2011, at 4:47 PM, "Khaled W. Chehab" 
> wrote:
> 
> >
> > I tried the ulimit
> >
> > [root@localhost ~]# ulimit
> > Unlimited
> >
> > Then
> > sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> > noservice)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> > SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
> >
> 

Re: [asterisk-users] Problem with ReceiveFAX app from FFA

[Bryant Zimmerman]

 From: "khalid touati" 
Sent: Tuesday, June 21, 2011 9:12 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] Problem with ReceiveFAX app from FFA

Ok, for the variables, I can retrieve some of them like the caller number 
and so on (I would assume that all the variables that last for duration of 
call are there), but I still think that I sould not use the h extension to 
continue after ReceiveFAX use, it's like not a lot of people use FFA, 
moreover very few came accross such an issue which is fine.

 


Here is a receivefax example. Note this is not a complete example just a 
snip. You have to use the "h" extension if you really want to make it all 
work. Don't fight it just do it. This is a standard process very similar to 
how you need to handle returns from "Dial" comands.  

f/F option is a special patch written by Kevin @ digium and will not be in 
the distro unitl 10.x
FAX-MASTER_CHK-FAILED(${CALLERID(number)) FAX-MASTER_DO-FAILED()  are 
database storage macro that stores and get's failed fax attempts so I can 
force a roll back. Store the fax vars in a database on success and fail. 
This lets me notify the user for either case.  

[fax_inbound_efax]
exten => PFax,1,Set(SIP_CODEC=ulaw)
exten => PFax,n,Set(l_faxoptions=f)
exten => 
PFax,n,Set(l_faxhasfailed=${FAX-MASTER_CHK-FAILED(${CALLERID(number)},${p_Ca
llSrcTrunk})})
exten => PFax,n,GotoIf($[${l_faxhasfailed}>0]?audioonly:tryt38)
exten => PFax,n(audioonly),Set(l_faxoptions=F)
exten => PFax,n(tryt38),Answer()
exten => PFax,n,Wait(2)
exten => PFax,n,Set(l_faxFile_Base=${STRFTIME(,,%Y%m%d-%H%M)}_${RAND(1)})
exten => PFax,n,Set(l_faxFile_Path=/var/spool/fax_in/)
exten => PFax,n,Set(l_faxFile_FullName=fax_${l_faxFile_Base}.tiff)
exten => PFax,n,Set(l_faxFile=${l_faxFile_Path}${l_faxFile_FullName})
exten => PFax,n,ReceiveFAX(${l_faxFile},${l_faxoptions}) 

exten => h,1,NoOp(Do Fax Hangup)
exten => h,n,Goto(Do-${FAXOPT(status)},1) 

exten => Do-SUCCESS,1,NoOp(Fax Success)
exten => Do-SUCCESS,n,Goto(Do-Store,1)
exten => Do-SUCCESS,n,NoOp(Return from System) 

exten => Do-FAILED,1,NoOp(Fax Failed)
exten => 
Do-FAILED,n,GotoIf($[${l_faxoptions}=f]?DoLogFallback:DoNoFallback)
exten => 
Do-FAILED,n(DoLogFallback),Set(FAX-MASTER_DO-FAILED()=${CALLERID(number)},${
p_CallSrcTrunk})
exten => Do-FAILED,n(DoNoFallback),Goto(Do-Store,1) 

exten => Do-Store,1,NoOp(Store Fax Data)
exten => Do-Store,n,GotoIf($["${FAXOPT(rate)}"=""]?DoRate0:DoRate)
exten => Do-Store,n(DoRate0),Set(l_storeRate=0)
exten => Do-Store,n,Goto(DoDisplay)
exten => Do-Store,n(DoRate),Set(l_storeRate=${FAXOPT(rate)})
exten => Do-Store,n(DoDisplay),NoOp(DT = ${STRFTIME(,,%Y-%m-%d %H:%M:%S)})
exten => Do-Store,n,NoOp(Fax Base = ${l_faxFile_Base})
exten => Do-Store,n,NoOp(Switch ID = ${gbl_switchid})
exten => Do-Store,n,NoOp(Account Code = ${gbl_actnumber})
exten => Do-Store,n,NoOp(Line Code  = ${gbl_actlineid})
exten => Do-Store,n,NoOp(Caller ID Num = ${CALLERID(number)})
exten => Do-Store,n,NoOp(Caller ID Name = ${CALLERID(name)})
exten => Do-Store,n,NoOp(File Path = ${l_faxFile_Path})
exten => Do-Store,n,NoOp(File Name = ${l_faxFile_FullName})
exten => Do-Store,n,NoOp(opt emc = ${FAXOPT(ecm)})
exten => Do-Store,n,NoOp(opt filename = ${FAXOPT(filename)})
exten => Do-Store,n,NoOp(opt localstationid = ${FAXOPT(localstationid)})
exten => Do-Store,n,NoOp(opt headerinfo = ${FAXOPT(headerinfo)})
exten => Do-Store,n,NoOp(opt remotestationid = ${FAXOPT(remotestationid)})
exten => Do-Store,n,NoOp(opt maxrate = ${FAXOPT(maxrate)})
exten => Do-Store,n,NoOp(opt minrate = ${FAXOPT(minrate)})
exten => Do-Store,n,NoOp(opt rate = ${l_storeRate})
exten => Do-Store,n,NoOp(opt pages = ${FAXOPT(pages)})
exten => Do-Store,n,NoOp(opt resolution = ${FAXOPT(resolution)})
exten => Do-Store,n,NoOp(opt error = ${FAXOPT(error)})
exten => Do-Store,n,NoOp(opt status = ${FAXOPT(status)})
exten => Do-Store,n,NoOp(opt statusstr = ${FAXOPT(statusstr)})
exten => Do-Store,n,"Run some kind of macro or storage/email script here" 
Good luck

Bryant
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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[Robert Huddleston]

If memory serves isn't that support contract include broken phones / parts
too?

 

I thought I read that if my phone Is broken - it is covered

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, June 20, 2011 9:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

 

On Mon, Jun 20, 2011 at 6:10 PM, Robert-iPhone 
wrote:

You are "supposed" to go via cisco and support contract BUT Google is your
friend (JFGI)


The support contract from Cisco is only US $8.99 on CDW

I really hate to link to my own blog, but I do have a post on there that
details how to setup a 79x1 phone using SIP firmware with asterisk.  Click
the link in my signature and go to the Blog and you should be able to easily
find the relevant post.  

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

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Re: [asterisk-users] menu issue

[salaheddine elharit]

Hello



I have created the menu below, with this menu when I call 520460XXX I can
hear the welcome message [home] context and when I press the # I can go to
the [menu] context and hear the menu message



When I press 1 in order to go to the [call] I can hear the call message


Now I have 2 client sip 222 and 223 with x-lite, how can I do in order to
receive the call in 223 with [call] and receive a call in 222 with [support]

thanks and best regards


exten => 520460XXX,1,Ringing()
exten => 520460XXX,2,Wait(4)
exten => 520460XXX,3,Goto(home,s,1)

[home]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,2,Background(${sounds_path}welcome)
exten => s,3,WaitExten(10)
exten => #,1,Goto(menu,s,1)
exten => i,1,Playback(${sounds_path}error-key)
exten => t,1,Goto(home,s,1)

[menu]
exten => s,1,Background(${sounds_path}menu)
exten => 0,1,Goto(menu,s,1)
exten => 1,1,Goto(call,s,1)
exten => 2,1,Goto(support,s,1)
exten => i,1,Playback(${sounds_path}error-key)
exten => i,2,Goto(call,s,1)
exten => t,1,Goto(call,s,1)
[call]
exten => s,1,Background(${sounds_path}call)
exten => 0,1,Goto(menu,s,1)
exten => 223,1,Dial(SIP/${EXTEN},20,tr)
exten => i,1,Playback(${sounds_path}error-key)
exten => t,1,Goto(appel,s,1)


[support]
exten => s,1,GoToIfTime(09:00-17:00|mon-fri|*|*?s,4)
exten => s,2,Playback(${sounds_path}no-relation-support)
exten => s,3,Goto(menu,s,1)
exten => s,4,Playback(${sounds_path}relation-support)
exten => s,5,Queue(default)
exten => t,1,Hangup()


2011/6/20 Warren Selby 

>  On Mon, Jun 20, 2011 at 12:17 PM, salaheddine elharit <
> salah.elharit...@gmail.com> wrote:
>
>> [home]
>> exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
>> exten => s,2,Background(${sounds_path}welcome)
>> exten => #,1,Goto(menu,s,1)
>> exten => i,1,Playback(${sounds_path}error-key)
>> exten => t,1,Goto(home,s,1)
>>
>>
> You need to add the following to the [home] context:
>
> exten => s,3,WaitExten(10)
>
> which will cause the call to wait 10 seconds for input, otherwise it will
> timeout and go to the 't' extension.  The way you currently have it, the
> call will end after the Background() app finishes playing because it has no
> additional steps and nothing that will tell it to go to the 't' extension.
>
> Also, consider switching your dialplan priorities away from "1,2,3..." and
> go to "1,n,n,n..." as this reduces headaches in the longrun.
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 
>
>
> --
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Re: [asterisk-users] Problem with ReceiveFAX app from FFA

[khalid touati]

Ok, for the variables, I can retrieve some of them like the caller number
and so on (I would assume that all the variables that last for duration of
call are there), but I still think that I sould not use the h extension to
continue after ReceiveFAX use, it's like not a lot of people use FFA,
moreover very few came accross such an issue which is fine.
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[asterisk-users] asteriks and ntt Japan

[Cristian Leonte]

Hi All,

Reposting this since since i was not a member of the list when i first sent
it

I'm trying to configure an asterisk serrver with a NTT T1 line, i get  the
error 'no D-Channel found', i searched the net and seems that NTT Japan uses
a special switch type called NTT (
http://en.wikipedia.org/wiki/Primary_Rate_Interface) there was sa patch in
here 
http://www.spanky-world.com/lab/asterisk/asterisk_lab_content_03.html(japanese
only, use google translate) but is for an older version of
asterisk. Do you have an idea on how this can be fixed?

asterisk-1.6.2.13-0
libpri-1.4.11.4-0
and a digium card Digium TE121


Thank you for your help

Cheers,
Cristi


-- 
Cristian Leonte
cristian.leo...@gmail.com
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[asterisk-users] Call to *2*999... : IP-phone configration

[Jonas Kellens]

Hello list,

I am unable to call *2*999... because my phone automatically sends the 
number after I press *. So my IP-phone calls *2.


Now this is a Cisco, but that's not my question. Does anyone know what 
setting I need to adjust so my phone (but actually any IP-phone) accepts 
an * in the middle of a number ?


At the moment, I don't really know what I'm looking for. So if anyone 
knows how to do it in a Cisco, Grandstream, Yealink or Snom IP-phone I 
can find out myself what settings to look for in other IP-phones.



Kind regards,
Jonas.
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Re: [asterisk-users] Inbound CallerID displays asterisk

[ERIC HERRON]

I set verbose to 10. I will let you know if capture it.

 

I would like to elaborate as well.

 

We use this MTE to test polycoms and routers and allow clients to demo the
hardware.

 

It's the polycoms that are displaying the "asterisk" caller ID.

 

There is also no inbound route to the tenant.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, June 20, 2011 10:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inbound CallerID displays asterisk

 

On Mon, Jun 20, 2011 at 6:33 PM, ERIC HERRON  wrote:

I have an asterisk 1.4.26 mte running.

 

Sometimes inbound caller ID displays "asterisk"

 

These calls do not show up on the CLI nor the CDR.

 

I read somewhere that these are asterisk hack attempts.

 

Is this true? 

 

What is the best way to defend from this?

 

I know a secure password and all but the client is getting annoyed with the
random inbound calls.

 





Crank you CLI verbosity up to 10 or so and wait for the next time this
happens.  You should see SOMETHING on the CLI during the call.  Post that
output to the list and we can help you from there.

This does not always indicate someone attempting to hack you, I've seen this
occur when there are line errors on FXO devices (among other things).

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

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[asterisk-users] Voice recognition recommendations?

[David Cunningham]

Hi all,

We have a project involving voice recognition, and will need a vocabulary of
10,000 words (actually names).

Can anyone recommend a product that works with Asterisk?

Thanks,

-- 
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
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Re: [asterisk-users] Asterisk call limitation

[Khaled W. Chehab]

Any update ?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Tuesday, June 21, 2011 12:40 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk call limitation


The problem remains  even when 

I add to /etc/init.d/asterisk
ulimit -n 65536

[root@localhost ~]# ulimit -a
core file size  (blocks, -c) 0
data seg size   (kbytes, -d) unlimited
scheduling priority (-e) 0
file size   (blocks, -f) unlimited
pending signals (-i) 65536
max locked memory   (kbytes, -l) 32
max memory size (kbytes, -m) unlimited
open files  (-n) 1024
pipe size(512 bytes, -p) 8
POSIX message queues (bytes, -q) 819200
real-time priority  (-r) 0
stack size  (kbytes, -s) 10240
cpu time   (seconds, -t) unlimited
max user processes  (-u) 65536
virtual memory  (kbytes, -v) unlimited
file locks  (-x) unlimited
[root@localhost ~]#

-Original Message-
From: Khaled W. Chehab [mailto:kche...@xplorium.com]
Sent: Tuesday, June 21, 2011 12:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Asterisk call limitation

Can  you please specify more 

1-how to set the ulimit on
[root@localhost ~]# ulimit
unlimited
[root@localhost ~]# ulimit --help
-bash: ulimit: --: invalid option
ulimit: usage: ulimit [-SHacdfilmnpqstuvx] [limit]
-
How to set the ulimit command on in  /etc/init.d/asterisk Since there is  no
parameter for ulimit in the file

Thanks in advance

Regards



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel
Sent: Tuesday, June 21, 2011 12:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk call limitation

Oh! Wait you set ulimit for running shellYou should set ulimit on  
asterisk. Also you can set ulimit command on asterisk startup file /
etc/init.d/asterisk and restart asterisk also you can set in limit.conf file

I had this issue before and I solved that way.

--
Sent from my iPhone

On Jun 20, 2011, at 4:47 PM, "Khaled W. Chehab" 
wrote:

>
> I tried the ulimit
>
> [root@localhost ~]# ulimit
> Unlimited
>
> Then
> sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150
>
> SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:4   Up  Playback(ss- 
> noservice)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/127.0.0.1:5061-0 s@from-trunk:5   Up  SayAlpha(2005)
>
> SIP/

[asterisk-users] AMI Suddenly not giving full response to 'Command'

[Ishfaq Malik]

We have a web server that connects to our asterisk server via the AMI
for simple polling of data.

This was all working fine yesterday but now, possibly after an asterisk
restart yesterday, the Command type command are not yielding any
results.

Here's an example, which is from a log file of the full response the web
server is receiving

-- Call started: 21/06/2011 11:00:53 --
action: command
command: sip show peers like 0001

Response: Follows
Privilege: Command
--END COMMAND--


Usually I would have a list of the peers between the 
Privilege: Command
and the 
--END COMMAND--

lines.

No changes have been made to the manager.conf between yesterday and
today. And yes, matching peers do exist. 

Other types of commands work fine such as:
-- Call started: 21/06/2011 09:42:46 --
action: MailboxCount
mailbox: 1030@default

Response: Success
Message: Mailbox Message Count
Mailbox: 1030@default
NewMessages: 2
OldMessages: 0

This is using Asterisk 1.4.17~dfsg-2ubuntu1

Has anyone ever experienced anything like this before?
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 01:12 PM, Karsten Wemheuer wrote:
> Hi randall,
> 
> Am Mittwoch, den 01.06.2011, 10:00 +0200 schrieb randall:
>>> > i get the following errors:
>>> > pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel
>>> > of span 2
>>> >
>>> >
>>> > Your telco provider has crc on or off , that is not matching with
>>> > your server cross check with them.
>>> > and this problem solve 4 problems
>>> >
>>> >
>>> thanks for the reply,
>>>
>>> what is crc (same as crc-4?)
>>> and where can i set this?
>>>
>>> same crc or crc4
>>>
>>> --
>> adding crc as follows, span=1,1,0,ccs,ami,crc, causes DAHDI to not load
>> at all
> 
> As I can see from Your first post, You are using BRI in
> point-to-multipoint mode. On BRI lines there is nothing like CRC/CRC4
> and that is the reason, why the config is not loading any more.
> 
> On a PTMP line there may be some CRC-errors from time to time, when the
> provider shuts down the line, which is normal in some countries. But
> this has nothing to do with Your initial problem.
> 
> Unfortunately I don't know a solution for Your problem. It may be a
> hardware issue.

is there a way to trace down the hardware causing this?


> 
> HTH,
> 
> Karsten
> 
> 
> 
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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[randall]

On 06/01/2011 06:28 PM, Steve Davies wrote:
> On 1 June 2011 15:10, randall  wrote:
>> On 06/01/2011 03:55 PM, randall wrote:
>>> On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
 On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
> Hi all,
>
> After running fine for a few months now asterisk seems to hang
> frequently , still functioning but the DAHDI channels seem busy  (users
> report a busy signal when calling or being called)
>
> A reboot will allow it to run for another day or maybe 2  or 3 till the
> problem occurs again.
>
>
> running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
> 2.6.32-5-686
>
> i get the following errors:
> pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of 
> span 2
>
> (happens on all 4 spans)
>
> and the following in dmesg:
> [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
> [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:current
> packet[0..2]: 55 55 FC
> [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71
>
>
> Channel 0/1, span 1 got hangup, cause 18

 Is this happening in the middle of a call? Or only a while after the
 call ended?

>>>
>>> the "bad fcs" messages seem to happen random
>> there seems to be a relation indeed, have seen them happen randomly
>> quite spurious, but they indeed tend to happen a while after the call is
>> made.
>>>
>>> the hangup happens when a call through DAHDI is attempted,
>>> (usually after it has been working fine for a while a day or 2)
> 
> In my experience, FCS errors are caused by line quality issues, and
> usually (not always) are in the telco's equipment. If they are only
> happening occasionally, it may be a marginal, but mostly-OK signal on
> the wire.
> 
> Do you also get occasional poor-quality audio on calls? The issue will
> happen more when a call is being setup, or is progressing because
> there are more frames being exchanged when a call is in progress.

audio quality seems to be fine ( e.g. had no complaints about that)

> 
> I have also seen a bad component or dry solder on a voice card cause
> this, and even a badly made ISDN cable can be part of the problem. If
> none of that helps, I would ask the telco to put a trace on the line.

would these FCS errors cause the system to become unresponsive leading
to "Channel 0/1, span 1 got hangup, cause 18" after a period of working
fine, or is this an unrelated issue?


> 
> Hope that helps,
> Steve
> 
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[asterisk-users] dropped calls on android voip connection

[Eric Smith]

Hi 

When using an extension to my android gingerbread nexus one,
calls drop after a n minutes of call due as per the following
 [Jun 21 09:34:37]   == Begin MixMonitor Recording SIP/nexusone-0a39
 [Jun 21 09:34:37] -- Executing [00310@default:4] Dial(" ...
 [Jun 21 09:34:37] -- Called 45xx:xx...@iax.voie.xl/00x ...
 [Jun 21 09:34:37] -- Call accepted by 84.243.247.100 (format al ...
 [Jun 21 09:34:37] -- Format for call is alaw
 [Jun 21 09:34:38] -- IAX2/4506-8090 is making progress passing
 [Jun 21 09:34:42] -- IAX2/4506-8090 answered SIP/nexusone-0a39a .c: 
Context 'macro-notifymobile' for macro 'notifymobile' lacks 's'
 logger.c: [Jun 21 09:35:44] -- Saved useragent "SIPAUA/0.1.001" for peer
 logger.c: [Jun 21 09:42:39] -- Unregistered SIP 'nexusone'
 logger.c: [Jun 21 09:42:39] -- Unregistered SIP 'nexusone'
 logger.c: [Jun 21 09:42:39] -- Registered SIP 'nexusone' at 77.165.58.113
 logger.c: [Jun 21 09:42:39] -- Saved useragent "SIPAUA/0.1.001" for peer
 logger.c: [Jun 21 09:45:57] -- Unregistered SIP 'nexusone'
 logger.c: [Jun 21 09:45:58] -- Registered SIP 'nexusone' at 77.165.58.113
 logger.c: [Jun 21 09:45:58] -- Saved useragent "SIPAUA/0.1.001" for peer
 logger.c: [Jun 21 10:13:23] -- Unregistered SIP 'nexusone'
 logger.c: [Jun 21 10:13:23] -- Unregistered SIP 'nexusone'

ex sip.conf;
[nexusone]
type=friend
context=default
username=nexusone
secret=Iainttelling
nat=no  
  disallow=all
allow=gsm   
  

*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status  
 
nexusone/nexusone  (Unspecified)D  0Unmonitored 

How could I prevent these calls from dropping?

---
Eric Smith

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Re: [asterisk-users] menu issue

[salaheddine elharit]

thanks a lot now it's work correctly :)

2011/6/20 Warren Selby 

>  On Mon, Jun 20, 2011 at 12:17 PM, salaheddine elharit <
> salah.elharit...@gmail.com> wrote:
>
>> [home]
>> exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
>> exten => s,2,Background(${sounds_path}welcome)
>> exten => #,1,Goto(menu,s,1)
>> exten => i,1,Playback(${sounds_path}error-key)
>> exten => t,1,Goto(home,s,1)
>>
>>
> You need to add the following to the [home] context:
>
> exten => s,3,WaitExten(10)
>
> which will cause the call to wait 10 seconds for input, otherwise it will
> timeout and go to the 't' extension.  The way you currently have it, the
> call will end after the Background() app finishes playing because it has no
> additional steps and nothing that will tell it to go to the 't' extension.
>
> Also, consider switching your dialplan priorities away from "1,2,3..." and
> go to "1,n,n,n..." as this reduces headaches in the longrun.
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 
>
>
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[asterisk-users] xorcom asterisk patch for sending rtp stream to remote oreka server

[Marcus Kvarsell]

Hello!

 

Im trying to setup the xorcom asterisk patch to be able of sending rtp
setrema to an oreka voip recording server but I get error messages.

 

"

"[2011-06-20 15:43:07] VERBOSE[22529] logger.c: [2011-06-20 15:43:07]
-- Reloading module 'res_monitor.so' (Call Monitoring Resource)

[2011-06-20 15:43:07] DEBUG[22529] res_monitor.c: LOAD (Channel module
load) [monitor.conf]

[2011-06-20 15:43:07] VERBOSE[22529] logger.c: [2011-06-20 15:43:07]
== Parsing '/etc/asterisk/monitor.conf': [2011-06-20 15:43:07]
VERBOSE[22529] logger.c: [2011-06-20 15:43:07] Found

[2011-06-20 15:43:07] NOTICE[22529] res_monitor.c: monitor.conf:18: Bad
config option 'rtp_server' - (ignoring)

[2011-06-20 15:43:07] DEBUG[22529] res_monitor.c: monitor to
bi-precision-06 [start=9000, end=]"

"

 

Any help appreciated!

 

Regards / Marcus

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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[bilal ghayyad]

Dear Dan;

I have to do something in the compilation to have chan_sccp? Because, I do not 
have this channel and I have only chan_skinny.

Even in the /usr/lib/asterisk/module/, I did not find chan_sccp.

Maybe that is the reason why I do not have the sccp.conf file?

So, using the sccp channel, will also face the same problem that the phones 
will restarted if I did reload?

Regards
Bilal


--- On Mon, 6/20/11, Dan Austin  wrote:

> From: Dan Austin 
> Subject: RE: Cisco IP Phones and Skinny in asterisk
> To: "bilal ghayyad" 
> Date: Monday, June 20, 2011, 7:09 PM
> It would be best to keep this on the
> list, I just had not
> had a chance to reply yet.
> 
> Your first issue is just how the SCCP protocol works. 
> Every
> keypress is relayed to the server, so the phones must
> maintain
> an active connection to the PBX.  You can avoid this
> by just
> reloading the modules you update and not the whole PBX-
>     ie- sip reload or module reload
> chan_sip
> 
> The second issue is likely a firmware issue on the phone,
> and
> Likely one where the phone software is too new.  You
> might also
> Not have the correct definition in skinny.conf
> 
> I did use chan_sccp years ago, but have not kept up with
> it.
> The configuration should be with the source package for
> that
> channel.  The configuration is similar, but you cannot
> rename
> the files as there are key differences.
> 
> Dan
> 
> -Original Message-
> From: bilal ghayyad [mailto:bilmar...@yahoo.com]
> 
> Sent: Monday, June 20, 2011 3:40 PM
> To: Dan Austin
> Subject: Re: Cisco IP Phones and Skinny in asterisk
> 
> Dear Dan;
> 
> Because you are using skinny with your Cisco IP Phones in
> the office, so I beleive you might help me really to resolve
> my problem (please).
> 
> First of all, are u using skinny channel or sccp channel?
> 
> Actually, I tried skinny and I faced two major problems (so
> if I am going to face same problems in sccp then no need to
> use sccp, so please advise).
> 
> The two problems that I faced them are:
> 
> 1) When I do reload then the skinny channel is reloaded and
> that will cause a restart for the Cisco IP Phones (that are
> registered to skinny channel). Is the same thing happening
> with u when u r using sccp channel?
> 
> 2) When I called the Phone, it is ringing, when we pickup
> the handset to answer the call, we hear
> t and we do not hear what source is
> talking and source does not hear us even .. but if we select
> music on hold, then caller will hear the music. Also, when
> we tried to use the Ciscp IP Phone to place a call, while we
> are dialing, the too tone is always existed and
> it is ringing at destination but no voice (always
> t).
> 
> So if I used sccp then I will not face these problems?
> 
> From the other side, if I need to use sccp (if we assumed
> the above problems are not existed) then can u please help
> for below:
> 
> 1) If i used sccp and I gave the IP Phone the IP address
> TFTP server, and no configuration files were existed on
> TFTP, then it will register on the asterisk sccp channel?
> 
> 2) The sccp.conf file, where I can find it? Is it the same
> as the skinny.conf file?
> 
> 3) To use sccp instead of the skinny channel, all what I
> need is to unload the skinny from the modules.conf file and
> load the sccp channel in the modules.conf, and I can use the
> skinny.conf file for the configuration? About the firmware
> on the Phone, it will stay the same?
> 
> I appreciate the kindly help please.
> Regards
> Bilal
> 

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Re: [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko

[randulo]

On Tue, Jun 21, 2011 at 5:47 AM, Alex Balashov
 wrote:
> I nominate this for most imaginative use of Asterisk-users of 2011.

It's already qualified to win in the grammar and spelling categories.

/r

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[asterisk-users] Easier to remember transfer and pick-up key sequences

[Sebastian Arcus]

I'm thinking of implementing some easier to remember key sequences in 
features.conf. Something like:


pickupexten = ##
atxfer => **


Can anybody think why this might be a bad idea, compared to the defaults 
*8 and *2? I have made sure that no other feature uses single * or #, to 
avoid matching the wrong feature.


Thanks,

Sebastian

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Re: [asterisk-users] Failover trunks

[Abid Saleem]

Dear Alex,
Thanks for the answer. Will it work in the trunk failover mode. I mean if 
trunk1 fails, the call will go through trunk2. Also I have another problem that 
my provider needs P-Preffered-Identity Header to be added for each trunk. So 
how can I add this header to each appropriate trunk with that trunks User name 
in Preferred-Identity header.
Regards---Abid Saleem

> Date: Wed, 18 May 2011 10:07:03 -0400
> From: abalas...@evaristesys.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Failover trunks
> 
> On 05/18/2011 10:05 AM, Abid Saleem wrote:
> 
> > Could you please help me with my following Scenario. I have a
> > softswitch where my carriers send calls from International to my
> > country for local termination. I route these calls to my Asterisk 1.8
> > which has a number of registered trunks from our SIP Provider. Please
> > guide me how should I configure extensions.conf for calls to be sent
> > to the next available trunk.
> 
> exten => ...,1,Dial(SIP/${EXTEN}@trunk1,...)
> exten => ...,n,Dial(SIP/${EXTEN}@trunk2,...)
> exten => ...,n,Dial(SIP/${EXTEN}@trunk3,...)
> 
> ?
> 
> -- 
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
> 
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Re: [asterisk-users] "pickupsound = beep" kills call pickup in Asterisk1.8.4.2

[Sebastian Arcus]

Thank you Alec,

I needed some confirmation that it wasn't something I was doing. I can 
live without pickupsound, and the bug is already reported - so it's all 
good.


Sebastian

On 21/06/11 00:29, Alec Davis wrote:

This has been fixed only last month, see
https://issues.asterisk.org/view.php?id=18654 and try bug18654.diff.txt
That will avoid the deadlock, but it's not the proper fix, there are other
issues that could trip you up,
mainly to do with race conditions with multiple channels picking up the same
ringing extensions.

Alec Davis






-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sebastian Arcus
Sent: Tuesday, 21 June 2011 11:10 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] "pickupsound = beep" kills call
pickup in Asterisk1.8.4.2

I have discovered that if I enable pickupsound = beep in
features.conf, if I try to do a pickup with *8, the calling
channel keeps on ringing, while the phone where I pick-up
from shows that the call has been answered (I don't know
where though). Also, it seems to completely bugger up my
outgoing IAX trunk (I really can't see the connection, as I'm
doing pick-up for a SIP channel). I can only shut Asterisk
down with killall asterisk -s9 - nothing else works.

I've tried starting the console with asterisk -rvv - but
there is nothing unusual there.

Could someone please confirm this behaviour on their box,
before I go and submit a bug - in case I am doing something wrong?

As soon as I comment out "pickupsound = beep" - everything
works just fine and I can do call pickup with *8.

Sebastian

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