Re: [asterisk-users] Rebooting a Grandstream

[Mike Diehl]

Well, it doesn't seem to work on my GXP2000's!Is there a 
configuration options that I need to set?

TIA,

Mike.

On Thursday 21 July 2011 6:59:19 pm Alec Davis wrote:
> That works for us with GXP2000's and GXP2010, but not the later HD series
> GXP21XX.
> 
> Alec
> 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > Mike Diehl
> > Sent: Friday, 22 July 2011 10:50 a.m.
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] Rebooting a Grandstream
> > 
> > Hi all,
> > 
> > I've got a number of Grandstream phones and I'd like to be
> > able to reboot them remotely, as I do my Polycoms...
> > 
> > I've got this in my sip_notify.cfg:
> > 
> > [grandstream-check-cfg]
> > Event=>sys-control
> > 
> > Doesn't seem to work.  Any ideas?
> 
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Re: [asterisk-users] Rebooting a Grandstream

[Alec Davis]

That works for us with GXP2000's and GXP2010, but not the later HD series
GXP21XX.

Alec

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
> Mike Diehl
> Sent: Friday, 22 July 2011 10:50 a.m.
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Rebooting a Grandstream
> 
> Hi all,
> 
> I've got a number of Grandstream phones and I'd like to be 
> able to reboot them remotely, as I do my Polycoms...
> 
> I've got this in my sip_notify.cfg:
> 
> [grandstream-check-cfg]
> Event=>sys-control
> 
> Doesn't seem to work.  Any ideas?
> 
> -- 
> 
> Take care and have fun,
> Mike Diehl.
> 
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[asterisk-users] Strange network issue

[Mike Diehl]

Hi all,

I've got a strange problem with a customer's phones.

They've got a bunch of Grandstreams that seem to be rock solid... until 
7:00pm.  At 7:00, some of the phones become unavailable, and stay down.  Call 
quality is solid almost all the time.  But right at 7:00, things go bad.  Only 
some of the phone lines go down and they stay down until the phone is 
rebooted.

I'm not even sure what to look for when I go to the site.  Any ideas?

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[asterisk-users] Per-line registration

[Mike Diehl]

Hi all,

I'm trying to figure out how it is that a couple lines on a given phone, with 3 
lines, can qualify as unavailable while the remaining lines can be available.  
I've got qualify=1000 in my sip.cfg.

Shouldn't this be an all-or-nothing proposition?

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[asterisk-users] Rebooting a Grandstream

[Mike Diehl]

Hi all,

I've got a number of Grandstream phones and I'd like to be able to reboot them 
remotely, as I do my Polycoms...

I've got this in my sip_notify.cfg:

[grandstream-check-cfg]
Event=>sys-control

Doesn't seem to work.  Any ideas?

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Re: [asterisk-users] FAX with SIP

[Paul Belanger]

On 11-07-21 05:52 PM, Joaquin Sosa wrote:

On Thu, Jul 21, 2011 at 17:39, Kevin P. Fleming  wrote:

We do this in our testing all the time, and it works fine. Since you didn't
specify any particular version of Asterisk, there's no way to associate your
"It won't work" statement with anything in particular. Given the variations
of T.38 implementations that exist in ATAs, carrier networks and other
places, *any* T.38 connection that involves implementations from more than
one vendor is (unfortunately) likely to have problems, whether any version
of Asterisk is involved or not.



Do you care to share your exact configuration? I would love to
reproduce it and demonstrate that T.38 in Asterisk indeed works. All
I've ever managed to do is have it drop calls when the T.38 switchover
is attempted. Does Asterisk even support T.38 AND NAT?



mnicholson recently added some tests[1] into the testsuite.

[1] http://svn.asterisk.org/svn/testsuite/asterisk/trunk/tests/fax/

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Re: [asterisk-users] FAX with SIP

[Israel Gottlieb]

On Fri, Jul 22, 2011 at 12:50 AM, Kevin P. Fleming wrote:

> On 07/21/2011 04:43 PM, Israel Gottlieb wrote:
>
>>
>>
>> On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming > > wrote:
>>
>>On 07/21/2011 04:34 PM, Joaquin Sosa wrote:
>>
>>On Mon, Jul 18, 2011 at 07:58, Steve Davies>>  wrote:
>>
>>
>>The magic sauce that you need is "T.38" - Asterisk 1.6
>>supports this
>>to a limited degree, and your ITSP will need to support it.
>>
>>The sip.conf.sample file and the voip-info wiki has all the
>>information you need to try it out.
>>
>>
>>Correct. However it would be helpful to note T.38 support in
>>Asterisk
>>is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and
>>try to send a fax. It won't work!
>>
>>
>>We do this in our testing all the time, and it works fine. Since you
>>didn't specify any particular version of Asterisk, there's no way to
>>associate your "It won't work" statement with anything in
>>particular. Given the variations of T.38 implementations that exist
>>in ATAs, carrier networks and other places, *any* T.38 connection
>>that involves implementations from more than one vendor is
>>(unfortunately) likely to have problems, whether any version of
>>Asterisk is involved or not
>>
>>
>>
>> well I tried  a linksys spa 8000 and 2102 thru
>> asterisk 1.8.3
>> 1.8.4
>> 1.6.2.16-19
>> sonus switch at itsp (012 israel)
>>
>> and no luck
>>
>
> We'd be happy to investigate why it failed, if you can capture the packet
> streams on both sides of Asterisk. Frequently, it's a configuration issue in
> at least one of the devices in the system.
>

NP I'll get that for you I have spent days trying to get it to work with no
luck

>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] FAX with SIP

[Joaquin Sosa]

On Thu, Jul 21, 2011 at 17:39, Kevin P. Fleming  wrote:
> We do this in our testing all the time, and it works fine. Since you didn't
> specify any particular version of Asterisk, there's no way to associate your
> "It won't work" statement with anything in particular. Given the variations
> of T.38 implementations that exist in ATAs, carrier networks and other
> places, *any* T.38 connection that involves implementations from more than
> one vendor is (unfortunately) likely to have problems, whether any version
> of Asterisk is involved or not.
>

Do you care to share your exact configuration? I would love to
reproduce it and demonstrate that T.38 in Asterisk indeed works. All
I've ever managed to do is have it drop calls when the T.38 switchover
is attempted. Does Asterisk even support T.38 AND NAT?

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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Joaquin Sosa]

On Sun, Jul 10, 2011 at 04:11, Tzafrir Cohen  wrote:
>
>
> The solution: IOMMU: http://en.wikipedia.org/wiki/Iommu .
> The CPU of the system has a Memory Management Unit (MMU) that
> maps virtual address spaces to processes. Likewise we know prevent the
> IO card from seeing physical addresses. Rather, it sees virtual
> addresses mapped by the IOMMU. Just like the operating system maps
> addresses for processes, the hypervisor maps address ranger to I

My understanding is PCI passthrough is supported on VMWare ESXi with
the proper hardware support.

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Re: [asterisk-users] FAX with SIP

[Kevin P. Fleming]

On 07/21/2011 04:43 PM, Israel Gottlieb wrote:



On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming mailto:kpflem...@digium.com>> wrote:

On 07/21/2011 04:34 PM, Joaquin Sosa wrote:

On Mon, Jul 18, 2011 at 07:58, Steve Daviesmailto:davies...@gmail.com>>  wrote:

The magic sauce that you need is "T.38" - Asterisk 1.6
supports this
to a limited degree, and your ITSP will need to support it.

The sip.conf.sample file and the voip-info wiki has all the
information you need to try it out.


Correct. However it would be helpful to note T.38 support in
Asterisk
is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and
try to send a fax. It won't work!


We do this in our testing all the time, and it works fine. Since you
didn't specify any particular version of Asterisk, there's no way to
associate your "It won't work" statement with anything in
particular. Given the variations of T.38 implementations that exist
in ATAs, carrier networks and other places, *any* T.38 connection
that involves implementations from more than one vendor is
(unfortunately) likely to have problems, whether any version of
Asterisk is involved or not



well I tried  a linksys spa 8000 and 2102 thru
asterisk 1.8.3
1.8.4
1.6.2.16-19
sonus switch at itsp (012 israel)

and no luck


We'd be happy to investigate why it failed, if you can capture the 
packet streams on both sides of Asterisk. Frequently, it's a 
configuration issue in at least one of the devices in the system.


--
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] FAX with SIP

[Israel Gottlieb]

On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming wrote:

> On 07/21/2011 04:34 PM, Joaquin Sosa wrote:
>
>> On Mon, Jul 18, 2011 at 07:58, Steve Davies  wrote:
>>
>>> The magic sauce that you need is "T.38" - Asterisk 1.6 supports this
>>> to a limited degree, and your ITSP will need to support it.
>>>
>>> The sip.conf.sample file and the voip-info wiki has all the
>>> information you need to try it out.
>>>
>>>
>> Correct. However it would be helpful to note T.38 support in Asterisk
>> is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and
>> try to send a fax. It won't work!
>>
>
> We do this in our testing all the time, and it works fine. Since you didn't
> specify any particular version of Asterisk, there's no way to associate your
> "It won't work" statement with anything in particular. Given the variations
> of T.38 implementations that exist in ATAs, carrier networks and other
> places, *any* T.38 connection that involves implementations from more than
> one vendor is (unfortunately) likely to have problems, whether any version
> of Asterisk is involved or not
>


well I tried  a linksys spa 8000 and 2102 thru
asterisk 1.8.3
1.8.4
1.6.2.16-19
sonus switch at itsp (012 israel)

and no luck
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Re: [asterisk-users] FAX with SIP

[Kevin P. Fleming]

On 07/21/2011 04:34 PM, Joaquin Sosa wrote:

On Mon, Jul 18, 2011 at 07:58, Steve Davies  wrote:

The magic sauce that you need is "T.38" - Asterisk 1.6 supports this
to a limited degree, and your ITSP will need to support it.

The sip.conf.sample file and the voip-info wiki has all the
information you need to try it out.



Correct. However it would be helpful to note T.38 support in Asterisk
is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and
try to send a fax. It won't work!


We do this in our testing all the time, and it works fine. Since you 
didn't specify any particular version of Asterisk, there's no way to 
associate your "It won't work" statement with anything in particular. 
Given the variations of T.38 implementations that exist in ATAs, carrier 
networks and other places, *any* T.38 connection that involves 
implementations from more than one vendor is (unfortunately) likely to 
have problems, whether any version of Asterisk is involved or not.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] FAX with SIP

[Joaquin Sosa]

On Mon, Jul 18, 2011 at 07:58, Steve Davies  wrote:
> The magic sauce that you need is "T.38" - Asterisk 1.6 supports this
> to a limited degree, and your ITSP will need to support it.
>
> The sip.conf.sample file and the voip-info wiki has all the
> information you need to try it out.
>

Correct. However it would be helpful to note T.38 support in Asterisk
is broken. Take a T.38 ITSP, T.38 enabled Asterisk and T.38 ATA and
try to send a fax. It won't work!

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Re: [asterisk-users] My Asterisk Box was hacked

[Joaquin Sosa]

Are you sure your box was actually hacked? Or did someone take
advantage of a configuration error?

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Re: [asterisk-users] asterisk's SDP

[Kevin P. Fleming]

On 07/21/2011 03:54 PM, vip killa wrote:

What if asterisk sends telephony events that are not in range of 0-15
though?


You are misunderstanding how SDP works; when an SDP offer or answer is 
sent, that indicates what the sender is willing to *receive*, not what 
it is going to send.


If the Sonus device sent "fmtp:101 0-15" in its SDP, then Asterisk 
should not send 'event 16' events to it. If it does, that's a bug, 
although standard programming practices would mean that it wouldn't be 
harmful, it would just be ignored by the Sonus device.


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Re: [asterisk-users] asterisk's SDP

[vip killa]

What if asterisk sends telephony events that are not in range of 0-15
though?

On Thu, Jul 21, 2011 at 4:47 PM, Kevin P. Fleming wrote:

> On 07/21/2011 03:30 PM, vip killa wrote:
>
>> We have a peer (a Sonus Media Gateway), that sends "a=fmtp:101 0-15"
>> Asterisk sends "0-16" back, is there anyway to have asterisk send a 0-15?
>>
>
> No, and it's completely unnecessary. Asterisk is willing to accept
> telephony-event codes 0 through 16, but the other endpoint is not obligated
> to send them if it doesn't want to.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] asterisk's SDP

[Kevin P. Fleming]

On 07/21/2011 03:30 PM, vip killa wrote:

We have a peer (a Sonus Media Gateway), that sends "a=fmtp:101 0-15"
Asterisk sends "0-16" back, is there anyway to have asterisk send a 0-15?


No, and it's completely unnecessary. Asterisk is willing to accept 
telephony-event codes 0 through 16, but the other endpoint is not 
obligated to send them if it doesn't want to.


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Digium, Inc. | Director of Software Technologies
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[asterisk-users] asterisk's SDP

[vip killa]

We have a peer (a Sonus Media Gateway), that sends "a=fmtp:101 0-15"
Asterisk sends "0-16" back, is there anyway to have asterisk send a 0-15?
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Re: [asterisk-users] My Asterisk Box was hacked

[Steve Edwards]

On Thu, 21 Jul 2011, Robert Huddleston wrote:


When I get hacked I typically run a rootkit checker
http://www.chkrootkit.org/


How often do you get hacked?

How are 'they' breaking in?

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Re: [asterisk-users] My Asterisk Box was hacked

[Robert Huddleston]

When I get hacked I typically run a rootkit checker
http://www.chkrootkit.org/

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chad Wallace
Sent: Thursday, July 21, 2011 2:18 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] My Asterisk Box was hacked

On Thu, 21 Jul 2011 13:29:09 +0800
Malvin Rito  wrote:

> My asterisk box was hacked! Can anyone help on how do I secure my 
> asterisk box, currently my box is installed with 2 NIC. 1st NIC is
> for LAN access and 2nd NIC has a public IP which is registered to our
> VoIP Provider.


Seven Steps to Better SIP Security with Asterisk
http://blogs.digium.com/2009/03/28/sip-security/


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Re: [asterisk-users] My Asterisk Box was hacked

[Chad Wallace]

On Thu, 21 Jul 2011 13:29:09 +0800
Malvin Rito  wrote:

> My asterisk box was hacked! Can anyone help on how do I secure my 
> asterisk box, currently my box is installed with 2 NIC. 1st NIC is
> for LAN access and 2nd NIC has a public IP which is registered to our
> VoIP Provider.


Seven Steps to Better SIP Security with Asterisk
http://blogs.digium.com/2009/03/28/sip-security/


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Re: [asterisk-users] Functions not autoloading

[--[ UxBoD ]--]

Just received a call and on checking messages I now see:

ERROR[14824] pbx.c: Function MASTER_CHANNEL not registered

Grrr, looks like time to go back to 1.8.3 as all the apps and functions exist 
in /usr/lib/asterisk/modules.

How could I help to debug this please ?
-- 
Thanks, Phil

- Original Message -
> On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote:
> > Since upgrading to 1.8.5.0 I have had to add into modules.conf:
> >
> > load =>  func_callerid.so
> > load =>  func_cdr.so
> >
> > otherwise they do not get loaded even though I have set
> > autoload=yes.
> >
> > Is this something you would expect as it is different behavior to
> > 1.8.3.0 and I do not see any issues in /var/log/asterisk/messages
> > ?
> 
> No, this is not expected behavior.
> 
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype:
> kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
> 
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Re: [asterisk-users] HELP - Client wants to reverse Asterisk Functionality

[Danny Nicholas]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten
Göllner
Sent: Thursday, July 21, 2011 1:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] HELP - Client wants to reverse Asterisk
Functionality

 

What does "but this didn't work" mean?

Am 20.07.2011 15:25, schrieb Danny Nicholas: 

Hello,

I’m putting Asterisk in to replace an existing IVR and that PBX
system uses * to terminate number input instead of #.  I thought it would be
a matter of simply making a new app_read that replaced \# with \* but this
didn’t work.  Any suggestions (besides bopping the client up side the head
for wanting this?).

 

I took apps/app_read.c in asterisk 1.4.37 (same source in all 1.4 modules
after that) and modified lines 71, 72 and 183 replacing # with * and save
the modified source as app_readstar.c.  I then did make menuselect and got
the new module app_readstar available for selection.  Selected it and built
Asterisk as normal.  In my dialplan I replaced Read(var,file,blah) with
Readstar(var,file,blah).  It gives the “right” message if you do core show
application readstar, but * does not terminate input as I had hoped/expected
it would.





 

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Re: [asterisk-users] Functions not autoloading

[Kevin P. Fleming]

On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote:

Since upgrading to 1.8.5.0 I have had to add into modules.conf:

load =>  func_callerid.so
load =>  func_cdr.so

otherwise they do not get loaded even though I have set autoload=yes.

Is this something you would expect as it is different behavior to 1.8.3.0 and I 
do not see any issues in /var/log/asterisk/messages ?


No, this is not expected behavior.

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] My Asterisk Box was hacked

[Terry Brummell]

Really, since you sound like a novice in the Asterisk world, maybe
rolling your own solution isn't a good idea.  Why not use an all-in-one
solution like PBX in a Flash?  

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Malvin
Rito
Sent: Thursday, July 21, 2011 1:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] My Asterisk Box was hacked

Hi List,

My asterisk box was hacked! Can anyone help on how do I secure my 
asterisk box, currently my box is installed with 2 NIC. 1st NIC is for 
LAN access and 2nd NIC has a public IP which is registered to our VoIP 
Provider.

As I remember I already tried putting our Box on NAT but unfortunately 
due to some issue like call is dropped after 30 seconds and sometimes 
voice are not heard. Then we disable again the NAT.

Your advise will be much appreciated. Thanks in advance.

Regards,
Malvin

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[asterisk-users] Functions not autoloading

[--[ UxBoD ]--]

Since upgrading to 1.8.5.0 I have had to add into modules.conf:

load => func_callerid.so
load => func_cdr.so

otherwise they do not get loaded even though I have set autoload=yes.

Is this something you would expect as it is different behavior to 1.8.3.0 and I 
do not see any issues in /var/log/asterisk/messages ?
-- 
Thanks, Phil

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[asterisk-users] Asterisk doesn't like OpenBTS!!!

[A.H. Jos]

HI list,
I have succeeded to establish calls using OpenBTS/USRP1/Asterisk. but the
problem is that my cell phone rings, I get 2 way audio but after a few
seconds the call is dropped. In my asterisk log I see this:

[Jul 18 11:25:48] WARNING[1221]: chan_sip.c:3622 retrans_pkt: Retransmission
timeout reached on transmission 1348333597@127.0.0.1 for seqno 94 (Critical
Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Jul 18 11:25:48] WARNING[1221]: chan_sip.c:3651 retrans_pkt: Hanging up
call 1348333597@127.0.0.1 - no reply to our critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

In the SIP Debug, I see always 10 Retransmissions of the same "SIP/2.0 200
Ok" message!!! after that the above "Retransmission timeout" message is
viewed!!!

Retransmitting #10 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK53934;received=127.0.0.
1
From: IMSI208012601160193

>;tag=lkbdg
To: ;tag=as7c57c466
Call-ID: 1348333597@127.0.0.1
CSeq: 94 INVITE
Server: Asterisk PBX 1.8.5.0-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: 
Content-Type: application/sdp
Content-Length: 213

I have established only a one call from my hardphone (connected to OpenBTS)
to my twinkle softphone. but after the call is dropped (T == 32 secondes) by
my softphone and after hanging up my hardphone (T == 60 seconds) I have
received automatically a call from my twinkle softphone!!!
In wireshark trace, I see that OpenBTS is trying to ACK the OK from
Asterisk, but Asterisk doesn't like it !!!

I have tried to modify the value of the SIP timers, that works only from a
hardphone to a softphone but not from hard to hard. can some one tell us
what's the definition of t1min and timert1?
t1min=1000
timert1=5000
timerb=32000

Any help will be appreciated.
A.H. Jos,
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Re: [asterisk-users] AsteriskNow install addons despite license conflict with FFA and G.729

[Michael]

Any suggestions on how to install Asterisk addons despite the license
conflict? BTW, on another system, we installed the addons first and then the
paid licenses from digium and there was absolutely no problem running and
installing both. It seems to happen only when Digium software is installed
BEFORE the addons are.

Thanks.

On Tue, Jul 19, 2011 at 10:46 PM, Michael  wrote:

> On Tue, Jul 19, 2011 at 9:49 PM, Jason Parker  wrote:
>
>> Yes, but you don't have to use cdr_mysql to insert into a MySQL database.
>>  The cdr_odbc module works just fine for that.
>
>
> So what's the procedure required to set FreePBX CDRs active, under these
> conditions? How do I activate/install/set the cdr_odbc module?
>
> Thanks.
>
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Re: [asterisk-users] a=sendonly Music On Hold ignored

[Michael]

On Tue, Jul 19, 2011 at 10:12 PM, Matthew J. Roth  wrote:

> Michael wrote:
> >
> > True. In the working system, LAN calls are also using G.729, while
> > in the non-working system, LAN calls are in G.711 (supported but
> > not prioritized by the phones) and only the SIP trunk to the ITSP
> > is set to G.729.
>
> Can you set the phone to G.711 and try making a LAN call on the non-
> working system.  If a call that is G.711 from end-to-end doesn't have
> the same problem it would be evidence of a codec translation issue.
>
That's the full trace of a call in G.711:

Reliably Transmitting (no NAT) to 192.168.1.109:5060:
INVITE sip:500@192.168.1.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4751ec69;rport
Max-Forwards: 70
From: "9000" ;tag=as40d1788b
To: 
Contact: 
Call-ID: 3b81833312062ead03c47919333e549c@192.168.1.10
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.6.2.19)
Date: Thu, 21 Jul 2011 07:12:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1017783905 1017783905 IN IP4 192.168.1.10
s=Asterisk PBX 1.6.2.19
c=IN IP4 192.168.1.10
t=0 0
m=audio 17696 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called 500

<--- SIP read from UDP:192.168.1.109:5060 --->
SIP/2.0 180 Ringing
From: "9000";tag=as40d1788b
To: ;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
Call-ID: 3b81833312062ead03c47919333e549c@192.168.1.10
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;branch=z9hG4bK4751ec69
Supported: replaces,100rel
User-Agent: SIP Phone
Contact: 
Content-Length: 0


<->
--- (10 headers 0 lines) ---
-- SIP/500-0330 is ringing

<--- SIP read from UDP:192.168.1.109:5060 --->
SIP/2.0 200 OK
From: "9000";tag=as40d1788b
To: ;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
Call-ID: 3b81833312062ead03c47919333e549c@192.168.1.10
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.10:5060;rport=5060;branch=z9hG4bK4751ec69
Supported: replaces,100rel
User-Agent: SIP Phone
Contact: 
Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 226

v=0
o=500 109600 0 IN IP4 192.168.1.109
s=Audio Session
i=Audio Session
c=IN IP4 192.168.1.109
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.109:16384
list_route: hop: 
set_destination: Parsing  for address/port to
send to
set_destination: set destination to 192.168.1.109, port 5060

Transmitting (no NAT) to 192.168.1.109:5060:
ACK sip:500@192.168.1.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1265e92e;rport
Max-Forwards: 70
From: "9000" ;tag=as40d1788b
To: ;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
Contact: 
Call-ID: 3b81833312062ead03c47919333e549c@192.168.1.10
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.6.2.19)
Content-Length: 0


---
-- SIP/500-0330 answered SIP/Smile-032f
[Jul 21 10:12:36] WARNING[13622]: dsp.c:1360 ast_dsp_process: Inband DTMF is
not supported on codec g729. Use RFC2833


<--- SIP read from UDP:192.168.1.109:5060 --->
INVITE sip:9000@192.168.1.10 SIP/2.0
From: ;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000";tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c@192.168.1.10
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK-313-c04d6-373f24ae
Max-Forwards: 70
Supported: replaces,100rel
User-Agent: SIP Phone
Contact: 
Allow: INVITE,ACK,BYE,REFER,NOTIFY,PRACK,CANCEL,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 294

v=0
o=500 109601 0 IN IP4 192.168.1.109
s=SIPPhone Session
i=Audio Session
c=IN IP4 0.0.0.0
t=0 0
m=audio 16384 RTP/AVP 0 8 18 101
a=sendonly
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<->
--- (13 headers 14 lines) ---
Sending to 192.168.1.109 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.109:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.109:5060
;branch=z9hG4bK-313-c04d6-373f24ae;received=192.168.1.109
From: ;tag=80d13448-c0a8016d-13c4-45026-2fe-5eeb0557-2fe
To: "9000";tag=as40d1788b
Call-ID: 3b81833312062ead03c47919333e549c@192.168.1.10
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.6.2.19)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: 
Conte

[asterisk-users] Prevent Asterisk from setting CALLERID(name) or unsetting CALLERID(name)

[Benoit Panizzon]

Hi all

We use a Genband Safari C3 Softswitch and have an attached Asterisk for some 
special funktions like SPOT Filtering.

Now then a call comes from PSTN to a SIP subscriber, the invite looks like:

From: ;tag=7f33ff47+1+68530003+e1367280

If the device is a Snom M9 (or many others) in the absence of a CALLERID(name) 
the CALLERID(num) is displayed.

If the same invite reaches the Asterisk server. Strangely the CALLERID(name) 
is set to the same Value as the CALLERID(num).

Now the CALLERID(num) is being changed (localized) during processing of the 
call on the asterisk server.

If now the call get's connected to a SIP customer.

From: "+4179***" ;tag=as40775516

So a SIP customer get's a non localized number on his display (which is 
CALLERID(name)).

Is there a way to get asterisk not to invent a CALLERID(name) if there is 
none?

Id did try to set ${CALLERID(name)=""} but that resulted in From: ""  
and the displaying of this empty string on the subscribers phone.

Is there a way to completely remove the CALLERID(name) like 
(UNSET({CALLERID(name))?

Kind regards

Benoit Panizzon
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Re: [asterisk-users] Multiple SIP trunks between same pair of asterisk box

[Faisal Hanif]

If it is just matter of billing you can pass billing related info in
additional SIP headers on single trunk.

 

If you must need multiple trunk you can add multiple IPs of different subnet
class to both interfaces and configure asterisk to listen of all IPs. Then
use one trunk per IP Subnet class.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Thursday, July 21, 2011 3:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Multiple SIP trunks between same pair of asterisk
box

 

Hello,
for billing purpose between a multitenant asterisk box and another asterisk,
I am in the need to maintain multiple SIP trunks between them. Usually I use
insecure=invite,port but I had to remove or the trunks will be selected
based on IP address and not with username/password. I had to use the
fromuser option or asterisk will try to authenticate the call using the CID
and not the username, but this break the outbound CID of the client.

Both are asterisk 1.6

Is there any other solution from multiple SIP trunks between two asterisk
boxes?

Leandro

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